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

[alsa-devel] [PATCH] ALSA: atmel_abdac: clk_round_rate() can return a zero upon error
by Paul Walmsley 10 Dec '13

10 Dec '13

Treat both negative and zero return values from clk_round_rate() as errors. This is needed since subsequent patches will convert clk_round_rate()'s return value to be an unsigned type, rather than a signed type, since some clock sources can generate rates higher than (2^31)-1 Hz. Eventually, when calling clk_round_rate(), only a return value of zero will be considered a error; all other values will be considered valid rates. The comparison against values less than 0 is kept to preserve the correct behavior in the meantime. Signed-off-by: Paul Walmsley <pwalmsley(a)nvidia.com> Cc: Hans-Christian Egtvedt <egtvedt(a)samfundet.no> Cc: Nicolas Ferre <nicolas.ferre(a)atmel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- Applies on v3.13-rc3. See also: http://marc.info/?l=linux-arm-kernel&m=138542591313620&w=2 sound/atmel/abdac.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 721d8fd..3519518 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -354,7 +354,7 @@ static int set_sample_rates(struct atmel_abdac *dac) /* we start at 192 kHz and work our way down to 5112 Hz */ while (new_rate >= RATE_MIN && index < (MAX_NUM_RATES + 1)) { new_rate = clk_round_rate(dac->sample_clk, 256 * new_rate); - if (new_rate < 0) + if (new_rate <= 0) break; /* make sure we are below the ABDAC clock */ if (index < MAX_NUM_RATES &&

2 1

Re: [alsa-devel] [PATCH V2] ASoC: add RT286 CODEC driver
by Bard Liao 10 Dec '13

10 Dec '13

> -----Original Message----- > From: Leon Romanovsky [mailto:leon@leon.nu] > Sent: Wednesday, December 04, 2013 10:37 PM > To: Bard Liao > Cc: Mark Brown; Liam Girdwood; Oder Chiou; alsa-devel(a)alsa-project.org; > Flove > Subject: Re: [alsa-devel] [PATCH V2] ASoC: add RT286 CODEC driver > > On Wed, Dec 4, 2013 at 3:41 PM, <bardliao(a)realtek.com> wrote: > > From: Bard Liao <bardliao(a)realtek.com> > > > > This patch adds the ALC286 codec driver. > > > > Signed-off-by: Bard Liao <bardliao(a)realtek.com> > > > > Please add change log while you are sending verX patch. Please see the change log as below. * Add platform data to determine if combo jack and/or IRQ is using or not. * Implement rt286_hw_read/write in regmap. * Configure codec in runtime instead of in probe. * Other minor changes (coding style, remove version control, remove empty functions, share code ...) Should I send another patch with these change log? > > > > + > > +static unsigned int rt286_reg_cache[TOTAL_NODE_ID + 1] = { > You don't need TOTAL_NODE_ID + 1 in array declaration. Without "TOTAL_NODE_ID + 1", I should always set [NODE_ID_MIXER_IN2] = 0x0000, right? Ie. Assign the latest array item a value. > > > + [NODE_ID_DAC_OUT1] = 0x7f7f, > > + [NODE_ID_DAC_OUT2] = 0x7f7f, > > + [NODE_ID_SPDIF] = 0x0000, > > + [NODE_ID_ADC_IN1] = 0x4343, > > + [NODE_ID_ADC_IN2] = 0x4343, > > + [NODE_ID_MIC1] = 0x0000, > > + [NODE_ID_MIXER_IN] = 0x000b, > > + [NODE_ID_MIXER_OUT1] = 0x0002, > > + [NODE_ID_MIXER_OUT2] = 0x0000, > > + [NODE_ID_SPK_OUT] = 0x0000, > > + [NODE_ID_HP_OUT] = 0x0000, > > + [NODE_ID_MIXER_IN1] = 0x0000, > > + [NODE_ID_MIXER_IN2] = 0x0000, > > +}; Thanks.

2 1

Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by Raymond Yau 10 Dec '13

10 Dec '13

2013/12/9 John Ettedgui <john.ettedgui(a)gmail.com> > > I am using the gitcompile script to configure, compile... > There is no configure by default, I was able to autogen it, but then I had > issues while building so I reverted back to gitcompile > [ 4.841881] ALSA hda_intel.c:3170 0000:01:00.1: Handle VGA-switcheroo audio client [ 4.841885] ALSA hda_intel.c:3364 0000:01:00.1: Using LPIB position fix [ 4.841886] ALSA hda_intel.c:3510 0000:01:00.1: Force to non-snoop mode [ 4.841920] snd_hda_intel 0000:01:00.1: irq 47 for MSI/MSI-X [ 4.841940] ALSA hda_intel.c:3635 0000:01:00.1: chipset global capabilities = 0x0 seem the max number of playback streams is hardcoded in the driver [ 4.842211] [drm] radeon kernel modesetting enabled. [ 4.844508] ALSA hda_intel.c:1244 0000:01:00.1: codec_mask = 0x1 [ 4.844630] ALSA hda_intel.c:1716 0000:01:00.1: codec #0 probed OK [ 4.844632] ALSA hda_intel.c:1837 0000:01:00.1: Enable sync_write for stable communication [ 4.851623] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=3 Presence_Detect=0 ELD_Valid=0 [ 4.851654] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=5 Presence_Detect=0 ELD_Valid=0 [ 4.851685] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=7 Presence_Detect=0 ELD_Valid=0 [ 4.851717] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=9 Presence_Detect=0 ELD_Valid=0 [ 4.851751] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=11 Presence_Detect=0 ELD_Valid=0 [ 4.851787] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=13 Presence_Detect=0 ELD_Valid=0 [ 4.851839] input: HDA ATI HDMI HDMI/DP,pcm=11 as /devices/pci0000:00/0000:00:03.0/0000:01:00.1/sound/card0/input12 [ 4.851898] input: HDA ATI HDMI HDMI/DP,pcm=10 as /devices/pci0000:00/0000:00:03.0/0000:01:00.1/sound/card0/input11 [ 4.851948] input: HDA ATI HDMI HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:03.0/0000:01:00.1/sound/card0/input10 [ 4.852014] input: HDA ATI HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/0000:01:00.1/sound/card0/input9 [ 4.852075] input: HDA ATI HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/0000:01:00.1/sound/card0/input8 [ 4.852129] input: HDA ATI HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/0000:01:00.1/sound/card0/input7 Does this mean monitor is plugged and graphic driver is requesting monitor to provide EDID ? 5.340797] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=0 [ 5.340812] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=1 [ 5.340825] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.340838] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.340844] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=5 Presence_Detect=1 ELD_Valid=0 [ 5.340857] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=5 Presence_Detect=1 ELD_Valid=1 [ 5.340870] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.340882] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.340885] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=7 Presence_Detect=1 ELD_Valid=0 [ 5.340898] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=7 Presence_Detect=1 ELD_Valid=1 [ 5.340911] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.340923] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.340926] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=9 Presence_Detect=1 ELD_Valid=0 [ 5.340939] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=9 Presence_Detect=1 ELD_Valid=1 [ 5.340952] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.340965] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.340967] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=11 Presence_Detect=1 ELD_Valid=0 [ 5.340980] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=11 Presence_Detect=1 ELD_Valid=1 [ 5.340993] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.341005] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.341008] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=13 Presence_Detect=1 ELD_Valid=0 [ 5.341021] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=13 Presence_Detect=1 ELD_Valid=1 [ 5.341034] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.341046] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.504525] [drm] Radeon Display Connectors [ 5.504527] [drm] Connector 0: [ 5.504528] [drm] DP-1 [ 5.504529] [drm] HPD4 [ 5.504530] [drm] DDC: 0x6530 0x6530 0x6534 0x6534 0x6538 0x6538 0x653c 0x653c [ 5.504531] [drm] Encoders: [ 5.504532] [drm] DFP1: INTERNAL_UNIPHY2 [ 5.504533] [drm] Connector 1: [ 5.504534] [drm] DP-2 [ 5.504535] [drm] HPD5 [ 5.504536] [drm] DDC: 0x6540 0x6540 0x6544 0x6544 0x6548 0x6548 0x654c 0x654c [ 5.504537] [drm] Encoders: [ 5.504538] [drm] DFP2: INTERNAL_UNIPHY2 [ 5.504538] [drm] Connector 2: [ 5.504539] [drm] HDMI-A-1 [ 5.504540] [drm] HPD1 [ 5.504541] [drm] DDC: 0x6550 0x6550 0x6554 0x6554 0x6558 0x6558 0x655c 0x655c [ 5.504542] [drm] Encoders: [ 5.504543] [drm] DFP3: INTERNAL_UNIPHY1 [ 5.504544] [drm] Connector 3: [ 5.504545] [drm] DVI-I-1 [ 5.504545] [drm] HPD6 [ 5.504547] [drm] DDC: 0x6580 0x6580 0x6584 0x6584 0x6588 0x6588 0x658c 0x658c [ 5.504547] [drm] Encoders: [ 5.504548] [drm] DFP4: INTERNAL_UNIPHY [ 5.504549] [drm] CRT1: INTERNAL_KLDSCP_DAC1 [ 5.640237] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=1 [ 5.640254] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.640267] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.640270] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=5 Presence_Detect=1 ELD_Valid=1 [ 5.640282] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.640295] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.640297] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=7 Presence_Detect=1 ELD_Valid=1 [ 5.640310] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.640322] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.640324] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=9 Presence_Detect=1 ELD_Valid=1 [ 5.640336] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.640349] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.640351] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=11 Presence_Detect=1 ELD_Valid=1 [ 5.640363] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.640376] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 5.640378] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=13 Presence_Detect=1 ELD_Valid=1 [ 5.640390] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 5.640403] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 7.906] (II) RADEON(0): EDID for output DisplayPort-0 [ 7.920] (II) RADEON(0): EDID for output DisplayPort-1 [ 7.983] (II) RADEON(0): EDID for output HDMI-0 [ 7.983] (II) RADEON(0): Manufacturer: DON Model: 26 Serial#: 16843009 [ 7.983] (II) RADEON(0): Year: 2011 Week: 0 [ 7.983] (II) RADEON(0): EDID Version: 1.3 [ 7.983] (II) RADEON(0): Monitor name: DENON-AVAMP [ 7.983] (II) RADEON(0): Number of EDID sections to follow: 1 [ 7.983] (II) RADEON(0): EDID (in hex): [ 7.983] (II) RADEON(0): 00ffffffffffff0011ee260001010101 [ 7.983] (II) RADEON(0): 00150103800000780adaffa3584aa229 [ 7.983] (II) RADEON(0): 17494b00000001010101010101010101 [ 7.983] (II) RADEON(0): 010101010101023a801871382d40582c [ 7.983] (II) RADEON(0): 4500ba882100001e011d8018711c1620 [ 7.983] (II) RADEON(0): 582c2500ba882100009e000000fc0044 [ 7.983] (II) RADEON(0): 454e4f4e2d4156414d500a20000000fd [ 7.983] (II) RADEON(0): 00173d0f440f000a202020202020010d [ 7.983] (II) RADEON(0): 020335714d9005200403020706010e23 [ 7.983] (II) RADEON(0): 0f24320f7f073d1ec01507505f7e0157 [ 7.983] (II) RADEON(0): 0600677e00835f0000e305030167030c [ 7.983] (II) RADEON(0): 001500b82d011d007251d01e206e2855 [ 7.983] (II) RADEON(0): 00ba882100001e8c0ad08a20e02d1010 [ 7.983] (II) RADEON(0): 3e9600ba88210000188c0ad08a20e02d [ 7.983] (II) RADEON(0): 10103e96000b88210000188c0aa01451 [ 7.983] (II) RADEON(0): f01600267c4300ba88210000980000e7 [ 8.003] (II) RADEON(0): EDID for output DVI-0 [ 8.003] (II) RADEON(0): Output DisplayPort-0 disconnected [ 8.003] (II) RADEON(0): Output DisplayPort-1 disconnected [ 8.003] (II) RADEON(0): Output HDMI-0 connected [ 8.003] (II) RADEON(0): Output DVI-0 disconnected Does this mean the graphic driver received EDID from DENON-AVAMP and pass ELD to audio driver ? [ 8.228364] ALSA patch_hdmi.c:972 HDMI hot plug event: Codec=0 Pin=3 Presence_Detect=0 ELD_Valid=1 [ 8.228380] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=1 [ 8.228393] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 8.228406] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 8.377652] type=1006 audit(1386558737.773:2): pid=515 uid=0 old auid=4294967295 new auid=1000 old ses=4294967295 new ses=1 res=1 [ 8.520795] ALSA hda_codec.c:1633 hda_codec_cleanup_stream: NID=0x2 [ 8.521113] ALSA hda_codec.c:1633 hda_codec_cleanup_stream: NID=0x2 [ 8.521434] ALSA hda_codec.c:1633 hda_codec_cleanup_stream: NID=0x2 [ 8.521727] ALSA hda_codec.c:1633 hda_codec_cleanup_stream: NID=0x2 [ 8.526625] ALSA patch_hdmi.c:1301 HDMI status: Codec=0 Pin=3 Presence_Detect=1 ELD_Valid=1 [ 8.526640] ALSA hda_eld.c:334 HDMI: ELD buf size is 0, force 128 [ 8.526652] ALSA hda_eld.c:351 HDMI: invalid ELD data byte 0 [ 8.261] (II) config/udev: Adding drm device (/dev/dri/card0) [ 8.262] (II) config/udev: Adding input device HDA ATI HDMI HDMI/DP,pcm=9 (/dev/input/event8) [ 8.262] (II) No input driver specified, ignoring this device. [ 8.262] (II) This device may have been added with another device file. [ 8.262] (II) config/udev: Adding input device HDA ATI HDMI HDMI/DP,pcm=10 (/dev/input/event7) [ 8.262] (II) No input driver specified, ignoring this device. [ 8.262] (II) This device may have been added with another device file. [ 8.262] (II) config/udev: Adding input device HDA ATI HDMI HDMI/DP,pcm=11 (/dev/input/event6) [ 8.262] (II) No input driver specified, ignoring this device. [ 8.262] (II) This device may have been added with another device file. [ 8.262] (II) config/udev: Adding input device HDA ATI HDMI HDMI/DP,pcm=3 (/dev/input/event11) [ 8.262] (II) No input driver specified, ignoring this device. [ 8.262] (II) This device may have been added with another device file. [ 8.262] (II) config/udev: Adding input device HDA ATI HDMI HDMI/DP,pcm=7 (/dev/input/event10) [ 8.262] (II) No input driver specified, ignoring this device. [ 8.262] (II) This device may have been added with another device file. [ 8.263] (II) config/udev: Adding input device HDA ATI HDMI HDMI/DP,pcm=8 (/dev/input/event9) [ 8.263] (II) No input driver specified, ignoring this device. [ 8.263] (II) This device may have been added with another device file. [ 8.263] (II) config/udev: Adding input device Logitech USB Receiver (/dev/input/event0)

5 6

[alsa-devel] [PATCH V2 1/3] ASoC: restructure dmaengine_pcm_request_chan_of()
by Stephen Warren 09 Dec '13

09 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> Restructure the internals of dmaengine_pcm_request_chan_of() as a loop over all channels to be allocated. This makes it easier to add logic that applies to all allocated channels, without having to duplicate that logic in each of the half-duplex/full-duplex paths. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2 (new as a separate patch): * Re-ordered patches, so that patch 3/3 in this series could be available in the ASoC tree without having to wait for the patches to the DMA tree the implement deferred probe during channel allocation. This whole series is a dependency for some patches that will go through the Tegra tree. If it could be applied to a topic branch in the ASoC tree that would be great. the overall dependency structure will be: a) This series. Can be applied on top of e.g. v3.13-rc1. b) Patches to the DMA tree, which implement deferred probe. Can be applied on top of e.g. v3.13-rc1. c) ASoC modifications to support deferred probe during DMA ch allocation This will depend on both (a) and (b). d) Tegra modifications that depend on (c). --- sound/soc/soc-generic-dmaengine-pcm.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index cbc9c96ce1f4..88b011b80058 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -288,21 +288,26 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, struct device *dev) { unsigned int i; + const char *name; if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || !dev->of_node) return; - if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { - pcm->chan[0] = dma_request_slave_channel(dev, "rx-tx"); - pcm->chan[1] = pcm->chan[0]; - } else { - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - pcm->chan[i] = dma_request_slave_channel(dev, - dmaengine_pcm_dma_channel_names[i]); - } + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; + i++) { + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + name = "rx-tx"; + else + name = dmaengine_pcm_dma_channel_names[i]; + pcm->chan[i] = dma_request_slave_channel(dev, name); + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + break; } + + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + pcm->chan[1] = pcm->chan[0]; } /** -- 1.8.1.5

3 7

Re: [alsa-devel] [PATCH v4] ASoC: tlv320aic3x: no mono controls 3007 model
by Mark Brown 09 Dec '13

09 Dec '13

On Thu, Dec 05, 2013 at 09:54:02AM +0100, Jan Weitzel wrote: > if codec driver is used for AIC3X_MODEL_3007 the mono iout controls overwrite > registers for class-d amplifier. > classd amplifier controls are only used for AIC3X_MODEL_3007. Applied, thanks.

1 0

[alsa-devel] [PATCH 1/2] ASoC: samsung: Provide helper for DMA init
by Mark Brown 09 Dec '13

09 Dec '13

From: Mark Brown <broonie(a)linaro.org> In preparation for using the dmaengine helpers in ASoC rather than the dmaengine wrappers for the Samsung API wrap the configuration of dma_data. The dmaengine code expects different data to that used by the legacy API. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 51 +++++++++++++++--------------------------------- sound/soc/samsung/dma.c | 8 ++++++++ sound/soc/samsung/dma.h | 3 +++ sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/pcm.c | 18 +++++++++-------- 5 files changed, 38 insertions(+), 44 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 350ba23..4a88e36 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = { .reset = s3c_ac97_cold_reset, }; -static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; +static int s3c_ac97_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in); + + return 0; +} + +static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in); + + return 0; +} + static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", @@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_dai_probe, .ops = &s3c_ac97_dai_ops, }, [S3C_AC97_DAI_MIC] = { @@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 1, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_mic_dai_probe, .ops = &s3c_ac97_mic_dai_ops, }, }; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index fe2748b..ee23194 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + snd_soc_dai_init_dma_data(dai, playback, capture); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0e86315..fb09a1c 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,6 +22,9 @@ struct s3c_dma_params { char *ch_name; }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture); int samsung_asoc_dma_platform_register(struct device *dev); void samsung_asoc_dma_platform_unregister(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4..67d9fa9 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -963,7 +963,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); - snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { other->addr = i2s->addr; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index e54256f..6a5e4bf 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); - /* Strictly check for sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_fmt = s3c_pcm_set_fmt, }; +static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture); + + return 0; +} + #define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 #define S3C_PCM_DAI_DECLARE \ .symmetric_rates = 1, \ + .probe = s3c_pcm_dai_probe, \ .ops = &s3c_pcm_dai_ops, \ .playback = { \ .channels_min = 2, \ -- 1.8.4.3

2 15

[alsa-devel] [PATCH v2 1/4] dt: Add bindings documentation for the ADI AXI-I2S controller
by Lars-Peter Clausen 09 Dec '13

09 Dec '13

This patch adds the devicetree documentation for the ADI AXI-SPDIF audio controller. The controller has: * One set of memory mapped register * Two clocks, one for the memory mapped register interface, one used as the audio reference clock * One DMA interface each for the transmit and receive data Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: devicetree(a)vger.kernel.org --- Changes since v1: * Use <&a ...>, <&b ...> instead of <&a ..., &b ...> in the examples --- .../devicetree/bindings/sound/adi,axi-i2s.txt | 31 ++++++++++++++++++++++ 1 file changed, 31 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,axi-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt new file mode 100644 index 0000000..5875ca4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt @@ -0,0 +1,31 @@ +ADI AXI-I2S controller + +Required properties: + - compatible : Must be "adi,axi-i2s-1.00.a" + - reg : Must contain I2S core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + The controller expects two clocks, the clock used for the AXI interface and + the clock used as the sampling rate reference clock sample. + - clock-names : "axi" for the clock to the AXI interface, "ref" for the sample + rate reference clock. + - dmas: Pairs of phandle and specifier for the DMA channels that are used by + the core. The core expects two dma channels, one for transmit and one for + receive. + - dma-names : "tx" for the transmit channel, "rx" for the receive channel. + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + + i2s: i2s@0x77600000 { + compatible = "adi,axi-i2s-1.00.a"; + reg = <0x77600000 0x1000>; + clocks = <&clk 15>, <&audio_clock>; + clock-names = "axi", "ref"; + dmas = <&ps7_dma 0>, <&ps7_dma 1>; + dma-names = "tx", "rx"; + }; -- 1.8.0

2 4

[alsa-devel] [PATCH] ASoC: tegra: fix uninitialized variables in set_fmt
by Stephen Warren 09 Dec '13

09 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> In tegra*_i2s_set_fmt(), in the (fmt == SND_SOC_DAIFMT_CBM_CFM) case, "val" is never assigned to, but left uninitialized. The other case does initialized it. Fix this by initializing val at the start of the function, and only ever ORing into it. Update the handling of "mask" so it works the same way for consistency. Update tegra20_spdif.c to use the same code-style for consistency, even though it doesn't happen to suffer from the same problem at present. Cc: <stable(a)vger.kernel.org> Fixes: 0f163546a772 ("ASoC: tegra: use regmap more directly") Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra20_i2s.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 10 +++++----- sound/soc/tegra/tegra30_i2s.c | 6 +++--- 3 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 364bf6a907e1..8c819f811470 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -74,7 +74,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -83,10 +83,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA20_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA20_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA20_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA20_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 08bc6931c7c7..8c7c1028e579 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -67,15 +67,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, { struct device *dev = dai->dev; struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; int ret, spdifclock; - mask = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; + mask |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - val = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; + val |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; break; default: return -EINVAL; diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 231a785b3921..02247fee1cf7 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -118,7 +118,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -127,10 +127,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA30_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA30_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA30_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA30_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; -- 1.8.1.5

3 2

[alsa-devel] [PATCH 0/3] dma: Indicate residue granularity in dma_slave_caps
by Lars-Peter Clausen 09 Dec '13

09 Dec '13

Hi, This patch series introduces a new field to the dma_slave_caps struct which exposes the granularity with which the reported residue is updated, updates the pl330 DMA driver to set this field and lets the generic dmaengine PCM audio driver check that field. The reason for introducing the field is that for audio we need to know the granularity with which the reported residue is updated. The uncertainty of the ALSA period pointer depends on the granularity of the reported residue. E.g. if residue is only updated once every period this means that the position in the buffer from/to which the hardware is currently reading/writing could be anywhere between the reported PCM pointer and the reported PCM pointer plus one period size (plus some extra margin for interrupt latency, etc). Certain algorithms (e.g. pulseaudio's timer based scheduling) only work correctly if the uncertainty of the PCM pointer is low enough. Exposing this information from the DMA driver makes it possible for applications to switch to alternative modes of operation which also work with a high granularity. The patch series introduces four levels of granularity, which I think will cover most DMA cores. I'd like some feedback though if anybody can think of a DMA controller for which none of the levels would be correct. The four levels are: * DESCRIPTOR: The DMA channel is only able to tell whether a descriptor has been completed or not, which means residue reporting is not supported by this channel. The residue field of the dma_tx_state field will always be 0. * SEGMENT: The DMA channel updates the residue field after each successfully completed segment of the transfer (For cyclic transfers this is after each period). This is typically implemented by having the hardware generate an interrupt after each transferred segment and then the driver updates the outstanding residue by the size of the segment. Another possibility is if the hardware supports SG and the segment descriptor has a field which gets set by the hardware after the segment has been completed. The driver then counts the number of segments which do not have the flag set to compute the residue. * BURST: The DMA channel updates the residue field after each transferred burst. This is typically only supported if the hardware has a progress register of some sort (E.g. a register with the current read/write address or a register with the amount of bursts/beats/bytes that have been transferred or still need to be transferred). * BYTE: Same as BURST but with a byte level granularity. I've only included the last one for completeness and I'm not sure if we really need it. Even if the hardware is able to report the amount of transferred data with a byte level granularity the lower bits will not carry too much meaning. When using burst transfers the lower bits will remain the same for longer periods and then rapidly increment during the burst transfer. The upper bits though will increment with the actual frequency with which the data is consumed. And that's what applications are typically interested in. - Lars Lars-Peter Clausen (3): dma: Indicate residue granularity in dma_slave_caps dma: pl330: Set residue_granularity ASoC: generic-dmaengine-pcm: Check DMA residue granularity drivers/dma/pl330.c | 1 + include/linux/dmaengine.h | 17 ++++++++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 37 +++++++++++++++++++++++++++++++++-- 3 files changed, 53 insertions(+), 2 deletions(-) -- 1.8.0

2 4

Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by Anssi Hannula 09 Dec '13

09 Dec '13

07.12.2013 09:18, Raymond Yau kirjoitti: > I just changed my RadeonHD 4670 for a 7870, and now when I type > "aplay -l" I get many HDMI codecs instead of just one like for the > previous > > > Do your card specification support Discrete Digital Multi-Point Audio > (DDMA) /?/ > > DDMA enables Radeon™HD 7700-7900 Series GPUs with the ability to output > multiple and independent audio streams simultaneously through digital > interfaces that support audio, such as DisplayPort and HDMI®. Each audio > stream can be multi-channel (up to 8 channels) AFAIK yes, the multi-hdmi-device AMD cards are "DDMA". [...] > monitor_present 1 > eld_valid 1 > monitor_name > connection_type HDMI > eld_version [0x2] CEA-861D or below > edid_version [0x0] no CEA EDID Timing Extension block present > manufacture_id 0x0 > product_id 0x0 > port_id 0x0 > support_hdcp 0 > support_ai 0 > audio_sync_delay 0 > speakers [0x5f] FL/FR LFE FC RL/RR RC RLC/RRC > sad_count 6 > sad0_coding_type [0x1] LPCM > sad0_channels 8 > sad0_rates [0x1ee0] 32000 44100 48000 88200 96000 176400 192000 > sad0_bits [0xe0000] 16 20 24 > sad1_coding_type [0x2] AC-3 > sad1_channels 6 > sad1_rates [0xe0] 32000 44100 48000 > sad1_max_bitrate 640000 > sad2_coding_type [0x7] DTS > sad2_channels 6 > sad2_rates [0x6c0] 44100 48000 88200 96000 > sad2_max_bitrate 1536000 > sad3_coding_type [0xa] E-AC-3/DD+ (Dolby Digital Plus) > sad3_channels 8 > sad3_rates [0xc0] 44100 48000 > sad4_coding_type [0xb] DTS-HD > sad4_channels 8 > sad4_rates [0x1ec0] 44100 48000 88200 96000 176400 192000 > sad5_coding_type [0xc] MLP (Dolby TrueHD) > sad5_channels 8 > sad5_rates [0x1ec0] 44100 48000 88200 96000 176400 192000 > monitor_present 1 > eld_valid 0 > monitor_present 1 > eld_valid 0 > monitor_present 1 > eld_valid 0 > monitor_present 1 > eld_valid 0 > monitor_present 1 > eld_valid 0 > > e.g. HDMI status message in function hdmi_present_sense in patch_hdmi.c > > need to find out why eld->monitor_present was set for all six complexes > > the value return by function atihdmi_pin_get_eld since it change > eld->eld_valid to false when it is negative When CONFIG_SND_DEBUG_VERBOSE is enabled, hdmi_present_sense() prints the actual hw Presence_Detect/ELD_Valid before eld->eld_valid is overridden to 0 (and hdmi_intrinsic_event() prints the hot plug event values). Alex, do you have information regarding why "Presence detect" (aka monitor_present above) seems to be set for all HDMI pins in the response of Pin Sense 0xF09 verb (or if it is even expected), and/or if we should use alternate means of getting jack status? CONFIG_SND_DEBUG_VERBOSE log might also be interesting, I wonder if eld_valid is also 1 for every pin or not (before ALSA forces 0). Also, I wonder if all monitor_present fields are 1 with fglrx driver as well, or when all displays are unplugged... [...] -- Anssi Hannula

2 1

[alsa-devel] [PATCH] ALSA: hda - Enable stereo mix as default for AD and VIA codecs
by Takashi Iwai 09 Dec '13

09 Dec '13

AD and VIA codecs had stereo mixer input enabled as default before moving to the generic parser, and people think the lack of such a regression. In this patch, the stereo mixer input is added back to the input selection if no auto-mic is available, and if it's not disabled explicitly via hint. This should satisfy most of demands, i.e. stereo mix on desktop machines like what it worked before, and it still keeps the new auto-mic feature on laptops. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 19 ++++++++++++++++++- sound/pci/hda/patch_analog.c | 1 + sound/pci/hda/patch_via.c | 1 + 3 files changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 058318de1af4..44be167d1cad 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3033,6 +3033,8 @@ static int parse_capture_source(struct hda_codec *codec, hda_nid_t pin, spec->imux_pins[imux->num_items] = pin; snd_hda_add_imux_item(imux, label, cfg_idx, NULL); imux_added = true; + if (spec->dyn_adc_switch) + spec->dyn_adc_idx[imux_idx] = c; } } @@ -3130,7 +3132,9 @@ static int create_input_ctls(struct hda_codec *codec) } } - if (mixer && spec->add_stereo_mix_input) { + /* add stereo mix when explicitly enabled via hint */ + if (mixer && spec->add_stereo_mix_input && + snd_hda_get_bool_hint(codec, "add_stereo_mix_input") > 0) { err = parse_capture_source(codec, mixer, CFG_IDX_MIX, num_adcs, "Stereo Mix", 0); if (err < 0) @@ -4403,6 +4407,19 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, if (err < 0) return err; + /* add stereo mix if available and not enabled yet */ + if (!spec->auto_mic && spec->mixer_nid && + spec->add_stereo_mix_input && + spec->input_mux.num_items > 1 && + snd_hda_get_bool_hint(codec, "add_stereo_mix_input") < 0) { + err = parse_capture_source(codec, spec->mixer_nid, + CFG_IDX_MIX, spec->num_all_adcs, + "Stereo Mix", 0); + if (err < 0) + return err; + } + + err = create_capture_mixers(codec); if (err < 0) return err; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 34d86ec5d3dd..05280033c302 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -194,6 +194,7 @@ static int ad198x_parse_auto_config(struct hda_codec *codec) codec->no_sticky_stream = 1; spec->gen.indep_hp = 1; + spec->gen.add_stereo_mix_input = 1; err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0); if (err < 0) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 0bc20ef5687a..f84195f3ea31 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -138,6 +138,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->gen.indep_hp = 1; spec->gen.keep_eapd_on = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; + spec->gen.add_stereo_mix_input = 1; return spec; } -- 1.8.5

1 0

[alsa-devel] [PATCH] ALSA: hda - Split verb definitions into sound/hda_verbs.h
by Takashi Iwai 09 Dec '13

09 Dec '13

Since there are more HD-audio compatible codecs, move the definitions of HD-audio verbs into common header location, include/sound, so that it can be included cleanly from other drivers than HD-audio driver. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_verbs.h | 554 ++++++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_codec.h | 547 +-------------------------------------------- 2 files changed, 555 insertions(+), 546 deletions(-) create mode 100644 include/sound/hda_verbs.h diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h new file mode 100644 index 000000000000..d0509db6d0ec --- /dev/null +++ b/include/sound/hda_verbs.h @@ -0,0 +1,554 @@ +/* + * HD-audio codec verbs + */ + +#ifndef __SOUND_HDA_VERBS_H +#define __SOUND_HDA_VERBS_H + +/* + * nodes + */ +#define AC_NODE_ROOT 0x00 + +/* + * function group types + */ +enum { + AC_GRP_AUDIO_FUNCTION = 0x01, + AC_GRP_MODEM_FUNCTION = 0x02, +}; + +/* + * widget types + */ +enum { + AC_WID_AUD_OUT, /* Audio Out */ + AC_WID_AUD_IN, /* Audio In */ + AC_WID_AUD_MIX, /* Audio Mixer */ + AC_WID_AUD_SEL, /* Audio Selector */ + AC_WID_PIN, /* Pin Complex */ + AC_WID_POWER, /* Power */ + AC_WID_VOL_KNB, /* Volume Knob */ + AC_WID_BEEP, /* Beep Generator */ + AC_WID_VENDOR = 0x0f /* Vendor specific */ +}; + +/* + * GET verbs + */ +#define AC_VERB_GET_STREAM_FORMAT 0x0a00 +#define AC_VERB_GET_AMP_GAIN_MUTE 0x0b00 +#define AC_VERB_GET_PROC_COEF 0x0c00 +#define AC_VERB_GET_COEF_INDEX 0x0d00 +#define AC_VERB_PARAMETERS 0x0f00 +#define AC_VERB_GET_CONNECT_SEL 0x0f01 +#define AC_VERB_GET_CONNECT_LIST 0x0f02 +#define AC_VERB_GET_PROC_STATE 0x0f03 +#define AC_VERB_GET_SDI_SELECT 0x0f04 +#define AC_VERB_GET_POWER_STATE 0x0f05 +#define AC_VERB_GET_CONV 0x0f06 +#define AC_VERB_GET_PIN_WIDGET_CONTROL 0x0f07 +#define AC_VERB_GET_UNSOLICITED_RESPONSE 0x0f08 +#define AC_VERB_GET_PIN_SENSE 0x0f09 +#define AC_VERB_GET_BEEP_CONTROL 0x0f0a +#define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c +#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ +#define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f +/* f10-f1a: GPIO */ +#define AC_VERB_GET_GPIO_DATA 0x0f15 +#define AC_VERB_GET_GPIO_MASK 0x0f16 +#define AC_VERB_GET_GPIO_DIRECTION 0x0f17 +#define AC_VERB_GET_GPIO_WAKE_MASK 0x0f18 +#define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 +#define AC_VERB_GET_GPIO_STICKY_MASK 0x0f1a +#define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c +/* f20: AFG/MFG */ +#define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 +#define AC_VERB_GET_CVT_CHAN_COUNT 0x0f2d +#define AC_VERB_GET_HDMI_DIP_SIZE 0x0f2e +#define AC_VERB_GET_HDMI_ELDD 0x0f2f +#define AC_VERB_GET_HDMI_DIP_INDEX 0x0f30 +#define AC_VERB_GET_HDMI_DIP_DATA 0x0f31 +#define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32 +#define AC_VERB_GET_HDMI_CP_CTRL 0x0f33 +#define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34 +#define AC_VERB_GET_DEVICE_SEL 0xf35 +#define AC_VERB_GET_DEVICE_LIST 0xf36 + +/* + * SET verbs + */ +#define AC_VERB_SET_STREAM_FORMAT 0x200 +#define AC_VERB_SET_AMP_GAIN_MUTE 0x300 +#define AC_VERB_SET_PROC_COEF 0x400 +#define AC_VERB_SET_COEF_INDEX 0x500 +#define AC_VERB_SET_CONNECT_SEL 0x701 +#define AC_VERB_SET_PROC_STATE 0x703 +#define AC_VERB_SET_SDI_SELECT 0x704 +#define AC_VERB_SET_POWER_STATE 0x705 +#define AC_VERB_SET_CHANNEL_STREAMID 0x706 +#define AC_VERB_SET_PIN_WIDGET_CONTROL 0x707 +#define AC_VERB_SET_UNSOLICITED_ENABLE 0x708 +#define AC_VERB_SET_PIN_SENSE 0x709 +#define AC_VERB_SET_BEEP_CONTROL 0x70a +#define AC_VERB_SET_EAPD_BTLENABLE 0x70c +#define AC_VERB_SET_DIGI_CONVERT_1 0x70d +#define AC_VERB_SET_DIGI_CONVERT_2 0x70e +#define AC_VERB_SET_VOLUME_KNOB_CONTROL 0x70f +#define AC_VERB_SET_GPIO_DATA 0x715 +#define AC_VERB_SET_GPIO_MASK 0x716 +#define AC_VERB_SET_GPIO_DIRECTION 0x717 +#define AC_VERB_SET_GPIO_WAKE_MASK 0x718 +#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 +#define AC_VERB_SET_GPIO_STICKY_MASK 0x71a +#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c +#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d +#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e +#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3 0x71f +#define AC_VERB_SET_EAPD 0x788 +#define AC_VERB_SET_CODEC_RESET 0x7ff +#define AC_VERB_SET_CVT_CHAN_COUNT 0x72d +#define AC_VERB_SET_HDMI_DIP_INDEX 0x730 +#define AC_VERB_SET_HDMI_DIP_DATA 0x731 +#define AC_VERB_SET_HDMI_DIP_XMIT 0x732 +#define AC_VERB_SET_HDMI_CP_CTRL 0x733 +#define AC_VERB_SET_HDMI_CHAN_SLOT 0x734 +#define AC_VERB_SET_DEVICE_SEL 0x735 + +/* + * Parameter IDs + */ +#define AC_PAR_VENDOR_ID 0x00 +#define AC_PAR_SUBSYSTEM_ID 0x01 +#define AC_PAR_REV_ID 0x02 +#define AC_PAR_NODE_COUNT 0x04 +#define AC_PAR_FUNCTION_TYPE 0x05 +#define AC_PAR_AUDIO_FG_CAP 0x08 +#define AC_PAR_AUDIO_WIDGET_CAP 0x09 +#define AC_PAR_PCM 0x0a +#define AC_PAR_STREAM 0x0b +#define AC_PAR_PIN_CAP 0x0c +#define AC_PAR_AMP_IN_CAP 0x0d +#define AC_PAR_CONNLIST_LEN 0x0e +#define AC_PAR_POWER_STATE 0x0f +#define AC_PAR_PROC_CAP 0x10 +#define AC_PAR_GPIO_CAP 0x11 +#define AC_PAR_AMP_OUT_CAP 0x12 +#define AC_PAR_VOL_KNB_CAP 0x13 +#define AC_PAR_DEVLIST_LEN 0x15 +#define AC_PAR_HDMI_LPCM_CAP 0x20 + +/* + * AC_VERB_PARAMETERS results (32bit) + */ + +/* Function Group Type */ +#define AC_FGT_TYPE (0xff<<0) +#define AC_FGT_TYPE_SHIFT 0 +#define AC_FGT_UNSOL_CAP (1<<8) + +/* Audio Function Group Capabilities */ +#define AC_AFG_OUT_DELAY (0xf<<0) +#define AC_AFG_IN_DELAY (0xf<<8) +#define AC_AFG_BEEP_GEN (1<<16) + +/* Audio Widget Capabilities */ +#define AC_WCAP_STEREO (1<<0) /* stereo I/O */ +#define AC_WCAP_IN_AMP (1<<1) /* AMP-in present */ +#define AC_WCAP_OUT_AMP (1<<2) /* AMP-out present */ +#define AC_WCAP_AMP_OVRD (1<<3) /* AMP-parameter override */ +#define AC_WCAP_FORMAT_OVRD (1<<4) /* format override */ +#define AC_WCAP_STRIPE (1<<5) /* stripe */ +#define AC_WCAP_PROC_WID (1<<6) /* Proc Widget */ +#define AC_WCAP_UNSOL_CAP (1<<7) /* Unsol capable */ +#define AC_WCAP_CONN_LIST (1<<8) /* connection list */ +#define AC_WCAP_DIGITAL (1<<9) /* digital I/O */ +#define AC_WCAP_POWER (1<<10) /* power control */ +#define AC_WCAP_LR_SWAP (1<<11) /* L/R swap */ +#define AC_WCAP_CP_CAPS (1<<12) /* content protection */ +#define AC_WCAP_CHAN_CNT_EXT (7<<13) /* channel count ext */ +#define AC_WCAP_DELAY (0xf<<16) +#define AC_WCAP_DELAY_SHIFT 16 +#define AC_WCAP_TYPE (0xf<<20) +#define AC_WCAP_TYPE_SHIFT 20 + +/* supported PCM rates and bits */ +#define AC_SUPPCM_RATES (0xfff << 0) +#define AC_SUPPCM_BITS_8 (1<<16) +#define AC_SUPPCM_BITS_16 (1<<17) +#define AC_SUPPCM_BITS_20 (1<<18) +#define AC_SUPPCM_BITS_24 (1<<19) +#define AC_SUPPCM_BITS_32 (1<<20) + +/* supported PCM stream format */ +#define AC_SUPFMT_PCM (1<<0) +#define AC_SUPFMT_FLOAT32 (1<<1) +#define AC_SUPFMT_AC3 (1<<2) + +/* GP I/O count */ +#define AC_GPIO_IO_COUNT (0xff<<0) +#define AC_GPIO_O_COUNT (0xff<<8) +#define AC_GPIO_O_COUNT_SHIFT 8 +#define AC_GPIO_I_COUNT (0xff<<16) +#define AC_GPIO_I_COUNT_SHIFT 16 +#define AC_GPIO_UNSOLICITED (1<<30) +#define AC_GPIO_WAKE (1<<31) + +/* Converter stream, channel */ +#define AC_CONV_CHANNEL (0xf<<0) +#define AC_CONV_STREAM (0xf<<4) +#define AC_CONV_STREAM_SHIFT 4 + +/* Input converter SDI select */ +#define AC_SDI_SELECT (0xf<<0) + +/* stream format id */ +#define AC_FMT_CHAN_SHIFT 0 +#define AC_FMT_CHAN_MASK (0x0f << 0) +#define AC_FMT_BITS_SHIFT 4 +#define AC_FMT_BITS_MASK (7 << 4) +#define AC_FMT_BITS_8 (0 << 4) +#define AC_FMT_BITS_16 (1 << 4) +#define AC_FMT_BITS_20 (2 << 4) +#define AC_FMT_BITS_24 (3 << 4) +#define AC_FMT_BITS_32 (4 << 4) +#define AC_FMT_DIV_SHIFT 8 +#define AC_FMT_DIV_MASK (7 << 8) +#define AC_FMT_MULT_SHIFT 11 +#define AC_FMT_MULT_MASK (7 << 11) +#define AC_FMT_BASE_SHIFT 14 +#define AC_FMT_BASE_48K (0 << 14) +#define AC_FMT_BASE_44K (1 << 14) +#define AC_FMT_TYPE_SHIFT 15 +#define AC_FMT_TYPE_PCM (0 << 15) +#define AC_FMT_TYPE_NON_PCM (1 << 15) + +/* Unsolicited response control */ +#define AC_UNSOL_TAG (0x3f<<0) +#define AC_UNSOL_ENABLED (1<<7) +#define AC_USRSP_EN AC_UNSOL_ENABLED + +/* Unsolicited responses */ +#define AC_UNSOL_RES_TAG (0x3f<<26) +#define AC_UNSOL_RES_TAG_SHIFT 26 +#define AC_UNSOL_RES_SUBTAG (0x1f<<21) +#define AC_UNSOL_RES_SUBTAG_SHIFT 21 +#define AC_UNSOL_RES_DE (0x3f<<15) /* Device Entry + * (for DP1.2 MST) + */ +#define AC_UNSOL_RES_DE_SHIFT 15 +#define AC_UNSOL_RES_IA (1<<2) /* Inactive (for DP1.2 MST) */ +#define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */ +#define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */ +#define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */ +#define AC_UNSOL_RES_CP_READY (1<<0) /* content protection */ + +/* Pin widget capabilies */ +#define AC_PINCAP_IMP_SENSE (1<<0) /* impedance sense capable */ +#define AC_PINCAP_TRIG_REQ (1<<1) /* trigger required */ +#define AC_PINCAP_PRES_DETECT (1<<2) /* presence detect capable */ +#define AC_PINCAP_HP_DRV (1<<3) /* headphone drive capable */ +#define AC_PINCAP_OUT (1<<4) /* output capable */ +#define AC_PINCAP_IN (1<<5) /* input capable */ +#define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */ +/* Note: This LR_SWAP pincap is defined in the Realtek ALC883 specification, + * but is marked reserved in the Intel HDA specification. + */ +#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */ +/* Note: The same bit as LR_SWAP is newly defined as HDMI capability + * in HD-audio specification + */ +#define AC_PINCAP_HDMI (1<<7) /* HDMI pin */ +#define AC_PINCAP_DP (1<<24) /* DisplayPort pin, can + * coexist with AC_PINCAP_HDMI + */ +#define AC_PINCAP_VREF (0x37<<8) +#define AC_PINCAP_VREF_SHIFT 8 +#define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ +#define AC_PINCAP_HBR (1<<27) /* High Bit Rate */ +/* Vref status (used in pin cap) */ +#define AC_PINCAP_VREF_HIZ (1<<0) /* Hi-Z */ +#define AC_PINCAP_VREF_50 (1<<1) /* 50% */ +#define AC_PINCAP_VREF_GRD (1<<2) /* ground */ +#define AC_PINCAP_VREF_80 (1<<4) /* 80% */ +#define AC_PINCAP_VREF_100 (1<<5) /* 100% */ + +/* Amplifier capabilities */ +#define AC_AMPCAP_OFFSET (0x7f<<0) /* 0dB offset */ +#define AC_AMPCAP_OFFSET_SHIFT 0 +#define AC_AMPCAP_NUM_STEPS (0x7f<<8) /* number of steps */ +#define AC_AMPCAP_NUM_STEPS_SHIFT 8 +#define AC_AMPCAP_STEP_SIZE (0x7f<<16) /* step size 0-32dB + * in 0.25dB + */ +#define AC_AMPCAP_STEP_SIZE_SHIFT 16 +#define AC_AMPCAP_MUTE (1<<31) /* mute capable */ +#define AC_AMPCAP_MUTE_SHIFT 31 + +/* driver-specific amp-caps: using bits 24-30 */ +#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */ + +/* Connection list */ +#define AC_CLIST_LENGTH (0x7f<<0) +#define AC_CLIST_LONG (1<<7) + +/* Supported power status */ +#define AC_PWRST_D0SUP (1<<0) +#define AC_PWRST_D1SUP (1<<1) +#define AC_PWRST_D2SUP (1<<2) +#define AC_PWRST_D3SUP (1<<3) +#define AC_PWRST_D3COLDSUP (1<<4) +#define AC_PWRST_S3D3COLDSUP (1<<29) +#define AC_PWRST_CLKSTOP (1<<30) +#define AC_PWRST_EPSS (1U<<31) + +/* Power state values */ +#define AC_PWRST_SETTING (0xf<<0) +#define AC_PWRST_ACTUAL (0xf<<4) +#define AC_PWRST_ACTUAL_SHIFT 4 +#define AC_PWRST_D0 0x00 +#define AC_PWRST_D1 0x01 +#define AC_PWRST_D2 0x02 +#define AC_PWRST_D3 0x03 +#define AC_PWRST_ERROR (1<<8) +#define AC_PWRST_CLK_STOP_OK (1<<9) +#define AC_PWRST_SETTING_RESET (1<<10) + +/* Processing capabilies */ +#define AC_PCAP_BENIGN (1<<0) +#define AC_PCAP_NUM_COEF (0xff<<8) +#define AC_PCAP_NUM_COEF_SHIFT 8 + +/* Volume knobs capabilities */ +#define AC_KNBCAP_NUM_STEPS (0x7f<<0) +#define AC_KNBCAP_DELTA (1<<7) + +/* HDMI LPCM capabilities */ +#define AC_LPCMCAP_48K_CP_CHNS (0x0f<<0) /* max channels w/ CP-on */ +#define AC_LPCMCAP_48K_NO_CHNS (0x0f<<4) /* max channels w/o CP-on */ +#define AC_LPCMCAP_48K_20BIT (1<<8) /* 20b bitrate supported */ +#define AC_LPCMCAP_48K_24BIT (1<<9) /* 24b bitrate supported */ +#define AC_LPCMCAP_96K_CP_CHNS (0x0f<<10) /* max channels w/ CP-on */ +#define AC_LPCMCAP_96K_NO_CHNS (0x0f<<14) /* max channels w/o CP-on */ +#define AC_LPCMCAP_96K_20BIT (1<<18) /* 20b bitrate supported */ +#define AC_LPCMCAP_96K_24BIT (1<<19) /* 24b bitrate supported */ +#define AC_LPCMCAP_192K_CP_CHNS (0x0f<<20) /* max channels w/ CP-on */ +#define AC_LPCMCAP_192K_NO_CHNS (0x0f<<24) /* max channels w/o CP-on */ +#define AC_LPCMCAP_192K_20BIT (1<<28) /* 20b bitrate supported */ +#define AC_LPCMCAP_192K_24BIT (1<<29) /* 24b bitrate supported */ +#define AC_LPCMCAP_44K (1<<30) /* 44.1kHz support */ +#define AC_LPCMCAP_44K_MS (1<<31) /* 44.1kHz-multiplies support */ + +/* Display pin's device list length */ +#define AC_DEV_LIST_LEN_MASK 0x3f +#define AC_MAX_DEV_LIST_LEN 64 + +/* + * Control Parameters + */ + +/* Amp gain/mute */ +#define AC_AMP_MUTE (1<<7) +#define AC_AMP_GAIN (0x7f) +#define AC_AMP_GET_INDEX (0xf<<0) + +#define AC_AMP_GET_LEFT (1<<13) +#define AC_AMP_GET_RIGHT (0<<13) +#define AC_AMP_GET_OUTPUT (1<<15) +#define AC_AMP_GET_INPUT (0<<15) + +#define AC_AMP_SET_INDEX (0xf<<8) +#define AC_AMP_SET_INDEX_SHIFT 8 +#define AC_AMP_SET_RIGHT (1<<12) +#define AC_AMP_SET_LEFT (1<<13) +#define AC_AMP_SET_INPUT (1<<14) +#define AC_AMP_SET_OUTPUT (1<<15) + +/* DIGITAL1 bits */ +#define AC_DIG1_ENABLE (1<<0) +#define AC_DIG1_V (1<<1) +#define AC_DIG1_VCFG (1<<2) +#define AC_DIG1_EMPHASIS (1<<3) +#define AC_DIG1_COPYRIGHT (1<<4) +#define AC_DIG1_NONAUDIO (1<<5) +#define AC_DIG1_PROFESSIONAL (1<<6) +#define AC_DIG1_LEVEL (1<<7) + +/* DIGITAL2 bits */ +#define AC_DIG2_CC (0x7f<<0) + +/* DIGITAL3 bits */ +#define AC_DIG3_ICT (0xf<<0) +#define AC_DIG3_KAE (1<<7) + +/* Pin widget control - 8bit */ +#define AC_PINCTL_EPT (0x3<<0) +#define AC_PINCTL_EPT_NATIVE 0 +#define AC_PINCTL_EPT_HBR 3 +#define AC_PINCTL_VREFEN (0x7<<0) +#define AC_PINCTL_VREF_HIZ 0 /* Hi-Z */ +#define AC_PINCTL_VREF_50 1 /* 50% */ +#define AC_PINCTL_VREF_GRD 2 /* ground */ +#define AC_PINCTL_VREF_80 4 /* 80% */ +#define AC_PINCTL_VREF_100 5 /* 100% */ +#define AC_PINCTL_IN_EN (1<<5) +#define AC_PINCTL_OUT_EN (1<<6) +#define AC_PINCTL_HP_EN (1<<7) + +/* Pin sense - 32bit */ +#define AC_PINSENSE_IMPEDANCE_MASK (0x7fffffff) +#define AC_PINSENSE_PRESENCE (1<<31) +#define AC_PINSENSE_ELDV (1<<30) /* ELD valid (HDMI) */ + +/* EAPD/BTL enable - 32bit */ +#define AC_EAPDBTL_BALANCED (1<<0) +#define AC_EAPDBTL_EAPD (1<<1) +#define AC_EAPDBTL_LR_SWAP (1<<2) + +/* HDMI ELD data */ +#define AC_ELDD_ELD_VALID (1<<31) +#define AC_ELDD_ELD_DATA 0xff + +/* HDMI DIP size */ +#define AC_DIPSIZE_ELD_BUF (1<<3) /* ELD buf size of packet size */ +#define AC_DIPSIZE_PACK_IDX (0x07<<0) /* packet index */ + +/* HDMI DIP index */ +#define AC_DIPIDX_PACK_IDX (0x07<<5) /* packet idnex */ +#define AC_DIPIDX_BYTE_IDX (0x1f<<0) /* byte index */ + +/* HDMI DIP xmit (transmit) control */ +#define AC_DIPXMIT_MASK (0x3<<6) +#define AC_DIPXMIT_DISABLE (0x0<<6) /* disable xmit */ +#define AC_DIPXMIT_ONCE (0x2<<6) /* xmit once then disable */ +#define AC_DIPXMIT_BEST (0x3<<6) /* best effort */ + +/* HDMI content protection (CP) control */ +#define AC_CPCTRL_CES (1<<9) /* current encryption state */ +#define AC_CPCTRL_READY (1<<8) /* ready bit */ +#define AC_CPCTRL_SUBTAG (0x1f<<3) /* subtag for unsol-resp */ +#define AC_CPCTRL_STATE (3<<0) /* current CP request state */ + +/* Converter channel <-> HDMI slot mapping */ +#define AC_CVTMAP_HDMI_SLOT (0xf<<0) /* HDMI slot number */ +#define AC_CVTMAP_CHAN (0xf<<4) /* converter channel number */ + +/* configuration default - 32bit */ +#define AC_DEFCFG_SEQUENCE (0xf<<0) +#define AC_DEFCFG_DEF_ASSOC (0xf<<4) +#define AC_DEFCFG_ASSOC_SHIFT 4 +#define AC_DEFCFG_MISC (0xf<<8) +#define AC_DEFCFG_MISC_SHIFT 8 +#define AC_DEFCFG_MISC_NO_PRESENCE (1<<0) +#define AC_DEFCFG_COLOR (0xf<<12) +#define AC_DEFCFG_COLOR_SHIFT 12 +#define AC_DEFCFG_CONN_TYPE (0xf<<16) +#define AC_DEFCFG_CONN_TYPE_SHIFT 16 +#define AC_DEFCFG_DEVICE (0xf<<20) +#define AC_DEFCFG_DEVICE_SHIFT 20 +#define AC_DEFCFG_LOCATION (0x3f<<24) +#define AC_DEFCFG_LOCATION_SHIFT 24 +#define AC_DEFCFG_PORT_CONN (0x3<<30) +#define AC_DEFCFG_PORT_CONN_SHIFT 30 + +/* Display pin's device list entry */ +#define AC_DE_PD (1<<0) +#define AC_DE_ELDV (1<<1) +#define AC_DE_IA (1<<2) + +/* device device types (0x0-0xf) */ +enum { + AC_JACK_LINE_OUT, + AC_JACK_SPEAKER, + AC_JACK_HP_OUT, + AC_JACK_CD, + AC_JACK_SPDIF_OUT, + AC_JACK_DIG_OTHER_OUT, + AC_JACK_MODEM_LINE_SIDE, + AC_JACK_MODEM_HAND_SIDE, + AC_JACK_LINE_IN, + AC_JACK_AUX, + AC_JACK_MIC_IN, + AC_JACK_TELEPHONY, + AC_JACK_SPDIF_IN, + AC_JACK_DIG_OTHER_IN, + AC_JACK_OTHER = 0xf, +}; + +/* jack connection types (0x0-0xf) */ +enum { + AC_JACK_CONN_UNKNOWN, + AC_JACK_CONN_1_8, + AC_JACK_CONN_1_4, + AC_JACK_CONN_ATAPI, + AC_JACK_CONN_RCA, + AC_JACK_CONN_OPTICAL, + AC_JACK_CONN_OTHER_DIGITAL, + AC_JACK_CONN_OTHER_ANALOG, + AC_JACK_CONN_DIN, + AC_JACK_CONN_XLR, + AC_JACK_CONN_RJ11, + AC_JACK_CONN_COMB, + AC_JACK_CONN_OTHER = 0xf, +}; + +/* jack colors (0x0-0xf) */ +enum { + AC_JACK_COLOR_UNKNOWN, + AC_JACK_COLOR_BLACK, + AC_JACK_COLOR_GREY, + AC_JACK_COLOR_BLUE, + AC_JACK_COLOR_GREEN, + AC_JACK_COLOR_RED, + AC_JACK_COLOR_ORANGE, + AC_JACK_COLOR_YELLOW, + AC_JACK_COLOR_PURPLE, + AC_JACK_COLOR_PINK, + AC_JACK_COLOR_WHITE = 0xe, + AC_JACK_COLOR_OTHER, +}; + +/* Jack location (0x0-0x3f) */ +/* common case */ +enum { + AC_JACK_LOC_NONE, + AC_JACK_LOC_REAR, + AC_JACK_LOC_FRONT, + AC_JACK_LOC_LEFT, + AC_JACK_LOC_RIGHT, + AC_JACK_LOC_TOP, + AC_JACK_LOC_BOTTOM, +}; +/* bits 4-5 */ +enum { + AC_JACK_LOC_EXTERNAL = 0x00, + AC_JACK_LOC_INTERNAL = 0x10, + AC_JACK_LOC_SEPARATE = 0x20, + AC_JACK_LOC_OTHER = 0x30, +}; +enum { + /* external on primary chasis */ + AC_JACK_LOC_REAR_PANEL = 0x07, + AC_JACK_LOC_DRIVE_BAY, + /* internal */ + AC_JACK_LOC_RISER = 0x17, + AC_JACK_LOC_HDMI, + AC_JACK_LOC_ATAPI, + /* others */ + AC_JACK_LOC_MOBILE_IN = 0x37, + AC_JACK_LOC_MOBILE_OUT, +}; + +/* Port connectivity (0-3) */ +enum { + AC_JACK_PORT_COMPLEX, + AC_JACK_PORT_NONE, + AC_JACK_PORT_FIXED, + AC_JACK_PORT_BOTH, +}; + +/* max. codec address */ +#define HDA_MAX_CODEC_ADDRESS 0x0f + +#endif /* __SOUND_HDA_VERBS_H */ diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 260b190adf74..3c484c27048b 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -25,552 +25,7 @@ #include <sound/control.h> #include <sound/pcm.h> #include <sound/hwdep.h> - -/* - * nodes - */ -#define AC_NODE_ROOT 0x00 - -/* - * function group types - */ -enum { - AC_GRP_AUDIO_FUNCTION = 0x01, - AC_GRP_MODEM_FUNCTION = 0x02, -}; - -/* - * widget types - */ -enum { - AC_WID_AUD_OUT, /* Audio Out */ - AC_WID_AUD_IN, /* Audio In */ - AC_WID_AUD_MIX, /* Audio Mixer */ - AC_WID_AUD_SEL, /* Audio Selector */ - AC_WID_PIN, /* Pin Complex */ - AC_WID_POWER, /* Power */ - AC_WID_VOL_KNB, /* Volume Knob */ - AC_WID_BEEP, /* Beep Generator */ - AC_WID_VENDOR = 0x0f /* Vendor specific */ -}; - -/* - * GET verbs - */ -#define AC_VERB_GET_STREAM_FORMAT 0x0a00 -#define AC_VERB_GET_AMP_GAIN_MUTE 0x0b00 -#define AC_VERB_GET_PROC_COEF 0x0c00 -#define AC_VERB_GET_COEF_INDEX 0x0d00 -#define AC_VERB_PARAMETERS 0x0f00 -#define AC_VERB_GET_CONNECT_SEL 0x0f01 -#define AC_VERB_GET_CONNECT_LIST 0x0f02 -#define AC_VERB_GET_PROC_STATE 0x0f03 -#define AC_VERB_GET_SDI_SELECT 0x0f04 -#define AC_VERB_GET_POWER_STATE 0x0f05 -#define AC_VERB_GET_CONV 0x0f06 -#define AC_VERB_GET_PIN_WIDGET_CONTROL 0x0f07 -#define AC_VERB_GET_UNSOLICITED_RESPONSE 0x0f08 -#define AC_VERB_GET_PIN_SENSE 0x0f09 -#define AC_VERB_GET_BEEP_CONTROL 0x0f0a -#define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c -#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d -#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ -#define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f -/* f10-f1a: GPIO */ -#define AC_VERB_GET_GPIO_DATA 0x0f15 -#define AC_VERB_GET_GPIO_MASK 0x0f16 -#define AC_VERB_GET_GPIO_DIRECTION 0x0f17 -#define AC_VERB_GET_GPIO_WAKE_MASK 0x0f18 -#define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 -#define AC_VERB_GET_GPIO_STICKY_MASK 0x0f1a -#define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c -/* f20: AFG/MFG */ -#define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 -#define AC_VERB_GET_CVT_CHAN_COUNT 0x0f2d -#define AC_VERB_GET_HDMI_DIP_SIZE 0x0f2e -#define AC_VERB_GET_HDMI_ELDD 0x0f2f -#define AC_VERB_GET_HDMI_DIP_INDEX 0x0f30 -#define AC_VERB_GET_HDMI_DIP_DATA 0x0f31 -#define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32 -#define AC_VERB_GET_HDMI_CP_CTRL 0x0f33 -#define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34 -#define AC_VERB_GET_DEVICE_SEL 0xf35 -#define AC_VERB_GET_DEVICE_LIST 0xf36 - -/* - * SET verbs - */ -#define AC_VERB_SET_STREAM_FORMAT 0x200 -#define AC_VERB_SET_AMP_GAIN_MUTE 0x300 -#define AC_VERB_SET_PROC_COEF 0x400 -#define AC_VERB_SET_COEF_INDEX 0x500 -#define AC_VERB_SET_CONNECT_SEL 0x701 -#define AC_VERB_SET_PROC_STATE 0x703 -#define AC_VERB_SET_SDI_SELECT 0x704 -#define AC_VERB_SET_POWER_STATE 0x705 -#define AC_VERB_SET_CHANNEL_STREAMID 0x706 -#define AC_VERB_SET_PIN_WIDGET_CONTROL 0x707 -#define AC_VERB_SET_UNSOLICITED_ENABLE 0x708 -#define AC_VERB_SET_PIN_SENSE 0x709 -#define AC_VERB_SET_BEEP_CONTROL 0x70a -#define AC_VERB_SET_EAPD_BTLENABLE 0x70c -#define AC_VERB_SET_DIGI_CONVERT_1 0x70d -#define AC_VERB_SET_DIGI_CONVERT_2 0x70e -#define AC_VERB_SET_VOLUME_KNOB_CONTROL 0x70f -#define AC_VERB_SET_GPIO_DATA 0x715 -#define AC_VERB_SET_GPIO_MASK 0x716 -#define AC_VERB_SET_GPIO_DIRECTION 0x717 -#define AC_VERB_SET_GPIO_WAKE_MASK 0x718 -#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 -#define AC_VERB_SET_GPIO_STICKY_MASK 0x71a -#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c -#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d -#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e -#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3 0x71f -#define AC_VERB_SET_EAPD 0x788 -#define AC_VERB_SET_CODEC_RESET 0x7ff -#define AC_VERB_SET_CVT_CHAN_COUNT 0x72d -#define AC_VERB_SET_HDMI_DIP_INDEX 0x730 -#define AC_VERB_SET_HDMI_DIP_DATA 0x731 -#define AC_VERB_SET_HDMI_DIP_XMIT 0x732 -#define AC_VERB_SET_HDMI_CP_CTRL 0x733 -#define AC_VERB_SET_HDMI_CHAN_SLOT 0x734 -#define AC_VERB_SET_DEVICE_SEL 0x735 - -/* - * Parameter IDs - */ -#define AC_PAR_VENDOR_ID 0x00 -#define AC_PAR_SUBSYSTEM_ID 0x01 -#define AC_PAR_REV_ID 0x02 -#define AC_PAR_NODE_COUNT 0x04 -#define AC_PAR_FUNCTION_TYPE 0x05 -#define AC_PAR_AUDIO_FG_CAP 0x08 -#define AC_PAR_AUDIO_WIDGET_CAP 0x09 -#define AC_PAR_PCM 0x0a -#define AC_PAR_STREAM 0x0b -#define AC_PAR_PIN_CAP 0x0c -#define AC_PAR_AMP_IN_CAP 0x0d -#define AC_PAR_CONNLIST_LEN 0x0e -#define AC_PAR_POWER_STATE 0x0f -#define AC_PAR_PROC_CAP 0x10 -#define AC_PAR_GPIO_CAP 0x11 -#define AC_PAR_AMP_OUT_CAP 0x12 -#define AC_PAR_VOL_KNB_CAP 0x13 -#define AC_PAR_DEVLIST_LEN 0x15 -#define AC_PAR_HDMI_LPCM_CAP 0x20 - -/* - * AC_VERB_PARAMETERS results (32bit) - */ - -/* Function Group Type */ -#define AC_FGT_TYPE (0xff<<0) -#define AC_FGT_TYPE_SHIFT 0 -#define AC_FGT_UNSOL_CAP (1<<8) - -/* Audio Function Group Capabilities */ -#define AC_AFG_OUT_DELAY (0xf<<0) -#define AC_AFG_IN_DELAY (0xf<<8) -#define AC_AFG_BEEP_GEN (1<<16) - -/* Audio Widget Capabilities */ -#define AC_WCAP_STEREO (1<<0) /* stereo I/O */ -#define AC_WCAP_IN_AMP (1<<1) /* AMP-in present */ -#define AC_WCAP_OUT_AMP (1<<2) /* AMP-out present */ -#define AC_WCAP_AMP_OVRD (1<<3) /* AMP-parameter override */ -#define AC_WCAP_FORMAT_OVRD (1<<4) /* format override */ -#define AC_WCAP_STRIPE (1<<5) /* stripe */ -#define AC_WCAP_PROC_WID (1<<6) /* Proc Widget */ -#define AC_WCAP_UNSOL_CAP (1<<7) /* Unsol capable */ -#define AC_WCAP_CONN_LIST (1<<8) /* connection list */ -#define AC_WCAP_DIGITAL (1<<9) /* digital I/O */ -#define AC_WCAP_POWER (1<<10) /* power control */ -#define AC_WCAP_LR_SWAP (1<<11) /* L/R swap */ -#define AC_WCAP_CP_CAPS (1<<12) /* content protection */ -#define AC_WCAP_CHAN_CNT_EXT (7<<13) /* channel count ext */ -#define AC_WCAP_DELAY (0xf<<16) -#define AC_WCAP_DELAY_SHIFT 16 -#define AC_WCAP_TYPE (0xf<<20) -#define AC_WCAP_TYPE_SHIFT 20 - -/* supported PCM rates and bits */ -#define AC_SUPPCM_RATES (0xfff << 0) -#define AC_SUPPCM_BITS_8 (1<<16) -#define AC_SUPPCM_BITS_16 (1<<17) -#define AC_SUPPCM_BITS_20 (1<<18) -#define AC_SUPPCM_BITS_24 (1<<19) -#define AC_SUPPCM_BITS_32 (1<<20) - -/* supported PCM stream format */ -#define AC_SUPFMT_PCM (1<<0) -#define AC_SUPFMT_FLOAT32 (1<<1) -#define AC_SUPFMT_AC3 (1<<2) - -/* GP I/O count */ -#define AC_GPIO_IO_COUNT (0xff<<0) -#define AC_GPIO_O_COUNT (0xff<<8) -#define AC_GPIO_O_COUNT_SHIFT 8 -#define AC_GPIO_I_COUNT (0xff<<16) -#define AC_GPIO_I_COUNT_SHIFT 16 -#define AC_GPIO_UNSOLICITED (1<<30) -#define AC_GPIO_WAKE (1<<31) - -/* Converter stream, channel */ -#define AC_CONV_CHANNEL (0xf<<0) -#define AC_CONV_STREAM (0xf<<4) -#define AC_CONV_STREAM_SHIFT 4 - -/* Input converter SDI select */ -#define AC_SDI_SELECT (0xf<<0) - -/* stream format id */ -#define AC_FMT_CHAN_SHIFT 0 -#define AC_FMT_CHAN_MASK (0x0f << 0) -#define AC_FMT_BITS_SHIFT 4 -#define AC_FMT_BITS_MASK (7 << 4) -#define AC_FMT_BITS_8 (0 << 4) -#define AC_FMT_BITS_16 (1 << 4) -#define AC_FMT_BITS_20 (2 << 4) -#define AC_FMT_BITS_24 (3 << 4) -#define AC_FMT_BITS_32 (4 << 4) -#define AC_FMT_DIV_SHIFT 8 -#define AC_FMT_DIV_MASK (7 << 8) -#define AC_FMT_MULT_SHIFT 11 -#define AC_FMT_MULT_MASK (7 << 11) -#define AC_FMT_BASE_SHIFT 14 -#define AC_FMT_BASE_48K (0 << 14) -#define AC_FMT_BASE_44K (1 << 14) -#define AC_FMT_TYPE_SHIFT 15 -#define AC_FMT_TYPE_PCM (0 << 15) -#define AC_FMT_TYPE_NON_PCM (1 << 15) - -/* Unsolicited response control */ -#define AC_UNSOL_TAG (0x3f<<0) -#define AC_UNSOL_ENABLED (1<<7) -#define AC_USRSP_EN AC_UNSOL_ENABLED - -/* Unsolicited responses */ -#define AC_UNSOL_RES_TAG (0x3f<<26) -#define AC_UNSOL_RES_TAG_SHIFT 26 -#define AC_UNSOL_RES_SUBTAG (0x1f<<21) -#define AC_UNSOL_RES_SUBTAG_SHIFT 21 -#define AC_UNSOL_RES_DE (0x3f<<15) /* Device Entry - * (for DP1.2 MST) - */ -#define AC_UNSOL_RES_DE_SHIFT 15 -#define AC_UNSOL_RES_IA (1<<2) /* Inactive (for DP1.2 MST) */ -#define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */ -#define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */ -#define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */ -#define AC_UNSOL_RES_CP_READY (1<<0) /* content protection */ - -/* Pin widget capabilies */ -#define AC_PINCAP_IMP_SENSE (1<<0) /* impedance sense capable */ -#define AC_PINCAP_TRIG_REQ (1<<1) /* trigger required */ -#define AC_PINCAP_PRES_DETECT (1<<2) /* presence detect capable */ -#define AC_PINCAP_HP_DRV (1<<3) /* headphone drive capable */ -#define AC_PINCAP_OUT (1<<4) /* output capable */ -#define AC_PINCAP_IN (1<<5) /* input capable */ -#define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */ -/* Note: This LR_SWAP pincap is defined in the Realtek ALC883 specification, - * but is marked reserved in the Intel HDA specification. - */ -#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */ -/* Note: The same bit as LR_SWAP is newly defined as HDMI capability - * in HD-audio specification - */ -#define AC_PINCAP_HDMI (1<<7) /* HDMI pin */ -#define AC_PINCAP_DP (1<<24) /* DisplayPort pin, can - * coexist with AC_PINCAP_HDMI - */ -#define AC_PINCAP_VREF (0x37<<8) -#define AC_PINCAP_VREF_SHIFT 8 -#define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ -#define AC_PINCAP_HBR (1<<27) /* High Bit Rate */ -/* Vref status (used in pin cap) */ -#define AC_PINCAP_VREF_HIZ (1<<0) /* Hi-Z */ -#define AC_PINCAP_VREF_50 (1<<1) /* 50% */ -#define AC_PINCAP_VREF_GRD (1<<2) /* ground */ -#define AC_PINCAP_VREF_80 (1<<4) /* 80% */ -#define AC_PINCAP_VREF_100 (1<<5) /* 100% */ - -/* Amplifier capabilities */ -#define AC_AMPCAP_OFFSET (0x7f<<0) /* 0dB offset */ -#define AC_AMPCAP_OFFSET_SHIFT 0 -#define AC_AMPCAP_NUM_STEPS (0x7f<<8) /* number of steps */ -#define AC_AMPCAP_NUM_STEPS_SHIFT 8 -#define AC_AMPCAP_STEP_SIZE (0x7f<<16) /* step size 0-32dB - * in 0.25dB - */ -#define AC_AMPCAP_STEP_SIZE_SHIFT 16 -#define AC_AMPCAP_MUTE (1<<31) /* mute capable */ -#define AC_AMPCAP_MUTE_SHIFT 31 - -/* driver-specific amp-caps: using bits 24-30 */ -#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */ - -/* Connection list */ -#define AC_CLIST_LENGTH (0x7f<<0) -#define AC_CLIST_LONG (1<<7) - -/* Supported power status */ -#define AC_PWRST_D0SUP (1<<0) -#define AC_PWRST_D1SUP (1<<1) -#define AC_PWRST_D2SUP (1<<2) -#define AC_PWRST_D3SUP (1<<3) -#define AC_PWRST_D3COLDSUP (1<<4) -#define AC_PWRST_S3D3COLDSUP (1<<29) -#define AC_PWRST_CLKSTOP (1<<30) -#define AC_PWRST_EPSS (1U<<31) - -/* Power state values */ -#define AC_PWRST_SETTING (0xf<<0) -#define AC_PWRST_ACTUAL (0xf<<4) -#define AC_PWRST_ACTUAL_SHIFT 4 -#define AC_PWRST_D0 0x00 -#define AC_PWRST_D1 0x01 -#define AC_PWRST_D2 0x02 -#define AC_PWRST_D3 0x03 -#define AC_PWRST_ERROR (1<<8) -#define AC_PWRST_CLK_STOP_OK (1<<9) -#define AC_PWRST_SETTING_RESET (1<<10) - -/* Processing capabilies */ -#define AC_PCAP_BENIGN (1<<0) -#define AC_PCAP_NUM_COEF (0xff<<8) -#define AC_PCAP_NUM_COEF_SHIFT 8 - -/* Volume knobs capabilities */ -#define AC_KNBCAP_NUM_STEPS (0x7f<<0) -#define AC_KNBCAP_DELTA (1<<7) - -/* HDMI LPCM capabilities */ -#define AC_LPCMCAP_48K_CP_CHNS (0x0f<<0) /* max channels w/ CP-on */ -#define AC_LPCMCAP_48K_NO_CHNS (0x0f<<4) /* max channels w/o CP-on */ -#define AC_LPCMCAP_48K_20BIT (1<<8) /* 20b bitrate supported */ -#define AC_LPCMCAP_48K_24BIT (1<<9) /* 24b bitrate supported */ -#define AC_LPCMCAP_96K_CP_CHNS (0x0f<<10) /* max channels w/ CP-on */ -#define AC_LPCMCAP_96K_NO_CHNS (0x0f<<14) /* max channels w/o CP-on */ -#define AC_LPCMCAP_96K_20BIT (1<<18) /* 20b bitrate supported */ -#define AC_LPCMCAP_96K_24BIT (1<<19) /* 24b bitrate supported */ -#define AC_LPCMCAP_192K_CP_CHNS (0x0f<<20) /* max channels w/ CP-on */ -#define AC_LPCMCAP_192K_NO_CHNS (0x0f<<24) /* max channels w/o CP-on */ -#define AC_LPCMCAP_192K_20BIT (1<<28) /* 20b bitrate supported */ -#define AC_LPCMCAP_192K_24BIT (1<<29) /* 24b bitrate supported */ -#define AC_LPCMCAP_44K (1<<30) /* 44.1kHz support */ -#define AC_LPCMCAP_44K_MS (1<<31) /* 44.1kHz-multiplies support */ - -/* Display pin's device list length */ -#define AC_DEV_LIST_LEN_MASK 0x3f -#define AC_MAX_DEV_LIST_LEN 64 - -/* - * Control Parameters - */ - -/* Amp gain/mute */ -#define AC_AMP_MUTE (1<<7) -#define AC_AMP_GAIN (0x7f) -#define AC_AMP_GET_INDEX (0xf<<0) - -#define AC_AMP_GET_LEFT (1<<13) -#define AC_AMP_GET_RIGHT (0<<13) -#define AC_AMP_GET_OUTPUT (1<<15) -#define AC_AMP_GET_INPUT (0<<15) - -#define AC_AMP_SET_INDEX (0xf<<8) -#define AC_AMP_SET_INDEX_SHIFT 8 -#define AC_AMP_SET_RIGHT (1<<12) -#define AC_AMP_SET_LEFT (1<<13) -#define AC_AMP_SET_INPUT (1<<14) -#define AC_AMP_SET_OUTPUT (1<<15) - -/* DIGITAL1 bits */ -#define AC_DIG1_ENABLE (1<<0) -#define AC_DIG1_V (1<<1) -#define AC_DIG1_VCFG (1<<2) -#define AC_DIG1_EMPHASIS (1<<3) -#define AC_DIG1_COPYRIGHT (1<<4) -#define AC_DIG1_NONAUDIO (1<<5) -#define AC_DIG1_PROFESSIONAL (1<<6) -#define AC_DIG1_LEVEL (1<<7) - -/* DIGITAL2 bits */ -#define AC_DIG2_CC (0x7f<<0) - -/* DIGITAL3 bits */ -#define AC_DIG3_ICT (0xf<<0) -#define AC_DIG3_KAE (1<<7) - -/* Pin widget control - 8bit */ -#define AC_PINCTL_EPT (0x3<<0) -#define AC_PINCTL_EPT_NATIVE 0 -#define AC_PINCTL_EPT_HBR 3 -#define AC_PINCTL_VREFEN (0x7<<0) -#define AC_PINCTL_VREF_HIZ 0 /* Hi-Z */ -#define AC_PINCTL_VREF_50 1 /* 50% */ -#define AC_PINCTL_VREF_GRD 2 /* ground */ -#define AC_PINCTL_VREF_80 4 /* 80% */ -#define AC_PINCTL_VREF_100 5 /* 100% */ -#define AC_PINCTL_IN_EN (1<<5) -#define AC_PINCTL_OUT_EN (1<<6) -#define AC_PINCTL_HP_EN (1<<7) - -/* Pin sense - 32bit */ -#define AC_PINSENSE_IMPEDANCE_MASK (0x7fffffff) -#define AC_PINSENSE_PRESENCE (1<<31) -#define AC_PINSENSE_ELDV (1<<30) /* ELD valid (HDMI) */ - -/* EAPD/BTL enable - 32bit */ -#define AC_EAPDBTL_BALANCED (1<<0) -#define AC_EAPDBTL_EAPD (1<<1) -#define AC_EAPDBTL_LR_SWAP (1<<2) - -/* HDMI ELD data */ -#define AC_ELDD_ELD_VALID (1<<31) -#define AC_ELDD_ELD_DATA 0xff - -/* HDMI DIP size */ -#define AC_DIPSIZE_ELD_BUF (1<<3) /* ELD buf size of packet size */ -#define AC_DIPSIZE_PACK_IDX (0x07<<0) /* packet index */ - -/* HDMI DIP index */ -#define AC_DIPIDX_PACK_IDX (0x07<<5) /* packet idnex */ -#define AC_DIPIDX_BYTE_IDX (0x1f<<0) /* byte index */ - -/* HDMI DIP xmit (transmit) control */ -#define AC_DIPXMIT_MASK (0x3<<6) -#define AC_DIPXMIT_DISABLE (0x0<<6) /* disable xmit */ -#define AC_DIPXMIT_ONCE (0x2<<6) /* xmit once then disable */ -#define AC_DIPXMIT_BEST (0x3<<6) /* best effort */ - -/* HDMI content protection (CP) control */ -#define AC_CPCTRL_CES (1<<9) /* current encryption state */ -#define AC_CPCTRL_READY (1<<8) /* ready bit */ -#define AC_CPCTRL_SUBTAG (0x1f<<3) /* subtag for unsol-resp */ -#define AC_CPCTRL_STATE (3<<0) /* current CP request state */ - -/* Converter channel <-> HDMI slot mapping */ -#define AC_CVTMAP_HDMI_SLOT (0xf<<0) /* HDMI slot number */ -#define AC_CVTMAP_CHAN (0xf<<4) /* converter channel number */ - -/* configuration default - 32bit */ -#define AC_DEFCFG_SEQUENCE (0xf<<0) -#define AC_DEFCFG_DEF_ASSOC (0xf<<4) -#define AC_DEFCFG_ASSOC_SHIFT 4 -#define AC_DEFCFG_MISC (0xf<<8) -#define AC_DEFCFG_MISC_SHIFT 8 -#define AC_DEFCFG_MISC_NO_PRESENCE (1<<0) -#define AC_DEFCFG_COLOR (0xf<<12) -#define AC_DEFCFG_COLOR_SHIFT 12 -#define AC_DEFCFG_CONN_TYPE (0xf<<16) -#define AC_DEFCFG_CONN_TYPE_SHIFT 16 -#define AC_DEFCFG_DEVICE (0xf<<20) -#define AC_DEFCFG_DEVICE_SHIFT 20 -#define AC_DEFCFG_LOCATION (0x3f<<24) -#define AC_DEFCFG_LOCATION_SHIFT 24 -#define AC_DEFCFG_PORT_CONN (0x3<<30) -#define AC_DEFCFG_PORT_CONN_SHIFT 30 - -/* Display pin's device list entry */ -#define AC_DE_PD (1<<0) -#define AC_DE_ELDV (1<<1) -#define AC_DE_IA (1<<2) - -/* device device types (0x0-0xf) */ -enum { - AC_JACK_LINE_OUT, - AC_JACK_SPEAKER, - AC_JACK_HP_OUT, - AC_JACK_CD, - AC_JACK_SPDIF_OUT, - AC_JACK_DIG_OTHER_OUT, - AC_JACK_MODEM_LINE_SIDE, - AC_JACK_MODEM_HAND_SIDE, - AC_JACK_LINE_IN, - AC_JACK_AUX, - AC_JACK_MIC_IN, - AC_JACK_TELEPHONY, - AC_JACK_SPDIF_IN, - AC_JACK_DIG_OTHER_IN, - AC_JACK_OTHER = 0xf, -}; - -/* jack connection types (0x0-0xf) */ -enum { - AC_JACK_CONN_UNKNOWN, - AC_JACK_CONN_1_8, - AC_JACK_CONN_1_4, - AC_JACK_CONN_ATAPI, - AC_JACK_CONN_RCA, - AC_JACK_CONN_OPTICAL, - AC_JACK_CONN_OTHER_DIGITAL, - AC_JACK_CONN_OTHER_ANALOG, - AC_JACK_CONN_DIN, - AC_JACK_CONN_XLR, - AC_JACK_CONN_RJ11, - AC_JACK_CONN_COMB, - AC_JACK_CONN_OTHER = 0xf, -}; - -/* jack colors (0x0-0xf) */ -enum { - AC_JACK_COLOR_UNKNOWN, - AC_JACK_COLOR_BLACK, - AC_JACK_COLOR_GREY, - AC_JACK_COLOR_BLUE, - AC_JACK_COLOR_GREEN, - AC_JACK_COLOR_RED, - AC_JACK_COLOR_ORANGE, - AC_JACK_COLOR_YELLOW, - AC_JACK_COLOR_PURPLE, - AC_JACK_COLOR_PINK, - AC_JACK_COLOR_WHITE = 0xe, - AC_JACK_COLOR_OTHER, -}; - -/* Jack location (0x0-0x3f) */ -/* common case */ -enum { - AC_JACK_LOC_NONE, - AC_JACK_LOC_REAR, - AC_JACK_LOC_FRONT, - AC_JACK_LOC_LEFT, - AC_JACK_LOC_RIGHT, - AC_JACK_LOC_TOP, - AC_JACK_LOC_BOTTOM, -}; -/* bits 4-5 */ -enum { - AC_JACK_LOC_EXTERNAL = 0x00, - AC_JACK_LOC_INTERNAL = 0x10, - AC_JACK_LOC_SEPARATE = 0x20, - AC_JACK_LOC_OTHER = 0x30, -}; -enum { - /* external on primary chasis */ - AC_JACK_LOC_REAR_PANEL = 0x07, - AC_JACK_LOC_DRIVE_BAY, - /* internal */ - AC_JACK_LOC_RISER = 0x17, - AC_JACK_LOC_HDMI, - AC_JACK_LOC_ATAPI, - /* others */ - AC_JACK_LOC_MOBILE_IN = 0x37, - AC_JACK_LOC_MOBILE_OUT, -}; - -/* Port connectivity (0-3) */ -enum { - AC_JACK_PORT_COMPLEX, - AC_JACK_PORT_NONE, - AC_JACK_PORT_FIXED, - AC_JACK_PORT_BOTH, -}; - -/* max. codec address */ -#define HDA_MAX_CODEC_ADDRESS 0x0f +#include <sound/hda_verbs.h> /* * generic arrays -- 1.8.5

1 0

[alsa-devel] [PATCH 0/2] ARM: Rename ARCH_SHMOBILE to ARCH_SHMOBILE_LEGACY
by Laurent Pinchart 09 Dec '13

09 Dec '13

Hello, This patch series enables compilation of SH-Mobile drivers on both the non-multiplatform (ARCH_SHMOBILE) and the multiplatform (ARCH_SHMOBILE_MULTI) SH-Mobile architectures. To do so it renames ARCH_SHMOBILE to ARCH_SHMOBILE_LEGACY and creates a new ARCH_SHMOBILE symbol selected by both ARCH_SHMOBILE_MULTI and ARCH_SHMOBILE_LEGACY. The second patch adds a dependency on ARCH_SHMOBILE_LEGACY to the ASoC R-Car driver. It can be squashed with the first patch or applied separately to minimize the risk of conflicts if desired. Laurent Pinchart (2): ARM: Rename ARCH_SHMOBILE to ARCH_SHMOBILE_LEGACY ASoC: sh: rcar: Depend on ARCH_SHMOBILE_LEGACY arch/arm/Kconfig | 14 ++++++++------ arch/arm/Makefile | 1 - arch/arm/boot/compressed/Makefile | 2 +- arch/arm/boot/dts/Makefile | 2 +- arch/arm/mach-shmobile/Kconfig | 8 ++++++-- drivers/Makefile | 2 +- sound/soc/sh/Kconfig | 1 + 7 files changed, 18 insertions(+), 12 deletions(-) -- Regards, Laurent Pinchart

4 7

[alsa-devel] issues with usb-audio and maudio quattro
by Patrick Shirkey 09 Dec '13

09 Dec '13

Hi, It seems that some changes were made to the usb-audio driver over the years that have resulted in partially broken behaviour with maudio quattro. We have run some tests on two different devices with different machines and found the following items: 1: There are no longer three pcm devices - this breaks previously working asoundrc files 2: pcm0 and pcm1 have different behaviours - this results in audio input working for pcm0 but white noise in pcm1 (probably due to endianess le vs be) 3: We can record 4 tracks with arecord. Tracks 1/2 have audio and track 3/4 have white noise 4: We can no longer run 4 playback tracks in JACK2 as it complains about the ports not matching (this might be a seperate issue with JACK2 and route/plug configs for multichannel devices but that would probably affect the delta 10/10, RME Hammerfall or other multichannel devices too.) jackd: ../common/JackGraphManager.cpp:44: void Jack::JackGraphManager::AssertPort(jack_port_id_t): Assertion `port_index < fPortMax' failed. So the main question is who made the decision to drop/consolidate the third pcm and break previously working asoundrc files? There must be a good reason in which case maybe we can get things working again relatively easily? Cheers -- Patrick Shirkey Boost Hardware Ltd

3 2

[alsa-devel] [PATCH][RESEND 0/8] Add and implement gen_pool_dma_alloc()
by Nicolin Chen 09 Dec '13

09 Dec '13

Previously, we don't have a specific gen_pool_alloc() for DMA usage; Instead, we need to use gen_pool_virt_to_phys() to convert the addr returned from gen_pool_alloc(). So each implementation of this has duplicated code. Thus add new helper function -- gen_pool_dma_alloc(). After gen_pool_dma_alloc() is introduced, we can replace the original combination of gen_pool_alloc()+gen_pool_virt_to_phys() with this new helper function. Thus this patch implement the helper function to all the current drivers which use gen_pool_virt_to_phys(). !!-------------------important------------------!! The later 7 patches need all related driver owner to test. We can here define a simple rule: 1, If one driver owner or maintainer doesn't like the mofication to his/her driver, just let me know. I would drop that patch. 2, If there's a bug and issue found after patch-testing, please reply the mail so that I can fix and refine the patch. 3, If one driver owner or maintainer is too busy and doesn't have bandwidth to test the patch, I would drop that patch from this series. We can reimplement it till there's someone test it. !!-------------------current progress-----------!! lib/genalloc: [Okay] ARM: davinci: [Untested] dma: mmp_tdma: [Untested] [media] coda: [Untested] uio: uio_pruss: [Untested] ALSA: memalloc: [Tested] by Nicolin Chen with i.MX6Q SabreSD ASoC: davinci: [Untested] ASoC: pxa: use [Untested] Nicolin Chen (8): lib/genalloc: add a helper function for DMA buffer allocation ARM: davinci: use gen_pool_dma_alloc() to sram.c dma: mmp_tdma: use gen_pool_dma_alloc() to allocate descriptor [media] coda: use gen_pool_dma_alloc() to allocate iram buffer uio: uio_pruss: use gen_pool_dma_alloc() to allocate sram memory ALSA: memalloc: use gen_pool_dma_alloc() to allocate iram buffer ASoC: davinci: use gen_pool_dma_alloc() in davinci-pcm.c ASoC: pxa: use gen_pool_dma_alloc() to allocate dma buffer arch/arm/mach-davinci/sram.c | 9 +-------- drivers/dma/mmp_tdma.c | 7 +------ drivers/media/platform/coda.c | 5 ++--- drivers/uio/uio_pruss.c | 6 ++---- include/linux/genalloc.h | 2 ++ lib/genalloc.c | 28 ++++++++++++++++++++++++++++ sound/core/memalloc.c | 6 +----- sound/soc/davinci/davinci-pcm.c | 3 +-- sound/soc/pxa/mmp-pcm.c | 3 +-- 9 files changed, 39 insertions(+), 30 deletions(-) -- 1.8.4

5 13

[alsa-devel] Resend: Incorrect sbits for snd-usb-audio
by Sander Jansen 08 Dec '13

08 Dec '13

Hi, I had send this out earlier back in September, but never got a response. I have a Focusrite Scarlett 2i4 USB Audio Interface. The audio format exposed by ALSA for this device is S32_LE. From the hardware specs and lsusb. it supposed to only have a 24 bit resolution (with the LSB bits set to 0): AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 When I query the sbits for this device, I expect it to return 24, but instead it returns 32. Looking through the usb audio driver code, it doesn't look like this information is actually exposed to userspace. The 'parse_audio_format_i_type' in usb/format.c merely returns the supported formats, but doesn't include any msbits info. (and it doesn't look like it can be easily changed to return the msbits info as well). Was this intentionally left out, or is this a missing feature in the driver? Thanks, Sander

2 4

Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by Raymond Yau 08 Dec '13

08 Dec '13

> I just changed my RadeonHD 4670 for a 7870, and now when I type > "aplay -l" I get many HDMI codecs instead of just one like for the > previous you have to specify CONFIG_SND_DYNAMIC_MINORS=y when building the driver [ 4.637413] ALSA hda_codec.c:4506 Too many HDMI devices [ 4.637414] ALSA hda_codec.c:4508 Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y [ 4.637644] ALSA control.c:352 control 0:0:0:HDMI/DP Jack:0 is already present [ 4.639220] hda_codec: cannot build controls for #0 (error -16) > can you post the system log if the graphic driver know hdmi, vga and two >> mini display ports are connected or not ? >> >> I cannot see anything in dmesg related, so attaching Xorg log. > >> - >> > 5.268590] [drm] Radeon Display Connectors [ 5.268592] [drm] Connector 0: [ 5.268593] [drm] DP-1 [ 5.268594] [drm] HPD4 [ 5.268595] [drm] DDC: 0x6530 0x6530 0x6534 0x6534 0x6538 0x6538 0x653c 0x653c [ 5.268596] [drm] Encoders: [ 5.268597] [drm] DFP1: INTERNAL_UNIPHY2 [ 5.268598] [drm] Connector 1: [ 5.268599] [drm] DP-2 [ 5.268600] [drm] HPD5 [ 5.268601] [drm] DDC: 0x6540 0x6540 0x6544 0x6544 0x6548 0x6548 0x654c 0x654c [ 5.268602] [drm] Encoders: [ 5.268603] [drm] DFP2: INTERNAL_UNIPHY2 [ 5.268603] [drm] Connector 2: [ 5.268604] [drm] HDMI-A-1 [ 5.268605] [drm] HPD1 [ 5.268606] [drm] DDC: 0x6550 0x6550 0x6554 0x6554 0x6558 0x6558 0x655c 0x655c [ 5.268607] [drm] Encoders: [ 5.268608] [drm] DFP3: INTERNAL_UNIPHY1 [ 5.268609] [drm] Connector 3: [ 5.268610] [drm] DVI-I-1 [ 5.268610] [drm] HPD6 [ 5.268612] [drm] DDC: 0x6580 0x6580 0x6584 0x6584 0x6588 0x6588 0x658c 0x658c [ 5.268612] [drm] Encoders: [ 5.268613] [drm] DFP4: INTERNAL_UNIPHY [ 5.268614] [drm] CRT1: INTERNAL_KLDSCP_DAC1 [ 7.613] (II) RADEON(0): Output DisplayPort-0 has no monitor section [ 7.626] (II) RADEON(0): Output DisplayPort-1 has no monitor section [ 7.693] (II) RADEON(0): Output HDMI-0 has no monitor section [ 7.713] (II) RADEON(0): Output DVI-0 has no monitor section [ 7.726] (II) RADEON(0): EDID for output DisplayPort-0 [ 7.740] (II) RADEON(0): EDID for output DisplayPort-1 [ 7.801] (II) RADEON(0): EDID for output HDMI-0 [ 7.801] (II) RADEON(0): Manufacturer: DON Model: 26 Serial#: 16843009 [ 7.801] (II) RADEON(0): Year: 2011 Week: 0 [ 7.801] (II) RADEON(0): EDID Version: 1.3 [ 7.823] (II) RADEON(0): EDID for output DVI-0 [ 7.823] (II) RADEON(0): Output DisplayPort-0 disconnected [ 7.823] (II) RADEON(0): Output DisplayPort-1 disconnected [ 7.823] (II) RADEON(0): Output HDMI-0 connected [ 7.823] (II) RADEON(0): Output DVI-0 disconnected the graphic driver knew only HDMI-0 is connected and the others are not connected

2 1

Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by John Ettedgui 08 Dec '13

08 Dec '13

and same logs but with -with-debug=detect instead of full.

1 0

[alsa-devel] [PATCH v2] ALSA: hda - Ignore small negative LPIB delay correction
by Takashi Iwai 06 Dec '13

06 Dec '13

Sometimes the hardware reports LPIB being advanced than POSBUF. When this happens, the driver adjusts to a positive value by adding the buffer size. Then the driver detects it as an error (greater than the period size), and stops the LPIB delay account from this point on. When I took a close look at these conditions, the values shown are all very small numbers, and it'd be better to just ignore these values instead of discontinuing the LPIB delay correction. In this patch, the driver checks a negative delay value and ignores if it's a significantly small error. Currently the threshold is set to 64 frames, but could be smaller. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: apply only when period size is greater than 64. sound/pci/hda/hda_intel.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 80c5f14e8ecd..af86c71f27bf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -431,6 +431,8 @@ struct azx_dev { struct timecounter azx_tc; struct cyclecounter azx_cc; + int delay_negative_threshold; + #ifdef CONFIG_SND_HDA_DSP_LOADER struct mutex dsp_mutex; #endif @@ -2195,6 +2197,15 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } + /* when LPIB delay correction gives a small negative value, + * we ignore it; currently set the threshold statically to + * 64 frames + */ + if (runtime->period_size > 64) + azx_dev->delay_negative_threshold = -frames_to_bytes(runtime, 64); + else + azx_dev->delay_negative_threshold = 0; + /* wallclk has 24Mhz clock source */ azx_dev->period_wallclk = (((runtime->period_size * 24000) / runtime->rate) * 1000); @@ -2447,8 +2458,12 @@ static unsigned int azx_get_position(struct azx *chip, delay = pos - lpib_pos; else delay = lpib_pos - pos; - if (delay < 0) - delay += azx_dev->bufsize; + if (delay < 0) { + if (delay >= azx_dev->delay_negative_threshold) + delay = 0; + else + delay += azx_dev->bufsize; + } if (delay >= azx_dev->period_bytes) { snd_printk(KERN_WARNING SFX "%s: Unstable LPIB (%d >= %d); " -- 1.8.5

1 0

[alsa-devel] [PATCH] ALSA: hda - Ignore small negative LPIB delay correction
by Takashi Iwai 06 Dec '13

06 Dec '13

Sometimes the hardware reports LPIB being advanced than POSBUF. When this happens, the driver adjusts to a positive value by adding the buffer size. Then the driver detects it as an error (greater than the period size), and stops the LPIB delay account from this point on. When I took a close look at these conditions, the values shown are all very small numbers, and it'd be better to just ignore these values instead of discontinuing the LPIB delay correction. In this patch, the driver checks a negative delay value and ignores if it's a significantly small error. Currently the threshold is set to 64 frames, but could be smaller. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 80c5f14e8ecd..fd4f2288a33f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -431,6 +431,8 @@ struct azx_dev { struct timecounter azx_tc; struct cyclecounter azx_cc; + int delay_negative_threshold; + #ifdef CONFIG_SND_HDA_DSP_LOADER struct mutex dsp_mutex; #endif @@ -2195,6 +2197,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } + /* when LPIB delay correction gives a small negative value, + * we ignore it; currently set the threshold statically to + * 64 frames + */ + azx_dev->delay_negative_threshold = -frames_to_bytes(runtime, 64); + /* wallclk has 24Mhz clock source */ azx_dev->period_wallclk = (((runtime->period_size * 24000) / runtime->rate) * 1000); @@ -2447,8 +2455,12 @@ static unsigned int azx_get_position(struct azx *chip, delay = pos - lpib_pos; else delay = lpib_pos - pos; - if (delay < 0) - delay += azx_dev->bufsize; + if (delay < 0) { + if (delay >= azx_dev->delay_negative_threshold) + delay = 0; + else + delay += azx_dev->bufsize; + } if (delay >= azx_dev->period_bytes) { snd_printk(KERN_WARNING SFX "%s: Unstable LPIB (%d >= %d); " -- 1.8.5

1 0

[alsa-devel] [PATCH 0/3] Enhance imx-wm8962 machine driver
by Nicolin Chen 06 Dec '13

06 Dec '13

Although there's no direct relationship between any of these three patches, I recommend to apply them in order so as to circumvent merging conflicts. Nicolin Chen (3): ASoC: fsl: imx-wm8962: Don't update bias_level in machine driver ASoC: fsl: imx-wm8962: Grant hw_params/free() permission to control FLL ASoC: fsl: imx-wm8962: Add headphone and microphone jack plugin/out detection .../devicetree/bindings/sound/imx-audio-wm8962.txt | 8 + Documentation/devicetree/bindings/sound/wm8962.txt | 8 + sound/soc/fsl/imx-wm8962.c | 361 ++++++++++++++++++--- 3 files changed, 326 insertions(+), 51 deletions(-) -- 1.8.4

2 9

[alsa-devel] [PATCH 1/2] ASoC: samsung: Provide helper for DMA init
by Mark Brown 06 Dec '13

06 Dec '13

From: Mark Brown <broonie(a)linaro.org> In preparation for using the dmaengine helpers in ASoC rather than the dmaengine wrappers for the Samsung API wrap the configuration of dma_data. The dmaengine code expects different data to that used by the legacy API. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 51 +++++++++++++++--------------------------------- sound/soc/samsung/dma.c | 8 ++++++++ sound/soc/samsung/dma.h | 3 +++ sound/soc/samsung/i2s.c | 7 +++++-- sound/soc/samsung/pcm.c | 18 +++++++++-------- 5 files changed, 42 insertions(+), 45 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 350ba23a9893..4a88e36c82ec 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = { .reset = s3c_ac97_cold_reset, }; -static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; +static int s3c_ac97_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in); + + return 0; +} + +static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in); + + return 0; +} + static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", @@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_dai_probe, .ops = &s3c_ac97_dai_ops, }, [S3C_AC97_DAI_MIC] = { @@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 1, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_mic_dai_probe, .ops = &s3c_ac97_mic_dai_ops, }, }; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index fe2748b494d4..ee23194f7ab8 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + snd_soc_dai_init_dma_data(dai, playback, capture); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0e86315a3eaf..fb09a1c5f75b 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,6 +22,9 @@ struct s3c_dma_params { char *ch_name; }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture); int samsung_asoc_dma_platform_register(struct device *dev); void samsung_asoc_dma_platform_unregister(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4f1655..eab0050d4579 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -946,8 +946,11 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; - if (other && other->clk) /* If this is probe on secondary */ + if (other && other->clk) { /* If this is probe on secondary */ + samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, + NULL); goto probe_exit; + } i2s->addr = ioremap(i2s->base, 0x100); if (i2s->addr == NULL) { @@ -963,7 +966,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); - snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { other->addr = i2s->addr; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index e54256fc4b2c..6a5e4bf6ac96 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); - /* Strictly check for sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_fmt = s3c_pcm_set_fmt, }; +static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture); + + return 0; +} + #define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 #define S3C_PCM_DAI_DECLARE \ .symmetric_rates = 1, \ + .probe = s3c_pcm_dai_probe, \ .ops = &s3c_pcm_dai_ops, \ .playback = { \ .channels_min = 2, \ -- 1.8.5.1

1 1

[alsa-devel] [PATCH] ASoC mc13783 codec: correct names
by Lambrecht Jürgen 06 Dec '13

06 Dec '13

Correct names to reality and for consistency: - "Speaker Amp Switch"->"Speaker Amp" - "Earpiece Amp Switch"->"Speaker Amp Switch" - "Loudspeaker enable"->"Loudspeaker Amp Switch" - "..Capture.."->"..Playback" (Line in..., CODEC..., ...Mixin Switch) - move "PCM Capture Volume" to bottom with other Capture controls Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/codecs/mc13783.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 08fb022..878e3e5 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -534,7 +534,7 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 0, &cdcout_ctl), - SND_SOC_DAPM_SWITCH("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 0, + SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, @@ -590,7 +590,7 @@ static struct snd_soc_dapm_route mc13783_routes[] = { { "HSR", NULL, "Headset Amp Right"}, { "RXOUTL", NULL, "Line out Amp Left"}, { "RXOUTR", NULL, "Line out Amp Right"}, - { "SP", "Speaker Amp Switch", "Speaker Amp Source MUX"}, + { "SP", "Speaker Amp", "Speaker Amp Source MUX"}, { "LSP", "Loudspeaker Amp", "Loudspeaker Amp Source MUX"}, { "HSL", "Headset Amp Left", "Headset Amp Source MUX"}, { "HSR", "Headset Amp Right", "Headset Amp Source MUX"}, @@ -609,26 +609,27 @@ static const struct soc_enum mc13783_enum_3d_mixer = mc13783_3d_mixer); static struct snd_kcontrol_new mc13783_control_list[] = { - SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), - SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0), - SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_SINGLE("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_SINGLE("Loudspeaker Amp Switch", MC13783_AUDIO_RX0, 5, 1, 0), SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0), SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0), - SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), - SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), - SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + SOC_SINGLE("CODEC Playback Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), + SOC_SINGLE("PCM Playback Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Playback Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("CODEC Playback Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("CODEC Playback Switch", MC13783_AUDIO_RX1, 0, 1, 0), - SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), - SOC_SINGLE("CODEC Capture Switch", MC13783_AUDIO_RX1, 0, 1, 0), + SOC_SINGLE("Line in Playback Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("Line in Playback Switch", MC13783_AUDIO_RX1, 10, 1, 0), - SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), - SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), -- 1.7.10.4

2 2

Re: [alsa-devel] Intermittent cut out on the video
by Dolevo Jay 06 Dec '13

06 Dec '13

From: cmst(a)live.com To: clemens(a)ladisch.de Subject: RE: [alsa-devel] Intermittent cut out on the video Date: Mon, 2 Dec 2013 09:44:16 +0000 > Date: Thu, 28 Nov 2013 09:29:45 +0100 > From: clemens(a)ladisch.de > To: cmst(a)live.com; alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Intermittent cut out on the video > > Dolevo Jay wrote: > > I have two sound cards on my system and I use the [multi plugin] to > > have audio from two jacks simultaneously. > > > > The audio coming from the jack on the pc is cutting out intermittently. > > It just cuts out for a fraction of a second and continues as normal > > and after some time it happens again. > > The two cards use different sample clocks, so it is not possible to feed > them reliably from the same stream. > > You would need software that measures that actual sample rate of both > devices and resamples its output appropriately. Try using PulseAudio. > > > Regards, > Clemens > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel Hi, I have tried to apply pulseaudio but couldn't really manage it. Could you please tell me if I can fix it in the configuration file? I mean, for example, would it be solved if I resample the audio for the appropriate sound card in the asound.conf? What other option do I have other than pulseaudio? Thanks a lot.

2 5

[alsa-devel] [PATCH] ALSA: hda - Allow capture-only configuration
by Takashi Iwai 06 Dec '13

06 Dec '13

We have blindly assumed that all valid configurations should have either analog or digital playback, but there can be capture-only configurations. The parser shouldn't escape in such a case. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 2729e881541d..5ba3106b9712 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4334,7 +4334,8 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, spec->no_analog = 1; goto dig_only; } - return 0; /* can't find valid BIOS pin config */ + if (!cfg->num_inputs && !cfg->dig_in_pin) + return 0; /* can't find valid BIOS pin config */ } if (!spec->no_primary_hp && -- 1.8.5

1 0

[alsa-devel] [PATCH] ALSA: hda - skip depop delay before D3 for Haswell and Valleyview2 display codec
by mengdong.lin＠intel.com 06 Dec '13

06 Dec '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch skips the default depop delay before D3 for Haswell (10 ms) and Valleyview2 (100 ms) display codec, to reduce codec suspend time. The analog part of display audio is implemented in the external display. Some displays have weak pop noise while others not when suspending, no matter there is the default delay or not. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2306645..e14359a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2022,6 +2022,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) intel_haswell_fixup_enable_dp12(codec); } + if (is_haswell(codec) || is_valleyview(codec)) { + codec->depop_delay = 0; + } + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ALSA: hda - Remove quirk for Dell Vostro 131
by David Henningsson 06 Dec '13

06 Dec '13

I've tested the old Dell Vostro 131 with the latest generic parser and it works just fine, and as a bonus we get better jack detection features in userspace. Therefore this quirk can be removed. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1f2717f..3fbf288 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2936,7 +2936,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS), SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - do delayed suspend for HD-audio devices
by mengdong.lin＠intel.com 06 Dec '13

06 Dec '13

From: Mengdong Lin <mengdong.lin(a)intel.com> HD-Audio codecs tends to take a long time to supend and is time-consuming part in driver suspend. In this patch, codec suspending and the left operations are delayed to a worker thead, so that they can be done asynchronously in parallel with suspending of other devices. And azx_suspend_late() is defined as PM .suspend_late ops to wait for completion of the delayed suspending. Credit goes to Liam Girdwood, who suggested this workaround. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 80c5f14..aed4cdf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -545,6 +545,11 @@ struct azx { struct work_struct probe_work; +#ifdef CONFIG_PM + struct workqueue_struct *pm_wq; /* workqueue to delay controller PM */ + struct work_struct suspend_work; /* task to delay controller suspend */ +#endif + /* reboot notifier (for mysterious hangup problem at power-down) */ struct notifier_block reboot_notifier; @@ -2908,20 +2913,11 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) /* * power management */ -static int azx_suspend(struct device *dev) -{ - struct pci_dev *pci = to_pci_dev(dev); - struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct azx_pcm *p; - if (chip->disabled) - return 0; +static int azx_suspend_continue(struct azx *chip) +{ + struct pci_dev *pci = chip->pci; - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - azx_clear_irq_pending(chip); - list_for_each_entry(p, &chip->pcm_list, list) - snd_pcm_suspend_all(p->pcm); if (chip->initialized) snd_hda_suspend(chip->bus); azx_stop_chip(chip); @@ -2937,6 +2933,41 @@ static int azx_suspend(struct device *dev) pci_set_power_state(pci, PCI_D3hot); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) hda_display_power(false); + + return 0; +} + +static void azx_suspend_work(struct work_struct *work) +{ + azx_suspend_continue(container_of(work, struct azx, suspend_work)); +} + + +static int azx_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct azx_pcm *p; + + if (chip->disabled) + return 0; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + azx_clear_irq_pending(chip); + list_for_each_entry(p, &chip->pcm_list, list) + snd_pcm_suspend_all(p->pcm); + + queue_work(chip->pm_wq, &chip->suspend_work); + return 0; +} + +static int azx_suspend_late(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + flush_workqueue(chip->pm_wq); + return 0; } @@ -3057,6 +3088,7 @@ static int azx_runtime_idle(struct device *dev) #ifdef CONFIG_PM static const struct dev_pm_ops azx_pm = { SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + .suspend_late = azx_suspend_late, SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, azx_runtime_idle) }; @@ -3272,6 +3304,12 @@ static int azx_free(struct azx *chip) hda_display_power(false); hda_i915_exit(); } + +#ifdef CONFIG_PM + if (chip->pm_wq) + destroy_workqueue(chip->pm_wq); +#endif + kfree(chip); return 0; @@ -3515,6 +3553,9 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, }; struct azx *chip; int err; +#ifdef CONFIG_PM + char wqname[24]; +#endif *rchip = NULL; @@ -3581,6 +3622,18 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, /* continue probing in work context as may trigger request module */ INIT_WORK(&chip->probe_work, azx_probe_work); +#ifdef CONFIG_PM + sprintf(wqname, "hda-azx-pm-wq%d", card->number); + chip->pm_wq = create_workqueue("wqname"); + if (!chip->pm_wq) { + snd_printk(KERN_ERR "cannot create PM workqueue\n"); + azx_free(chip); + return -ENOMEM; + } + + INIT_WORK(&chip->suspend_work, azx_suspend_work); +#endif + *rchip = chip; return 0; -- 1.8.1.2

3 5

Re: [alsa-devel] [PATCH] usb-audio: fix warning
by Takashi Iwai 05 Dec '13

05 Dec '13

At Thu, 5 Dec 2013 14:32:43 -0500 (EST), Mikulas Patocka wrote: > > Fix the following warning when optimizing for size with gcc-4.6.4: > sound/usb/mixer_quirks.c:1514:6: warning: 'err' may be used uninitialized in this function [-Wuninitialized] > > Signed-off-by: Mikulas Patocka <mpatocka(a)redhat.com> Thanks, applied. Takashi > > --- > sound/usb/mixer_quirks.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > Index: linux-3.12.3/sound/usb/mixer_quirks.c > =================================================================== > --- linux-3.12.3.orig/sound/usb/mixer_quirks.c 2013-12-05 20:21:18.000000000 +0100 > +++ linux-3.12.3/sound/usb/mixer_quirks.c 2013-12-05 20:21:23.000000000 +0100 > @@ -1520,7 +1520,7 @@ static int snd_microii_controls_create(s > return err; > } > > - return err; > + return 0; > } > > int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) >

1 0

[alsa-devel] [PATCH] ASoC: wm8962: Enable SYSCLK provisonally before fetching generated DSPCLK_DIV
by Nicolin Chen 05 Dec '13

05 Dec '13

DSPCLK_DIV can be only generated correctly after enabling SYSCLK. But if the current bias_level hasn't reached SND_SOC_BIAS_ON, DAPM won't enable SYSCLK, which would cause the calculation result from DSPCLK_DIV invalid since bit DSPCLK_DIV will be finally turned to its true value after DAPM enables SYSCLK while the driver won't calculate it again for the current instance. In this circumstance, a playback which needs non-zero DSPCLK_DIV would be distorted due to unexpected clock frequency resulted from an invalid DSPCLK_DIV value. So this patch provisionally enables the SYSCLK to get a valid DSPCLK_DIV for calculation and then disables it afterward. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/codecs/wm8962.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 07da601..97db3b4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2439,7 +2439,20 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8962_CLOCKING_4, WM8962_SYSCLK_RATE_MASK, clocking4); + /* DSPCLK_DIV can be only generated correctly after enabling SYSCLK. + * So we here provisionally enable it and then disable it afterward + * if current bias_level hasn't reached SND_SOC_BIAS_ON. + */ + if (codec->dapm.bias_level != SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA); + dspclk = snd_soc_read(codec, WM8962_CLOCKING1); + + if (codec->dapm.bias_level != SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA_MASK, 0); + if (dspclk < 0) { dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk); return; -- 1.8.4

3 3

[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: Do FLL configuration in hw_params() and hw_free()
by Nicolin Chen 05 Dec '13

05 Dec '13

Previously, we couldn't use hw_params() and hw_free() to open and close FLL becuase there might be race between two simmultaneous substreams and cause FLL configuration being changed and accordingly mulfunction. So we adopted DAPM to control it since there won't be any race in DAPM. However, if we want to playback a different sample rate file, we need to wait for DAPM to change its bias_level to reconfigure FLL. But after we introduced full symmetry protection in the soc-pcm, we don't need to worry about the race any more. And the instance by using hw_params() and hw_free() to control FLL would allow us to support flexible use cases, 'aplay -Dhw:0 44k16bit.wav 48k24bit.wav 32k32bit.wav' for example. Thus this patch mainly moves FLL configuration from set_bias_level() to hw_params() and hw_free() so as to enchance the sound card's capability. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/imx-wm8962.c | 147 +++++++++++++++++++++------------------------ 1 file changed, 70 insertions(+), 77 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 61e4885..df67a27 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -28,6 +28,9 @@ #define DAI_NAME_SIZE 32 +#define IMX_WM8962_DAI_FMT \ + (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM) + struct imx_wm8962_data { struct snd_soc_dai_link dai; struct snd_soc_card card; @@ -49,92 +52,84 @@ static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; -static int sample_rate = 44100; -static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE; static int imx_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - sample_rate = params_rate(params); - sample_format = params_format(params); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + snd_pcm_format_t sample_format = params_format(params); + u32 pll_out, sample_rate = params_rate(params); + int ret = 0; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, IMX_WM8962_DAI_FMT); + if (ret) { + dev_err(dev, "failed to set codec dai fmt: %d\n", ret); + return ret; + } + + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, + data->clk_frequency, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, + pll_out, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_hifi_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 imx_priv *priv = &card_priv; + struct device *dev = &priv->pdev->dev; + int ret; + + /* Don't diable FLL if still having multiple substreams running */ + if (codec_dai->active != 1) + return 0; + + /* Switch to MCLK as sysclk once so as to disable FLL */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + /* Disable FLL so that we can reset its output freq later */ + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + WM8962_FLL_MCLK, 0, 0); + if (ret < 0) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } return 0; } static struct snd_soc_ops imx_hifi_ops = { .hw_params = imx_hifi_hw_params, + .hw_free = imx_hifi_hw_free, }; -static int imx_wm8962_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); - struct device *dev = &priv->pdev->dev; - unsigned int pll_out; - int ret; - - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = sample_rate * 384; - else - pll_out = sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, - pll_out); - if (ret < 0) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, - "failed to switch away from FLL: %d\n", - ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - } - break; - - default: - break; - } - - dapm->bias_level = level; - - return 0; -} - static int imx_wm8962_late_probe(struct snd_soc_card *card) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; @@ -248,8 +243,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); data->dai.platform_of_node = ssi_np; data->dai.ops = &imx_hifi_ops; - data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; + data->dai.dai_fmt = IMX_WM8962_DAI_FMT; data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); @@ -264,7 +258,6 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); data->card.late_probe = imx_wm8962_late_probe; - data->card.set_bias_level = imx_wm8962_set_bias_level; ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { -- 1.8.4

2 6

[alsa-devel] [PATCH] ASoC: samsung: Ensure DMA data is initialised for secondary DAI
by Mark Brown 05 Dec '13

05 Dec '13

From: Mark Brown <broonie(a)linaro.org> While testing the upcoming dmaengine conversion for the driver Padma noticed that the DMA data for the secondard DAI is not being correctly initialised and provided a fix on top of the conversion. Also do this fix for mainline. Reported-by: Padmavathi Venna <padma.v(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> Cc: stable(a)vger.kernel.org --- Compile tested only, I don't have hardware to test. sound/soc/samsung/i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4f1655..f9f2227421ea 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -946,8 +946,11 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; - if (other && other->clk) /* If this is probe on secondary */ + if (other && other->clk) { /* If this is probe on secondary */ + snd_soc_dai_init_dma_data(dai, &other->sec_dai->dma_playback, + NULL); goto probe_exit; + } i2s->addr = ioremap(i2s->base, 0x100); if (i2s->addr == NULL) { -- 1.8.5.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: add DT binding document for MAX98090 CODEC
by Stephen Warren 05 Dec '13

05 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> This binding mainly serves to document the list of input and output pins that may be used in a sound card's audio routing table. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This series doesn't depend on anything; it could be applied anywhere in the ASoC tree. .../devicetree/bindings/sound/max98090.txt | 43 ++++++++++++++++++++++ 1 file changed, 43 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98090.txt diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt new file mode 100644 index 000000000000..e4c8b36dcf89 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -0,0 +1,43 @@ +MAX98090 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "maxim,max98090". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Pins on the device (for linking into audio routes): + + * MIC1 + * MIC2 + * DMICL + * DMICR + * IN1 + * IN2 + * IN3 + * IN4 + * IN5 + * IN6 + * IN12 + * IN34 + * IN56 + * HPL + * HPR + * SPKL + * SPKR + * RCVL + * RCVR + * MICBIAS + +Example: + +audio-codec@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(H, 4) GPIO_ACTIVE_HIGH>; +}; -- 1.8.1.5

2 6

Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by Raymond Yau 05 Dec '13

05 Dec '13

> >>>>>>>>> I just changed my RadeonHD 4670 for a 7870, and now when I type >>>>>>>>> "aplay -l" I get many HDMI codecs instead of just one like for the >>>>>>>>> previous card. >>>>>>>>> >>>>>>>> >>>>> How many hdmi , dvi or display port connectors do your card have ? >>>>> >>>>> 1 hdmi, 1 vga, 2 tiny ports that I cannot name. >>>> >>> are these mini display port ? http://en.wikipedia.org/wiki/Mini_DisplayPort > >>>>> Specification seem support up to 6 displays >>>>> >>>> I believe these are virtual HDMI ouputs, not physical ones and total >>>> for 6 in aplay -l and for 4 in aplay -L. >>>> >>> >>> are you using latest alsa-lib ? >>> >>> >>> http://git.alsa-project.org/?p=alsa-lib.git;a=commit;h=383912ef83cb2d583523… >>> >>> Good point, I'm on a 3.13-rc2 kernel but totally forgot about the lib >> part. >> I have the 1.0.27.2 lib, let me get the git version. >> > > Alright exactly the same behavior with -git. >

1 0

[alsa-devel] [PATCH] ASoC: samsung: Initialize the dma_data for secondary dai
by Padmavathi Venna 05 Dec '13

05 Dec '13

This patch initializes the dma_data for the secondary dai. Without this patch there is crash due to NULL pointer dereference during boot time on systems where secondary dai is initialized. Below is the crash log: [ 1.630000] smdk-audio-wm8894 sound.7: wm8994-aif1 <-> 3830000.i2s mapping ok [ 1.630000] Unable to handle kernel NULL pointer dereference at virtual address 00000010 [ 1.630000] pgd = c0004000 [ 1.630000] [00000010] *pgd=00000000 [ 1.630000] Internal error: Oops: 5 [#1] PREEMPT SMP ARM [ 1.630000] Modules linked in: ... ... [<c0342d6c>] (dmaengine_pcm_new+0x284/0x2ac) from [<c0341ae8>] (soc_new_pcm+0x2fc/0x3f4) [<c0341ae8>] (soc_new_pcm+0x2fc/0x3f4) from [<c0338558>] (snd_soc_register_card+0x1898/0x18d8) [<c0338558>] (snd_soc_register_card+0x1898/0x18d8) from [<c034cc34>] (smdk_audio_probe+0xa8/0xe8) [<c034cc34>] (smdk_audio_probe+0xa8/0xe8) from [<c0252368>] (platform_drv_probe+0x18/0x1c) [<c0252368>] (platform_drv_probe+0x18/0x1c) from [<c02510a4>] (driver_probe_device+0xf4/0x204) [<c02510a4>] (driver_probe_device+0xf4/0x204) from [<c0251240>] (__driver_attach+0x8c/0x90) [<c0251240>] (__driver_attach+0x8c/0x90) from [<c024f70c>] (bus_for_each_dev+0x54/0x88) [<c024f70c>] (bus_for_each_dev+0x54/0x88) from [<c0250724>] (bus_add_driver+0xe0/0x2b8) [<c0250724>] (bus_add_driver+0xe0/0x2b8) from [<c0251868>] (driver_register+0x78/0xf4) [<c0251868>] (driver_register+0x78/0xf4) from [<c00086e8>] (do_one_initcall+0x34/0x150) [<c00086e8>] (do_one_initcall+0x34/0x150) from [<c0543c30>] (kernel_init_freeable+0x12c/0x1d0) [<c0543c30>] (kernel_init_freeable+0x12c/0x1d0) from [<c03ebf9c>] (kernel_init+0x8/0xe8) [<c03ebf9c>] (kernel_init+0x8/0xe8) from [<c000e978>] (ret_from_fork+0x14/0x3c) Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- This is made based on Mark Brown for-next branch of sound.git This patch is based on Mark Brown below patches for generic dma engine support for samsung audio IPs. ASoC: samsung: Use ASoC dmaengine code where possible ASoC: samsung: Provide helper for DMA init sound/soc/samsung/dmaengine.c | 3 ++- sound/soc/samsung/i2s.c | 5 ++++- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index ad0a371..3f60479 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -67,7 +67,8 @@ EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); int samsung_asoc_dma_platform_register(struct device *dev) { return snd_dmaengine_pcm_register(dev, &samsung_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_COMPAT); + SND_DMAENGINE_PCM_FLAG_COMPAT | + SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME); } EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 67d9fa9..0bcd7fb 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -946,8 +946,11 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; - if (other && other->clk) /* If this is probe on secondary */ + if (other && other->clk) { /* If this is probe on secondary */ + samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, + NULL); goto probe_exit; + } i2s->addr = ioremap(i2s->base, 0x100); if (i2s->addr == NULL) { -- 1.7.4.4

3 3

Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by Raymond Yau 05 Dec '13

05 Dec '13

>>>> I just changed my RadeonHD 4670 for a 7870, and now when I type "aplay >>>> -l" I get many HDMI codecs instead of just one like for the previous card. >>>> >>> How many hdmi , dvi or display port connectors do your card have ? Specification seem support up to 6 displays >>> >>> Oh interesting, but I was just using the standard HDMI connector of the >> VGA card without connecting my card to the motherboard by more than PCI-E. >> Also, spdif would not get me 6 channels in LCPM, so I don't think that >> was it but I could be wrong. >> >>> >>> >>> Is there any reason for you to disable on-board hda controller and alc888 as you don't have any capture device anymore ? pactl list Do pulseaudio server select null sink or hdmi.monitor as the default source ? >>>> When I type "aplay -L" I get one more codec that I can see it in the >>>> pavucontrol profile (not sure which one as the names are irrelevant) >>>> >>>> >>>> if in default.pa I specify the alsa sink device=hw:0,3 directly, I do >>>> get sound, but then in pavucontrol I do not get the advanced sub-menu >>>> anymore which makes me think something is not working properly... >>>> >>>> For now I have settled on this hack instead: >>>> >>>> >>>> I changed the extra-hdmi.conf in >>>> /usr/share/pulseaudio/alsa-mixer/profile-sets file to : >>>> >>>> [Mapping hdmi-surround-extra0] >>>> description = Digital Surround 5.1 (HDMI 0) >>>> device-strings = hdmi:%f,0 >>>> paths-output = hdmi-output-0 >>>> channel-map = >>>> front-left,front-right,rear-left,rear-right,front-center,lfe >>>> priority = 1 >>>> direction = output >>>> >>>> [Mapping hdmi-surround-extra3] >>>> description = Digital Surround 5.1 (HDMI 3) >>>> device-strings = hdmi:%f,0 >>>> paths-output = hdmi-output-3 >>>> channel-map = >>>> front-left,front-right,rear-left,rear-right,front-center,lfe >>>> priority = 1 >>>> direction = output >>>> >>>> I only need one of these two, somehow they both work... >>>> >>>> Now everything works for me, but since it's not out of the box, I am >>>> guessing there might be something out there to fix. >>>> >>>> >>>> >>> >>> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >>> >>> Hmmm I am not sure what you meant by that. >> >> > Here is the alsa-info. > strange that all jacks return true control.1 { iface CARD name 'HDMI/DP,pcm=3 Jack' value true comment { access read type BOOLEAN count 1 } } ... control.31 { iface CARD name 'HDMI/DP,pcm=11 Jack' value true comment { access read type BOOLEAN count 1 } } cat /proc/asound/card0/eld#*.0 does your hdmi montior name appear in one of the six eld [ 5.468413] [drm] Connector 2: [ 5.468414] [drm] HDMI-A-1 [ 5.468415] [drm] HPD1 Does the graphic driver show which hdmi monitor connect to which ports in system log ?

1 0

[alsa-devel] [PATCH] ALSA: hda - fix mic issues on Acer Aspire E-572
by Oleksij Rempel 05 Dec '13

05 Dec '13

This patch add quirk for Acer Aspire E-572: - fix external mic - limit mic boost for internal mic with maximal noise level of -24dB Signed-off-by: Oleksij Rempel <linux(a)rempel-privat.de> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 59c36de..f3041a0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3902,6 +3902,7 @@ enum { ALC269_FIXUP_ASUS_X101, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, + ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572, ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ASUS_ZENBOOK, @@ -4164,6 +4165,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC271_FIXUP_AMIC_MIC2, }, + [ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC271_FIXUP_HP_GATE_MIC_JACK, + }, [ALC269_FIXUP_ACER_AC700] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -4261,6 +4268,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), + SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), -- 1.8.3.2

2 1

[alsa-devel] alsa/asoundlib.h file not found
by Appalayagari Sreedhar 05 Dec '13

05 Dec '13

Hi, I have integrated the codec driver and the kernel log shows the sound card is detected. now i want to test the codec. i tried the sample test code for capture and playback. i am getting a compilation errors due to the asoundlib.h. please find the error log for my application. main.c:2:23: error: asoundlib.h: No such file or directory main.c: In function 'main': main.c:7: error: 'snd_pcm_t' undeclared (first use in this function) main.c:7: error: (Each undeclared identifier is reported only once main.c:7: error: for each function it appears in.) main.c:7: error: 'pcm_handle' undeclared (first use in this function) main.c:10: error: 'snd_pcm_stream_t' undeclared (first use in this function) main.c:10: error: expected ';' before 'stream' main.c:15: error: 'snd_pcm_hw_params_t' undeclared (first use in this function) main.c:15: error: 'hwparams' undeclared (first use in this function) main.c:22: warning: 'return' with a value, in function returning void main.c:34: error: 'snd_pcm_uframes_t' undeclared (first use in this function) main.c:34: error: expected ';' before 'periodsize' main.c:46: error: 'SND_PCM_ACCESS_RW_INTERLEAVED' undeclared (first use in this function) main.c:48: warning: 'return' with a value, in function returning void main.c:52: error: 'SND_PCM_FORMAT_S16_LE' undeclared (first use in this function) main.c:54: warning: 'return' with a value, in function returning void main.c:62: warning: 'return' with a value, in function returning void main.c:65:19: warning: missing terminating " character main.c:65: error: missing terminating " character main.c:66: error: expected expression before '==' token main.c:66: error: stray '\' in program main.c:66:31: warning: missing terminating " character main.c:66: error: missing terminating " character main.c:67: error: expected ';' before '}' token main.c:72: warning: 'return' with a value, in function returning void main.c:78: warning: 'return' with a value, in function returning void main.c:86: error: 'periodsize' undeclared (first use in this function) main.c:88: warning: 'return' with a value, in function returning void main.c:99: warning: 'return' with a value, in function returning void main.c:112: warning: incompatible implicit declaration of built-in function 'malloc' main.c:115: error: 'num_frames' undeclared (first use in this function) main.c:5: warning: return type of 'main' is not 'int' please suggest what needs to be done to resolve this issue asap. Thank you, Sreedhar.

5 4

[alsa-devel] [PATCH] ASoC: imx-ssi: Check the return value from clk_prepare_enable()
by Fabio Estevam 05 Dec '13

05 Dec '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> clk_prepare_enable() may fail, so let's check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-ssi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 6336757..df552fa 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -535,7 +535,9 @@ static int imx_ssi_probe(struct platform_device *pdev) ret); goto failed_clk; } - clk_prepare_enable(ssi->clk); + ret = clk_prepare_enable(ssi->clk); + if (ret) + goto failed_clk; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); ssi->base = devm_ioremap_resource(&pdev->dev, res); -- 1.8.1.2

2 1

[alsa-devel] [PATCH v2 0/8] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Tomasz Figa 04 Dec '13

04 Dec '13

Since the patches extending support of amba-pl08x DMA engine driver to PL080S DMA engine (PL080 modified by Samsung) found in Samsung S3C64xx SoCs got merged into Linux 3.12, it is a good time to migrate the platform to use this driver and drop the legacy DMA code. Due to changes scattered across different areas of kernel, patches are based on merged 4 branches: - for-next of Kukjin Kim's Samsung tree, - clk-next of Mike Turquette's clock tree, - for-next of Mark Brown's SPI tree, - for-next of Mark Brown's ASoC tree. To ease testing I have prepared a branch in my private tree for anyone willing to check the patches out: git://github.com/tom3q/linux.git s3c64xx-pl08x-patches On S3C64xx-based Mini6410 and SMDK6410 boards, with I2S audio playback and capture (including full duplex operation) and also SPI using spidev: Tested-by: Tomasz Figa <tomasz.figa(a)gmail.com> Changes since v1: - Dropped the now merged patches adding PL080s support amba-pl08x driver, - Rebased onto current for-next branches of respective trees. - Stopped hijacking pl330_filter() symbol to plug pl08_filter_id() as channel filtering callback on S3C64xx. Changes since RFC v2: - Added clkdev lookups to old clock driver. - Added patches removing the old DMA driver and any remaining code needed by it. - Fixed DMA support for SPI. - Added a word about PL080S to amba-pl08x.c file header. - Changed definition of LLI words from enums to macros. - Extended debugging messages to handle PL080S variant as well. - Little cleanup of LLI dumping code. - Added check for peripheral flow control, which is unsupported by PL080S to dma_set_runtime_config. - Corrected transfer size mask of PL080S. Changes since RFC v1: - Returned to original way of storing quirks as booleans, as suggested by Russell, Linus and Arnd. - Added reg_config field to pl08x_phy_chan struct, which stores variant-specific address of channel config register, as suggested by Russell. - Simplified handling of extended maximum transfer size of PL080S (no more conditional passing of 0 as length to pl08x_cctl_bits()). - Reworked LLI handling in the driver to stop casting arbitrary memory to a struct and allow different word count of LLI entry, as suggested by Linus. - Removed AMBA ID override from S3C64xx PL080 initialization code. - Fixed brokenness of Samsung DMA wrapper API, which caused cyclic buffers to be queued multiple times when DMA engine is used. - Included patch adding clock aliases for DMA engines (depends on Common Clock Framework driver for S3C64xx). - Fixed several minor stylistic issues. Tomasz Figa (8): spi: s3c64xx: Do not require legacy DMA API in case of S3C64XX ASoC: Samsung: Do not require legacy DMA API in case of S3C64XX ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver clk: samsung: s3c64xx: Add aliases for DMA clocks ARM: s3c64xx: Remove legacy DMA driver clk: samsung: s3c64xx: Remove clock aliases of old DMA driver spi: s3c64xx: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled ASoC: Samsung: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled arch/arm/Kconfig | 1 + arch/arm/mach-s3c64xx/Kconfig | 7 +- arch/arm/mach-s3c64xx/Makefile | 2 +- arch/arm/mach-s3c64xx/common.h | 5 + arch/arm/mach-s3c64xx/dma.c | 762 ------------------------------- arch/arm/mach-s3c64xx/include/mach/dma.h | 144 ++---- arch/arm/mach-s3c64xx/pl080.c | 244 ++++++++++ arch/arm/plat-samsung/devs.c | 10 +- arch/arm/plat-samsung/dma-ops.c | 8 +- drivers/clk/samsung/clk-s3c64xx.c | 4 +- drivers/spi/Kconfig | 2 +- sound/soc/samsung/Kconfig | 2 +- 12 files changed, 316 insertions(+), 875 deletions(-) delete mode 100644 arch/arm/mach-s3c64xx/dma.c create mode 100644 arch/arm/mach-s3c64xx/pl080.c -- 1.8.3.2

5 16

[alsa-devel] 3.13 regression on panda board
by Takashi Iwai 04 Dec '13

04 Dec '13

Hi Peter (or anyone who is responsible): could you update the bugzilla report below? https://bugzilla.kernel.org/show_bug.cgi?id=66451 thanks, Takashi

3 6

[alsa-devel] [PATCH 07/10] ASoC: ux500: Store DMA data in the DAI differently in the pdata and DT case
by Lee Jones 04 Dec '13

04 Dec '13

In this patch we do two things. Firstly, instead of open coding the store of DMA data in to the DAI for later use, we use the API provided. Secondly we create and store similar DMA data for the DT case, only this time we use 'struct snd_dmaengine_dai_dma_data' which is provided by the core for this very reason. Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_msp_dai.c | 42 +++++++++++++++++++++++++++++++++++++++-- 1 file changed, 40 insertions(+), 2 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index c6fb5cc..8d1b01f 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -17,12 +17,14 @@ #include <linux/bitops.h> #include <linux/platform_device.h> #include <linux/clk.h> +#include <linux/of.h> #include <linux/regulator/consumer.h> #include <linux/mfd/dbx500-prcmu.h> #include <linux/platform_data/asoc-ux500-msp.h> #include <sound/soc.h> #include <sound/soc-dai.h> +#include <sound/dmaengine_pcm.h> #include "ux500_msp_i2s.h" #include "ux500_msp_dai.h" @@ -654,16 +656,52 @@ static int ux500_msp_dai_trigger(struct snd_pcm_substream *substream, return ret; } +static int ux500_msp_dai_of_probe(struct snd_soc_dai *dai) +{ + struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct snd_dmaengine_dai_dma_data *playback_dma_data; + struct snd_dmaengine_dai_dma_data *capture_dma_data; + + playback_dma_data = devm_kzalloc(dai->dev, + sizeof(*playback_dma_data), + GFP_KERNEL); + if (!playback_dma_data) + return -ENOMEM; + + capture_dma_data = devm_kzalloc(dai->dev, + sizeof(*capture_dma_data), + GFP_KERNEL); + if (!capture_dma_data) + return -ENOMEM; + + playback_dma_data->addr = drvdata->msp->playback_dma_data.tx_rx_addr; + capture_dma_data->addr = drvdata->msp->capture_dma_data.tx_rx_addr; + + playback_dma_data->maxburst = 4; + capture_dma_data->maxburst = 4; + + snd_soc_dai_init_dma_data(dai, playback_dma_data, capture_dma_data); + + return 0; +} + static int ux500_msp_dai_probe(struct snd_soc_dai *dai) { struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); + struct device_node *np = dai->dev->of_node; + int ret; - dai->playback_dma_data = &drvdata->msp->playback_dma_data; - dai->capture_dma_data = &drvdata->msp->capture_dma_data; + if (np) { + ret = ux500_msp_dai_of_probe(dai); + return ret; + } drvdata->msp->playback_dma_data.data_size = drvdata->slot_width; drvdata->msp->capture_dma_data.data_size = drvdata->slot_width; + snd_soc_dai_init_dma_data(dai, + &drvdata->msp->playback_dma_data, + &drvdata->msp->capture_dma_data); return 0; } -- 1.8.1.2

4 8

[alsa-devel] [PATCH 0/2] ASoC: sam9x5_wm8731: fix left right channel swap issue
by Bo Shen 04 Dec '13

04 Dec '13

This patch series fix sam9x5 wm8731 playback left right channel swap issue However, sam9x5 and wm8731 only can work in DSP A mode while can not work in I2S mode. Bo Shen (2): ASoC: atmel_ssc_dai: add dai trigger ops ASoC: sam9x5_wm8731: change to work in DSP A mode sound/soc/atmel/atmel_ssc_dai.c | 30 +++++++++++++++++++++++++++++- sound/soc/atmel/sam9x5_wm8731.c | 2 +- 2 files changed, 30 insertions(+), 2 deletions(-) -- 1.7.9.5

4 8

[alsa-devel] ALSA - No sound on MBA2,1
by Bruno Prémont 04 Dec '13

04 Dec '13

Hi Takashi, Since I started using Linux on a MBA2,1 I never had sound working. recently searching for some solution to get sound working I saw that some users had sound with 3.2.x kernels and reported it stopped working somewhere between 3.2 and 3.4. Thus I tried bisecting this and found two steps of losing sound support, a first one being losing sound output on built-in speaker, a second point where sound through headphone jack stopped working. Even with 3.2 non of the mute options is working (I did not try recording via build-in microphone) First commit where speaker stops working: commit 3a93897ea37cbb8277f8a4232c12c0c18168a7db Author: Takashi Iwai <tiwai(a)suse.de> Date: Fri Oct 28 01:16:55 2011 +0200 ALSA: hda - Manage unsol tags in hda_jack.c Manage the tags assigned for unsolicited events dynamically together with the jack-detection routines. Basically this is almost same as what we've done in patch_sigmatel.c. Assign the new tag number for each new unsol event, associate with the given NID and the action type, etc. With this change, now all pins looked over in snd_hda_jack_add_kctls() are actually enabled for detection now even if the pins aren't used for jack-retasking by the driver. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> First commit where headphone stops working (while speaker already does not work anymore): commit 1a97b7f22774b454531f013638b181803fba470f Author: Takashi Iwai <tiwai(a)suse.de> Date: Tue Feb 21 11:11:48 2012 +0100 ALSA: hda/realtek - Remove the last static quirks for ALC882 Resitance is futile. The remaining static model quirks for Apple machines with ALC882-compatible codecs are converted to the auto-parser now. We can remove all alc*_quirks.c finally. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Attached is the output of `alsa-info --with-devices` under current 3.12.2 kernel and under 3.2.53. As this all seems related to quirks being replaced with auto-parsing. Trying to revert the commits on top of 3.12.x looks unworkable. Is there some extra data I should try to provide or a workaround/fix I could try out? I'm booting in EFI mode using EFI stub (did backport it from 3.3 to 3.2 for bisecting/testing 3.2). Thanks, Bruno

2 7

[alsa-devel] [PATCH V2] ASoC: add RT286 CODEC driver
by bardliao＠realtek.com 04 Dec '13

04 Dec '13

From: Bard Liao <bardliao(a)realtek.com> This patch adds the ALC286 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- This is the initial codec driver of ALC286. I tested Headset recording and Speaker playback. I tested with snd_soc_register_codec because my kernel doesn't support devm_snd_soc_register_codec yet. --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1367 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 89 +++ 5 files changed, 1481 insertions(+) create mode 100644 include/sound/rt286.h create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 0000000..6c0c1c9 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool irq_en; /*IRQ enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..755a39d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -61,6 +61,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C + select SND_SOC_RT286 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -319,6 +320,9 @@ config SND_SOC_RT5631 config SND_SOC_RT5640 tristate +config SND_SOC_RT286 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..b31615c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -48,6 +48,7 @@ snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o +snd-soc-rt286-objs := rt286.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -181,6 +182,7 @@ obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 0000000..e7aef0d --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1367 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt286.h> + +#include "../../pci/hda/hda_codec.h" +#include "rt286.h" + +struct rt286_priv { + struct regmap *regmap; + struct rt286_platform_data pdata; + int sys_clk; +}; + +static unsigned int rt286_reg_cache[TOTAL_NODE_ID + 1] = { + [NODE_ID_DAC_OUT1] = 0x7f7f, + [NODE_ID_DAC_OUT2] = 0x7f7f, + [NODE_ID_SPDIF] = 0x0000, + [NODE_ID_ADC_IN1] = 0x4343, + [NODE_ID_ADC_IN2] = 0x4343, + [NODE_ID_MIC1] = 0x0000, + [NODE_ID_MIXER_IN] = 0x000b, + [NODE_ID_MIXER_OUT1] = 0x0002, + [NODE_ID_MIXER_OUT2] = 0x0000, + [NODE_ID_SPK_OUT] = 0x0000, + [NODE_ID_HP_OUT] = 0x0000, + [NODE_ID_MIXER_IN1] = 0x0000, + [NODE_ID_MIXER_IN2] = 0x0000, +}; + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + unsigned int buf = 0x0; + + if (reg <= 0xff) { /*read cache*/ + *value = rt286_reg_cache[reg]; + return 0; + } + reg = cpu_to_be32(reg); + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + u8 data[4]; + int ret; + + if (reg <= 0xff) { /*write cache*/ + rt286_reg_cache[reg] = value; + return 0; + } + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + data[2] = (reg >> 8) & 0xff; + data[3] = reg & 0xff; + + ret = i2c_master_send(client, data, 4); + if (ret == 4) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static unsigned int rt286_read(struct snd_soc_codec *codec, + unsigned int vid, + unsigned int nid, + unsigned int data) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb; + unsigned int buf = 0x0; + + verb = (nid << 20) | (vid << 8) | data; + + regmap_read(rt286->regmap, verb, &buf); + return buf; +} + +static int rt286_write(struct snd_soc_codec *codec, + unsigned int vid, + unsigned int nid, + unsigned int value) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + reg = (nid << 20) | (vid << 8) | value; + + return regmap_write(rt286->regmap, reg, 0); +} + +static int rt286_update_bits(struct snd_soc_codec *codec, unsigned int vid, + unsigned int nid, unsigned int data, + unsigned int mask, unsigned int value) +{ + unsigned int old, new; + int change, ret; + + old = rt286_read(codec, vid | 0x800, nid, data); + new = (old & ~mask) | (value & mask); + change = old != new; + + if (change) { + ret = rt286_write(codec, vid, nid, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_index_write(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, unsigned int data) +{ + int ret; + + ret = rt286_write(codec, AC_VERB_SET_COEF_INDEX, + wid, index); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + goto err; + } + ret = rt286_write(codec, AC_VERB_SET_PROC_COEF, + wid, data); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + + return 0; +err: + return ret; +} + +static unsigned int rt286_index_read(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index) +{ + int ret; + + ret = rt286_write(codec, AC_VERB_SET_COEF_INDEX, + wid, index); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + return ret; + } + return rt286_read(codec, AC_VERB_GET_PROC_COEF, + wid, index); +} + +static int rt286_index_update_bits(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, + unsigned int mask, unsigned int data) +{ + unsigned int old, new; + int change, ret; + + old = rt286_index_read(codec, wid, index); + new = (old & ~mask) | (data & mask); + change = old != new; + + if (change) { + ret = rt286_index_write(codec, wid, index, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_support_power_controls[] = { + NODE_ID_DAC_OUT1, + NODE_ID_DAC_OUT2, + NODE_ID_ADC_IN1, + NODE_ID_ADC_IN2, + NODE_ID_MIC1, + NODE_ID_DMIC1, + NODE_ID_DMIC2, + NODE_ID_SPK_OUT, + NODE_ID_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static const struct reg_default rt286_reg[] = { + { NODE_ID_DAC_OUT1, 0x7f7f }, + { NODE_ID_DAC_OUT2, 0x7f7f }, + { NODE_ID_SPDIF, 0x0000 }, + { NODE_ID_ADC_IN1, 0x4343 }, + { NODE_ID_ADC_IN2, 0x4343 }, + { NODE_ID_MIC1, 0x0000 }, + { NODE_ID_MIXER_IN, 0x000b }, + { NODE_ID_MIXER_OUT1, 0x0002 }, + { NODE_ID_MIXER_OUT2, 0x0000 }, + { NODE_ID_SPK_OUT, 0x0000 }, + { NODE_ID_HP_OUT, 0x0000 }, + { NODE_ID_MIXER_IN1, 0x0005 }, + { NODE_ID_MIXER_IN2, 0x0005 }, +}; + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NODE_ID_DAC_OUT1: + case NODE_ID_DAC_OUT2: + case NODE_ID_ADC_IN1: + case NODE_ID_ADC_IN2: + case NODE_ID_MIC1: + case NODE_ID_SPDIF: + case NODE_ID_MIXER_IN: + case NODE_ID_MIXER_OUT1: + case NODE_ID_MIXER_OUT2: + case NODE_ID_DMIC1: + case NODE_ID_DMIC2: + case NODE_ID_SPK_OUT: + case NODE_ID_HP_OUT: + case NODE_ID_MIXER_IN1: + case NODE_ID_MIXER_IN2: + return true; + default: + return false; + } +} + +int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + *hp = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_HP_OUT, 0); + if (*hp) { + /* power on HV,VERF */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x03, 0x1001, 0x0); + /* power LDO1 */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x08, 0x4, 0x4); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_MIC1, 0x24); + rt286_write(codec, AC_VERB_SET_COEF_INDEX, + 0x20, 0x50); + val = rt286_read(codec, AC_VERB_GET_PROC_COEF, + 0x20, 0x50); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + rt286_write(codec, AC_VERB_SET_COEF_INDEX, + 0x20, 0x50); + val = rt286_read(codec, + AC_VERB_GET_PROC_COEF, 0x20, 0x50); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + rt286_write(codec, + AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_MIC1, 0x20); + /* power off HV,VERF */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x03, 0x1001, 0x1001); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x0000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + } else { + *mic = false; + } + + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x20, 0x0060, 0x0000); + } else { + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x20, 0x0060, 0x0020); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x8000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0020); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x8000); + + *mic = false; + } + } else { + *hp = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_HP_OUT, 0); + *mic = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_MIC1, 0); + } + + + /* Clear IRQ */ + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x33, 0x1, 0x1); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_jack_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static int rt286_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int io = 0x8000; + + if (NODE_ID_ADC_IN1 == mc->reg) + io = 0; + ucontrol->value.integer.value[0] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x2000 | io) & 0x7f; + ucontrol->value.integer.value[1] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, io) & 0x7f; + + return 0; +} + +static int rt286_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vall, valr, io; + + if (NODE_ID_ADC_IN1 == mc->reg) + io = 0x4000; + else + io = 0x8000; + vall = ucontrol->value.integer.value[0]; + valr = ucontrol->value.integer.value[1]; + if (vall == valr) { + vall = vall | 0x3000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + } else { + vall = vall | 0x2000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + valr = valr | 0x1000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + } + + return 0; +} + +static int rt286_mic_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x0) & 0x3; + + return 0; +} + +static int rt286_mic_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int val; + + val = ucontrol->value.integer.value[0]; + val = val | 0x7000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, val); + + return 0; +} + +static int rt286_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + + ucontrol->value.integer.value[0] = + !(rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0xa000) & 0x80); + ucontrol->value.integer.value[1] = + !(rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x8000) & 0x80); + + return 0; +} + +static int rt286_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vall, valr, val = 0; + + vall = (!ucontrol->value.integer.value[0] << 7); + valr = (!ucontrol->value.integer.value[1] << 7); + if (vall) + val |= 0x8000; + if (valr) + val |= 0x0080; + + if (vall == valr) { + vall = vall | 0xb000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + } else { + vall = vall | 0xa000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + valr = valr | 0x9000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + } + snd_soc_update_bits(codec, mc->reg, 0x8080, val); + + return 0; +} + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_EXT_TLV("DAC0 Playback Volume", NODE_ID_DAC_OUT1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_DOUBLE_EXT_TLV("ADC0 Capture Volume", NODE_ID_ADC_IN1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_SINGLE_EXT_TLV("AMIC Volume", NODE_ID_MIC1, 0, 0x3, 0, + rt286_mic_gain_get, rt286_mic_gain_put, + mic_vol_tlv), + SOC_DOUBLE_EXT("Headphone Playback Switch", NODE_ID_HP_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), + SOC_DOUBLE_EXT("Speaker Playback Switch", NODE_ID_SPK_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), +}; + +static void hw_configure(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i; + + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D3); + for (i = 0; i < RT286_POWER_REG_LEN; i++) { + rt286_write(codec, AC_VERB_SET_POWER_STATE, + rt286_support_power_controls[i], AC_PWRST_D3); + } + + if (rt286->pdata.irq_en) + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x33, 0x0002, 0x0002); + + if (!rt286->pdata.cbj_en) { + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x50, 0x0000); + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x19, 0x0816); + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x20, 0x0000); + } + + mdelay(10); + + /* Set configuration default */ + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_MIC1, 0x00); + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_DMIC1, 0x00); + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_DMIC2, 0x00); + + mdelay(10); + + if (rt286->pdata.cbj_en) + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x4f, 0x5029); + else + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x4f, 0xb029); + + /*Power down LDO2*/ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x08, 0x0008, 0x0000); +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", NODE_ID_MIXER_OUT1, + RT286_M_FRONT_DAC_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", NODE_ID_MIXER_OUT1, + RT286_M_FRONT_REC_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Beep Switch", NODE_ID_MIXER_IN, + RT286_M_REC_BEEP_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", NODE_ID_MIXER_IN, + RT286_M_REC_LINE1_SFT, 1, 1), + SOC_DAPM_SINGLE("Mic1 Switch", NODE_ID_MIXER_IN, + RT286_M_REC_MIC1_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", NODE_ID_MIXER_IN, + RT286_M_REC_I2S_SFT, 1, 1), +}; + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "Dmic" +}; + +static int rt286_adc_values[] = { + 0, 5, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, NODE_ID_MIXER_IN1, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_VALUE_ENUM("ADC 0 source", rt286_adc0_enum); + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, NODE_ID_MIXER_IN2, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_VALUE_ENUM("ADC 1 source", rt286_adc1_enum); + +/* HP-OUT source */ +static const char * const rt286_hpo_src[] = { + "Front", "Surround" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, NODE_ID_HP_OUT, + RT286_HP_SEL_SFT, rt286_hpo_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static const char * const rt286_spo_src[] = { + "Front", "Surround" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_spo_enum, NODE_ID_SPK_OUT, + RT286_SPK_SEL_SFT, rt286_spo_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +/* SPDIF source */ +static const char * const rt286_spdif_src[] = { + "PCM-IN 0", "PCM-IN 1", "SP-OUT", "PP" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_spdif_enum, NODE_ID_SPDIF, + RT286_SPDIF_SEL_SFT, rt286_spdif_src); + +static const struct snd_kcontrol_new rt286_spdif_mux = +SOC_DAPM_ENUM("SPDIF source", rt286_spdif_enum); + + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_SPK_OUT, AC_PWRST_D0); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_SPK_OUT, 0x40); + rt286_write(codec, AC_VERB_SET_EAPD_BTLENABLE, + NODE_ID_SPK_OUT, 0x0002); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_EAPD_BTLENABLE, + NODE_ID_SPK_OUT, 0x0000); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_SPK_OUT, 0x00); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_SPK_OUT, AC_PWRST_D3); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + val = snd_soc_read(codec, NODE_ID_HP_OUT) & 0x8080; + switch (val) { + case 0x0: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xb000); + break; + case 0x8000: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0x9000); + break; + case 0x0080: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xa000); + break; + } + if (val != 0x8080) + mdelay(100); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xb080); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D3); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_DMIC1, 0x20); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC1, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC1, AC_PWRST_D3); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_DMIC1, 0x0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_hp_pow_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_HP_OUT, AC_PWRST_D0); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_HP_OUT, 0x40); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_HP_OUT, 0x00); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_HP_OUT, AC_PWRST_D3); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt286->pdata.cbj_en) { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x8000); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0020); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x8000); + } + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0f, 0x0020, 0x0000); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D0); + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7080); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D3); + break; + case SND_SOC_DAPM_POST_REG: + e = (struct soc_enum *)kcontrol->private_value; + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, e->reg, + snd_soc_read(codec, e->reg) & RT286_ADC_SEL_MASK); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_out_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + + switch (event) { + case SND_SOC_DAPM_POST_REG: + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, + e->reg, snd_soc_read(codec, e->reg) & 0x1); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", NODE_ID_DMIC1, 15, 0, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("DMIC2", NODE_ID_DMIC2, 15, 0, + NULL, 0, rt286_set_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", NODE_ID_ADC_IN1, 15, 0, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", NODE_ID_ADC_IN2, 15, 0, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + + /* Output Side */ + + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC 0", NULL, NODE_ID_DAC_OUT1, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC 1", NULL, NODE_ID_DAC_OUT2, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Output Mux */ + SND_SOC_DAPM_MUX_E("SPK Mux", SND_SOC_NOPM, 0, 0, + &rt286_spo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_MUX_E("HPO Mux", SND_SOC_NOPM, 0, 0, + &rt286_hpo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_SUPPLY("HP Power", SND_SOC_NOPM, + 0, 0, rt286_hp_pow_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_MUX("SPDIF Mux", SND_SOC_NOPM, 0, 0, + &rt286_spdif_mux), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", SND_SOC_NOPM, 0, 0, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_PGA_E("SPO", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("HPO", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", NULL, "SPK Mux"}, + {"HPO", NULL, "HPO Mux"}, + {"HPO", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO"}, +}; + +static int rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_format(params)) { + /* bit 6:4 Bits per Sample */ + case SNDRV_PCM_FORMAT_S16_LE: + d_len_code = 0; + val |= (0x1 << 4); + break; + case SNDRV_PCM_FORMAT_S32_LE: + d_len_code = 2; + val |= (0x4 << 4); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + d_len_code = 1; + val |= (0x2 << 4); + break; + case SNDRV_PCM_FORMAT_S24_LE: + d_len_code = 2; + val |= (0x3 << 4); + break; + case SNDRV_PCM_FORMAT_S8: + break; + default: + return -EINVAL; + } + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0018, d_len_code << 3); + pr_debug("format val = 0x%lx\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, 0, 0x407f, val); + else + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_ADC_IN1, 0, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, 0, 0x8000, 0x0); + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_ADC_IN1, 0, 0x8000, 0x0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + pr_debug("%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x0100, 0x0); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x49, 0x0020, 0x0020); + } else { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x0100, 0x0100); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x63, 0x4, 0x4); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x49, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + break; + case 12288000: + case 11289600: + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x8, 0x0); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0b, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x8, 0x8); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0b, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_dai_dfs(struct snd_soc_dai *dai, unsigned int fs) +{ + struct snd_soc_codec *codec = dai->codec; + + pr_debug("%s fs=%d\n", __func__, fs); + if (50 == fs) + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x1000, 0x1000); + else + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D0); + break; + + case SND_SOC_BIAS_OFF: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D3); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + hw_configure(codec); + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + rt286_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_dfs = rt286_set_dai_dfs, +}; + +struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .readable_reg = rt286_readable_register, + .cache_type = REGCACHE_NONE, +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt5640", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static struct acpi_device_id rt286_acpi_match[] = { + { "INT33CA", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + struct device *dev = &i2c->dev; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, + sizeof(struct rt286_priv), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + ret = devm_snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt5640", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .id_table = rt286_i2c_id, +}; + +static int __init rt286_modinit(void) +{ + return i2c_add_driver(&rt286_i2c_driver); +} + +module_init(rt286_modinit); + +static void __exit rt286_modexit(void) +{ + i2c_del_driver(&rt286_i2c_driver); +} + +module_exit(rt286_modexit); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 0000000..c579c53 --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,89 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT286_H__ +#define __RT286_H__ + +#define NODE_ID_AUDIO_FUNCTION_GROUP 0x01 +#define NODE_ID_DAC_OUT1 0x02 +#define NODE_ID_DAC_OUT2 0x03 +#define NODE_ID_ADC_IN1 0x09 +#define NODE_ID_ADC_IN2 0x08 +#define NODE_ID_MIXER_IN 0x0b +#define NODE_ID_MIXER_OUT1 0x0c +#define NODE_ID_MIXER_OUT2 0x0d +#define NODE_ID_DMIC1 0x12 +#define NODE_ID_DMIC2 0x13 +#define NODE_ID_SPK_OUT 0x14 +#define NODE_ID_MIC1 0x18 +#define NODE_ID_LINE1 0x1a +#define NODE_ID_BEEP 0x1d +#define NODE_ID_SPDIF 0x1e +#define NODE_ID_VENDOR_REGISTERS 0x20 +#define NODE_ID_HP_OUT 0x21 +#define NODE_ID_MIXER_IN1 0x22 +#define NODE_ID_MIXER_IN2 0x23 +#define TOTAL_NODE_ID 0x23 + +#define CONNECTION_INDEX_MIC1 0X0 +#define CONNECTION_INDEX_DMIC 0X5 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIFS, +}; + +int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic); + +#endif /* __RT286_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] ASUS Phoebus (CMI8888) HDA Support
by Geoffrey McRae 04 Dec '13

04 Dec '13

Hi, I have spent the better part of the last two days trying to add support to the hda-intel driver for this card and I have finally got it functional, not through any extra code though :( To get it going I added the extra device info into the device list and specified it as a generic device with no msi and LPIB position fix. Initially this did not work, probing for codec info failed, the interrupt was getting received, but the DMA address contained no data (all 0x0). After messing around a heap trying to find out why there was no DMA data, I discovered that if I forced the device into single_cmd mode it would operate, with very nasty scratchy audio, seems the IRQs were not getting through and the circular buffer was under running. If I made did something involving IRQs such as cat /proc/interrupts, it would briefly fix the problem... if I wrote a while loop to do just that, I would get mostly clean audio. There is another thread in this mailing list where someone else has tried all of this and ended up with the same output. I decided to try to identify why the device was not returning RIRB data in the DMA responses, and out of desperation setup a KVM with windows to try to perform mmio tracing to identify where the hda-intel driver was going wrong. This did not work, windows would not detect the codecs attached to the device, I think it is due to a lack of MSI support for passthrough devices. A positive thing did come out of this though, to do this I enabled MMIO in my BIOS, which instantly fixed the missing RIRB DMA data issue... I do not know enough about MMIO to understand why this fixed this. The device now detects the codec without single_cmd mode, but alas, still no audio, but also no errors... further experimentation I found that if I started the device in single_cmd mode and then went back to normal, I would get perfect audio now. Since I did this, I get perfect audio even without entering single_cmd mode. I have dumped the registers under windows and can confirm that during its initialization of the HDA device it does indeed use single_cmd mode before switching over to IRQ/DMA mode, there might be some standard init the driver is missing here. So, to make this card work, very little extra code will be required, just need to understand why enabling MMIO fixed the DMA issue as this is not a viable fix for the problem, not everyone has MMIO. Here is playback of a stereo stream while the device is working: [ 66.697114] hda_intel: Disabling MSI [ 66.697116] ALSA hda_intel.c:2553 Using LPIB position fix [ 66.697146] snd_hda_intel 0000:06:00.0: setting latency timer to 64 [ 66.697151] ALSA hda_intel.c:2796 chipset global capabilities = 0x4300 [ 66.697159] ALSA hda_intel.c:1175 Clearing TCSEL [ 66.724049] ALSA hda_intel.c:1031 codec_mask = 0x2 [ 66.724181] ALSA hda_intel.c:1484 codec #1 probed OK [ 66.734419] ALSA hda_generic.c:586 AUD_IN = 8 [ 66.734422] ALSA hda_generic.c:239 UNMUTE IN: NID=0x8 IDX=0x0 [ 66.734879] ALSA hda_generic.c:239 UNMUTE IN: NID=0x12 IDX=0x0 [ 66.735565] ALSA hda_generic.c:619 [Capture Source] NID=0x8, #SRC=1 [ 66.735567] ALSA hda_generic.c:622 [Front Mic] IDX=0x0 [ 66.735568] ALSA hda_generic.c:291 AUD_OUT found 2 [ 66.735570] ALSA hda_generic.c:239 UNMUTE IN: NID=0xc IDX=0x0 [ 66.735903] ALSA hda_generic.c:224 UNMUTE OUT: NID=0xc [ 66.736136] ALSA hda_generic.c:224 UNMUTE OUT: NID=0xc [ 66.736139] ALSA hda_generic.c:291 AUD_OUT found 6 [ 66.736569] ALSA hda_generic.c:254 CONNECT: NID=0x8 IDX=0x0 [ 66.736572] ALSA hda_generic.c:699 [Capture Switch] NID=0x8, DIR=IN, IDX=0x0 [ 66.736575] ALSA hda_generic.c:725 [Capture Volume] NID=0x8, DIR=IN, IDX=0x0 [ 66.736578] ALSA hda_generic.c:710 [Master Playback Switch] NID=0x2, DIR=OUT [ 66.736580] ALSA hda_generic.c:734 [Master Playback Volume] NID=0x2, DIR=OUT [ 76.125606] ALSA hda_intel.c:1848 azx_pcm_prepare: bufsize=0x10000, format=0x31 [ 76.125615] ALSA hda_codec.c:1414 hda_codec_setup_stream: NID=0x2, stream=0x7, channel=0, format=0x31 [ 93.308666] ALSA hda_codec.c:1477 hda_codec_cleanup_stream: NID=0x2 - Geoffrey McRae

2 8

[alsa-devel] [PATCH] ASoC: rcar: use devm_clk_get() instead of clk_get()
by Kuninori Morimoto 04 Dec '13

04 Dec '13

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index a821b14..85936ae 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -653,7 +653,7 @@ int rsnd_ssi_probe(struct platform_device *pdev, snprintf(name, RSND_SSI_NAME_SIZE, "ssi.%d", i); - clk = clk_get(dev, name); + clk = devm_clk_get(dev, name); if (IS_ERR(clk)) return PTR_ERR(clk); @@ -714,7 +714,6 @@ void rsnd_ssi_remove(struct platform_device *pdev, int i; for_each_rsnd_ssi(ssi, priv, i) { - clk_put(ssi->clk); if (rsnd_ssi_dma_available(ssi)) rsnd_dma_quit(priv, rsnd_mod_to_dma(&ssi->mod)); } -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 0/2] Only do snd_soc_dai_digital_mute() in hw_free()
by Nicolin Chen 04 Dec '13

04 Dec '13

This series of patches mainly fixes an issue of snd_soc_dai_digital_mute() operation after playback/capture is done. The current system has two of them in a normal PCM instance while one of them has an invalid condition. Thus we first fix the condition and then drop the redundant one. Changelog: v2: * Bisected into two patches. * Fixed indentation in PATCH-1. Nicolin Chen (2): ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free() ASoC: soc-pcm: Drop the redundant snd_soc_dai_digital_mute() in soc_pcm_close() sound/soc/soc-pcm.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) -- 1.8.4

2 4

[alsa-devel] ASoC fixes for v3.13
by Mark Brown 04 Dec '13

04 Dec '13

The following changes since commit dc1ccc48159d63eca5089e507c82c7d22ef60839: Linux 3.13-rc2 (2013-11-29 12:57:14 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc2 for you to fetch changes up to 29e248829dc7d44248c69bbd5d40eca152a50cab: Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/kirkwood', 'asoc/fix/omap', 'asoc/fix/rcar', 'asoc/fix/wm8731' and 'asoc/fix/wm8990' into asoc-linus (2013-12-03 18:09:00 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A smattering of fixes here, some core ones for the rate combination issues for things other than simple bitmasks, for readback of byte controls and for updating the power of value muxes plus a bunch of driver fixes of varying severity. The warning fix in the i.MX FIQ driver is fixing a warning introduced by a previous fix. ---------------------------------------------------------------- Bo Shen (2): ASoC: atmel: sam9x5_wm8731: fix oops when unload module ASoC: wm8731: fix dsp mode configuration Charles Keepax (2): ASoC: wm5110: Remove output OSR and PGA volume controls ASoC: core: Use consistent byte ordering in snd_soc_bytes_get Fabio Estevam (1): ASoC: fsl: imx-pcm-fiq: Remove unused 'runtime' variable Jarkko Nikula (1): ASoC: omap: n810: Convert to clk_prepare_enable/clk_disable_unprepare Jean-Francois Moine (2): ASoC: kirkwood: Fix invalid S/PDIF format ASoC: kirkwood: Fix erroneous double output while playing Kuninori Morimoto (1): ASoC: rcar: select REGMAP Lars-Peter Clausen (2): ASoC: pcm: Fix rate_max calculation ASoC: pcm: Always honor DAI min and max sample rate constraints Mark Brown (4): ASoC: wm8990: Mark the register map as dirty when powering down Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/kirkwood', 'asoc/fix/omap', 'asoc/fix/rcar', 'asoc/fix/wm8731' and 'asoc/fix/wm8990' into asoc-linus Oskar Schirmer (1): ASoC: fsl: imx-pcm-fiq: remove bogus period delta calculation Shawn Guo (1): ASoC: core: fix devres parameter in devm_snd_soc_register_card() Stephen Warren (1): ASoC: dapm: Use SND_SOC_DAPM_INIT_REG_VAL in SND_SOC_DAPM_MUX Wei Yongjun (1): ASoC: fsl: set correct platform drvdata in pcm030_fabric_probe() include/sound/soc-dapm.h | 3 ++- sound/soc/atmel/sam9x5_wm8731.c | 2 ++ sound/soc/codecs/wm5110.c | 25 ------------------------- sound/soc/codecs/wm8731.c | 4 ++-- sound/soc/codecs/wm8990.c | 2 ++ sound/soc/fsl/pcm030-audio-fabric.c | 3 ++- sound/soc/kirkwood/kirkwood-i2s.c | 22 +++++++++++++--------- sound/soc/omap/n810.c | 4 ++-- sound/soc/sh/Kconfig | 1 + sound/soc/soc-core.c | 4 ++-- sound/soc/soc-devres.c | 4 ++-- sound/soc/soc-pcm.c | 18 ++++++++++++------ 12 files changed, 42 insertions(+), 50 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Implement symmetric_channels and symmetric_samplebits
by Nicolin Chen 04 Dec '13

04 Dec '13

Since we introduced symmetric_channels and symmetric_samplebits, we implement these two features to fsl_ssi so as to drop some no-more-needed code and make the driver neat and clean. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 79 ++++++------------------------------------------- 1 file changed, 9 insertions(+), 70 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 90ff107..f9f4569 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -119,8 +119,6 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI - * @first_stream: pointer to the stream that was opened first - * @second_stream: pointer to second stream * @playback: the number of playback streams opened * @capture: the number of capture streams opened * @cpu_dai: the CPU DAI for this device @@ -132,8 +130,6 @@ struct fsl_ssi_private { struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; - struct snd_pcm_substream *first_stream; - struct snd_pcm_substream *second_stream; unsigned int fifo_depth; struct snd_soc_dai_driver cpu_dai_drv; struct device_attribute dev_attr; @@ -438,54 +434,13 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; - /* - * If this is the first stream opened, then request the IRQ - * and initialize the SSI registers. + /* First, we only do fsl_ssi_setup() when SSI is going to be active. + * Second, fsl_ssi_setup was already called by ac97_init earlier if + * the driver is in ac97 mode. */ - if (!ssi_private->first_stream) { - ssi_private->first_stream = substream; - - /* - * fsl_ssi_setup was already called by ac97_init earlier if - * the driver is in ac97 mode. - */ - if (!ssi_private->imx_ac97) - fsl_ssi_setup(ssi_private); - } else { - if (synchronous) { - struct snd_pcm_runtime *first_runtime = - ssi_private->first_stream->runtime; - /* - * This is the second stream open, and we're in - * synchronous mode, so we need to impose sample - * sample size constraints. This is because STCCR is - * used for playback and capture in synchronous mode, - * so there's no way to specify different word - * lengths. - * - * Note that this can cause a race condition if the - * second stream is opened before the first stream is - * fully initialized. We provide some protection by - * checking to make sure the first stream is - * initialized, but it's not perfect. ALSA sometimes - * re-initializes the driver with a different sample - * rate or size. If the second stream is opened - * before the first stream has received its final - * parameters, then the second stream may be - * constrained to the wrong sample rate or size. - */ - if (first_runtime->sample_bits) { - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - first_runtime->sample_bits, - first_runtime->sample_bits); - } - } - - ssi_private->second_stream = substream; - } + if (!dai->active && !ssi_private->imx_ac97) + fsl_ssi_setup(ssi_private); return 0; } @@ -615,23 +570,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -/** - * fsl_ssi_shutdown: shutdown the SSI - * - * Shutdown the SSI if there are no other substreams open. - */ -static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - - if (ssi_private->first_stream == substream) - ssi_private->first_stream = ssi_private->second_stream; - - ssi_private->second_stream = NULL; -} - static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(dai); @@ -647,7 +585,6 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, - .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, }; @@ -722,7 +659,6 @@ static int fsl_ssi_ac97_trigger(struct snd_pcm_substream *substream, int cmd, static const struct snd_soc_dai_ops fsl_ssi_ac97_dai_ops = { .startup = fsl_ssi_startup, - .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_ac97_trigger, }; @@ -947,8 +883,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) } /* Are the RX and the TX clocks locked? */ - if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) + if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; + ssi_private->cpu_dai_drv.symmetric_samplebits = 1; + } /* Determine the FIFO depth. */ iprop = of_get_property(np, "fsl,fifo-depth", NULL); -- 1.8.4

2 4

[alsa-devel] [PATCH] Add no-mmu support for alsa-utils
by Aaron Wu 04 Dec '13

04 Dec '13

From: Aaron Wu <Aaron.wu(a)analog.com> Signed-off-by: Aaron Wu <Aaron.wu(a)analog.com> Add no-mmu support for alsa-utils mainly for Blackfin --- alsactl/init_utils_run.c | 4 ++++ alsaloop/alsaloop.c | 8 ++++++++ configure.in | 1 + 3 files changed, 13 insertions(+) diff --git a/alsactl/init_utils_run.c b/alsactl/init_utils_run.c index dde490b..ac074e6 100644 --- a/alsactl/init_utils_run.c +++ b/alsactl/init_utils_run.c @@ -89,7 +89,11 @@ int run_program0(struct space *space, argv[0] = program; } + #ifdef HAVE_FORK pid = fork(); + #else + pid = vfork(); + #endif switch(pid) { case 0: /* child closes parent ends of pipes */ diff --git a/alsaloop/alsaloop.c b/alsaloop/alsaloop.c index 3b20682..1b1e540 100644 --- a/alsaloop/alsaloop.c +++ b/alsaloop/alsaloop.c @@ -861,14 +861,22 @@ int main(int argc, char *argv[]) logit(LOG_CRIT, "daemon() failed: %s\n", strerror(errno)); exit(EXIT_FAILURE); } + #ifdef HAVE_FORK i = fork(); + #else + i = vfork(); + #endif if (i < 0) { logit(LOG_CRIT, "fork() failed: %s\n", strerror(errno)); exit(EXIT_FAILURE); } if (i > 0) { /* wait(&i); */ + #ifdef HAVE_FORK exit(EXIT_SUCCESS); + #else + _exit(EXIT_SUCCESS); + #endif } } diff --git a/configure.in b/configure.in index 60da1fa..f289f9b 100644 --- a/configure.in +++ b/configure.in @@ -25,6 +25,7 @@ then AC_MSG_RESULT($CC) fi +AC_CHECK_FUNC([fork]) AC_PROG_CC dnl AC_PROG_CXX AC_PROG_INSTALL -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix missing ELD info when using jackpoll_ms parameter
by David Henningsson 04 Dec '13

04 Dec '13

In the case of using jackpoll_ms instead of unsol events, the jack was correctly detected, but ELD info was not refreshed on plug-in. And without ELD info, no proper restriction of pcm, which can in turn break sound output on some devices. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) @Takashi; feel free to add stable if you think it's worth it. (After all using the jackpoll_ms parameter is not something we usually do.) It's still a bug fix so sending to Linus tree should be okay, I believe. diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4098196..f0a8ea8 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1142,32 +1142,34 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); -static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +static void jack_callback(struct hda_codec *codec, struct hda_jack_tbl *jack) { struct hdmi_spec *spec = codec->spec; + int pin_idx = pin_nid_to_pin_index(spec, jack->nid); + if (pin_idx < 0) + return; + + if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) + snd_hda_jack_report_sync(codec); +} + +static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +{ int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int pin_nid; - int pin_idx; struct hda_jack_tbl *jack; int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; jack = snd_hda_jack_tbl_get_from_tag(codec, tag); if (!jack) return; - pin_nid = jack->nid; jack->jack_dirty = 1; _snd_printd(SND_PR_VERBOSE, "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA), + codec->addr, jack->nid, dev_entry, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); - pin_idx = pin_nid_to_pin_index(spec, pin_nid); - if (pin_idx < 0) - return; - - if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) - snd_hda_jack_report_sync(codec); + jack_callback(codec, jack); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -2080,7 +2082,8 @@ static int generic_hdmi_init(struct hda_codec *codec) hda_nid_t pin_nid = per_pin->pin_nid; hdmi_init_pin(codec, pin_nid); - snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); + snd_hda_jack_detect_enable_callback(codec, pin_nid, pin_nid, + codec->jackpoll_interval > 0 ? jack_callback : NULL); } return 0; } -- 1.7.9.5

2 5

[alsa-devel] Independent of model for HP
by Kailang 04 Dec '13

04 Dec '13

Hi Takashi, HP Chrome Book, the headset jack was difference between other chrome book. It need to manual control Mic jack detect. Thanks. Kailang

2 6

Re: [alsa-devel] [pulseaudio-discuss] What-did-you-plugin-dialog
by Raymond Yau 04 Dec '13

04 Dec '13

: > > > can post the output of alsa-info.sh after jack retasking ? > > Before retasking: > > http://www.alsa-project.org/db/?f=fd97b8194e205b4f91127bca74ad7c9b2cd93ab9 > > After retasking blue line in, pin ID 0x1a as Line out (Side): > > http://www.alsa-project.org/db/?f=d56f9a6d0692c9a8d5c88f43100a6cb78407c163 > > Note that retasking did not work: attempting to play a 8-channel wav > file through hw:1,0 resulted in an "Invalid argument" error. > > FWIW, here is what hdajackretask printed to the console: > > 0x11 0x99430130 > 0x12 0x4037c540 > 0x14 0x01014010 > 0x15 0x01011012 > 0x16 0x01016011 > 0x17 0x411111f0 > 0x18 0x01a19050 > 0x19 0x02a19060 > 0x1a 0x01014413 > 0x1b 0x02214020 > 0x1c 0x411111f0 > 0x1d 0x4045e601 > 0x1e 0x01452140 > 0x1f 0x411111f0 > 1 > > you need root privilege for hdajackretask to write the firmware file to /lib/firmware and modify /etc/modprobe.d/alsa-lbase.conf to use the firmware file can you try the patch which add "channel mode" control which allow you to switch from 6ch to 8ch it is not clear that the "Line jack" control is used for detect the blue jack state or reflect the line In is no longer available after it is retasked as output jack

2 7

[alsa-devel] [PATCH - plugins] oss: avoid conflict between reclev and igain mixer ctls
by Luca Bayer 04 Dec '13

04 Dec '13

To fix the following error that happens with Intel HDA driver. $ amixer ALSA lib simple_none.c:1549:(simple_add1) helem (MIXER,'Capture Volume',0,0,0) appears twice or more amixer: Mixer default load error: Invalid argument Signed-off-by: Luca Bayer <kometto(a)web.de> diff --git a/oss/ctl_oss.c b/oss/ctl_oss.c index b8847b1..b05ba18 100644 --- a/oss/ctl_oss.c +++ b/oss/ctl_oss.c @@ -52,7 +52,7 @@ static const char *const vol_devices[SOUND_MIXER_NRDEVICES] = { [SOUND_MIXER_CD] = "CD Playback Volume", [SOUND_MIXER_IMIX] = "Monitor Mix Playback Volume", [SOUND_MIXER_ALTPCM] = "Headphone Playback Volume", - [SOUND_MIXER_RECLEV] = "Capture Volume", + [SOUND_MIXER_RECLEV] = "Master Capture Volume", [SOUND_MIXER_IGAIN] = "Capture Volume", [SOUND_MIXER_OGAIN] = "Playback Volume", [SOUND_MIXER_LINE1] = "Aux Playback Volume",

1 0

[alsa-devel] [PATCH] ASoC: dmaengine: remove SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME
by Stephen Warren 04 Dec '13

04 Dec '13

From: Stephen Warren <swarren(a)nvidia.com> struct snd_dmaengine_pcm_config's new chan_names field implements the same feature, and nothing uses SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME, so remove it. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- This follows on from the patch "ASoC: dmaengine: add custom DMA config to snd_dmaengine_pcm_config" which I posted earlier today. include/sound/dmaengine_pcm.h | 6 ------ sound/soc/soc-generic-dmaengine-pcm.c | 13 +------------ 2 files changed, 1 insertion(+), 18 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 87e9d481d7b6..634b0fca3afa 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -61,7 +61,6 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @slave_id: Slave requester id for the DMA channel. * @filter_data: Custom DMA channel filter data, this will usually be used when * requesting the DMA channel. - * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes */ struct snd_dmaengine_dai_dma_data { @@ -70,7 +69,6 @@ struct snd_dmaengine_dai_dma_data { u32 maxburst; unsigned int slave_id; void *filter_data; - const char *chan_name; unsigned int fifo_size; }; @@ -100,10 +98,6 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * playback. */ #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) -/* - * The PCM streams have custom channel names specified. - */ -#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index ac16259490ee..819d5997eece 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -188,8 +188,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); const struct snd_dmaengine_pcm_config *config = pcm->config; - struct device *dev = rtd->platform->dev; - struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_substream *substream; size_t prealloc_buffer_size; size_t max_buffer_size; @@ -210,13 +208,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (!pcm->chan[i] && - (pcm->flags & SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) - pcm->chan[i] = dma_request_slave_channel(dev, - dma_data->chan_name); - if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, substream); @@ -290,9 +281,7 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, unsigned int i; const char *name; - if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | - SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || - !dev->of_node) + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node) return; if (config->dma_dev) { -- 1.8.1.5

1 0

[alsa-devel] [PATCH 1/2] ASoC: jz4740-i2s: Use managed resources
by Lars-Peter Clausen 03 Dec '13

03 Dec '13

Makes the code a bit shorter. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/jz4740-i2s.c | 85 ++++++++--------------------------------- 1 file changed, 15 insertions(+), 70 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 4c849a4..a0b6a85 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -432,91 +432,36 @@ static const struct snd_soc_component_driver jz4740_i2s_component = { static int jz4740_i2s_dev_probe(struct platform_device *pdev) { struct jz4740_i2s *i2s; + struct resource *mem; int ret; - i2s = kzalloc(sizeof(*i2s), GFP_KERNEL); - + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); if (!i2s) return -ENOMEM; - i2s->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!i2s->mem) { - ret = -ENOENT; - goto err_free; - } - - i2s->mem = request_mem_region(i2s->mem->start, resource_size(i2s->mem), - pdev->name); - if (!i2s->mem) { - ret = -EBUSY; - goto err_free; - } - - i2s->base = ioremap_nocache(i2s->mem->start, resource_size(i2s->mem)); - if (!i2s->base) { - ret = -EBUSY; - goto err_release_mem_region; - } + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->base)) + return PTR_ERR(i2s->base); - i2s->phys_base = i2s->mem->start; + i2s->phys_base = mem->start; - i2s->clk_aic = clk_get(&pdev->dev, "aic"); - if (IS_ERR(i2s->clk_aic)) { - ret = PTR_ERR(i2s->clk_aic); - goto err_iounmap; - } + i2s->clk_aic = devm_clk_get(&pdev->dev, "aic"); + if (IS_ERR(i2s->clk_aic)) + return PTR_ERR(i2s->clk_aic); - i2s->clk_i2s = clk_get(&pdev->dev, "i2s"); - if (IS_ERR(i2s->clk_i2s)) { - ret = PTR_ERR(i2s->clk_i2s); - goto err_clk_put_aic; - } + i2s->clk_i2s = devm_clk_get(&pdev->dev, "i2s"); + if (IS_ERR(i2s->clk_i2s)) + return PTR_ERR(i2s->clk_i2s); platform_set_drvdata(pdev, i2s); - ret = snd_soc_register_component(&pdev->dev, &jz4740_i2s_component, - &jz4740_i2s_dai, 1); - - if (ret) { - dev_err(&pdev->dev, "Failed to register DAI\n"); - goto err_clk_put_i2s; - } - - return 0; - -err_clk_put_i2s: - clk_put(i2s->clk_i2s); -err_clk_put_aic: - clk_put(i2s->clk_aic); -err_iounmap: - iounmap(i2s->base); -err_release_mem_region: - release_mem_region(i2s->mem->start, resource_size(i2s->mem)); -err_free: - kfree(i2s); - - return ret; -} - -static int jz4740_i2s_dev_remove(struct platform_device *pdev) -{ - struct jz4740_i2s *i2s = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - - clk_put(i2s->clk_i2s); - clk_put(i2s->clk_aic); - iounmap(i2s->base); - release_mem_region(i2s->mem->start, resource_size(i2s->mem)); - - kfree(i2s); - - return 0; + return devm_snd_soc_register_component(&pdev->dev, + &jz4740_i2s_component, &jz4740_i2s_dai, 1); } static struct platform_driver jz4740_i2s_driver = { .probe = jz4740_i2s_dev_probe, - .remove = jz4740_i2s_dev_remove, .driver = { .name = "jz4740-i2s", .owner = THIS_MODULE, -- 1.7.10.4

2 2

[alsa-devel] [PATCH - oxygen 1/1] Patch for the Xonar DG Patch by Roman Volkov
by Marius Graefe 03 Dec '13

03 Dec '13

I just tested Roman Volkovs oxygen patch for the xonar dg and it works like a charm, however I needed to change two lines in xonar_dg_mixer.c: 1) #include <linux/export.h> was missing which resulted in a warning when compiling. 2) I had to assign a .shortname in xonar_dg_model, otherwise my kernel (unmodified ubuntu 3.2.0-56-generic-pae) throws an oops, the backtrace led me to the missing string. Actually this is weird because the shortname is assigned in get_oxygen_model in oxygen.c, and alsamixer also shows that name later, but at some point in the kernel startup sequence it's NULL. Volkovs Xonar DG 4-Patch series is required! Signed-off-by: Marius Graefe <alsa(a)mgraefe.de> diff --git a/sound/pci/oxygen/xonar_dg_mixer.c b/sound/pci/oxygen/xonar_dg_mixer.c index 39a0da7..102cfce 100644 --- a/sound/pci/oxygen/xonar_dg_mixer.c +++ b/sound/pci/oxygen/xonar_dg_mixer.c @@ -26,6 +26,7 @@ #include <linux/pci.h> #include <linux/delay.h> +#include <linux/export.h> #include <sound/control.h> #include <sound/core.h> #include <sound/info.h> @@ -456,6 +457,7 @@ static int xonar_control_filter(struct snd_kcontrol_new *template) } struct oxygen_model model_xonar_dg = { + .shortname = "Xonar DG/DGX", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8786", .init = dg_init, -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ALSA: hda - fixup ALC262 to skip depop delay before D3 on Intel BayleyBay
by mengdong.lin＠intel.com 03 Dec '13

03 Dec '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch sets a 0ms depop delay in fixup funtion 'alc_fixup_no_depop_delay'. And Realteck ALC262 applies this on Intel Baytrail BayleyBay platform to reduce codec suspend time. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Reviewed-by: Kailang Yang <kailang(a)realtek.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 21dcb7b..69f11e2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -374,8 +374,10 @@ static void alc_fixup_no_depop_delay(struct hda_codec *codec, { struct alc_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PROBE) + if (action == HDA_FIXUP_ACT_PROBE) { spec->no_depop_delay = 1; + codec->depop_delay = 0; + } } static int alc_auto_parse_customize_define(struct hda_codec *codec) -- 1.8.1.2

2 1

[alsa-devel] [TINYCOMPRESS PATCH 1/2] crec: Initial version of a compressed capture utility
by Charles Keepax 03 Dec '13

03 Dec '13

This version only supports capture of PCM streams over a compressed device and saves these as WAV files. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Android.mk | 10 ++ crec.c | 425 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ makefile.linux | 20 ++- 3 files changed, 448 insertions(+), 7 deletions(-) create mode 100644 crec.c diff --git a/Android.mk b/Android.mk index 104dfef..3cdd3d0 100644 --- a/Android.mk +++ b/Android.mk @@ -20,3 +20,13 @@ LOCAL_MODULE_TAGS := optional include $(BUILD_EXECUTABLE) +include $(CLEAR_VARS) + +LOCAL_C_INCLUDES:= $(LOCAL_PATH)/include +LOCAL_SRC_FILES:= crec.c +LOCAL_MODULE := crec +LOCAL_SHARED_LIBRARIES:= libcutils libutils libtinycompress +LOCAL_MODULE_TAGS := optional + +include $(BUILD_EXECUTABLE) + diff --git a/crec.c b/crec.c new file mode 100644 index 0000000..d2fd873 --- /dev/null +++ b/crec.c @@ -0,0 +1,425 @@ +/* + * BSD LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013, Wolfson Microelectronic Ltd. + * All rights reserved. + * + * Author: Vinod Koul <vinod.koul(a)linux.intel.com> + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * Neither the name of Intel Corporation nor the names of its contributors + * may be used to endorse or promote products derived from this software + * without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" + * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE + * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR + * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF + * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS + * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN + * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) + * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF + * THE POSSIBILITY OF SUCH DAMAGE. + * + * LGPL LICENSE + * + * crec command line recorder for compress audio record in alsa + * Copyright (c) 2011-2012, Intel Corporation + * Copyright (c) 2013, Wolfson Microelectronic Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdint.h> +#include <linux/types.h> +#include <fcntl.h> +#include <errno.h> +#include <unistd.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <signal.h> +#include <stdbool.h> +#include <getopt.h> +#include <sys/time.h> +#define __force +#define __bitwise +#define __user +#include "sound/compress_params.h" +#include "sound/compress_offload.h" +#include "tinycompress/tinycompress.h" + +static int verbose; + +static const unsigned int DEFAULT_CHANNELS = 1; +static const unsigned int DEFAULT_RATE = 44100; +static const unsigned int DEFAULT_FORMAT = SNDRV_PCM_FORMAT_S16_LE; + +struct riff_chunk { + char desc[4]; + uint32_t size; +} __attribute__((__packed__)); + +struct wave_header { + struct { + struct riff_chunk chunk; + char format[4]; + } __attribute__((__packed__)) riff; + + struct { + struct riff_chunk chunk; + uint16_t type; + uint16_t channels; + uint32_t rate; + uint32_t byterate; + uint16_t blockalign; + uint16_t samplebits; + } __attribute__((__packed__)) fmt; + + struct { + struct riff_chunk chunk; + } __attribute__((__packed__)) data; +} __attribute__((__packed__)); + +const struct wave_header blank_wave_header = { + .riff = { + .chunk = { + .desc = "RIFF", + }, + .format = "WAVE", + }, + .fmt = { + .chunk = { + .desc = "fmt ", /* Note the space is important here */ + .size = sizeof(blank_wave_header.fmt) - + sizeof(blank_wave_header.fmt.chunk), + }, + .type = 0x01, /* PCM */ + }, + .data = { + .chunk = { + .desc = "data", + }, + }, +}; + +static void init_wave_header(struct wave_header *header, uint16_t channels, + uint32_t rate, uint16_t samplebits) +{ + memcpy(header, &blank_wave_header, sizeof(blank_wave_header)); + + header->fmt.channels = channels; + header->fmt.rate = rate; + header->fmt.byterate = channels * rate * (samplebits / 8); + header->fmt.blockalign = channels * (samplebits / 8); + header->fmt.samplebits = samplebits; +} + +static void size_wave_header(struct wave_header *header, uint32_t size) +{ + header->riff.chunk.size = sizeof(*header) - + sizeof(header->riff.chunk) + size; + header->data.chunk.size = size; +} + +static void usage(void) +{ + fprintf(stderr, "usage: crec [OPTIONS] filename\n" + "-c\tcard number\n" + "-d\tdevice node\n" + "-b\tbuffer size\n" + "-f\tfragments\n" + "-v\tverbose mode\n" + "-l\tlength of record in seconds\n" + "-h\tPrints this help list\n\n" + "-C\tSpecify the number of channels (default %u)\n" + "-R\tSpecify the sample rate (default %u)\n" + "-F\tSpecify the format: S16_LE, S32_LE (default S16_LE)\n\n" + "Example:\n" + "\tcrec -c 1 -d 2 test.wav\n" + "\tcrec -f 5 test.wav\n", + DEFAULT_CHANNELS, DEFAULT_RATE); + + exit(EXIT_FAILURE); +} + +static int print_time(struct compress *compress) +{ + unsigned int avail; + struct timespec tstamp; + + if (compress_get_hpointer(compress, &avail, &tstamp) != 0) { + fprintf(stderr, "Error querying timestamp\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + return -1; + } else { + printf("DSP recorded %jd.%jd\n", + (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); + } + return 0; +} + +void capture_samples(char *name, unsigned int card, unsigned int device, + unsigned long buffer_size, unsigned int frag, + unsigned int length, unsigned int rate, + unsigned int channels, unsigned int format) +{ + struct compr_config config; + struct snd_codec codec; + struct compress *compress; + struct wave_header header; + FILE *file; + char *buffer; + size_t written; + int read, ret; + unsigned int size, total_read = 0; + unsigned int samplebits; + + switch (format) { + case SNDRV_PCM_FORMAT_S32_LE: + samplebits = 32; + break; + default: + samplebits = 16; + break; + } + + /* Convert length from seconds to bytes */ + length = length * rate * (samplebits / 8) * channels; + + if (verbose) + printf("%s: entry, reading %u bytes\n", __func__, length); + + file = fopen(name, "w+b"); + if (!file) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + + /* Write a header, will update with size once record is complete */ + init_wave_header(&header, channels, rate, samplebits); + written = fwrite(&header, sizeof(header), 1, file); + if (written != 1) { + fprintf(stderr, "Error writing output file header: %d\n", + ferror(file)); + goto file_exit; + } + + codec.id = SND_AUDIOCODEC_PCM; + codec.ch_in = channels; + codec.ch_out = channels; + codec.sample_rate = compress_get_alsa_rate(rate); + if (!codec.sample_rate) { + fprintf(stderr, "invalid sample rate %d\n", rate); + goto file_exit; + } + codec.bit_rate = 0; + codec.rate_control = 0; + codec.profile = 0; + codec.level = 0; + codec.ch_mode = 0; + codec.format = format; + if ((buffer_size != 0) && (frag != 0)) { + config.fragment_size = buffer_size/frag; + config.fragments = frag; + } else { + /* use driver defaults */ + config.fragment_size = 0; + config.fragments = 0; + } + config.codec = &codec; + + compress = compress_open(card, device, COMPRESS_OUT, &config); + if (!compress || !is_compress_ready(compress)) { + fprintf(stderr, "Unable to open Compress device %d:%d\n", + card, device); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto file_exit; + }; + + if (verbose) + printf("%s: Opened compress device\n", __func__); + + size = config.fragment_size; + buffer = malloc(size * config.fragments); + if (!buffer) { + fprintf(stderr, "Unable to allocate %d bytes\n", size); + goto comp_exit; + } + + printf("Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + name, card, device, buffer_size); + printf("Format %u Channels %u, %u Hz\n", + codec.id, codec.ch_out, rate); + + compress_start(compress); + + if (verbose) + printf("%s: Capturing audio NOW!!!\n", __func__); + + do { + if (length && size > length - total_read) + size = length - total_read; + + read = compress_read(compress, buffer, size); + if (read < 0) { + fprintf(stderr, "Error reading sample\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + goto buf_exit; + } + if ((unsigned int)read != size) { + fprintf(stderr, "We read %d, DSP sent %d\n", + size, read); + } + + if (read > 0) { + total_read += read; + + written = fwrite(buffer, 1, read, file); + if (written != (size_t)read) { + fprintf(stderr, "Error writing output file: %d\n", + ferror(file)); + goto buf_exit; + } + if (verbose) { + print_time(compress); + printf("%s: read %d\n", __func__, read); + } + } + } while (!length || total_read < length); + + ret = compress_stop(compress); + if (ret < 0) { + fprintf(stderr, "Error closing stream\n"); + fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); + } + + /* Update file header now we know file size */ + size_wave_header(&header, total_read); + ret = fseek(file, 0, SEEK_SET); + if (ret < 0) { + fprintf(stderr, "Error seeking: %s\n", stderror(errno)); + goto buf_exit; + } + written = fwrite(&header, sizeof(header), 1, file); + if (written != 1) { + fprintf(stderr, "Error updating output file header: %d\n", + ferror(file)); + goto buf_exit; + } + + if (verbose) + printf("%s: exit success\n", __func__); + + free(buffer); + fclose(file); + + compress_close(compress); + + return; +buf_exit: + free(buffer); +comp_exit: + compress_close(compress); +file_exit: + fclose(file); + + if (verbose) + printf("%s: exit failure\n", __func__); + + exit(EXIT_FAILURE); +} + +int main(int argc, char **argv) +{ + char *file; + unsigned long buffer_size = 0; + int c; + unsigned int card = 0, device = 0, frag = 0, length = 0; + unsigned int rate = DEFAULT_RATE, channels = DEFAULT_CHANNELS; + unsigned int format = DEFAULT_FORMAT; + + if (argc < 2) + usage(); + + verbose = 0; + while ((c = getopt(argc, argv, "hvl:R:C:F:b:f:c:d:")) != -1) { + switch (c) { + case 'h': + usage(); + break; + case 'b': + buffer_size = strtol(optarg, NULL, 0); + break; + case 'f': + frag = strtol(optarg, NULL, 10); + break; + case 'c': + card = strtol(optarg, NULL, 10); + break; + case 'd': + device = strtol(optarg, NULL, 10); + break; + case 'v': + verbose = 1; + break; + case 'l': + length = strtol(optarg, NULL, 10); + break; + case 'R': + rate = strtol(optarg, NULL, 10); + break; + case 'C': + channels = strtol(optarg, NULL, 10); + break; + case 'F': + if (strcmp(optarg, "S16_LE") == 0) { + format = SNDRV_PCM_FORMAT_S16_LE; + } else if (strcmp(optarg, "S32_LE") == 0) { + format = SNDRV_PCM_FORMAT_S32_LE; + } else { + fprintf(stderr, "Unrecognised format: %s\n", + optarg); + usage(); + } + break; + default: + exit(EXIT_FAILURE); + } + } + if (optind >= argc) + usage(); + + file = argv[optind]; + + capture_samples(file, card, device, buffer_size, frag, length, + rate, channels, format); + + printf("Finish capturing... Close Normally\n"); + exit(EXIT_SUCCESS); +} + diff --git a/makefile.linux b/makefile.linux index 677878d..81782d0 100644 --- a/makefile.linux +++ b/makefile.linux @@ -1,10 +1,13 @@ LIB = libtinycompress -BIN = cplay VER = 0.1 LIBSRC = compress.c -SRC = cplay.c utils.c +PLAYBIN = cplay +RECBIN = crec +PLAYSRC = cplay.c utils.c +RECSRC = crec.c utils.c LIBOBJ = ${LIBSRC:.c=.o} -OBJ = ${SRC:.c=.o} +PLAYOBJ = ${PLAYSRC:.c=.o} +RECOBJ = ${RECSRC:.c=.o} CC = gcc CROSS_COMPILE = @@ -13,23 +16,26 @@ CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress LIBLDFLAGS = -lasound -all: libtinycompress cplay +all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} -$(BIN): ${OBJ} ${LIB} +$(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${OBJ} ${LDFLAGS} + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} +$(RECBIN): ${RECOBJ} ${LIB} + @echo " LD "$@ + @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: - @rm -rf ${BIN} ${OBJ} ${LIB}.so ${LIBOBJ} + @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} .PHONY: all clean -- 1.7.2.5

2 3

[alsa-devel] [PATCH] ASoC: wm8731: fix dsp mode configuration
by Bo Shen 03 Dec '13

03 Dec '13

According to WM8731 "PD, Rev 4.9 October 2012" datasheet, when it works in DSP mode A, LRP = 1, while works in DSP mode B, LRP = 0. So, fix LRP for DSP mode as the datesheet specification. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8731.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 456bb8c..bc7472c 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -447,10 +447,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0001; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x0013; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x0003; break; default: return -EINVAL; -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: soc-pcm: Only do snd_soc_dai_digital_mute() in hw_free()
by Nicolin Chen 03 Dec '13

03 Dec '13

This patch removed the redundant snd_soc_dai_digital_mute() in close() since it is better to mute in hw_free() which is slightly earlier and symmetrical for the case of reconfiguration, 'aplay 44100.wav 48000.wav' for example: open()->hw_params()->prepare(unmute)->playing->hw_free(mute)->hw_params()-> parepare(unmute)->playing->hw_free(mute)->close() This patch also corrects the condition of the snd_soc_dai_digital_mute() in hw_free() because it would not be executed as we only decrease codec->active in snd_soc_close(). Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/soc-pcm.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index eb340a8..eac82858 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -474,11 +474,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; - /* Muting the DAC suppresses artifacts caused during digital - * shutdown, for example from stopping clocks. - */ - snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); - if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); @@ -689,7 +684,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -707,7 +702,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } /* apply codec digital mute */ - if (!codec->active) + if ((playback && codec_dai->playback_active == 1) || + (!playback && codec_dai->capture_active == 1)) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ -- 1.8.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: ux500_pcm: Differentiate between pdata and DT initialisation
by Lee Jones 03 Dec '13

03 Dec '13

If booting with full DT support (i.e. DMA too, the last piece of the puzzle), then we don't need to use the compatible_request_channel call back or require some of the historical bumph which probably isn't required by a platform data start-up now either. This will also be ripped out in upcoming commits. Cc: alsa-devel(a)alsa-project.org Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_pcm.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 2f1bdb7..ad0d420 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -134,15 +134,24 @@ static const struct snd_dmaengine_pcm_config ux500_dmaengine_pcm_config = { .prepare_slave_config = ux500_pcm_prepare_slave_config, }; +static const struct snd_dmaengine_pcm_config ux500_dmaengine_of_pcm_config = { + .prepare_slave_config = ux500_pcm_prepare_slave_config, +}; + int ux500_pcm_register_platform(struct platform_device *pdev) { + const struct snd_dmaengine_pcm_config *pcm_config; + struct device_node *np = pdev->dev.of_node; int ret; - ret = snd_dmaengine_pcm_register(&pdev->dev, - &ux500_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_COMPAT | - SND_DMAENGINE_PCM_FLAG_NO_DT); + if (np) + pcm_config = &ux500_dmaengine_of_pcm_config; + else + pcm_config = &ux500_dmaengine_pcm_config; + + ret = snd_dmaengine_pcm_register(&pdev->dev, pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_COMPAT); if (ret < 0) { dev_err(&pdev->dev, "%s: ERROR: Failed to register platform '%s' (%d)!\n", -- 1.8.3.2

3 7

[alsa-devel] [PATCH] ASoC: ssm2602: Use core for applying symmetry constraints
by Lars-Peter Clausen 03 Dec '13

03 Dec '13

Let the core take care of applying sample rate and sample bits constraints instead of open-coding this in the driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm2602.c | 50 ++-------------------------------------------- 1 file changed, 2 insertions(+), 48 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 480074d..c6dd485 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -53,8 +53,6 @@ enum ssm2602_type { struct ssm2602_priv { unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; struct regmap *regmap; @@ -277,11 +275,6 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params)); unsigned int iface; - if (substream == ssm2602->slave_substream) { - dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n"); - return 0; - } - if (srate < 0) return srate; @@ -314,33 +307,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct snd_pcm_runtime *master_runtime; - - /* The DAI has shared clocks so if we already have a playback or - * capture going then constrain this substream to match it. - * TODO: the ssm2602 allows pairs of non-matching PB/REC rates - */ - if (ssm2602->master_substream) { - master_runtime = ssm2602->master_substream->runtime; - dev_dbg(codec->dev, "Constraining to %d bits at %dHz\n", - master_runtime->sample_bits, - master_runtime->rate); - - if (master_runtime->rate != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - if (master_runtime->sample_bits != 0) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); - - ssm2602->slave_substream = substream; - } else - ssm2602->master_substream = substream; if (ssm2602->sysclk_constraints) { snd_pcm_hw_constraint_list(substream->runtime, 0, @@ -351,19 +317,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, return 0; } -static void ssm2602_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - - if (ssm2602->master_substream == substream) - ssm2602->master_substream = ssm2602->slave_substream; - - ssm2602->slave_substream = NULL; -} - - static int ssm2602_mute(struct snd_soc_dai *dai, int mute) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec); @@ -530,7 +483,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, static const struct snd_soc_dai_ops ssm2602_dai_ops = { .startup = ssm2602_startup, .hw_params = ssm2602_hw_params, - .shutdown = ssm2602_shutdown, .digital_mute = ssm2602_mute, .set_sysclk = ssm2602_set_dai_sysclk, .set_fmt = ssm2602_set_dai_fmt, @@ -551,6 +503,8 @@ static struct snd_soc_dai_driver ssm2602_dai = { .rates = SSM2602_RATES, .formats = SSM2602_FORMATS,}, .ops = &ssm2602_dai_ops, + .symmetric_rates = 1, + .symmetric_samplebits = 1, }; static int ssm2602_suspend(struct snd_soc_codec *codec) -- 1.8.0

2 1

Re: [alsa-devel] [pulseaudio-discuss] [PATCH] alsa: Disable timer-scheduling for PCMs with the BATCH flag
by Raymond Yau 03 Dec '13

03 Dec '13

> > > > PCM Devices which have the BATCH flag set update the PCM pointer > only with > > period size granularity. Using timer based scheduling does not have > any > > advantage in this mode. For one devices which have that flag set > usually > > update > > the position pointer in software after getting the period interrupt. > So > > disabling the period interrupt is not possible for this kind of > devices. > > Furthermore writing to or reading from the buffer slice for the > current > > period > > is not possible since the position inside the buffer is not known. On > > the other > > hand the tsched algorithm seems to get easily confused for this kind > of > > hardware, which results in garbled audio output. This typically means > > that timer > > based scheduling needs to be manually disabled on systems with such > devices. > > Auto disabling tsched in this case allows these systems to run with > the > > default > > configuration. > > > > > > If the playback position is reported in steps instead of monotonic > increasing ? > > > > does this mean that you also need to increase rewind_safeguard to one > period > > or modifiy snd_pcm_rewindable ? > > > Yes that makes sense. The safeguard should probably be 1.5 periods or 1 > period + fixed value, since it will always take some time from when the > hardware reaches the next period to when userspace is notified about this. > > I'm unfortunately not that familiar with pulseaudio. Maybe there is a > better > way to fix this problem then to disable tsched. If anybody wants to > recreate > the issue it can easily be emulated by applying the following patch to your > kernel: > > does all alsa drivers with SNDRV_PCM_INFO_PATCH update the playback/capture position in period interrupt ? usb-audio seem update when each URB send/receive (in 1ms intervals) ymfpci driver also update hwptr in 5ms interval but no SNDRV_PCM_INFO_PATCH and no constraint in period time must be interval of 5 ms the rewind_safeguard is affected by the size of FIFO buffer as you cannot rewind any data in FIFO buffer of DSP/ sound controller

3 4

[alsa-devel] [PATCHv8] dmaengine: Add support for BCM2835
by Florian Meier 03 Dec '13

03 Dec '13

Add support for DMA controller of BCM2835 as used in the Raspberry Pi. Currently it only supports cyclic DMA. Signed-off-by: Florian Meier <florian.meier(a)koalo.de> --- This version includes some more style improvements suggested in the previous thread. .../devicetree/bindings/dma/bcm2835-dma.txt | 57 ++ drivers/dma/Kconfig | 6 + drivers/dma/Makefile | 1 + drivers/dma/bcm2835-dma.c | 727 +++++++++++++++++++++ 4 files changed, 791 insertions(+) create mode 100644 Documentation/devicetree/bindings/dma/bcm2835-dma.txt create mode 100644 drivers/dma/bcm2835-dma.c diff --git a/Documentation/devicetree/bindings/dma/bcm2835-dma.txt b/Documentation/devicetree/bindings/dma/bcm2835-dma.txt new file mode 100644 index 0000000..e7c31c1 --- /dev/null +++ b/Documentation/devicetree/bindings/dma/bcm2835-dma.txt @@ -0,0 +1,57 @@ +* BCM2835 DMA controller + +The BCM2835 DMA controller has 16 channels in total. +Only the lower 13 channels have an associated IRQ. +Some arbitrary channels are used by the firmware +(1,3,6,7 in the current firmware version). +The channels 0,2 and 3 have special functionality +and should not be used by the driver. + +Required properties: +- compatible: Should be "brcm,bcm2835-dma". +- reg: Should contain DMA registers location and length. +- interrupts: Should contain the DMA interrupts associated + to the DMA channels in ascending order. +- #dma-cells: Must be <1>, the cell in the dmas property of the + client device represents the DREQ number. +- brcm,dma-channel-mask: Bit mask representing the channels + not used by the firmware in ascending order, + i.e. first channel corresponds to LSB. + +Example: + +dma: dma@7e007000 { + compatible = "brcm,bcm2835-dma"; + reg = <0x7e007000 0xf00>; + interrupts = <1 16> + <1 17> + <1 18> + <1 19> + <1 20> + <1 21> + <1 22> + <1 23> + <1 24> + <1 25> + <1 26> + <1 27> + <1 28>; + + #dma-cells = <1>; + brcm,dma-channel-mask = <0x7f35>; +}; + +DMA clients connected to the BCM2835 DMA controller must use the format +described in the dma.txt file, using a two-cell specifier for each channel. + +Example: + +bcm2835_i2s: i2s@7e203000 { + compatible = "brcm,bcm2835-i2s"; + reg = < 0x7e203000 0x20> + < 0x7e101098 0x02>; + + dmas = <&dma 2> + <&dma 3>; + dma-names = "tx", "rx"; +}; diff --git a/drivers/dma/Kconfig b/drivers/dma/Kconfig index c61a6ec..880e723 100644 --- a/drivers/dma/Kconfig +++ b/drivers/dma/Kconfig @@ -300,6 +300,12 @@ config DMA_OMAP select DMA_ENGINE select DMA_VIRTUAL_CHANNELS +config DMA_BCM2835 + tristate "BCM2835 DMA engine support" + depends on (ARCH_BCM2835 || MACH_BCM2708) + select DMA_ENGINE + select DMA_VIRTUAL_CHANNELS + config TI_CPPI41 tristate "AM33xx CPPI41 DMA support" depends on ARCH_OMAP diff --git a/drivers/dma/Makefile b/drivers/dma/Makefile index 0ce2da9..0a6f08e 100644 --- a/drivers/dma/Makefile +++ b/drivers/dma/Makefile @@ -38,6 +38,7 @@ obj-$(CONFIG_EP93XX_DMA) += ep93xx_dma.o obj-$(CONFIG_DMA_SA11X0) += sa11x0-dma.o obj-$(CONFIG_MMP_TDMA) += mmp_tdma.o obj-$(CONFIG_DMA_OMAP) += omap-dma.o +obj-$(CONFIG_DMA_BCM2835) += bcm2835-dma.o obj-$(CONFIG_MMP_PDMA) += mmp_pdma.o obj-$(CONFIG_DMA_JZ4740) += dma-jz4740.o obj-$(CONFIG_TI_CPPI41) += cppi41.o diff --git a/drivers/dma/bcm2835-dma.c b/drivers/dma/bcm2835-dma.c new file mode 100644 index 0000000..a9d34df --- /dev/null +++ b/drivers/dma/bcm2835-dma.c @@ -0,0 +1,727 @@ +/* + * BCM2835 DMA engine support + * + * This driver only supports cyclic DMA transfers + * as needed for the I2S module. + * + * Author: Florian Meier <florian.meier(a)koalo.de> + * Copyright 2013 + * + * Based on + * OMAP DMAengine support by Russell King + * + * BCM2708 DMA Driver + * Copyright (C) 2010 Broadcom + * + * Raspberry Pi PCM I2S ALSA Driver + * Copyright (c) by Phil Poole 2013 + * + * MARVELL MMP Peripheral DMA Driver + * Copyright 2012 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ +#include <linux/dmaengine.h> +#include <linux/dma-mapping.h> +#include <linux/err.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/list.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/spinlock.h> +#include <linux/of.h> +#include <linux/of_dma.h> + +#include "virt-dma.h" + +struct bcm2835_dmadev { + struct dma_device ddev; + spinlock_t lock; + void __iomem *base; + struct device_dma_parameters dma_parms; +}; + +struct bcm2835_dma_cb { + uint32_t info; + uint32_t src; + uint32_t dst; + uint32_t length; + uint32_t stride; + uint32_t next; + uint32_t pad[2]; +}; + +struct bcm2835_chan { + struct virt_dma_chan vc; + struct list_head node; + + struct dma_slave_config cfg; + bool cyclic; + unsigned int dreq; + + int ch; + struct bcm2835_desc *desc; + + void __iomem *chan_base; + int irq_number; +}; + +struct bcm2835_desc { + struct virt_dma_desc vd; + enum dma_transfer_direction dir; + + unsigned int control_block_size; + struct bcm2835_dma_cb *control_block_base; + dma_addr_t control_block_base_phys; + + unsigned int frames; + size_t size; +}; + +#define BCM2835_DMA_CS 0x00 +#define BCM2835_DMA_ADDR 0x04 +#define BCM2835_DMA_SOURCE_AD 0x0c +#define BCM2835_DMA_DEST_AD 0x10 +#define BCM2835_DMA_NEXTCB 0x1C + +/* DMA CS Control and Status bits */ +#define BCM2835_DMA_ACTIVE BIT(0) +#define BCM2835_DMA_INT BIT(2) +#define BCM2835_DMA_ISPAUSED BIT(4) /* Pause requested or not active */ +#define BCM2835_DMA_ISHELD BIT(5) /* Is held by DREQ flow control */ +#define BCM2835_DMA_ERR BIT(8) +#define BCM2835_DMA_ABORT BIT(30) /* Stop current CB, go to next, WO */ +#define BCM2835_DMA_RESET BIT(31) /* WO, self clearing */ + +#define BCM2835_DMA_INT_EN BIT(0) +#define BCM2835_DMA_D_INC BIT(4) +#define BCM2835_DMA_D_DREQ BIT(6) +#define BCM2835_DMA_S_INC BIT(8) +#define BCM2835_DMA_S_DREQ BIT(10) + +#define BCM2835_DMA_PER_MAP(x) ((x) << 16) + +#define BCM2835_DMA_DATA_TYPE_S8 1 +#define BCM2835_DMA_DATA_TYPE_S16 2 +#define BCM2835_DMA_DATA_TYPE_S32 4 +#define BCM2835_DMA_DATA_TYPE_S128 16 + +#define BCM2835_DMA_BULK_MASK BIT(0) +#define BCM2835_DMA_FIQ_MASK (BIT(2) | BIT(3)) + +/* Valid only for channels 0 - 14, 15 has its own base address */ +#define BCM2835_DMA_CHAN(n) ((n) << 8) /* Base address */ +#define BCM2835_DMA_CHANIO(base, n) ((base) + BCM2835_DMA_CHAN(n)) + +static inline struct bcm2835_dmadev *to_bcm2835_dma_dev(struct dma_device *d) +{ + return container_of(d, struct bcm2835_dmadev, ddev); +} + +static inline struct bcm2835_chan *to_bcm2835_dma_chan(struct dma_chan *c) +{ + return container_of(c, struct bcm2835_chan, vc.chan); +} + +static inline struct bcm2835_desc *to_bcm2835_dma_desc( + struct dma_async_tx_descriptor *t) +{ + return container_of(t, struct bcm2835_desc, vd.tx); +} + +static void bcm2835_dma_desc_free(struct virt_dma_desc *vd) +{ + struct bcm2835_desc *desc = container_of(vd, struct bcm2835_desc, vd); + dma_free_coherent(desc->vd.tx.chan->device->dev, + desc->control_block_size, + desc->control_block_base, + desc->control_block_base_phys); + kfree(desc); +} + +static int bcm2835_dma_abort(void __iomem *chan_base) +{ + unsigned long cs; + long int timeout = 10000; + + cs = readl(chan_base + BCM2835_DMA_CS); + if (!(cs & BCM2835_DMA_ACTIVE)) + return 0; + + /* Write 0 to the active bit - Pause the DMA */ + writel(0, chan_base + BCM2835_DMA_CS); + + /* Wait for any current AXI transfer to complete */ + while ((cs & BCM2835_DMA_ISPAUSED) && --timeout) { + cpu_relax(); + cs = readl(chan_base + BCM2835_DMA_CS); + } + + /* We'll un-pause when we set of our next DMA */ + if (!timeout) + return -ETIMEDOUT; + + if (!(cs & BCM2835_DMA_ACTIVE)) + return 0; + + /* Terminate the control block chain */ + writel(0, chan_base + BCM2835_DMA_NEXTCB); + + /* Abort the whole DMA */ + writel(BCM2835_DMA_ABORT | BCM2835_DMA_ACTIVE, + chan_base + BCM2835_DMA_CS); + + return 0; +} + +static void bcm2835_dma_start_desc(struct bcm2835_chan *c) +{ + struct virt_dma_desc *vd = vchan_next_desc(&c->vc); + struct bcm2835_desc *d; + + if (!vd) { + c->desc = NULL; + return; + } + + list_del(&vd->node); + + c->desc = d = to_bcm2835_dma_desc(&vd->tx); + + writel(d->control_block_base_phys, c->chan_base + BCM2835_DMA_ADDR); + writel(BCM2835_DMA_ACTIVE, c->chan_base + BCM2835_DMA_CS); +} + +static irqreturn_t bcm2835_dma_callback(int irq, void *data) +{ + struct bcm2835_chan *c = data; + struct bcm2835_desc *d; + unsigned long flags; + + spin_lock_irqsave(&c->vc.lock, flags); + + /* Acknowledge interrupt */ + writel(BCM2835_DMA_INT, c->chan_base + BCM2835_DMA_CS); + + d = c->desc; + + if (d) { + /* TODO Only works for cyclic DMA */ + vchan_cyclic_callback(&d->vd); + } + + /* Keep the DMA engine running */ + writel(BCM2835_DMA_ACTIVE, c->chan_base + BCM2835_DMA_CS); + + spin_unlock_irqrestore(&c->vc.lock, flags); + + return IRQ_HANDLED; +} + +static int bcm2835_dma_alloc_chan_resources(struct dma_chan *chan) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + + dev_dbg(c->vc.chan.device->dev, + "Allocating DMA channel %d\n", c->ch); + + return request_irq(c->irq_number, + bcm2835_dma_callback, 0, "DMA IRQ", c); +} + +static void bcm2835_dma_free_chan_resources(struct dma_chan *chan) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + + vchan_free_chan_resources(&c->vc); + free_irq(c->irq_number, c); + + dev_dbg(c->vc.chan.device->dev, "Freeing DMA channel %u\n", c->ch); +} + +static size_t bcm2835_dma_desc_size(struct bcm2835_desc *d) +{ + return d->size; +} + +static size_t bcm2835_dma_desc_size_pos(struct bcm2835_desc *d, dma_addr_t addr) +{ + unsigned int i; + size_t size; + + for (size = i = 0; i < d->frames; i++) { + struct bcm2835_dma_cb *control_block = + &d->control_block_base[i]; + size_t this_size = control_block->length; + dma_addr_t dma; + + if (d->dir == DMA_DEV_TO_MEM) + dma = control_block->dst; + else + dma = control_block->src; + + if (size) + size += this_size; + else if (addr >= dma && addr < dma + this_size) + size += dma + this_size - addr; + } + + return size; +} + +static enum dma_status bcm2835_dma_tx_status(struct dma_chan *chan, + dma_cookie_t cookie, struct dma_tx_state *txstate) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + struct virt_dma_desc *vd; + enum dma_status ret; + unsigned long flags; + + ret = dma_cookie_status(chan, cookie, txstate); + if (ret == DMA_SUCCESS || !txstate) + return ret; + + spin_lock_irqsave(&c->vc.lock, flags); + vd = vchan_find_desc(&c->vc, cookie); + if (vd) { + txstate->residue = + bcm2835_dma_desc_size(to_bcm2835_dma_desc(&vd->tx)); + } else if (c->desc && c->desc->vd.tx.cookie == cookie) { + struct bcm2835_desc *d = c->desc; + dma_addr_t pos; + + if (d->dir == DMA_MEM_TO_DEV) + pos = readl(c->chan_base + BCM2835_DMA_SOURCE_AD); + else if (d->dir == DMA_DEV_TO_MEM) + pos = readl(c->chan_base + BCM2835_DMA_DEST_AD); + else + pos = 0; + + txstate->residue = bcm2835_dma_desc_size_pos(d, pos); + } else { + txstate->residue = 0; + } + + spin_unlock_irqrestore(&c->vc.lock, flags); + + return ret; +} + +static void bcm2835_dma_issue_pending(struct dma_chan *chan) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + unsigned long flags; + + c->cyclic = true; /* Nothing else is implemented */ + + spin_lock_irqsave(&c->vc.lock, flags); + if (vchan_issue_pending(&c->vc) && !c->desc) + bcm2835_dma_start_desc(c); + + spin_unlock_irqrestore(&c->vc.lock, flags); +} + +static struct dma_async_tx_descriptor *bcm2835_dma_prep_dma_cyclic( + struct dma_chan *chan, dma_addr_t buf_addr, size_t buf_len, + size_t period_len, enum dma_transfer_direction direction, + unsigned long flags, void *context) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + enum dma_slave_buswidth dev_width; + struct bcm2835_desc *d; + dma_addr_t dev_addr; + unsigned int es, sync_type; + unsigned int frame; + + /* Grab configuration */ + if (!is_slave_direction(direction)) { + dev_err(chan->device->dev, "%s: bad direction?\n", __func__); + return NULL; + } + + if (direction == DMA_DEV_TO_MEM) { + dev_addr = c->cfg.src_addr; + dev_width = c->cfg.src_addr_width; + sync_type = BCM2835_DMA_S_DREQ; + } else { + dev_addr = c->cfg.dst_addr; + dev_width = c->cfg.dst_addr_width; + sync_type = BCM2835_DMA_D_DREQ; + } + + /* Bus width translates to the element size (ES) */ + switch (dev_width) { + case DMA_SLAVE_BUSWIDTH_4_BYTES: + es = BCM2835_DMA_DATA_TYPE_S32; + break; + default: + return NULL; + } + + /* Now allocate and setup the descriptor. */ + d = kzalloc(sizeof(*d), GFP_NOWAIT); + if (!d) + return NULL; + + d->dir = direction; + d->frames = buf_len / period_len; + + /* Allocate memory for control blocks */ + d->control_block_size = d->frames * sizeof(struct bcm2835_dma_cb); + d->control_block_base = dma_zalloc_coherent(chan->device->dev, + d->control_block_size, &d->control_block_base_phys, + GFP_NOWAIT); + + if (!d->control_block_base) { + kfree(d); + return NULL; + } + + /* + * Iterate over all frames, create a control block + * for each frame and link them together. + */ + for (frame = 0; frame < d->frames; frame++) { + struct bcm2835_dma_cb *control_block = + &d->control_block_base[frame]; + + /* Setup adresses */ + if (d->dir == DMA_DEV_TO_MEM) { + control_block->info = BCM2835_DMA_D_INC; + control_block->src = dev_addr; + control_block->dst = buf_addr + frame * period_len; + } else { + control_block->info = BCM2835_DMA_S_INC; + control_block->src = buf_addr + frame * period_len; + control_block->dst = dev_addr; + } + + /* Enable interrupt */ + control_block->info |= BCM2835_DMA_INT_EN; + + /* Setup synchronization */ + if (sync_type != 0) + control_block->info |= sync_type; + + /* Setup DREQ channel */ + if (c->dreq != 0) + control_block->info |= + BCM2835_DMA_PER_MAP(c->dreq); + + /* Length of a frame */ + control_block->length = period_len; + d->size += control_block->length; + + /* + * Next block is the next frame. + * This DMA engine driver currently only supports cyclic DMA. + * Therefore, wrap around at number of frames. + */ + control_block->next = d->control_block_base_phys + + sizeof(struct bcm2835_dma_cb) + * ((frame + 1) % d->frames); + } + + return vchan_tx_prep(&c->vc, &d->vd, flags); +} + +static int bcm2835_dma_slave_config(struct bcm2835_chan *c, + struct dma_slave_config *cfg) +{ + if ((cfg->direction == DMA_DEV_TO_MEM && + cfg->src_addr_width != DMA_SLAVE_BUSWIDTH_4_BYTES) || + (cfg->direction == DMA_MEM_TO_DEV && + cfg->dst_addr_width != DMA_SLAVE_BUSWIDTH_4_BYTES) || + !is_slave_direction(cfg->direction)) { + return -EINVAL; + } + + c->cfg = *cfg; + + return 0; +} + +static int bcm2835_dma_terminate_all(struct bcm2835_chan *c) +{ + struct bcm2835_dmadev *d = to_bcm2835_dma_dev(c->vc.chan.device); + unsigned long flags; + int timeout = 10000; + LIST_HEAD(head); + + spin_lock_irqsave(&c->vc.lock, flags); + + /* Prevent this channel being scheduled */ + spin_lock(&d->lock); + list_del_init(&c->node); + spin_unlock(&d->lock); + + /* + * Stop DMA activity: we assume the callback will not be called + * after bcm_dma_abort() returns (even if it does, it will see + * c->desc is NULL and exit.) + */ + if (c->desc) { + c->desc = NULL; + bcm2835_dma_abort(c->chan_base); + + /* Wait for stopping */ + while (--timeout) { + if (!(readl(c->chan_base + BCM2835_DMA_CS) & + BCM2835_DMA_ACTIVE)) + break; + + cpu_relax(); + } + + if (!timeout) + dev_err(d->ddev.dev, "DMA transfer could not be terminated\n"); + } + + vchan_get_all_descriptors(&c->vc, &head); + spin_unlock_irqrestore(&c->vc.lock, flags); + vchan_dma_desc_free_list(&c->vc, &head); + + return 0; +} + +static int bcm2835_dma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, + unsigned long arg) +{ + struct bcm2835_chan *c = to_bcm2835_dma_chan(chan); + + switch (cmd) { + case DMA_SLAVE_CONFIG: + return bcm2835_dma_slave_config(c, + (struct dma_slave_config *)arg); + + case DMA_TERMINATE_ALL: + return bcm2835_dma_terminate_all(c); + + default: + return -ENXIO; + } +} + +static int bcm2835_dma_chan_init(struct bcm2835_dmadev *d, int chan_id, int irq) +{ + struct bcm2835_chan *c; + + c = devm_kzalloc(d->ddev.dev, sizeof(*c), GFP_KERNEL); + if (!c) + return -ENOMEM; + + c->vc.desc_free = bcm2835_dma_desc_free; + vchan_init(&c->vc, &d->ddev); + INIT_LIST_HEAD(&c->node); + + d->ddev.chancnt++; + + c->chan_base = BCM2835_DMA_CHANIO(d->base, chan_id); + c->ch = chan_id; + c->irq_number = irq; + + return 0; +} + +static void bcm2835_dma_free(struct bcm2835_dmadev *od) +{ + while (!list_empty(&od->ddev.channels)) { + struct bcm2835_chan *c = list_first_entry(&od->ddev.channels, + struct bcm2835_chan, vc.chan.device_node); + + list_del(&c->vc.chan.device_node); + tasklet_kill(&c->vc.task); + } +} + +#if defined(CONFIG_OF) +static const struct of_device_id bcm2835_dma_of_match[] = { + { .compatible = "brcm,bcm2835-dma", }, + {}, +}; +MODULE_DEVICE_TABLE(of, bcm2835_dma_of_match); +#endif + +static struct dma_chan *bcm2835_dma_xlate(struct of_phandle_args *spec, + struct of_dma *ofdma) +{ + struct bcm2835_dmadev *d = ofdma->of_dma_data; + struct dma_chan *chan, *candidate; + +retry: + candidate = NULL; + + /* Walk the list of channels registered with the current instance and + * find one that is currently unused */ + list_for_each_entry(chan, &d->ddev.channels, device_node) + if (chan->client_count == 0) { + candidate = chan; + break; + } + + if (!candidate) + return NULL; + + /* dma_get_slave_channel will return NULL if we lost a race between + * the lookup and the reservation */ + chan = dma_get_slave_channel(candidate); + + if (!chan) + goto retry; + + /* Set DREQ from param */ + to_bcm2835_dma_chan(chan)->dreq = spec->args[0]; + + return chan; +} + +static int bcm2835_dma_device_slave_caps(struct dma_chan *dchan, + struct dma_slave_caps *caps) +{ + caps->src_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + caps->dstn_addr_widths = BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + caps->directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); + caps->cmd_pause = false; + caps->cmd_terminate = true; + + return 0; +} + +static int bcm2835_dma_probe(struct platform_device *pdev) +{ + struct bcm2835_dmadev *od; + struct resource *res; + void __iomem *base; + int rc; + int i; + int irq; + uint32_t chans_available; + + if (!pdev->dev.dma_mask) + pdev->dev.dma_mask = &pdev->dev.coherent_dma_mask; + + rc = dma_set_mask(&pdev->dev, DMA_BIT_MASK(32)); + if (rc) + return rc; + dma_set_coherent_mask(&pdev->dev, DMA_BIT_MASK(32)); + + od = devm_kzalloc(&pdev->dev, sizeof(*od), GFP_KERNEL); + if (!od) + return -ENOMEM; + + pdev->dev.dma_parms = &od->dma_parms; + dma_set_max_seg_size(&pdev->dev, 0x3FFFFFFF); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + od->base = base; + + dma_cap_set(DMA_SLAVE, od->ddev.cap_mask); + dma_cap_set(DMA_CYCLIC, od->ddev.cap_mask); + od->ddev.device_alloc_chan_resources = bcm2835_dma_alloc_chan_resources; + od->ddev.device_free_chan_resources = bcm2835_dma_free_chan_resources; + od->ddev.device_tx_status = bcm2835_dma_tx_status; + od->ddev.device_issue_pending = bcm2835_dma_issue_pending; + od->ddev.device_slave_caps = bcm2835_dma_device_slave_caps; + od->ddev.device_prep_dma_cyclic = bcm2835_dma_prep_dma_cyclic; + od->ddev.device_control = bcm2835_dma_control; + od->ddev.dev = &pdev->dev; + INIT_LIST_HEAD(&od->ddev.channels); + spin_lock_init(&od->lock); + + platform_set_drvdata(pdev, od); + + /* Request DMA channel mask from device tree */ + if (of_property_read_u32(pdev->dev.of_node, + "brcm,dma-channel-mask", + &chans_available)) { + dev_err(&pdev->dev, "Failed to get channel mask\n"); + rc = -EINVAL; + goto err_no_dma; + } + + /* Do not use the FIQ and BULK channels, + * because they are used by the GPU. + */ + chans_available &= ~(BCM2835_DMA_FIQ_MASK | BCM2835_DMA_BULK_MASK); + + for (i = 0; i < pdev->num_resources; i++) { + irq = platform_get_irq(pdev, i); + if (irq < 0) + break; + + if (chans_available & (1 << i)) { + rc = bcm2835_dma_chan_init(od, i, irq); + if (rc) + goto err_no_dma; + } + } + + dev_dbg(&pdev->dev, "Initialized %i DMA channels\n", i); + + if (pdev->dev.of_node) { + /* Device-tree DMA controller registration */ + rc = of_dma_controller_register(pdev->dev.of_node, + bcm2835_dma_xlate, od); + if (rc) { + dev_err(&pdev->dev, "Failed to register DMA controller\n"); + goto err_no_dma; + } + } + + rc = dma_async_device_register(&od->ddev); + if (rc) { + dev_err(&pdev->dev, + "Failed to register slave DMA engine device: %d\n", rc); + goto err_no_dma; + } + + dev_dbg(&pdev->dev, "Load BCM2835 DMA engine driver\n"); + + return 0; + +err_no_dma: + bcm2835_dma_free(od); + return rc; +} + +static int bcm2835_dma_remove(struct platform_device *pdev) +{ + struct bcm2835_dmadev *od = platform_get_drvdata(pdev); + + dma_async_device_unregister(&od->ddev); + bcm2835_dma_free(od); + + return 0; +} + +static struct platform_driver bcm2835_dma_driver = { + .probe = bcm2835_dma_probe, + .remove = bcm2835_dma_remove, + .driver = { + .name = "bcm2835-dma", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(bcm2835_dma_of_match), + }, +}; + +module_platform_driver(bcm2835_dma_driver); + +MODULE_ALIAS("platform:bcm2835-dma"); +MODULE_DESCRIPTION("BCM2835 DMA engine driver"); +MODULE_AUTHOR("Florian Meier <florian.meier(a)koalo.de>"); +MODULE_LICENSE("GPL v2"); -- 1.8.3.2

3 2

[alsa-devel] How should SGTL5000 codec driver adapt to dummy regulator change?
by Shawn Guo 03 Dec '13

03 Dec '13

Mark, The sgtl5000_enable_regulators() function in SGTL5000 codec driver makes the decision whether an external VDDD supply is available by checking the return of regulator_bulk_get() call. It used to work because dummy regulator did not play in there, and if regulator_bulk_get() fails, most likely it's only because there is no external VDDD supply. But with commit 4ddfebd (regulator: core: Provide a dummy regulator with full constraints) merged, the regulator_bulk_get() will always succeed. It breaks the driver's assumption. What's your suggestion to the problem? Shawn

2 1

[alsa-devel] [PATCH 1/3] ASoC: ux500_pcm: Stop pretending that we support varying address widths
by Lee Jones 03 Dec '13

03 Dec '13

The Slave Config's addr_width attribute is populated by data_width of dma_cfg, which in turn is derived from dma_params' data_size attribute and that comes from the slot_width which is always 16 bits (2 Bytes). We're cutting out the middle man here and just setting the DMA Slave Config directly. Cc: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_pcm.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 6162f70..2f1bdb7 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -104,26 +104,25 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct dma_slave_config *slave_config) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct ux500_msp_dma_params *dma_params; - struct stedma40_chan_cfg *dma_cfg; + struct snd_dmaengine_dai_dma_data *dma_params; int ret; dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - dma_cfg = dma_params->dma_cfg; ret = snd_hwparams_to_dma_slave_config(substream, params, slave_config); if (ret) return ret; slave_config->dst_maxburst = 4; - slave_config->dst_addr_width = dma_cfg->dst_info.data_width; slave_config->src_maxburst = 4; - slave_config->src_addr_width = dma_cfg->src_info.data_width; + + slave_config->src_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - slave_config->dst_addr = dma_params->tx_rx_addr; + slave_config->dst_addr = dma_params->addr; else - slave_config->src_addr = dma_params->tx_rx_addr; + slave_config->src_addr = dma_params->addr; return 0; } -- 1.8.3.2

3 13

[alsa-devel] [PATCH] ALSA: hda - Add missing initialization of aamix paths
by Takashi Iwai 03 Dec '13

03 Dec '13

The loopback mixing paths aren't initialized correctly at init callback. Mostly this is harmless as codecs usually set the mute state as default, but we still should make sure. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index c4671d00babd..28baeff99c62 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5138,6 +5138,23 @@ static void init_multi_io(struct hda_codec *codec) } } +static void init_aamix_paths(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + + if (!spec->have_aamix_ctl) + return; + update_aamix_paths(codec, spec->aamix_mode, spec->out_paths[0], + spec->aamix_out_paths[0], + spec->autocfg.line_out_type); + update_aamix_paths(codec, spec->aamix_mode, spec->hp_paths[0], + spec->aamix_out_paths[1], + AUTO_PIN_HP_OUT); + update_aamix_paths(codec, spec->aamix_mode, spec->speaker_paths[0], + spec->aamix_out_paths[2], + AUTO_PIN_SPEAKER_OUT); +} + /* set up input pins and loopback paths */ static void init_analog_input(struct hda_codec *codec) { @@ -5240,6 +5257,7 @@ int snd_hda_gen_init(struct hda_codec *codec) init_multi_out(codec); init_extra_out(codec); init_multi_io(codec); + init_aamix_paths(codec); init_analog_input(codec); init_input_src(codec); init_digital(codec); -- 1.8.5

1 0

Re: [alsa-devel] [PATCH] ALSA: hda - Fix headset mic input after muted internal mic (Dell/Realtek)
by Takashi Iwai 03 Dec '13

03 Dec '13

At Tue, 3 Dec 2013 15:52:37 +0800, Kailang wrote: > > Hi David, > > The verb was ADC digital mute(bit 6 default 1). > Switch internal mic and headset mic will run alc_headset_mode_default. > The coef index 0x11 will set to 0x0041. > Because headset mode was fixed type. It doesn't need to run > alc_determine_headset_type. > So, the value still keep 0x0041. ADC was muted. > Your patch will solve this errors. OK, I added your comment in the changelog and add Cc to stable, then applied to for-linus branch now. thanks, Takashi > > Thanks. > Kailang > > > > By trial and error, I found this patch could work around an issue > > where the headset mic would stop working if you switch between the > > internal mic and the headset mic, and the internal mic was muted. > > > > It still takes a second or two before the headset mic actually starts > > working, but still better than nothing. > > > > I'd like to add stable to this patch, but I'd like Kailang's ack first > > because I don't know what the verbs really do. > > > > BugLink: https://bugs.launchpad.net/bugs/1256840 > > Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> > > --- > > sound/pci/hda/patch_realtek.c | 2 ++ > > 1 file changed, 2 insertions(+) > > > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > > index 537991c..85d1c8b 100644 > > --- a/sound/pci/hda/patch_realtek.c > > +++ b/sound/pci/hda/patch_realtek.c > > @@ -3338,6 +3338,7 @@ static void alc_headset_mode_ctia(struct hda_codec > > *codec) > > alc_write_coef_idx(codec, 0x18, 0x7388); > > break; > > case 0x10ec0668: > > + alc_write_coef_idx(codec, 0x11, 0x0001); > > alc_write_coef_idx(codec, 0x15, 0x0d60); > > alc_write_coef_idx(codec, 0xc3, 0x0000); > > break; > > @@ -3366,6 +3367,7 @@ static void alc_headset_mode_omtp(struct hda_codec > > *codec) > > alc_write_coef_idx(codec, 0x18, 0x7388); > > break; > > case 0x10ec0668: > > + alc_write_coef_idx(codec, 0x11, 0x0001); > > alc_write_coef_idx(codec, 0x15, 0x0d50); > > alc_write_coef_idx(codec, 0xc3, 0x0000); > > break; > > -- > > 1.7.9.5 > > > > > > ------Please consider the environment before printing this e-mail. >

1 0

Re: [alsa-devel] compressed audio support (maruthi srinivas)
by anish singh 03 Dec '13

03 Dec '13

> Hi, > > I saw that compressed audio framework being used > with Asoc (alsa soc). > Can the framework be used with other than Asoc ? > I mean with ALSA pci based cards. Never worked on this so don't know. > I see the implementation in the following files. which one > is to be used ? > Does tinycompress in userspace can be used with either > approaches ? This will not be relevant after you see below answer. > >http://lxr.free-electrons.com/source/sound/core/compress_offload.c >http://lxr.free-electrons.com/source/sound/soc/soc-compress.c Both are compress framework driver files. compress_offload.c file is the one which interacts with userspace by using ioctl's, registers the compressed device with alsa framework(/dev/compress* gets created because of that), calls the corresponding functions such as trigger, pointer update, copy and others when userspace pauses, plays and stops the playback or recording. soc_compress.c file is the file which interacts with compress_offload.c file to create a 'dev' node for compress driver.Basically when you register your compress driver with compress framework,alsa core will come to know that your dai support compress operations and it will create the dev node by using soc_compress.c file.The functions in this file gets called by compress_offload.c functions and who inturn gets called by the userspace. userspace->compress_offload.c->soc_compress.c->your_compress_driver.

2 1

Re: [alsa-devel] [PATCH v2] ASoC: simple-card: add Device Tree support
by Mark Brown 03 Dec '13

03 Dec '13

On Thu, Oct 03, 2013 at 02:33:55AM -0700, Kuninori Morimoto wrote: > ping ? Don't send contentless pings. I'm waiting for review from the device tree guys but I see now that you didn't CC them so they're unlikely to respond - please resend CCing them.

4 31

[alsa-devel] [PATCH 37/39] sound: remove DEFINE_PCI_DEVICE_TABLE macro
by Jingoo Han 03 Dec '13

03 Dec '13

Don't use DEFINE_PCI_DEVICE_TABLE macro, because this macro is not preferred. Signed-off-by: Jingoo Han <jg1.han(a)samsung.com> --- sound/oss/kahlua.c | 2 +- sound/pci/ad1889.c | 2 +- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/als300.c | 2 +- sound/pci/als4000.c | 2 +- sound/pci/asihpi/asihpi.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au8810.c | 2 +- sound/pci/au88x0/au8820.c | 2 +- sound/pci/au88x0/au8830.c | 2 +- sound/pci/aw2/aw2-alsa.c | 2 +- sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 4 ++-- sound/pci/ca0106/ca0106_main.c | 2 +- sound/pci/cmipci.c | 4 ++-- sound/pci/cs4281.c | 2 +- sound/pci/cs46xx/cs46xx.c | 2 +- sound/pci/cs5530.c | 2 +- sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/ctxfi/xfi.c | 2 +- sound/pci/echoaudio/darla20.c | 2 +- sound/pci/echoaudio/darla24.c | 2 +- sound/pci/echoaudio/echo3g.c | 2 +- sound/pci/echoaudio/gina20.c | 2 +- sound/pci/echoaudio/gina24.c | 2 +- sound/pci/echoaudio/indigo.c | 2 +- sound/pci/echoaudio/indigodj.c | 2 +- sound/pci/echoaudio/indigodjx.c | 2 +- sound/pci/echoaudio/indigoio.c | 2 +- sound/pci/echoaudio/indigoiox.c | 2 +- sound/pci/echoaudio/layla20.c | 2 +- sound/pci/echoaudio/layla24.c | 2 +- sound/pci/echoaudio/mia.c | 2 +- sound/pci/echoaudio/mona.c | 2 +- sound/pci/emu10k1/emu10k1.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 2 +- sound/pci/ens1370.c | 2 +- sound/pci/es1938.c | 2 +- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- sound/pci/intel8x0.c | 2 +- sound/pci/intel8x0m.c | 2 +- sound/pci/korg1212/korg1212.c | 2 +- sound/pci/lola/lola.c | 2 +- sound/pci/lx6464es/lx6464es.c | 2 +- sound/pci/maestro3.c | 2 +- sound/pci/mixart/mixart.c | 2 +- sound/pci/nm256/nm256.c | 2 +- sound/pci/oxygen/oxygen.c | 2 +- sound/pci/oxygen/virtuoso.c | 2 +- sound/pci/pcxhr/pcxhr.c | 2 +- sound/pci/riptide/riptide.c | 4 ++-- sound/pci/rme32.c | 2 +- sound/pci/rme96.c | 2 +- sound/pci/rme9652/hdsp.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- sound/pci/rme9652/rme9652.c | 2 +- sound/pci/sis7019.c | 2 +- sound/pci/sonicvibes.c | 2 +- sound/pci/trident/trident.c | 2 +- sound/pci/via82xx.c | 2 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/vx222/vx222.c | 2 +- sound/pci/ymfpci/ymfpci.c | 2 +- 68 files changed, 71 insertions(+), 71 deletions(-) diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index 12be1fb..c4b0434 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -197,7 +197,7 @@ MODULE_LICENSE("GPL"); * 5530 only. The 5510/5520 decode is different. */ -static DEFINE_PCI_DEVICE_TABLE(id_tbl) = { +static const struct pci_device_id id_tbl[] = { { PCI_VDEVICE(CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO), 0 }, { } }; diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index b680d03..793cf1e 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -1048,7 +1048,7 @@ snd_ad1889_remove(struct pci_dev *pci) snd_card_free(pci_get_drvdata(pci)); } -static DEFINE_PCI_DEVICE_TABLE(snd_ad1889_ids) = { +static const struct pci_device_id snd_ad1889_ids[] = { { PCI_DEVICE(PCI_VENDOR_ID_ANALOG_DEVICES, PCI_DEVICE_ID_AD1889JS) }, { 0, }, }; diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index c6835a3..f7bdcc6 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -275,7 +275,7 @@ struct snd_ali { #endif }; -static DEFINE_PCI_DEVICE_TABLE(snd_ali_ids) = { +static const struct pci_device_id snd_ali_ids[] = { {PCI_DEVICE(PCI_VENDOR_ID_AL, PCI_DEVICE_ID_AL_M5451), 0, 0, 0}, {0, } }; diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 591efb6..7bfeed8 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -152,7 +152,7 @@ struct snd_als300_substream_data { int block_counter_register; }; -static DEFINE_PCI_DEVICE_TABLE(snd_als300_ids) = { +static const struct pci_device_id snd_als300_ids[] = { { 0x4005, 0x0300, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300 }, { 0x4005, 0x0308, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300_PLUS }, { 0, } diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index ffc821b..6c9429e 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -116,7 +116,7 @@ struct snd_card_als4000 { #endif }; -static DEFINE_PCI_DEVICE_TABLE(snd_als4000_ids) = { +static const struct pci_device_id snd_als4000_ids[] = { { 0x4005, 0x4000, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ALS4000 */ { 0, } }; diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 5f2acd3..623f1ae 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -2960,7 +2960,7 @@ static void snd_asihpi_remove(struct pci_dev *pci_dev) asihpi_adapter_remove(pci_dev); } -static DEFINE_PCI_DEVICE_TABLE(asihpi_pci_tbl) = { +static const struct pci_device_id asihpi_pci_tbl[] = { {HPI_PCI_VENDOR_ID_TI, HPI_PCI_DEV_ID_DSP6205, HPI_PCI_VENDOR_ID_AUDIOSCIENCE, PCI_ANY_ID, 0, 0, (kernel_ulong_t)HPI_6205}, diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index f6dec3e..144d594 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -286,7 +286,7 @@ struct atiixp { /* */ -static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = { +static const struct pci_device_id snd_atiixp_ids[] = { { PCI_VDEVICE(ATI, 0x4341), 0 }, /* SB200 */ { PCI_VDEVICE(ATI, 0x4361), 0 }, /* SB300 */ { PCI_VDEVICE(ATI, 0x4370), 0 }, /* SB400 */ diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 289563e..c6d1148 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -261,7 +261,7 @@ struct atiixp_modem { /* */ -static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = { +static const struct pci_device_id snd_atiixp_ids[] = { { PCI_VDEVICE(ATI, 0x434d), 0 }, /* SB200 */ { PCI_VDEVICE(ATI, 0x4378), 0 }, /* SB400 */ { 0, } diff --git a/sound/pci/au88x0/au8810.c b/sound/pci/au88x0/au8810.c index aa51cc7..1b2e340 100644 --- a/sound/pci/au88x0/au8810.c +++ b/sound/pci/au88x0/au8810.c @@ -1,6 +1,6 @@ #include "au8810.h" #include "au88x0.h" -static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { +static const struct pci_device_id snd_vortex_ids[] = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE), 1,}, {0,} }; diff --git a/sound/pci/au88x0/au8820.c b/sound/pci/au88x0/au8820.c index 2f321e7..74c53fa 100644 --- a/sound/pci/au88x0/au8820.c +++ b/sound/pci/au88x0/au8820.c @@ -1,6 +1,6 @@ #include "au8820.h" #include "au88x0.h" -static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { +static const struct pci_device_id snd_vortex_ids[] = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1), 0,}, {0,} }; diff --git a/sound/pci/au88x0/au8830.c b/sound/pci/au88x0/au8830.c index 279b78f..56f675a 100644 --- a/sound/pci/au88x0/au8830.c +++ b/sound/pci/au88x0/au8830.c @@ -1,6 +1,6 @@ #include "au8830.h" #include "au88x0.h" -static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { +static const struct pci_device_id snd_vortex_ids[] = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2), 0,}, {0,} }; diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 2925220..f05b843 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -160,7 +160,7 @@ MODULE_PARM_DESC(id, "ID string for the Audiowerk2 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); -static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = { +static const struct pci_device_id snd_aw2_ids[] = { {PCI_VENDOR_ID_PHILIPS, PCI_DEVICE_ID_PHILIPS_SAA7146, 0, 0, 0, 0, 0}, {0} diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 1aef712..4d5b97b 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -376,7 +376,7 @@ struct snd_azf3328 { #endif }; -static DEFINE_PCI_DEVICE_TABLE(snd_azf3328_ids) = { +static const struct pci_device_id snd_azf3328_ids[] = { { 0x122D, 0x50DC, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* PCI168/3328 */ { 0x122D, 0x80DA, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* 3328 */ { 0, } diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 1880203..55a3d1e 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -789,7 +789,7 @@ fail: .driver_data = SND_BT87X_BOARD_ ## id } /* driver_data is the card id for that device */ -static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_ids) = { +static const struct pci_device_id snd_bt87x_ids[] = { /* Hauppauge WinTV series */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, GENERIC), /* Hauppauge WinTV series */ @@ -957,7 +957,7 @@ static void snd_bt87x_remove(struct pci_dev *pci) /* default entries for all Bt87x cards - it's not exported */ /* driver_data is set to 0 to call detection */ -static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_default_ids) = { +static const struct pci_device_id snd_bt87x_default_ids[] = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), { } diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index f4db558..bbf9f39 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1959,7 +1959,7 @@ static SIMPLE_DEV_PM_OPS(snd_ca0106_pm, snd_ca0106_suspend, snd_ca0106_resume); #endif // PCI IDs -static DEFINE_PCI_DEVICE_TABLE(snd_ca0106_ids) = { +static const struct pci_device_id snd_ca0106_ids[] = { { PCI_VDEVICE(CREATIVE, 0x0007), 0 }, /* Audigy LS or Live 24bit */ { 0, } }; diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 2755ec5..fb73198 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2803,7 +2803,7 @@ static inline void snd_cmipci_proc_init(struct cmipci *cm) {} #endif -static DEFINE_PCI_DEVICE_TABLE(snd_cmipci_ids) = { +static const struct pci_device_id snd_cmipci_ids[] = { {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A), 0}, {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B), 0}, {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738), 0}, @@ -3025,7 +3025,7 @@ static int snd_cmipci_create(struct snd_card *card, struct pci_dev *pci, int integrated_midi = 0; char modelstr[16]; int pcm_index, pcm_spdif_index; - static DEFINE_PCI_DEVICE_TABLE(intel_82437vx) = { + static const struct pci_device_id intel_82437vx[] = { { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82437VX) }, { }, }; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 1dc793e..3eef727 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -494,7 +494,7 @@ struct cs4281 { static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id); -static DEFINE_PCI_DEVICE_TABLE(snd_cs4281_ids) = { +static const struct pci_device_id snd_cs4281_ids[] = { { PCI_VDEVICE(CIRRUS, 0x6005), 0, }, /* CS4281 */ { 0, } }; diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index b034983..8234fbb 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -64,7 +64,7 @@ MODULE_PARM_DESC(thinkpad, "Force to enable Thinkpad's CLKRUN control."); module_param_array(mmap_valid, bool, NULL, 0444); MODULE_PARM_DESC(mmap_valid, "Support OSS mmap."); -static DEFINE_PCI_DEVICE_TABLE(snd_cs46xx_ids) = { +static const struct pci_device_id snd_cs46xx_ids[] = { { PCI_VDEVICE(CIRRUS, 0x6001), 0, }, /* CS4280 */ { PCI_VDEVICE(CIRRUS, 0x6003), 0, }, /* CS4612 */ { PCI_VDEVICE(CIRRUS, 0x6004), 0, }, /* CS4615 */ diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index c6b82c8..1f86a42 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -66,7 +66,7 @@ struct snd_cs5530 { unsigned long pci_base; }; -static DEFINE_PCI_DEVICE_TABLE(snd_cs5530_ids) = { +static const struct pci_device_id snd_cs5530_ids[] = { {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0}, {0,} diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 902bebd..0276e12 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -66,7 +66,7 @@ MODULE_PARM_DESC(id, "ID string for " DRIVER_NAME); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " DRIVER_NAME); -static DEFINE_PCI_DEVICE_TABLE(snd_cs5535audio_ids) = { +static const struct pci_device_id snd_cs5535audio_ids[] = { { PCI_DEVICE(PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO) }, { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO) }, {} diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index d464ad2..568b365 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -44,7 +44,7 @@ MODULE_PARM_DESC(enable, "Enable Creative X-Fi driver"); module_param_array(subsystem, int, NULL, 0444); MODULE_PARM_DESC(subsystem, "Override subsystem ID for Creative X-Fi driver"); -static DEFINE_PCI_DEVICE_TABLE(ct_pci_dev_ids) = { +static const struct pci_device_id ct_pci_dev_ids[] = { /* only X-Fi is supported, so... */ { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K1), .driver_data = ATC20K1, diff --git a/sound/pci/echoaudio/darla20.c b/sound/pci/echoaudio/darla20.c index d47e72a..4632946 100644 --- a/sound/pci/echoaudio/darla20.c +++ b/sound/pci/echoaudio/darla20.c @@ -63,7 +63,7 @@ static const struct firmware card_fw[] = { {0, "darla20_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x1801, 0xECC0, 0x0010, 0, 0, 0}, /* DSP 56301 Darla20 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/darla24.c b/sound/pci/echoaudio/darla24.c index 413acf7..f81c839 100644 --- a/sound/pci/echoaudio/darla24.c +++ b/sound/pci/echoaudio/darla24.c @@ -67,7 +67,7 @@ static const struct firmware card_fw[] = { {0, "darla24_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x1801, 0xECC0, 0x0040, 0, 0, 0}, /* DSP 56301 Darla24 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0041, 0, 0, 0}, /* DSP 56301 Darla24 rev.1 */ {0,} diff --git a/sound/pci/echoaudio/echo3g.c b/sound/pci/echoaudio/echo3g.c index 1ec4edc..3a5346c 100644 --- a/sound/pci/echoaudio/echo3g.c +++ b/sound/pci/echoaudio/echo3g.c @@ -81,7 +81,7 @@ static const struct firmware card_fw[] = { {0, "3g_asic.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x0100, 0, 0, 0}, /* Echo 3G */ {0,} }; diff --git a/sound/pci/echoaudio/gina20.c b/sound/pci/echoaudio/gina20.c index 039125b..9cb81c5 100644 --- a/sound/pci/echoaudio/gina20.c +++ b/sound/pci/echoaudio/gina20.c @@ -67,7 +67,7 @@ static const struct firmware card_fw[] = { {0, "gina20_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x1801, 0xECC0, 0x0020, 0, 0, 0}, /* DSP 56301 Gina20 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/gina24.c b/sound/pci/echoaudio/gina24.c index 5e966f6..35d3e6e 100644 --- a/sound/pci/echoaudio/gina24.c +++ b/sound/pci/echoaudio/gina24.c @@ -85,7 +85,7 @@ static const struct firmware card_fw[] = { {0, "gina24_361_asic.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x1801, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56301 Gina24 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0051, 0, 0, 0}, /* DSP 56301 Gina24 rev.1 */ {0x1057, 0x3410, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56361 Gina24 rev.0 */ diff --git a/sound/pci/echoaudio/indigo.c b/sound/pci/echoaudio/indigo.c index c166b7e..8d91842 100644 --- a/sound/pci/echoaudio/indigo.c +++ b/sound/pci/echoaudio/indigo.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x0090, 0, 0, 0}, /* Indigo */ {0,} }; diff --git a/sound/pci/echoaudio/indigodj.c b/sound/pci/echoaudio/indigodj.c index a3ef3b9..289cb96 100644 --- a/sound/pci/echoaudio/indigodj.c +++ b/sound/pci/echoaudio/indigodj.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_dj_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x00B0, 0, 0, 0}, /* Indigo DJ*/ {0,} }; diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c index f516444..201688e 100644 --- a/sound/pci/echoaudio/indigodjx.c +++ b/sound/pci/echoaudio/indigodjx.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_djx_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx*/ {0,} }; diff --git a/sound/pci/echoaudio/indigoio.c b/sound/pci/echoaudio/indigoio.c index c22c82f..405a3f2 100644 --- a/sound/pci/echoaudio/indigoio.c +++ b/sound/pci/echoaudio/indigoio.c @@ -69,7 +69,7 @@ static const struct firmware card_fw[] = { {0, "indigo_io_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x00A0, 0, 0, 0}, /* Indigo IO*/ {0,} }; diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c index 86cf2d0..e145b68 100644 --- a/sound/pci/echoaudio/indigoiox.c +++ b/sound/pci/echoaudio/indigoiox.c @@ -69,7 +69,7 @@ static const struct firmware card_fw[] = { {0, "indigo_iox_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ {0,} }; diff --git a/sound/pci/echoaudio/layla20.c b/sound/pci/echoaudio/layla20.c index 6a027f3..b392dd7 100644 --- a/sound/pci/echoaudio/layla20.c +++ b/sound/pci/echoaudio/layla20.c @@ -76,7 +76,7 @@ static const struct firmware card_fw[] = { {0, "layla20_asic.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x1801, 0xECC0, 0x0030, 0, 0, 0}, /* DSP 56301 Layla20 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0031, 0, 0, 0}, /* DSP 56301 Layla20 rev.1 */ {0,} diff --git a/sound/pci/echoaudio/layla24.c b/sound/pci/echoaudio/layla24.c index 96a5991..bc7f730 100644 --- a/sound/pci/echoaudio/layla24.c +++ b/sound/pci/echoaudio/layla24.c @@ -87,7 +87,7 @@ static const struct firmware card_fw[] = { {0, "layla24_2S_asic.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x0060, 0, 0, 0}, /* DSP 56361 Layla24 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c index b8ce27e..27a9a6e 100644 --- a/sound/pci/echoaudio/mia.c +++ b/sound/pci/echoaudio/mia.c @@ -77,7 +77,7 @@ static const struct firmware card_fw[] = { {0, "mia_dsp.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x3410, 0xECC0, 0x0080, 0, 0, 0}, /* DSP 56361 Mia rev.0 */ {0x1057, 0x3410, 0xECC0, 0x0081, 0, 0, 0}, /* DSP 56361 Mia rev.1 */ {0,} diff --git a/sound/pci/echoaudio/mona.c b/sound/pci/echoaudio/mona.c index 1283bfb..3d13875 100644 --- a/sound/pci/echoaudio/mona.c +++ b/sound/pci/echoaudio/mona.c @@ -92,7 +92,7 @@ static const struct firmware card_fw[] = { {0, "mona_2_asic.fw"} }; -static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { +static const struct pci_device_id snd_echo_ids[] = { {0x1057, 0x1801, 0xECC0, 0x0070, 0, 0, 0}, /* DSP 56301 Mona rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0071, 0, 0, 0}, /* DSP 56301 Mona rev.1 */ {0x1057, 0x1801, 0xECC0, 0x0072, 0, 0, 0}, /* DSP 56301 Mona rev.2 */ diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 9e1bd0c..7feeb78 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -79,7 +79,7 @@ MODULE_PARM_DESC(delay_pcm_irq, "Delay PCM interrupt by specified number of samp /* * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400 */ -static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1_ids) = { +static const struct pci_device_id snd_emu10k1_ids[] = { { PCI_VDEVICE(CREATIVE, 0x0002), 0 }, /* EMU10K1 */ { PCI_VDEVICE(CREATIVE, 0x0004), 1 }, /* Audigy */ { PCI_VDEVICE(CREATIVE, 0x0008), 1 }, /* Audigy 2 Value SB0400 */ diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 56ad9d6..6a80ab8 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1626,7 +1626,7 @@ static void snd_emu10k1x_remove(struct pci_dev *pci) } // PCI IDs -static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1x_ids) = { +static const struct pci_device_id snd_emu10k1x_ids[] = { { PCI_VDEVICE(CREATIVE, 0x0006), 0 }, /* Dell OEM version (EMU10K1) */ { 0, } }; diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 61262f3..f5963b6 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -446,7 +446,7 @@ struct ensoniq { static irqreturn_t snd_audiopci_interrupt(int irq, void *dev_id); -static DEFINE_PCI_DEVICE_TABLE(snd_audiopci_ids) = { +static const struct pci_device_id snd_audiopci_ids[] = { #ifdef CHIP1370 { PCI_VDEVICE(ENSONIQ, 0x5000), 0, }, /* ES1370 */ #endif diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 9213fb3..7eb87f2 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -243,7 +243,7 @@ struct es1938 { static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id); -static DEFINE_PCI_DEVICE_TABLE(snd_es1938_ids) = { +static const struct pci_device_id snd_es1938_ids[] = { { PCI_VDEVICE(ESS, 0x1969), 0, }, /* Solo-1 */ { 0, } }; diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index b0e3d92..cd1d947 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -570,7 +570,7 @@ struct es1968 { static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id); -static DEFINE_PCI_DEVICE_TABLE(snd_es1968_ids) = { +static const struct pci_device_id snd_es1968_ids[] = { /* Maestro 1 */ { 0x1285, 0x0100, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, TYPE_MAESTRO }, /* Maestro 2 */ diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 45bc8a9..739f224 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -210,7 +210,7 @@ struct fm801 { #endif }; -static DEFINE_PCI_DEVICE_TABLE(snd_fm801_ids) = { +static const struct pci_device_id snd_fm801_ids[] = { { 0x1319, 0x0801, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* FM801 */ { 0x5213, 0x0510, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* Gallant Odyssey Sound 4 */ { 0, } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c6d2301..d1f11a5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3969,7 +3969,7 @@ static void azx_remove(struct pci_dev *pci) } /* PCI IDs */ -static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { +static const struct pci_device_id azx_ids[] = { /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 28ec872..b8378e7 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -105,7 +105,7 @@ module_param_array(dxr_enable, int, NULL, 0444); MODULE_PARM_DESC(dxr_enable, "Enable DXR support for Terratec DMX6FIRE."); -static DEFINE_PCI_DEVICE_TABLE(snd_ice1712_ids) = { +static const struct pci_device_id snd_ice1712_ids[] = { { PCI_VDEVICE(ICE, PCI_DEVICE_ID_ICE_1712), 0 }, /* ICE1712 */ { 0, } }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 5004717..4a3165b 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -94,7 +94,7 @@ MODULE_PARM_DESC(model, "Use the given board model."); /* Both VT1720 and VT1724 have the same PCI IDs */ -static DEFINE_PCI_DEVICE_TABLE(snd_vt1724_ids) = { +static const struct pci_device_id snd_vt1724_ids[] = { { PCI_VDEVICE(ICE, PCI_DEVICE_ID_VT1724), 0 }, { 0, } }; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 08d8733..e5cb94e 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -430,7 +430,7 @@ struct intel8x0 { u32 int_sta_mask; /* interrupt status mask */ }; -static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0_ids) = { +static const struct pci_device_id snd_intel8x0_ids[] = { { PCI_VDEVICE(INTEL, 0x2415), DEVICE_INTEL }, /* 82801AA */ { PCI_VDEVICE(INTEL, 0x2425), DEVICE_INTEL }, /* 82901AB */ { PCI_VDEVICE(INTEL, 0x2445), DEVICE_INTEL }, /* 82801BA */ diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 3573c11..ab7edfd 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -219,7 +219,7 @@ struct intel8x0m { unsigned int pcm_pos_shift; }; -static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0m_ids) = { +static const struct pci_device_id snd_intel8x0m_ids[] = { { PCI_VDEVICE(INTEL, 0x2416), DEVICE_INTEL }, /* 82801AA */ { PCI_VDEVICE(INTEL, 0x2426), DEVICE_INTEL }, /* 82901AB */ { PCI_VDEVICE(INTEL, 0x2446), DEVICE_INTEL }, /* 82801BA */ diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 9cf9829..9413b8a 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -418,7 +418,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Korg 1212 soundcard."); MODULE_AUTHOR("Haroldo Gamal <gamal(a)alternex.com.br>"); -static DEFINE_PCI_DEVICE_TABLE(snd_korg1212_ids) = { +static const struct pci_device_id snd_korg1212_ids[] = { { .vendor = 0x10b5, .device = 0x906d, diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 0568540..a7beffe 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -762,7 +762,7 @@ static void lola_remove(struct pci_dev *pci) } /* PCI IDs */ -static DEFINE_PCI_DEVICE_TABLE(lola_ids) = { +static const struct pci_device_id lola_ids[] = { { PCI_VDEVICE(DIGIGRAM, 0x0001) }, { 0, } }; diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 5fcaaa6..a016e39 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -56,7 +56,7 @@ static const char card_name[] = "LX6464ES"; #define PCI_DEVICE_ID_PLX_LX6464ES PCI_DEVICE_ID_PLX_9056 -static DEFINE_PCI_DEVICE_TABLE(snd_lx6464es_ids) = { +static const struct pci_device_id snd_lx6464es_ids[] = { { PCI_DEVICE(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES), .subvendor = PCI_VENDOR_ID_DIGIGRAM, .subdevice = PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index d541736..836ce95 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -800,7 +800,7 @@ struct snd_m3 { /* * pci ids */ -static DEFINE_PCI_DEVICE_TABLE(snd_m3_ids) = { +static const struct pci_device_id snd_m3_ids[] = { {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO_1, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO, PCI_ANY_ID, PCI_ANY_ID, diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 1e0f6ee..c9fe3e2 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -61,7 +61,7 @@ MODULE_PARM_DESC(enable, "Enable Digigram " CARD_NAME " soundcard."); /* */ -static DEFINE_PCI_DEVICE_TABLE(snd_mixart_ids) = { +static const struct pci_device_id snd_mixart_ids[] = { { PCI_VDEVICE(MOTOROLA, 0x0003), 0, }, /* MC8240 */ { 0, } }; diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index fe79fff..97eac98 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -262,7 +262,7 @@ struct nm256 { /* * PCI ids */ -static DEFINE_PCI_DEVICE_TABLE(snd_nm256_ids) = { +static const struct pci_device_id snd_nm256_ids[] = { {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO), 0}, {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO), 0}, {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO), 0}, diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index ada6c25..74afb6b 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -97,7 +97,7 @@ enum { MODEL_XONAR_DGX, }; -static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { +static const struct pci_device_id oxygen_ids[] = { /* C-Media's reference design */ { OXYGEN_PCI_SUBID(0x10b0, 0x0216), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0217), .driver_data = MODEL_CMEDIA_REF }, diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 64b9fda..ce33544 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -41,7 +41,7 @@ MODULE_PARM_DESC(id, "ID string"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); -static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { +static const struct pci_device_id xonar_ids[] = { { OXYGEN_PCI_SUBID(0x1043, 0x8269) }, { OXYGEN_PCI_SUBID(0x1043, 0x8275) }, { OXYGEN_PCI_SUBID(0x1043, 0x82b7) }, diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index d379b28..a4b05a2 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -102,7 +102,7 @@ enum { PCI_ID_LAST }; -static DEFINE_PCI_DEVICE_TABLE(pcxhr_ids) = { +static const struct pci_device_id pcxhr_ids[] = { { 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, }, { 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, }, { 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, }, diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 56cc891..d29fa04 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -508,7 +508,7 @@ static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip); /* */ -static DEFINE_PCI_DEVICE_TABLE(snd_riptide_ids) = { +static const struct pci_device_id snd_riptide_ids[] = { { PCI_DEVICE(0x127a, 0x4310) }, { PCI_DEVICE(0x127a, 0x4320) }, { PCI_DEVICE(0x127a, 0x4330) }, @@ -517,7 +517,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_riptide_ids) = { }; #ifdef SUPPORT_JOYSTICK -static DEFINE_PCI_DEVICE_TABLE(snd_riptide_joystick_ids) = { +static const struct pci_device_id snd_riptide_joystick_ids[] = { { PCI_DEVICE(0x127a, 0x4312) }, { PCI_DEVICE(0x127a, 0x4322) }, { PCI_DEVICE(0x127a, 0x4332) }, diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index cc26346..ef93838 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -226,7 +226,7 @@ struct rme32 { struct snd_kcontrol *spdif_ctl; }; -static DEFINE_PCI_DEVICE_TABLE(snd_rme32_ids) = { +static const struct pci_device_id snd_rme32_ids[] = { {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32), 0,}, {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8), 0,}, {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_PRO), 0,}, diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 0236363..1211d36 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -263,7 +263,7 @@ struct rme96 { struct snd_kcontrol *spdif_ctl; }; -static DEFINE_PCI_DEVICE_TABLE(snd_rme96_ids) = { +static const struct pci_device_id snd_rme96_ids[] = { { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96), 0, }, { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8), 0, }, { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO), 0, }, diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index f59a321..fd348ce 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -603,7 +603,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static DEFINE_PCI_DEVICE_TABLE(snd_hdsp_ids) = { +static const struct pci_device_id snd_hdsp_ids[] = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index e98dc00..c6bf321 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1077,7 +1077,7 @@ struct hdspm { }; -static DEFINE_PCI_DEVICE_TABLE(snd_hdspm_ids) = { +static const struct pci_device_id snd_hdspm_ids[] = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP_MADI, diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index b96d9e1..18cf23d 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -313,7 +313,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static DEFINE_PCI_DEVICE_TABLE(snd_rme9652_ids) = { +static const struct pci_device_id snd_rme9652_ids[] = { { .vendor = 0x10ee, .device = 0x3fc4, diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index e413b4e..196df18 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -52,7 +52,7 @@ MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); module_param(codecs, int, 0444); MODULE_PARM_DESC(codecs, "Set bit to indicate that codec number is expected to be present (default 1)"); -static DEFINE_PCI_DEVICE_TABLE(snd_sis7019_ids) = { +static const struct pci_device_id snd_sis7019_ids[] = { { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, { 0, } }; diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 2a46bf9..bc929f3 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -242,7 +242,7 @@ struct sonicvibes { #endif }; -static DEFINE_PCI_DEVICE_TABLE(snd_sonic_ids) = { +static const struct pci_device_id snd_sonic_ids[] = { { PCI_VDEVICE(S3, 0xca00), 0, }, { 0, } }; diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index b3b588b..29a0d32 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -62,7 +62,7 @@ MODULE_PARM_DESC(pcm_channels, "Number of hardware channels assigned for PCM."); module_param_array(wavetable_size, int, NULL, 0444); MODULE_PARM_DESC(wavetable_size, "Maximum memory size in kB for wavetable synth."); -static DEFINE_PCI_DEVICE_TABLE(snd_trident_ids) = { +static const struct pci_device_id snd_trident_ids[] = { {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_DX), PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_NX), diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 5ae6f04..7029142 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -404,7 +404,7 @@ struct via82xx { #endif }; -static DEFINE_PCI_DEVICE_TABLE(snd_via82xx_ids) = { +static const struct pci_device_id snd_via82xx_ids[] = { /* 0x1106, 0x3058 */ { PCI_VDEVICE(VIA, PCI_DEVICE_ID_VIA_82C686_5), TYPE_CARD_VIA686, }, /* 686A */ /* 0x1106, 0x3059 */ diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index ca19028..7424c13 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -260,7 +260,7 @@ struct via82xx_modem { struct snd_info_entry *proc_entry; }; -static DEFINE_PCI_DEVICE_TABLE(snd_via82xx_modem_ids) = { +static const struct pci_device_id snd_via82xx_modem_ids[] = { { PCI_VDEVICE(VIA, 0x3068), TYPE_CARD_VIA82XX_MODEM, }, { 0, } }; diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index ab8a9b1..c4fa2f7 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -60,7 +60,7 @@ enum { VX_PCI_VX222_NEW }; -static DEFINE_PCI_DEVICE_TABLE(snd_vx222_ids) = { +static const struct pci_device_id snd_vx222_ids[] = { { 0x10b5, 0x9050, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_OLD, }, /* PLX */ { 0x10b5, 0x9030, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_NEW, }, /* PLX */ { 0, } diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index e8932b2..3e14fd7 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -66,7 +66,7 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address"); module_param_array(rear_switch, bool, NULL, 0444); MODULE_PARM_DESC(rear_switch, "Enable shared rear/line-in switch"); -static DEFINE_PCI_DEVICE_TABLE(snd_ymfpci_ids) = { +static const struct pci_device_id snd_ymfpci_ids[] = { { PCI_VDEVICE(YAMAHA, 0x0004), 0, }, /* YMF724 */ { PCI_VDEVICE(YAMAHA, 0x000d), 0, }, /* YMF724F */ { PCI_VDEVICE(YAMAHA, 0x000a), 0, }, /* YMF740 */ -- 1.7.10.4

1 0

[alsa-devel] Bug: plugging in a new sound card causes mixer state of other sound cards to be changed
by Angus McInnes 02 Dec '13

02 Dec '13

Hi, When I plug in my USB sound card, the mixer settings of the internal sound card are changed. I don't want this to happen. I believe the problem is with this commit: http://git.alsa-project.org/?p=alsa-utils.git;a=blobdiff;f=alsactl/90-alsa-… The card number $attr{number} comes from the card* device. Since KERNELS=="card*" was changed to KERNELS!="card*", alsactl restore is run with no card number attribute, so it restores the state of all cards. If I change the != back to ==, alsactl restore is run with a card number and the problem seems to be resolved. (By the way, your bug tracker at https://bugtrack.alsa-project.org/alsa-bug seems to be down.) Regards, Angus McInnes

1 0

[alsa-devel] [PATCH] ASoC: omap: n810: Convert to clk_prepare_enable/clk_disable_unprepare
by Jarkko Nikula 02 Dec '13

02 Dec '13

N810 audio driver has stopped working at some point. Probably when OMAP2 was converted to common clock framework since now call to clk_enable dumps the stack trace in drivers/clk/clk.c: __clk_enable() due clk->prepare_count is zero. Fix this by converting clk_enable/_disable calls to those that take care of clock prepare/unprepare. I'm not queueing this to linux-stable since OMAP2 common clock framework conversion in commit ed1ebc4948fd ("ARM: OMAP2: clock: Convert to common clk") happened before N810 was really usable in mainline and user base for N810 is anyway small. Potential linux-stable candidates are only those after commit 3d3a6d18abc6 ("watchdog: introduce retu_wdt driver"). Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- sound/soc/omap/n810.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 6d216cb6c19b..3fde9e402710 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -100,12 +100,12 @@ static int n810_startup(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); n810_ext_control(&codec->dapm); - return clk_enable(sys_clkout2); + return clk_prepare_enable(sys_clkout2); } static void n810_shutdown(struct snd_pcm_substream *substream) { - clk_disable(sys_clkout2); + clk_disable_unprepare(sys_clkout2); } static int n810_hw_params(struct snd_pcm_substream *substream, -- 1.8.4.3

2 4

[alsa-devel] [PATCH v3] ASoC: fsl_ssi: Add monaural audio support for non-ac97 interface
by Nicolin Chen 02 Dec '13

02 Dec '13

The normal mode of SSI allows it to send/receive data to/from the first slot of each period. So we can use this normal mode to trick I2S signal by puting/getting data to/from the first slot only (the left channel) so as to support monaural audio playback and recording. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v3: * Fixed conflict for git apply. v2: * Moved i2s_mode to ssi_private so that we can save and retore it as needed * And dropped the horrible static thing. sound/soc/fsl/fsl_ssi.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index fb8f52a..83e4ed8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -143,6 +143,7 @@ struct fsl_ssi_private { bool ssi_on_imx; bool imx_ac97; bool use_dma; + u8 i2s_mode; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -354,14 +355,13 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u8 i2s_mode; u8 wm; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; if (ssi_private->imx_ac97) - i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; else - i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; /* * Section 16.5 of the MPC8610 reference manual says that the SSI needs @@ -378,7 +378,7 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | - i2s_mode | + ssi_private->i2s_mode | (synchronous ? CCSR_SSI_SCR_SYN : 0)); write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -508,6 +508,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int channels = params_channels(hw_params); unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); u32 wl = CCSR_SSI_SxCCR_WL(sample_size); @@ -537,6 +538,11 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, else write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + if (!ssi_private->imx_ac97) + write_ssi_mask(&ssi->scr, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : ssi_private->i2s_mode); + return 0; } @@ -649,14 +655,13 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { static struct snd_soc_dai_driver fsl_ssi_dai_template = { .probe = fsl_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, -- 1.8.4

2 1

[alsa-devel] Why does `aplay -L' not show PCM's defined in ~/.asoundrc ?
by Gabriel M. Beddingfield 02 Dec '13

02 Dec '13

Hi guys, Why does `aplay -L' not list PCM's that are defined in ~/.asoundrc (i.e. PCM's implemented by plugins)? How do you get a listing that includes these? I was helping a user today who needed to set up a device using the `multi' plugin -- and the fact that the new virtual pcm device didn't show up in any standard listing made things very confusing. -gabriel

2 2

[alsa-devel] [PATCH] ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free()
by Nicolin Chen 02 Dec '13

02 Dec '13

The snd_soc_dai_digital_mute() here will be never executed because we only decrease codec->active in snd_soc_close(). Thus correct it. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 22af038..7d19117 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -689,7 +689,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) } /* apply codec digital mute */ - if (!codec->active) + if (codec->active == 1) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ -- 1.8.4

3 4

[alsa-devel] [PATCH v2] ASoC: fsl_ssi: Add monaural audio support for non-ac97 interface
by Nicolin Chen 02 Dec '13

02 Dec '13

The normal mode of SSI allows it to send/receive data to/from the first slot of each period. So we can use this normal mode to trick I2S signal by puting/getting data to/from the first slot only (the left channel) so as to support monaural audio playback and recording. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v2: * Moved i2s_mode to ssi_private so that we can save and retore it as needed * And dropped the horrible static thing. sound/soc/fsl/fsl_ssi.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f43be6d..cba5c4d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -144,6 +144,7 @@ struct fsl_ssi_private { bool imx_ac97; bool use_dma; bool use_dual_fifo; + u8 i2s_mode; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -325,14 +326,13 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) { struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u8 i2s_mode; u8 wm; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; if (ssi_private->imx_ac97) - i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_NET; else - i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; /* * Section 16.5 of the MPC8610 reference manual says that the SSI needs @@ -349,7 +349,7 @@ static int fsl_ssi_setup(struct fsl_ssi_private *ssi_private) write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, CCSR_SSI_SCR_TFR_CLK_DIS | - i2s_mode | + ssi_private->i2s_mode | (synchronous ? CCSR_SSI_SCR_SYN : 0)); write_ssi(CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -517,6 +517,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int channels = params_channels(hw_params); unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); u32 wl = CCSR_SSI_SxCCR_WL(sample_size); @@ -546,6 +547,11 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, else write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + if (!ssi_private->imx_ac97) + write_ssi_mask(&ssi->scr, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : ssi_private->i2s_mode); + return 0; } @@ -658,14 +664,13 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { static struct snd_soc_dai_driver fsl_ssi_dai_template = { .probe = fsl_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, -- 1.8.4

2 2

[alsa-devel] [PATCH 1/2] ALSA: hda/analog - Handle inverted EAPD properly in vmaster hook
by Takashi Iwai 02 Dec '13

02 Dec '13

ad_vmaster_eapd_hook() needs to handle the inverted EAPD case properly, too. Otherwise the output gets broken on Lenovo N100 with AD1986A codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 34d86ec5d3dd..f6351b8e0f5d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -147,6 +147,8 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled) if (!spec->eapd_nid) return; + if (codec->inv_eapd) + enabled = !enabled; snd_hda_codec_update_cache(codec, spec->eapd_nid, 0, AC_VERB_SET_EAPD_BTLENABLE, enabled ? 0x02 : 0x00); -- 1.8.4.4

1 1

[alsa-devel] [PATCH] ALSA: hda - Another fixup for ASUS laptop with ALC660 codec
by Takashi Iwai 02 Dec '13

02 Dec '13

ASUS Z35HL laptop also needs the very same fix as the previous one that was applied to ASUS W7J. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66231 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a98f3296b891..9ca83001fc37 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4559,6 +4559,7 @@ static const struct hda_fixup alc861_fixups[] = { static const struct snd_pci_quirk alc861_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1253, "ASUS W7J", ALC660_FIXUP_ASUS_W7J), + SND_PCI_QUIRK(0x1043, 0x1263, "ASUS Z35HL", ALC660_FIXUP_ASUS_W7J), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT), -- 1.8.4.4

1 0

[alsa-devel] HD-Audio: How to reduce driver initializaton time if multiple codecs present on the bus?
by Lin, Mengdong 02 Dec '13

02 Dec '13

Hi Takashi, We're trying to reduce the HD-A driver initialization time when more than one codecs are connected to the bus, but are blocked. Would you please share some advices on this? Usually, there is one HD-A controller connecting to two codecs: one on-board codec and one integrated display codec. During initialization, the codecs are created and configured in a serial way. Creating a codec may cost 6~20ms, and then building controls make cost about 15~30ms. So I had thought it's helpful to build controls for different codecs in parallel in function snd_hda_build_controls(), as we did in driver suspend/resume. But the test shows this doesn't work: (1) Using a workqueue without WQ_UNBOUND, the work items, which calls snd_hda_codec_build_controls(), are processed one after one (I think because there is no explicit sleep in the work function). So the total time does not change. (2) Using an unbouned work queue for the bus or using separate kthreads for each codec, the work items for different codecs are processed in parallel. However, the total time is still the sum of two codec as that without parallel. I think that because snd_hda_codec_read() are used during initialization, the command execution on bus is serialized in the low level. So even if we parallel initializing two codecs in high level, commands on bus are still processed one by one so the total time cannot reduce. We're blocked here. Is there any other means to reduce the time? Thanks Mengdong

3 15

[alsa-devel] [PATCH] ALSA: hda - Fix bad EAPD setup for HP machines with AD1984A
by Takashi Iwai 02 Dec '13

02 Dec '13

It seems that EAPD on NID 0x16 is the only control over all outputs on HP machines with AD1984A while turning EAPD on NID 0x12 breaks the output. Thus we need to avoid fiddling EAPD on NID. As a quick workaround, just set own_eapd_ctrl flag for the wrong EAPD, then implement finer EAPD controls. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66321 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1a83559f4cbd..34d86ec5d3dd 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -962,6 +962,7 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook; + spec->gen.own_eapd_ctl = 1; snd_hda_sequence_write_cache(codec, gpio_init_verbs); break; case HDA_FIXUP_ACT_PROBE: -- 1.8.4.4

1 0

[alsa-devel] [PATCH] ASoC: fsl: imx-ssi: omit ssi counter to avoid harm in unbalanced situation
by Oskar Schirmer 02 Dec '13

02 Dec '13

Unbalanced calls to imx_ssi_trigger() may result in endless SSI activity and thus provoke eternal sound. While on the first glance, the switch statement looks pretty symmetric, the SUSPEND/RESUME pair is not: the suspend case comes along snd_pcm_suspend_all(), which for fsl/imx-pcm-fiq is called only at snd_soc_suspend(), but the resume case originates straight from the SNDRV_PCM_IOCTL_RESUME. This way userland may provoke an unbalanced resume, which might cause the ssi->enabled counter to increase and never return to zero again, so eventually SSI_SCR_SSIEN is never disabled. Simply removing the ssi->enabled will solve the problem, as long as one never goes play and capture game simultaneously, but beware trying both at once, the early TRIGGER_STOP will cut off the other activity prematurely. So now playing and capturing is scrutinized separately, instead of by counting. This is essentially the same stuff as in sound/soc/fsl/imx-pcm-fiq.c which I send a patch for three days ago. Astonishing enough this highly fragile scheme is used twice in parallel to serve the very same control function, synchronously: Once out of sync you are lost until reboot. Note, that these fixes wont prevent state machine distortion on alsa level to cut sound or the like. It just makes sure we have a chance to synchronise again later on. Signed-off-by: Oskar Schirmer <oskar(a)scara.com> --- sound/soc/fsl/imx-ssi.c | 19 ++++++++++++------- sound/soc/fsl/imx-ssi.h | 3 ++- 2 files changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f5f248c..c68499b 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -298,27 +298,32 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { scr |= SSI_SCR_TE; - else + ssi->playing = 1; + } else { scr |= SSI_SCR_RE; + ssi->capturing = 1; + } sier |= sier_bits; - if (++ssi->enabled == 1) - scr |= SSI_SCR_SSIEN; + scr |= SSI_SCR_SSIEN; break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { scr &= ~SSI_SCR_TE; - else + ssi->playing = 0; + } else { scr &= ~SSI_SCR_RE; + ssi->capturing = 0; + } sier &= ~sier_bits; - if (--ssi->enabled == 0) + if (!ssi->playing && !ssi->capturing) scr &= ~SSI_SCR_SSIEN; break; diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index 560c40f..a3c90d5 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -213,7 +213,8 @@ struct imx_ssi { int fiq_init; int dma_init; - int enabled; + int playing; + int capturing; }; #endif /* _IMX_SSI_H */ -- 1.7.9.5

2 3

[alsa-devel] [PATCH 1/1] ASoC: soc_compress: Add set_metadata
by Richard Fitzgerald 02 Dec '13

02 Dec '13

Pass the set_metadata() calls through to the codec driver. Signed-off-by: Zhao Weijia <weijia.zhao(a)capelabs.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 18 +++++++++++++++++- 1 files changed, 17 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 53c9ecd..186802b 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -318,6 +318,21 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, mutex_unlock(&rtd->pcm_mutex); return 0; } +static int soc_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + if (platform->driver->compr_ops && platform->driver->compr_ops->set_metadata) + platform->driver->compr_ops->set_metadata(cstream, metadata); + + mutex_unlock(&rtd->pcm_mutex); + return 0; +} + static int soc_compr_copy(struct snd_compr_stream *cstream, char __user *buf, size_t count) @@ -372,7 +387,8 @@ static struct snd_compr_ops soc_compr_ops = { .pointer = soc_compr_pointer, .ack = soc_compr_ack, .get_caps = soc_compr_get_caps, - .get_codec_caps = soc_compr_get_codec_caps + .get_codec_caps = soc_compr_get_codec_caps, + .set_metadata = soc_compr_set_metadata, }; /* create a new compress */ -- 1.7.2.5

2 2

[alsa-devel] Using devm_snd_soc_register_card() causes problem with imx-sgtl5000
by Shawn Guo 02 Dec '13

02 Dec '13

I'm trying to test Lars' devm_snd_dmaengine_pcm_register patches with imx-sgtl5000 machine driver, but running into the rmmod problem below on v3.13-rc1. Reverting commit 01984a4 (ASoC: imx-sgtl5000: Use devm_snd_soc_register_card) seems to fix the problem for me. Any hint on what's going wrong here? Shawn root@arm:~# lsmod Module Size Used by snd_soc_imx_sgtl5000 3506 0 snd_soc_sgtl5000 13677 2 snd_soc_imx_audmux 5324 1 snd_soc_imx_sgtl5000 snd_soc_fsl_ssi 8139 2 imx_pcm_dma 1380 1 snd_soc_fsl_ssi root@arm:~# rmmod snd_soc_imx_sgtl5000 Unable to handle kernel paging request at virtual address e594025c pgd = be134000 [e594025c] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: snd_soc_imx_sgtl5000(-) snd_soc_sgtl5000 snd_soc_imx_audmux snd_soc_fsl_ssi imx_pcm_dma CPU: 0 PID: 1793 Comm: rmmod Not tainted 3.13.0-rc1 #1570 task: bee28900 ti: bfbec000 task.ti: bfbec000 PC is at debugfs_remove_recursive+0x28/0x154 LR is at snd_soc_unregister_card+0xa0/0xcc pc : [<80252b38>] lr : [<80496ac4>] psr: a0000013 sp : bfbede00 ip : bfbede28 fp : bfbede24 r10: 803281d4 r9 : bfbec000 r8 : 803271ac r7 : bef54440 r6 : 00000004 r5 : bf9a4010 r4 : bf9a4010 r3 : e5940224 r2 : 00000000 r1 : bef54450 r0 : 803271ac Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c53c7d Table: 4e13404a DAC: 00000015 Process rmmod (pid: 1793, stack limit = 0xbfbec240) Stack: (0xbfbede00 to 0xbfbee000) de00: 00000000 bf9a4010 bf9a4010 00000004 bef54440 bec89000 bfbede44 bfbede28 de20: 80496ac4 80252b1c 804a4b60 bfbede60 bf9a4010 00000004 bfbede54 bfbede48 de40: 804a4b74 80496a30 bfbede94 bfbede58 80328728 804a4b6c bfbede94 a0000013 de60: bf1b5800 bef54440 00000002 bf9a4010 7f0169f8 bf9a4044 00000081 8000e9c4 de80: bfbec000 00000000 bfbedeac bfbede98 80328cb0 80328618 7f016000 bf9a4010 dea0: bfbedec4 bfbedeb0 8032561c 80328c84 bf9a4010 7f0169f8 bfbedee4 bfbedec8 dec0: 80325e84 803255a8 bee28900 7f0169f8 00000000 78208d30 bfbedefc bfbedee8 dee0: 80325410 80325dd4 beca8100 7f0169f8 bfbedf14 bfbedf00 803264f8 803253c8 df00: 7f01635c 7f016a3c bfbedf24 bfbedf18 80327098 803264d4 bfbedf34 bfbedf28 df20: 7f016370 80327090 bfbedfa4 bfbedf38 80085ef0 7f016368 bfbedf54 5f646e73 df40: 5f636f73 5f786d69 6c746773 30303035 00000000 78208008 bfbedf84 bfbedf68 df60: 800613b0 80061194 fffffffe 78208d00 7efc2f07 00000081 7f016a3c 00000800 df80: bfbedf84 00000000 00000000 fffffffe 78208d00 7efc2f07 00000000 bfbedfa8 dfa0: 8000e800 80085dcc fffffffe 78208d00 78208d30 00000800 a8c82400 a8c82400 dfc0: fffffffe 78208d00 7efc2f07 00000081 00000002 00000000 78208008 00000800 dfe0: 7efc2e1c 7efc2ba8 76f5ca47 76edec7c 80000010 78208d30 00000000 00000000 Backtrace: [<80252b10>] (debugfs_remove_recursive+0x0/0x154) from [<80496ac4>] (snd_soc_unregister_card+0xa0/0xcc) r8:bec89000 r7:bef54440 r6:00000004 r5:bf9a4010 r4:bf9a4010 r3:00000000 [<80496a24>] (snd_soc_unregister_card+0x0/0xcc) from [<804a4b74>] (devm_card_release+0x14/0x18) r6:00000004 r5:bf9a4010 r4:bfbede60 r3:804a4b60 [<804a4b60>] (devm_card_release+0x0/0x18) from [<80328728>] (release_nodes+0x11c/0x1dc) [<8032860c>] (release_nodes+0x0/0x1dc) from [<80328cb0>] (devres_release_all+0x38/0x54) [<80328c78>] (devres_release_all+0x0/0x54) from [<8032561c>] (__device_release_driver+0x80/0xd4) r4:bf9a4010 r3:7f016000 [<8032559c>] (__device_release_driver+0x0/0xd4) from [<80325e84>] (driver_detach+0xbc/0xc0) r5:7f0169f8 r4:bf9a4010 [<80325dc8>] (driver_detach+0x0/0xc0) from [<80325410>] (bus_remove_driver+0x54/0x98) r6:78208d30 r5:00000000 r4:7f0169f8 r3:bee28900 [<803253bc>] (bus_remove_driver+0x0/0x98) from [<803264f8>] (driver_unregister+0x30/0x50) r4:7f0169f8 r3:beca8100 [<803264c8>] (driver_unregister+0x0/0x50) from [<80327098>] (platform_driver_unregister+0x14/0x18) r4:7f016a3c r3:7f01635c [<80327084>] (platform_driver_unregister+0x0/0x18) from [<7f016370>] (imx_sgtl5000_driver_exit+0x14/0x1c [snd_soc_imx_sgtl5000]) [<7f01635c>] (imx_sgtl5000_driver_exit+0x0/0x1c [snd_soc_imx_sgtl5000]) from [<80085ef0>] (SyS_delete_module+0x130/0x18c) [<80085dc0>] (SyS_delete_module+0x0/0x18c) from [<8000e800>] (ret_fast_syscall+0x0/0x48) r6:7efc2f07 r5:78208d00 r4:fffffffe Code: 889da9f8 e5983020 e3530000 089da9f8 (e5933038) ---[ end trace 825e7e125251a225 ]---

2 3

[alsa-devel] [PATCH 1/6] ASoC: Add resource managed snd_dmaengine_pcm_register()
by Lars-Peter Clausen 02 Dec '13

02 Dec '13

For many drivers using the generic dmaengine PCM driver one of the few (or the only) things left to do in the drivers remove function is to unregister the PCM device. This patch adds a resource managed version of snd_dmaengine_pcm_register() which makes it possible to simplify the remove function as well as the error path in the probe function for those drivers. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Moved snd_dmaengine_pcm_register() to soc-devres.c --- include/sound/dmaengine_pcm.h | 4 ++++ sound/soc/soc-devres.c | 41 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 45 insertions(+) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 1501731..4ef986c 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -140,6 +140,10 @@ int snd_dmaengine_pcm_register(struct device *dev, unsigned int flags); void snd_dmaengine_pcm_unregister(struct device *dev); +int devm_snd_dmaengine_pcm_register(struct device *dev, + const struct snd_dmaengine_pcm_config *config, + unsigned int flags); + int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config); diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index b1d7322..9998619 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> static void devm_component_release(struct device *dev, void *res) { @@ -84,3 +85,43 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) return ret; } EXPORT_SYMBOL_GPL(devm_snd_soc_register_card); + +#ifdef CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM + +static void devm_dmaengine_pcm_release(struct device *dev, void *res) +{ + snd_dmaengine_pcm_unregister(*(struct device **)res); +} + +/** + * devm_snd_dmaengine_pcm_register - resource managed dmaengine PCM registration + * @dev: The parent device for the PCM device + * @config: Platform specific PCM configuration + * @flags: Platform specific quirks + * + * Register a dmaengine based PCM device with automatic unregistration when the + * device is unregistered. + */ +int devm_snd_dmaengine_pcm_register(struct device *dev, + const struct snd_dmaengine_pcm_config *config, unsigned int flags) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_dmaengine_pcm_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_dmaengine_pcm_register(dev, config, flags); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_dmaengine_pcm_register); + +#endif -- 1.8.0

2 11

[alsa-devel] [PATCH] ASoC: generic-dmaengine-pcm: Set BATCH flag when residue reporting is not supported
by Lars-Peter Clausen 02 Dec '13

02 Dec '13

For dmaengine drivers which do not support transfer residue reporting we update the PCM pointer with period granularity. Set the SNDRV_PCM_INFO_BATCH flag in this case to let userspace know about this. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-generic-dmaengine-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index cbc9c96..87e8635 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -137,6 +137,9 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea hw.buffer_bytes_max = SIZE_MAX; hw.fifo_size = dma_data->fifo_size; + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) + hw.info |= SNDRV_PCM_INFO_BATCH; + ret = dma_get_slave_caps(chan, &dma_caps); if (ret == 0) { if (dma_caps.cmd_pause) -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: Set SNDRV_PCM_INFO_JOINT_DUPLEX for PCMs with symmetry constraints
by Lars-Peter Clausen 02 Dec '13

02 Dec '13

If there are symmetry constraints between the playback and the capture channel set the SNDRV_PCM_INFO_JOINT_DUPLEX flag to let userspace know about this. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-pcm.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 22af038..eb340a8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -180,6 +180,21 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, return 0; } +static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver; + struct snd_soc_dai_driver *codec_driver = rtd->codec_dai->driver; + struct snd_soc_dai_link *link = rtd->dai_link; + + return cpu_driver->symmetric_rates || codec_driver->symmetric_rates || + link->symmetric_rates || cpu_driver->symmetric_channels || + codec_driver->symmetric_channels || link->symmetric_channels || + cpu_driver->symmetric_samplebits || + codec_driver->symmetric_samplebits || + link->symmetric_samplebits; +} + /* * List of sample sizes that might go over the bus for parameter * application. There ought to be a wildcard sample size for things @@ -316,6 +331,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) &cpu_dai_drv->capture); } + if (soc_pcm_has_symmetry(substream)) + runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; + ret = -EINVAL; if (!runtime->hw.rates) { printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n", -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: imx-spdif: Remove error message upon devm_kzalloc() failure
by Fabio Estevam 02 Dec '13

02 Dec '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> No need to have a specific OOM message, since there is generic MM out of memory message in place. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-spdif.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 8499d52..980dd1f 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -35,7 +35,6 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { - dev_err(&pdev->dev, "failed to allocate memory\n"); ret = -ENOMEM; goto end; } -- 1.8.1.2

2 1

[alsa-devel] MAXIM 98089 codec support in Linux
by VS 02 Dec '13

02 Dec '13

Hi, We are trying to integrate MAXIM 98089 (Audio codec) into our Linux product build. This has 2 DAI's available with it - DAI1 with both Playback and Capture facility and DAI2 with Playback capability alone. When we use this codec in our build, we find that the dai1_hw_params and dai1_set_fmt functions never gets called but it is always dai2_hw_params and dai2_set_fmt function that gets called because of which apparently, we are able to hear the voice that is incoming but not able to capture the voice and send it out externally. Here are our definitions: static struct snd_soc_dai_driver max98089_dai[] = { { .name = "max98089.0", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = MAX98089_RATES, .formats = MAX98089_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = MAX98089_RATES, .formats = MAX98089_FORMATS, }, .ops = &max98089_dai1_ops, }, { .name = "max98089.1", .playback = { .stream_name = "Aux Playback", .channels_min = 1, .channels_max = 2, .rates = MAX98089_RATES, .formats = MAX98089_FORMATS, }, .ops = &max98089_dai2_ops, .symmetric_rates = 1, } }; The codec specific registers only when we put in dai2_ops functions, we are able to see the effects but there is no effect if we put it in dai1_ops functions. Is there any specific way of referring to DAI when we have multiple DAI support in the codec that we are using? Thanks in advance for your inputs VS

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix headset mic input after muted internal mic (Dell/Realtek)
by David Henningsson 02 Dec '13

02 Dec '13

By trial and error, I found this patch could work around an issue where the headset mic would stop working if you switch between the internal mic and the headset mic, and the internal mic was muted. It still takes a second or two before the headset mic actually starts working, but still better than nothing. I'd like to add stable to this patch, but I'd like Kailang's ack first because I don't know what the verbs really do. BugLink: https://bugs.launchpad.net/bugs/1256840 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 537991c..85d1c8b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3338,6 +3338,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) alc_write_coef_idx(codec, 0x18, 0x7388); break; case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); alc_write_coef_idx(codec, 0x15, 0x0d60); alc_write_coef_idx(codec, 0xc3, 0x0000); break; @@ -3366,6 +3367,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) alc_write_coef_idx(codec, 0x18, 0x7388); break; case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); alc_write_coef_idx(codec, 0x15, 0x0d50); alc_write_coef_idx(codec, 0xc3, 0x0000); break; -- 1.7.9.5

2 1

[alsa-devel] Remove fixup list for HP
by Kailang 02 Dec '13

02 Dec '13

Hi Takashi, Chrome OS was loaded driver by models. The patch was remove the alc269 fixup entry for HP chrome book. Thanks. Kialang

2 2

[alsa-devel] compressed audio support
by maruthi srinivas 02 Dec '13

02 Dec '13

Hi, I saw that compressed audio framework being used with Asoc (alsa soc). Can the framework be used with other than Asoc ? I mean with ALSA pci based cards. I see the implementation in the following files. which one is to be used ? Does tinycompress in userspace can be used with either approaches ? http://lxr.free-electrons.com/source/sound/core/compress_offload.c http://lxr.free-electrons.com/source/sound/soc/soc-compress.c Regards, Maruthi

2 1

[alsa-devel] [PATCH] alsa-lib: dlmisc: Add a fake dl lib head file when dl lib is not supported by the toolchain.
by Sonic Zhang 02 Dec '13

02 Dec '13

From: sonic <sonic(a)htpc.(none)> The FLAT GNU toolchain doesn't include the dlfcn.h header file. Signed-off-by: Sonic Zhang <sonic.adi(a)gmail.com> --- include/dlmisc.h | 24 ++++++++++++++++++++++++ modules/mixer/simple/sbasedl.c | 4 ++++ src/mixer/simple_abst.c | 4 ++++ 3 files changed, 32 insertions(+) create mode 100644 include/dlmisc.h diff --git a/include/dlmisc.h b/include/dlmisc.h new file mode 100644 index 0000000..3be8d43 --- /dev/null +++ b/include/dlmisc.h @@ -0,0 +1,24 @@ +/* + * This library is free software; you can redistribute it and/or modify + * it under the terms of the GNU Lesser General Public License as + * published by the Free Software Foundation; either version 2.1 of + * the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#ifndef __ALSA_DLMISC_H +#define __ALSA_DLMISC_H + +#define RTLD_NOW 0x00002 +#define RTLD_GLOBAL 0x00100 + +#endif diff --git a/modules/mixer/simple/sbasedl.c b/modules/mixer/simple/sbasedl.c index 494802f..511bebf 100644 --- a/modules/mixer/simple/sbasedl.c +++ b/modules/mixer/simple/sbasedl.c @@ -27,7 +27,11 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#include <dlmisc.h> +#endif #include "config.h" #include "asoundlib.h" #include "mixer_abst.h" diff --git a/src/mixer/simple_abst.c b/src/mixer/simple_abst.c index 9e9aaf5..dd951fe 100644 --- a/src/mixer/simple_abst.c +++ b/src/mixer/simple_abst.c @@ -34,7 +34,11 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <math.h> +#ifdef HAVE_LIBDL #include <dlfcn.h> +#else +#include <dlmisc.h> +#endif #include "config.h" #include "asoundlib.h" #include "mixer_simple.h" -- 1.7.9.5

1 1

[alsa-devel] Bug: plugging in a new sound card causes mixer state of other sound cards to be changed
by Angus McInnes 01 Dec '13

01 Dec '13

Hi, When I plug in my USB sound card, the mixer settings of the internal sound card are changed. I don't want this to happen. I believe the problem is with this commit: http://git.alsa-project.org/?p=alsa-utils.git;a=blobdiff;f=alsactl/90-alsa-… The card number $attr{number} comes from the card* device. Since KERNELS=="card*" was changed to KERNELS!="card*", alsactl restore is run with no card number attribute, so it restores the state of all cards. If I change the != back to ==, alsactl restore is run with a card number and the problem seems to be resolved. (By the way, your bug tracker at https://bugtrack.alsa-project.org/alsa-bug seems to be down.) Regards, Angus McInnes

1 0

[alsa-devel] [RFC 0/4] DT support for rx51-audio
by Sebastian Reichel 30 Nov '13

30 Nov '13

Hi, This patchset adds DT support in rx51-audio. I tested it on the Nokia N900 and was able to play sound with aplay using earphones and earspeaker. The Loudspeakers did not work. I don't know the reason. The patchset consists of 4 patches: 1. convert rx51-audio to snd_soc_register_card() 2. init audio in rx51 platform file 3. add some DT helpers to ASoC core 4. add DT support to rx51-audio Please note, that the patch is RFC level for now. For example non DT boot does not work, because I did not yet add pdata in boardcode. Proposed DTS node would look like this: sound: rx51-audio { compatible = "nokia,rx51-audio"; ti,mcbsp = <&mcbsp2>; ti,codec = <&tlv320aic3x>; ti,codec_aux = <&tlv320aic3x_aux>; ti,headset_amp = <&tpa6130a2>; tvout-selection-gpio = <&gpio2 8 GPIO_ACTIVE_HIGH>; /* 40 */ jack-detection-gpio = <&gpio6 17 GPIO_ACTIVE_HIGH>; /* 177 */ eci-sw-gpio = <&gpio6 22 GPIO_ACTIVE_HIGH>; /* 182 */ speaker-amp-gpio = <&twl_gpio 7 GPIO_ACTIVE_HIGH>; }; -- Sebastian Pali Rohár (2): ASoC: omap: rx51: Use snd_soc_register_card ARM: OMAP: rx51: Register audio device Sebastian Reichel (2): ASoC: Allow Aux Codecs to be specified using DT ASoC: RX-51: Add DT support to sound driver .../devicetree/bindings/sound/nokia,rx51.txt | 28 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + arch/arm/mach-omap2/board-rx51-peripherals.c | 11 ++ include/sound/rx51.h | 18 ++ include/sound/soc.h | 13 +- sound/soc/omap/omap-mcbsp.c | 5 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/rx51.c | 212 +++++++++++++++------ sound/soc/soc-core.c | 68 +++++-- 9 files changed, 277 insertions(+), 81 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nokia,rx51.txt create mode 100644 include/sound/rx51.h -- 1.8.4.rc3

6 17

[alsa-devel] [PATCH] ALSA: hda - Fix silent output on ASUS W7J laptop
by Takashi Iwai 29 Nov '13

29 Nov '13

The recent kernels got regressions on ASUS W7J with ALC660 codec where no sound comes out. After a long debugging session, we found out that setting the pin control on the unused NID 0x10 is mandatory for the outputs. And, it was found out that another magic of NID 0x0f that is required for other ASUS laptops isn't needed on this machine. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66081 Reported-and-tested-by: Andrey Lipaev <lipaev(a)mail.ru> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6366a6683e22..a98f3296b891 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4494,6 +4494,7 @@ enum { ALC861_FIXUP_AMP_VREF_0F, ALC861_FIXUP_NO_JACK_DETECT, ALC861_FIXUP_ASUS_A6RP, + ALC660_FIXUP_ASUS_W7J, }; /* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */ @@ -4543,10 +4544,21 @@ static const struct hda_fixup alc861_fixups[] = { .v.func = alc861_fixup_asus_amp_vref_0f, .chained = true, .chain_id = ALC861_FIXUP_NO_JACK_DETECT, + }, + [ALC660_FIXUP_ASUS_W7J] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* ASUS W7J needs a magic pin setup on unused NID 0x10 + * for enabling outputs + */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + { } + }, } }; static const struct snd_pci_quirk alc861_fixup_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x1253, "ASUS W7J", ALC660_FIXUP_ASUS_W7J), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT), -- 1.8.4.4

1 0

[alsa-devel] [PATCH] ASoC: core: Use consistent byte ordering in snd_soc_bytes_get
by Charles Keepax 29 Nov '13

29 Nov '13

snd_soc_bytes_put treats the data in the binary control as big endian words, however snd_soc_bytes_get uses the endian of the host machine. This causes the two functions to be inconsistant with how the mask is applied on little endian machines. This patch applies the big_endian format used in snd_soc_bytes_put to snd_soc_bytes_get. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e53d87..a66783e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3212,11 +3212,11 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, break; case 2: ((u16 *)(&ucontrol->value.bytes.data))[0] - &= ~params->mask; + &= cpu_to_be16(~params->mask); break; case 4: ((u32 *)(&ucontrol->value.bytes.data))[0] - &= ~params->mask; + &= cpu_to_be32(~params->mask); break; default: return -EINVAL; -- 1.7.2.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: atmel: sam9x5_wm8731: fix oops when unload module
by Bo Shen 29 Nov '13

29 Nov '13

As the priv is not assigned to card->drvdata, it is NULL, so when unload module, it will cause NULL pointer oops. Assign priv to card->drvdata to fix this issue. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/atmel/sam9x5_wm8731.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 992ae38..6dc33ac 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -99,6 +99,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) card->dev = &pdev->dev; card->owner = THIS_MODULE; + card->drvdata = priv; card->dai_link = dai; card->num_links = 1; card->dapm_widgets = sam9x5_dapm_widgets; -- 1.7.9.5

3 9

[alsa-devel] [PATCH 1/4] dt: Add bindings documentation for the ADI AXI-I2S controller
by Lars-Peter Clausen 29 Nov '13

29 Nov '13

This patch adds the devicetree documentation for the ADI AXI-SPDIF audio controller. The controller has: * One set of memory mapped register * Two clocks, one for the memory mapped register interface, one used as the audio reference clock * One DMA interface each for the transmit and receive data Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: devicetree(a)vger.kernel.org --- .../devicetree/bindings/sound/adi,axi-i2s.txt | 29 ++++++++++++++++++++++ 1 file changed, 29 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,axi-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt new file mode 100644 index 0000000..d9c0e53 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,axi-i2s.txt @@ -0,0 +1,29 @@ +ADI AXI-I2S controller + +Required properties: + - compatible : Must be "adi,axi-i2s-1.00.a" + - reg : Must contain I2S registers location and length + - clocks : Clock specifier providing a handle to the controller's clocks. The + controller expects two clocks, the clock used for the AXI interface and the + clock used as thei sampling rate reference clock sample. + - clock-names : "axi" for the clock to the AXI interface, "ref" for the sample + rate reference clock. + - dma : Specifiers for the DMA channels that are used by the core. + - dma-names : "tx" for the transmit channel, "rx" for the receive channel. + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + + i2s: i2s@0x77600000 { + compatible = "adi,axi-i2s-1.00.a"; + reg = <0x77600000 0x1000>; + clocks = <&clk 15 &audio_clock>; + clock-names = "axi", "ref"; + dmas = <&ps7_dma 0 &ps7_dma 1>; + dma-names = "tx", "rx"; + }; -- 1.8.0

2 8

[alsa-devel] [PATCH] ALSA: hda - allow a codec to define its own depop delay time
by mengdong.lin＠intel.com 29 Nov '13

29 Nov '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds 'depop_delay' to struct hda_codec, to indicate a depop delay time in ms when power-down, in function set_power_state() to D3. Default value is -1, for a default delay time. Machine fixup can set a suitable value according to the codec chip and HW audio design. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 1e04a20..75ec91e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1444,6 +1444,7 @@ int snd_hda_codec_new(struct hda_bus *bus, INIT_LIST_HEAD(&codec->conn_list); INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); + codec->depop_delay = -1; #ifdef CONFIG_PM spin_lock_init(&codec->power_lock); @@ -3901,8 +3902,10 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, /* this delay seems necessary to avoid click noise at power-down */ if (power_state == AC_PWRST_D3) { - /* transition time less than 10ms for power down */ - msleep(codec->epss ? 10 : 100); + if (codec->depop_delay < 0) + msleep(codec->epss ? 10 : 100); + else if (codec->depop_delay > 0) + msleep(codec->depop_delay); flags = HDA_RW_NO_RESPONSE_FALLBACK; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index bedb436..efd0edc 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -943,6 +943,8 @@ struct hda_codec { struct snd_array jacks; #endif + int depop_delay; /* depop delay in ms, -1 for default delay time */ + /* fix-up list */ int fixup_id; const struct hda_fixup *fixup_list; -- 1.8.1.2

2 1

[alsa-devel] [RFC] ALSA: add new alsa control byte extended
by Vinod Koul 29 Nov '13

29 Nov '13

As discussed in the last Audio uConf we need to improve byte controls to allow larger size data to be sent to DSPs The modern DSPs require alsa byte controls size which far exceeds the today 512bytes limit. In order for usermode to send larger sizes (few 100s of KBs) along with size information we add extended byte control interface which sends any size bytes parameter buffer for DSPs to use Obviosly the size must be supported by the device and would be required to inform the max size allowed for the control. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/asound.h | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 9fc6219..8c10359 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -782,7 +782,8 @@ typedef int __bitwise snd_ctl_elem_type_t; #define SNDRV_CTL_ELEM_TYPE_BYTES ((__force snd_ctl_elem_type_t) 4) /* byte array */ #define SNDRV_CTL_ELEM_TYPE_IEC958 ((__force snd_ctl_elem_type_t) 5) /* IEC958 (S/PDIF) setup */ #define SNDRV_CTL_ELEM_TYPE_INTEGER64 ((__force snd_ctl_elem_type_t) 6) /* 64-bit integer type */ -#define SNDRV_CTL_ELEM_TYPE_LAST SNDRV_CTL_ELEM_TYPE_INTEGER64 +#define SNDRV_CTL_ELEM_TYPE_BYTES_EXT ((__force snd_ctl_elem_type_t) 7) /* bytes array extended */ +#define SNDRV_CTL_ELEM_TYPE_LAST SNDRV_CTL_ELEM_TYPE_BYTES_EXT typedef int __bitwise snd_ctl_elem_iface_t; #define SNDRV_CTL_ELEM_IFACE_CARD ((__force snd_ctl_elem_iface_t) 0) /* global control */ @@ -891,6 +892,11 @@ struct snd_ctl_elem_value { unsigned char data[512]; unsigned char *data_ptr; /* obsoleted */ } bytes; + struct { + unsigned long size; /* size of buffer */ + __u64 data; i /* user data pointer */ + } bytes_ext; + struct snd_aes_iec958 iec958; } value; /* RO */ struct timespec tstamp; -- 1.7.0.4

6 19

[alsa-devel] How to properly restart after config change
by Dolevo Jay 29 Nov '13

29 Nov '13

Hi experts, In my application, I some parameters in the asound.conf at runtime. However, after my software edits and changes the necessary parameters, the application still runs with the old configuration. If I run aplay, it works fine and runs with the new config but not my application. So, could anyone tell me what I should do extra after changing the parameters? Perhaps a small restart of something? thanks.

2 1

Re: [alsa-devel] ALSA: Add ALSA driver for Atmel Audio Bitstream DAC
by Dan Carpenter 29 Nov '13

29 Nov '13

Hello Hans-Christian Egtvedt, The patch e4967d6016b7: "ALSA: Add ALSA driver for Atmel Audio Bitstream DAC" from Feb 5, 2009, leads to the following static checker warning: "sound/atmel/abdac.c:373 set_sample_rates() error: buffer overflow 'dac->rates' 6 <= 6" sound/atmel/abdac.c 354 /* we start at 192 kHz and work our way down to 5112 Hz */ 355 while (new_rate >= RATE_MIN && index < (MAX_NUM_RATES + 1)) { index == MAX_NUM_RATES + 1 so index is 7. 356 new_rate = clk_round_rate(dac->sample_clk, 256 * new_rate); 357 if (new_rate < 0) 358 break; 359 /* make sure we are below the ABDAC clock */ 360 if (new_rate <= clk_get_rate(dac->pclk)) { 361 dac->rates[index] = new_rate / 256; index == MAX_NUM_RATES is off by one. 362 index++; 363 } 364 /* divide by 256 and then by two to get next rate */ 365 new_rate /= 256 * 2; 366 } 367 368 if (index) { 369 int i; 370 371 /* reverse array, smallest go first */ 372 for (i = 0; i < (index / 2); i++) { 373 unsigned int tmp = dac->rates[index - 1 - i]; 7 - 1 - 0 is 6, but dac->rates[] only has 6 elements so we are potentially reading beyond the end of the array here. 374 dac->rates[index - 1 - i] = dac->rates[i]; 375 dac->rates[i] = tmp; 376 } 377 378 dac->constraints_rates.count = index; 379 dac->constraints_rates.list = dac->rates; 380 dac->constraints_rates.mask = 0; 381 dac->rates_num = index; 382 383 retval = 0; 384 } regards, dan carpenter

2 2

[alsa-devel] Remove depop delay for suspend and resume
by Kailang 29 Nov '13

29 Nov '13

Hi Takashi, Intel test alc262. They find the suspend and resume time were > 50ms. So, I delete the delay for alc_eapd_shutup and alc_resume. If we want to have depop functions, we will follow like alc283. Thanks. Kailang

3 6

[alsa-devel] [PATCH 1/2] ALSA: hda/realtek - Remove depop delay for suspend and resume if applicable
by mengdong.lin＠intel.com 29 Nov '13

29 Nov '13

From: Kailang Yang <kailang(a)realtek.com> This patch defines a flag "no_depop_delay" in alc_spec. If this flag is set, delay in alc_eapd_shutup and alc_resume will be skipped. Machine-specific fixup can set this flag to reduce suspend/resume time, if the codec and hardware analog design can avoid pop noise without this delay. Signed-off-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6366a66..503cf78 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -118,7 +118,8 @@ struct alc_spec { int init_amp; int codec_variant; /* flag for other variants */ - bool has_alc5505_dsp; + unsigned int has_alc5505_dsp:1; + unsigned int no_depop_delay:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -280,8 +281,11 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on) */ static void alc_eapd_shutup(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; + alc_auto_setup_eapd(codec, false); - msleep(200); + if (!spec->no_depop_delay) + msleep(200); snd_hda_shutup_pins(codec); } @@ -863,7 +867,10 @@ static int alc_suspend(struct hda_codec *codec) #ifdef CONFIG_PM static int alc_resume(struct hda_codec *codec) { - msleep(150); /* to avoid pop noise */ + struct alc_spec *spec = codec->spec; + + if (!spec->no_depop_delay) + msleep(150); /* to avoid pop noise */ codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); @@ -4448,7 +4455,7 @@ static int patch_alc269(struct hda_codec *codec) } if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { - spec->has_alc5505_dsp = true; + spec->has_alc5505_dsp = 1; spec->init_hook = alc5505_dsp_init; } -- 1.8.1.2

2 2

[alsa-devel] [PATCH 0/6] ASoC: rcar: add Gen2 support
by Kuninori Morimoto 29 Nov '13

29 Nov '13

Hi Mark These patches add Gen2 PIO transfer support for Renesas sound driver. DMA transfer will be supported. Kuninori Morimoto (6): ASoC: rcar: remove unused register settings ASoC: rcar: separate regmap init common field ASoC: rcar: add rsnd_is_accessible_reg() ASoC: rcar: remove .path_init/exit from rsnd_gen_ops ASoC: rcar: remove rcar_gen_ops ASoC: rcar: add Gen2 sound support sound/soc/sh/rcar/gen.c | 248 +++++++++++++++++++++++++++++----------------- sound/soc/sh/rcar/rsnd.h | 6 +- sound/soc/sh/rcar/ssi.c | 3 + 3 files changed, 162 insertions(+), 95 deletions(-) Best regards --- Kuninori Morimoto

2 7

[alsa-devel] [PATCH] ASoC: wm_adsp: Stop region iteration when the desired region is found
by Charles Keepax 29 Nov '13

29 Nov '13

When locating the memory region relating to a coefficient block written through a bin file we keep processing the list of regions even after we have found the region we require. This patch adds a break, so we don't process redundant list items. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 46ec0e9..b42f9af 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1286,6 +1286,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) reg = wm_adsp_region_to_reg(mem, reg); reg += offset; + break; } } -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ASoC: cs42l52: Fix build
by Mark Brown 29 Nov '13

29 Nov '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4a47a634f376..0bac6d5a4ac8 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1232,7 +1232,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, if (of_property_read_u32(i2c_client->dev.of_node, "cirrus,chgfreq-divisor", &val32) >= 0) - pdata->chgfreq_divisor = val32; + pdata->chgfreq = val32; pdata->reset_gpio = of_get_named_gpio(i2c_client->dev.of_node, -- 1.8.4.4

1 0

[alsa-devel] [PATCH V2] ASoC: tegra: use reset framework
by Stephen Warren 29 Nov '13

29 Nov '13

From: Stephen Warren <swarren(a)nvidia.com> Tegra's clock driver now provides an implementation of the common reset API (include/linux/reset.h). Use this instead of the old Tegra- specific API; that will soon be removed. This change also renames "clock"/"clk" to "modules"/"mod" in symbols related to entries in configlink_clocks[], since: - We don't care about clock handles any more, but rather reset handles, so the old name isn't applicable. - It really is a list of modules on the bus, about which we currently only care about reset handles. If we start caring about any other aspect of the modules in the future, we won't have to rename all these symbols again. Note: The addition of "depends COMMON_CLOCK" is something that was missing before, not a new requirement. Cc: treding(a)nvidia.com Cc: pdeschrijver(a)nvidia.com Cc: linux-tegra(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Added note about COMMON_CLOCK to commit description. This patch is part of a series with strong internal depdendencies. I'm looking for an ack so that I can take the entire series through the Tegra and arm-soc trees. The series will be part of a stable branch that can be merged into other subsystems if needed to avoid/resolve dependencies. --- sound/soc/tegra/Kconfig | 2 ++ sound/soc/tegra/tegra30_ahub.c | 70 +++++++++++++++++++++++------------------- sound/soc/tegra/tegra30_ahub.h | 2 +- 3 files changed, 41 insertions(+), 33 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 8fc653ca3ab4..896292bb853f 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -1,6 +1,8 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" depends on (ARCH_TEGRA && TEGRA20_APB_DMA) || COMPILE_TEST + depends on COMMON_CLK + depends on RESET_CONTROLLER select REGMAP_MMIO select SND_SOC_GENERIC_DMAENGINE_PCM help diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 31154338c1eb..5ce00dc48c44 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -24,6 +24,7 @@ #include <linux/platform_device.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <linux/slab.h> #include <linux/clk/tegra.h> #include <sound/soc.h> @@ -301,27 +302,27 @@ int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif) } EXPORT_SYMBOL_GPL(tegra30_ahub_unset_rx_cif_source); -#define CLK_LIST_MASK_TEGRA30 BIT(0) -#define CLK_LIST_MASK_TEGRA114 BIT(1) +#define MOD_LIST_MASK_TEGRA30 BIT(0) +#define MOD_LIST_MASK_TEGRA114 BIT(1) -#define CLK_LIST_MASK_TEGRA30_OR_LATER \ - (CLK_LIST_MASK_TEGRA30 | CLK_LIST_MASK_TEGRA114) +#define MOD_LIST_MASK_TEGRA30_OR_LATER \ + (MOD_LIST_MASK_TEGRA30 | MOD_LIST_MASK_TEGRA114) static const struct { - const char *clk_name; - u32 clk_list_mask; -} configlink_clocks[] = { - { "i2s0", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "i2s1", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "i2s2", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "i2s3", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "i2s4", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "dam0", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "dam1", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "dam2", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "spdif_in", CLK_LIST_MASK_TEGRA30_OR_LATER }, - { "amx", CLK_LIST_MASK_TEGRA114 }, - { "adx", CLK_LIST_MASK_TEGRA114 }, + const char *rst_name; + u32 mod_list_mask; +} configlink_mods[] = { + { "i2s0", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s1", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s2", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s3", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s4", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "dam0", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "dam1", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "dam2", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "spdif", MOD_LIST_MASK_TEGRA30_OR_LATER }, + { "amx", MOD_LIST_MASK_TEGRA114 }, + { "adx", MOD_LIST_MASK_TEGRA114 }, }; #define LAST_REG(name) \ @@ -450,17 +451,17 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = { }; static struct tegra30_ahub_soc_data soc_data_tegra30 = { - .clk_list_mask = CLK_LIST_MASK_TEGRA30, + .mod_list_mask = MOD_LIST_MASK_TEGRA30, .set_audio_cif = tegra30_ahub_set_cif, }; static struct tegra30_ahub_soc_data soc_data_tegra114 = { - .clk_list_mask = CLK_LIST_MASK_TEGRA114, + .mod_list_mask = MOD_LIST_MASK_TEGRA114, .set_audio_cif = tegra30_ahub_set_cif, }; static struct tegra30_ahub_soc_data soc_data_tegra124 = { - .clk_list_mask = CLK_LIST_MASK_TEGRA114, + .mod_list_mask = MOD_LIST_MASK_TEGRA114, .set_audio_cif = tegra124_ahub_set_cif, }; @@ -475,7 +476,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) { const struct of_device_id *match; const struct tegra30_ahub_soc_data *soc_data; - struct clk *clk; + struct reset_control *rst; int i; struct resource *res0, *res1, *region; u32 of_dma[2]; @@ -495,19 +496,24 @@ static int tegra30_ahub_probe(struct platform_device *pdev) * operate correctly, all devices on this bus must be out of reset. * Ensure that here. */ - for (i = 0; i < ARRAY_SIZE(configlink_clocks); i++) { - if (!(configlink_clocks[i].clk_list_mask & - soc_data->clk_list_mask)) + for (i = 0; i < ARRAY_SIZE(configlink_mods); i++) { + if (!(configlink_mods[i].mod_list_mask & + soc_data->mod_list_mask)) continue; - clk = clk_get(&pdev->dev, configlink_clocks[i].clk_name); - if (IS_ERR(clk)) { - dev_err(&pdev->dev, "Can't get clock %s\n", - configlink_clocks[i].clk_name); - ret = PTR_ERR(clk); + + rst = reset_control_get(&pdev->dev, + configlink_mods[i].rst_name); + if (IS_ERR(rst)) { + dev_err(&pdev->dev, "Can't get reset %s\n", + configlink_mods[i].rst_name); + ret = PTR_ERR(rst); goto err; } - tegra_periph_reset_deassert(clk); - clk_put(clk); + + ret = reset_control_deassert(rst); + reset_control_put(rst); + if (ret) + goto err; } ahub = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_ahub), diff --git a/sound/soc/tegra/tegra30_ahub.h b/sound/soc/tegra/tegra30_ahub.h index d67321d90faa..1383f8cd3572 100644 --- a/sound/soc/tegra/tegra30_ahub.h +++ b/sound/soc/tegra/tegra30_ahub.h @@ -502,7 +502,7 @@ void tegra124_ahub_set_cif(struct regmap *regmap, unsigned int reg, struct tegra30_ahub_cif_conf *conf); struct tegra30_ahub_soc_data { - u32 clk_list_mask; + u32 mod_list_mask; void (*set_audio_cif)(struct regmap *regmap, unsigned int reg, struct tegra30_ahub_cif_conf *conf); -- 1.8.1.5

3 2

[alsa-devel] [PATCH] ASoC mc13783 codec: Add 'loudspeaker amp' and 'lineout amp'
by Lambrecht Jürgen 29 Nov '13

29 Nov '13

Add Alsp (loudspeaker amp) and Arxout (lineout amp) in the same way as Asp (speaker amp) and Ahs (headphone amp) - Remove "LSPL": all output controls are added now; this one doesn't exist - Fix: "LSP" is loudspeaker amp; they indeed share the same mux according to the datasheet figure but they have separate registers. And testing shows the Asp and Alsp muxes are separated. - Add "CODEC Capture Switch" control and move "CODEC Capture Mixin Switch" to its correct place Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/codecs/mc13783.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index af7bc55..08fb022 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -437,6 +437,10 @@ static const SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4, speaker_amp_source_text); static const struct snd_kcontrol_new speaker_amp_source_mux = SOC_DAPM_ENUM("Speaker Amp Source MUX", speaker_amp_source); +static const SOC_ENUM_SINGLE_DECL(loudspeaker_amp_source, MC13783_AUDIO_RX0, 7, + speaker_amp_source_text); +static const struct snd_kcontrol_new loudspeaker_amp_source_mux = + SOC_DAPM_ENUM("Loudspeaker Amp Source MUX", loudspeaker_amp_source); static const char * const headset_amp_source_text[] = { "CODEC", "Mixer" @@ -447,6 +451,11 @@ static const SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11, static const struct snd_kcontrol_new headset_amp_source_mux = SOC_DAPM_ENUM("Headset Amp Source MUX", headset_amp_source); +static const SOC_ENUM_SINGLE_DECL(lineout_amp_source, MC13783_AUDIO_RX0, 17, + headset_amp_source_text); +static const struct snd_kcontrol_new lineout_amp_source_mux = + SOC_DAPM_ENUM("Lineout Amp Source MUX", lineout_amp_source); + static const struct snd_kcontrol_new cdcout_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 18, 1, 0); @@ -493,10 +502,15 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_MUX("Speaker Amp Source MUX", SND_SOC_NOPM, 0, 0, &speaker_amp_source_mux), + SND_SOC_DAPM_MUX("Loudspeaker Amp Source MUX", SND_SOC_NOPM, 0, 0, + &loudspeaker_amp_source_mux), SND_SOC_DAPM_MUX("Headset Amp Source MUX", SND_SOC_NOPM, 0, 0, &headset_amp_source_mux), + SND_SOC_DAPM_MUX("Lineout Amp Source MUX", SND_SOC_NOPM, 0, 0, + &lineout_amp_source_mux), + SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -514,7 +528,6 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), SND_SOC_DAPM_OUTPUT("HSR"), - SND_SOC_DAPM_OUTPUT("LSPL"), SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), SND_SOC_DAPM_OUTPUT("CDCOUT"), @@ -563,21 +576,26 @@ static struct snd_soc_dapm_route mc13783_routes[] = { { "Speaker Amp Source MUX", "CODEC", "Voice CODEC PGA"}, { "Speaker Amp Source MUX", "Right", "DAC PGA"}, + { "Loudspeaker Amp Source MUX", "CODEC", "Voice CODEC PGA"}, + { "Loudspeaker Amp Source MUX", "Right", "DAC PGA"}, { "Headset Amp Source MUX", "CODEC", "Voice CODEC PGA"}, { "Headset Amp Source MUX", "Mixer", "DAC PGA"}, + { "Lineout Amp Source MUX", "CODEC", "Voice CODEC PGA"}, + { "Lineout Amp Source MUX", "Mixer", "DAC PGA"}, + /* Output */ { "HSL", NULL, "Headset Amp Left" }, { "HSR", NULL, "Headset Amp Right"}, { "RXOUTL", NULL, "Line out Amp Left"}, { "RXOUTR", NULL, "Line out Amp Right"}, { "SP", "Speaker Amp Switch", "Speaker Amp Source MUX"}, - { "LSP", "Loudspeaker Amp", "Speaker Amp Source MUX"}, + { "LSP", "Loudspeaker Amp", "Loudspeaker Amp Source MUX"}, { "HSL", "Headset Amp Left", "Headset Amp Source MUX"}, { "HSR", "Headset Amp Right", "Headset Amp Source MUX"}, - { "Line out Amp Left", NULL, "DAC PGA"}, - { "Line out Amp Right", NULL, "DAC PGA"}, + { "RXOUTL", "Line out Amp Left", "Lineout Amp Source MUX"}, + { "RXOUTR", "Line out Amp Right", "Lineout Amp Source MUX"}, { "DAC PGA", NULL, "DAC"}, { "DAC", NULL, "DAC_E"}, { "CDCOUT", "CDCOUT Switch", "Voice CODEC PGA"}, @@ -602,11 +620,12 @@ static struct snd_kcontrol_new mc13783_control_list[] = { SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0), SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0), + SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), - SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), + SOC_SINGLE("CODEC Capture Switch", MC13783_AUDIO_RX1, 0, 1, 0), SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC mc13783 codec: complete capture mux
by Lambrecht Jürgen 29 Nov '13

29 Nov '13

codec adc (capture) mux was not complete: * fix naming of TXIN/TX * reorder RXIN and TXIN to show muxing priority * implement RXIN/RX (R and L are the same control) Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/codecs/mc13783.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index bae6016..af7bc55 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -401,11 +401,14 @@ static const struct snd_kcontrol_new mc2_amp_ctl = static const struct snd_kcontrol_new atx_amp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 11, 1, 0); +static const struct snd_kcontrol_new rx_sel_ctl = + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 13, 1, 0); + /* Virtual mux. The chip does the input selection automatically * as soon as we enable one input. */ static const char * const adcl_enum_text[] = { - "MC1L", "RXINL", + "MC1L", "RXL", }; static const struct soc_enum adcl_enum = @@ -415,7 +418,7 @@ static const struct snd_kcontrol_new left_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcl_enum); static const char * const adcr_enum_text[] = { - "MC1R", "MC2", "RXINR", "TXIN", + "MC1R", "MC2", "TX", "RXR", }; static const struct soc_enum adcr_enum = @@ -470,9 +473,9 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MC1LIN"), SND_SOC_DAPM_INPUT("MC1RIN"), SND_SOC_DAPM_INPUT("MC2IN"), + SND_SOC_DAPM_INPUT("TXIN"), SND_SOC_DAPM_INPUT("RXINR"), SND_SOC_DAPM_INPUT("RXINL"), - SND_SOC_DAPM_INPUT("TXIN"), SND_SOC_DAPM_SUPPLY("MC1 Bias", MC13783_AUDIO_TX, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MC2 Bias", MC13783_AUDIO_TX, 1, 0, NULL, 0), @@ -480,7 +483,8 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("MC1L Amp", MC13783_AUDIO_TX, 7, 0, &mc1l_amp_ctl), SND_SOC_DAPM_SWITCH("MC1R Amp", MC13783_AUDIO_TX, 5, 0, &mc1r_amp_ctl), SND_SOC_DAPM_SWITCH("MC2 Amp", MC13783_AUDIO_TX, 9, 0, &mc2_amp_ctl), - SND_SOC_DAPM_SWITCH("TXIN Amp", MC13783_AUDIO_TX, 11, 0, &atx_amp_ctl), + SND_SOC_DAPM_SWITCH("TX Amp", MC13783_AUDIO_TX, 11, 0, &atx_amp_ctl), + SND_SOC_DAPM_SWITCH("RX Sel", MC13783_AUDIO_TX, 13, 0, &rx_sel_ctl), SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, &left_input_mux), @@ -537,14 +541,16 @@ static struct snd_soc_dapm_route mc13783_routes[] = { { "MC1L Amp", NULL, "MC1LIN"}, { "MC1R Amp", NULL, "MC1RIN" }, { "MC2 Amp", NULL, "MC2IN" }, - { "TXIN Amp", NULL, "TXIN"}, + { "TX Amp", NULL, "TXIN"}, + { "RX Sel", NULL, "RXINR"}, + { "RX Sel", NULL, "RXINL"}, /* indeed same control bit for RXINR and RXINL */ { "PGA Left Input Mux", "MC1L", "MC1L Amp" }, - { "PGA Left Input Mux", "RXINL", "RXINL"}, + { "PGA Left Input Mux", "RXL", "RX Sel"}, { "PGA Right Input Mux", "MC1R", "MC1R Amp" }, { "PGA Right Input Mux", "MC2", "MC2 Amp"}, - { "PGA Right Input Mux", "TXIN", "TXIN Amp"}, - { "PGA Right Input Mux", "RXINR", "RXINR"}, + { "PGA Right Input Mux", "TX", "TX Amp"}, + { "PGA Right Input Mux", "RXR", "RX Sel"}, { "PGA Left Input", NULL, "PGA Left Input Mux"}, { "PGA Right Input", NULL, "PGA Right Input Mux"}, -- 1.7.10.4

1 0

[alsa-devel] [patch] ALSA: dice: fix array limits in dice_proc_read()
by Dan Carpenter 29 Nov '13

29 Nov '13

The array limits are supposed to be in units of u32 instead of in bytes. The current code has a potential array overflow. Fixes: c614475b0ea9 ('ALSA: dice: add a proc file to show device information') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 57bcd31fcc12..c0aa64941cee 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -1019,7 +1019,7 @@ static void dice_proc_read(struct snd_info_entry *entry, if (dice_proc_read_mem(dice, &tx_rx_header, sections[2], 2) < 0) return; - quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx)); + quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx) / 4); for (stream = 0; stream < tx_rx_header.number; ++stream) { if (dice_proc_read_mem(dice, &buf.tx, sections[2] + 2 + stream * tx_rx_header.size, @@ -1045,7 +1045,7 @@ static void dice_proc_read(struct snd_info_entry *entry, if (dice_proc_read_mem(dice, &tx_rx_header, sections[4], 2) < 0) return; - quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx)); + quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx) / 4); for (stream = 0; stream < tx_rx_header.number; ++stream) { if (dice_proc_read_mem(dice, &buf.rx, sections[4] + 2 + stream * tx_rx_header.size,

4 4

[alsa-devel] [PATCH] ALSA: hda - Add mono speaker quirk for Dell Inspiron 5439
by David Henningsson 29 Nov '13

29 Nov '13

This machine also has mono output if run through DAC node 0x03. Cc: stable(a)vger.kernel.org (v3.10+) BugLink: https://bugs.launchpad.net/bugs/1256212 Tested-by: David Chen <david.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2ad568e..8099200 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4243,6 +4243,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), -- 1.7.9.5

2 1

[alsa-devel] [PATCH 00/11] codecs: Use IS_ENABLED() macro
by Fabio Estevam 29 Nov '13

29 Nov '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Using the IS_ENABLED() macro can make the code shorter and simpler. Mark, There are still more ocurrences that can be converted to IS_ENABLED. If you are happy with this series I will send another one with the remaining pieces.

3 15

[alsa-devel] [PATCH] hda-emu: Add workqueue_set_max_active function
by David Henningsson 29 Nov '13

29 Nov '13

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/linux/workqueue.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/include/linux/workqueue.h b/include/linux/workqueue.h index 1a8834e..ca49b4c 100644 --- a/include/linux/workqueue.h +++ b/include/linux/workqueue.h @@ -53,4 +53,7 @@ static inline struct delayed_work *to_delayed_work(struct work_struct *work) return container_of(work, struct delayed_work, work); } +static inline void workqueue_set_max_active(struct workqueue_struct *wq, + int max_active) { } + #endif /* __LINUX_WORKQUEUE_H */ -- 1.7.9.5

2 1

[alsa-devel] HD-Audio: can msleep in hda_set_power_state() to D3 be optional?
by Lin, Mengdong 29 Nov '13

29 Nov '13

Hi Takashi, In hda_set_power_state(), there is a mandatory sleep up to 100ms before setting state to D3. /* this delay seems necessary to avoid click noise at power-down */ if (power_state == AC_PWRST_D3) { msleep(codec->epss ? 10 : 100); ... Could this be optional for specific codecs? flags = HDA_RW_NO_RESPONSE_FALLBACK; } Some codecs implement their own sleep in patch_ops.suspend() to avoid the noise. So is it okay to add a codec flag to decide whether the sleep is necessary here? Thanks Mengdong

2 3

[alsa-devel] [PATCH] ASoC: adsp: Use async writes where possible
by Mark Brown 28 Nov '13

28 Nov '13

From: Mark Brown <broonie(a)linaro.org> This will allow a marginal speed improvement when used with a bus that supports async I/O by reducing the amount of context thrashing between writes, allowing the bus to be more fully utilised. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm_adsp.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 46ec0e9744d4..6b1c01c66e64 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1468,8 +1468,8 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) unsigned int val; int ret, count; - ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_SYS_ENA, ADSP2_SYS_ENA); + ret = regmap_update_bits_async(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_SYS_ENA, ADSP2_SYS_ENA); if (ret != 0) return ret; @@ -1521,9 +1521,9 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, val = (val & ARIZONA_SYSCLK_FREQ_MASK) >> ARIZONA_SYSCLK_FREQ_SHIFT; - ret = regmap_update_bits(dsp->regmap, - dsp->base + ADSP2_CLOCKING, - ADSP2_CLK_SEL_MASK, val); + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CLOCKING, + ADSP2_CLK_SEL_MASK, val); if (ret != 0) { adsp_err(dsp, "Failed to set clock rate: %d\n", ret); @@ -1586,10 +1586,10 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = regmap_update_bits(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_CORE_ENA | ADSP2_START, - ADSP2_CORE_ENA | ADSP2_START); + ret = regmap_update_bits_async(dsp->regmap, + dsp->base + ADSP2_CONTROL, + ADSP2_CORE_ENA | ADSP2_START, + ADSP2_CORE_ENA | ADSP2_START); if (ret != 0) goto err; -- 1.8.4.4

1 0

[alsa-devel] [PATCH 1/3] ASoC: ak4642: Convert to module_i2c_driver()
by Mark Brown 28 Nov '13

28 Nov '13

From: Mark Brown <broonie(a)linaro.org> The device does not support anything other than I2C (at least with the current driver) so save code. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak4642.c | 22 +--------------------- 1 file changed, 1 insertion(+), 21 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2f861c9b1d69..5bb18e913fa5 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -511,7 +511,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { .num_dapm_routes = ARRAY_SIZE(ak4642_intercon), }; -#if IS_ENABLED(CONFIG_I2C) static struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -571,27 +570,8 @@ static struct i2c_driver ak4642_i2c_driver = { .remove = ak4642_i2c_remove, .id_table = ak4642_i2c_id, }; -#endif -static int __init ak4642_modinit(void) -{ - int ret = 0; -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&ak4642_i2c_driver); -#endif - return ret; - -} -module_init(ak4642_modinit); - -static void __exit ak4642_exit(void) -{ -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&ak4642_i2c_driver); -#endif - -} -module_exit(ak4642_exit); +module_i2c_driver(ak4642_i2c_driver); MODULE_DESCRIPTION("Soc AK4642 driver"); MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); -- 1.8.4.4

1 2

[alsa-devel] [PATCH 1/2] ALSA: hda - Initialize missing bass speaker pin for ASUS AIO ET2700
by Takashi Iwai 28 Nov '13

28 Nov '13

Add a fixup entry for the missing bass speaker pin 0x16 on ASUS ET2700 AiO desktop. The channel map will be added in the next patch, so that this can be backported easily to stable kernels. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a9a1b0c1dbe..fd835c52618e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1782,6 +1782,7 @@ enum { ALC889_FIXUP_IMAC91_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, + ALC887_FIXUP_ASUS_BASS, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2105,6 +2106,13 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc882_fixup_no_primary_hp, }, + [ALC887_FIXUP_ASUS_BASS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x16, 0x99130130}, /* bass speaker */ + {} + }, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2138,6 +2146,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), + SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), -- 1.8.4.4

1 1

[alsa-devel] [PATCH] Add nommu support for alsa-utils
by Aaron Wu 28 Nov '13

28 Nov '13

Signed-off-by: Aaron Wu <aaron.wu(a)analog.com> Add nommu support for alsa-utils mainly for Blackfin --- package/alsa-utils/alsa-utils-nommu.patch | 54 +++++++++++++++++++++++++++++ package/alsa-utils/alsa-utils.mk | 1 + 2 files changed, 55 insertions(+) create mode 100644 package/alsa-utils/alsa-utils-nommu.patch diff --git a/package/alsa-utils/alsa-utils-nommu.patch b/package/alsa-utils/alsa-utils-nommu.patch new file mode 100644 index 0000000..c0227f5 --- /dev/null +++ b/package/alsa-utils/alsa-utils-nommu.patch @@ -0,0 +1,54 @@ +diff -Nur alsa-utils-1.0.26/alsactl/init_utils_run.c alsa-utils-1.0.26-new/alsactl/init_utils_run.c +--- alsa-utils-1.0.26/alsactl/init_utils_run.c 2012-09-06 15:52:57.000000000 +0800 ++++ alsa-utils-1.0.26-new/alsactl/init_utils_run.c 2013-10-22 02:19:06.210649921 +0800 +@@ -88,8 +88,11 @@ + strlcat(program, argv[0], sizeof(program)); + argv[0] = program; + } +- ++#ifdef HAVE_FORK + pid = fork(); ++#else ++ pid = vfork(); ++#endif + switch(pid) { + case 0: + /* child closes parent ends of pipes */ +diff -Nur alsa-utils-1.0.26/alsaloop/alsaloop.c alsa-utils-1.0.26-new/alsaloop/alsaloop.c +--- alsa-utils-1.0.26/alsaloop/alsaloop.c 2012-09-06 15:52:57.000000000 +0800 ++++ alsa-utils-1.0.26-new/alsaloop/alsaloop.c 2013-10-22 02:21:26.994650695 +0800 +@@ -850,14 +850,22 @@ + logit(LOG_CRIT, "daemon() failed: %s\n", strerror(errno)); + exit(EXIT_FAILURE); + } ++ #ifdef HAVE_FORK + i = fork(); ++ #else ++ i = vfork(); ++ #endif + if (i < 0) { + logit(LOG_CRIT, "fork() failed: %s\n", strerror(errno)); + exit(EXIT_FAILURE); + } + if (i > 0) { + /* wait(&i); */ ++ #ifdef HAVE_FORK + exit(EXIT_SUCCESS); ++ #else ++ _exit(EXIT_SUCCESS); ++ #endif + } + } + +diff -Nur alsa-utils-1.0.26/configure.in alsa-utils-1.0.26-new/configure.in +--- alsa-utils-1.0.26/configure.in 2012-09-06 16:45:44.000000000 +0800 ++++ alsa-utils-1.0.26-new/configure.in 2013-10-23 02:25:40.391127379 +0800 +@@ -23,6 +23,8 @@ + AC_MSG_RESULT($CC) + fi + ++ ++AC_CHECK_FUNC([fork]) + AC_PROG_CC + dnl AC_PROG_CXX + AC_PROG_INSTALL diff --git a/package/alsa-utils/alsa-utils.mk b/package/alsa-utils/alsa-utils.mk index b951112..4956e04 100644 --- a/package/alsa-utils/alsa-utils.mk +++ b/package/alsa-utils/alsa-utils.mk @@ -9,6 +9,7 @@ ALSA_UTILS_SOURCE = alsa-utils-$(ALSA_UTILS_VERSION).tar.bz2 ALSA_UTILS_SITE = http://alsa.cybermirror.org/utils ALSA_UTILS_LICENSE = GPLv2 ALSA_UTILS_LICENSE_FILES = COPYING +ALSA_UTILS_AUTORECONF = YES ALSA_UTILS_INSTALL_STAGING = YES ALSA_UTILS_DEPENDENCIES = host-gettext alsa-lib \ $(if $(BR2_PACKAGE_NCURSES),ncurses) -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Add zenbook-boost model for ASUS Zenbook
by Takashi Iwai 28 Nov '13

28 Nov '13

There's been a complaint about the too soft output on ASUS Zenbook with ALC269VB in comparison with Windows. It turned out, however, that the actual hardware output level is same between Linux and Windows unless software plugin is used. Still, it'd be convenient to give a capability for increasing the output level via Realtek codec's D-class amp. As a compromise, here provides a new model option, zenbook-boost, to enable the COEFs for the output boost amp. It's not used unless the option is given explicitly just for safety reason. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=62651 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 2 files changed, 14 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 85c362d8ea34..6fbd108bcb27 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -32,6 +32,7 @@ ALC269/270/275/276/28x/29x lenovo-dock Enables docking station I/O for some Lenovos dell-headset-multi Headset jack, which can also be used as mic-in dell-headset-dock Headset jack (without mic-in), and also dock I/O + zenbook-boost Add +6dB amp to output of ASUS Zenbook ALC66x/67x/892 ============== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a9a1b0c1dbe..284952c37929 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3799,6 +3799,7 @@ enum { ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ASUS_ZENBOOK, + ALC269VB_FIXUP_ASUS_ZENBOOK_BOOST, ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, @@ -4082,6 +4083,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269VB_FIXUP_DMIC, }, + [ALC269VB_FIXUP_ASUS_ZENBOOK_BOOST] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* class-D output boost +6dB */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x12}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3000}, + { } + }, + .chained = true, + .chain_id = ALC269VB_FIXUP_ASUS_ZENBOOK, + }, [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, @@ -4299,6 +4311,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, + {.id = ALC269VB_FIXUP_ASUS_ZENBOOK_BOOST, .name = "zenbook-boost"}, {} }; -- 1.8.4.4

1 0

[alsa-devel] [PATCH] ALSA: hda - Check leaf nodes to find aamix amps
by Takashi Iwai 28 Nov '13

28 Nov '13

The current generic parser assumes blindly that the volume and mute amps are found in the aamix node itself. But on some codecs, typically Analog Devices ones, the aamix amps are separately implemented in each leaf node of the aamix node, and the current driver can't establish the correct amp controls. This is a regression compared with the previous static quirks. This patch extends the search for the amps to the leaf nodes for allowing the aamix controls again on such codecs. In this implementation, I didn't code to loop through the whole paths, since usually one depth should suffice, and we can't search too deeply, as it may result in the conflicting control assignments. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65641 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 57 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 45 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index faefff1d3e81..c4671d00babd 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -2808,6 +2808,42 @@ static int add_loopback_list(struct hda_gen_spec *spec, hda_nid_t mix, int idx) return 0; } +/* return true if either a volume or a mute amp is found for the given + * aamix path; the amp has to be either in the mixer node or its direct leaf + */ +static bool look_for_mix_leaf_ctls(struct hda_codec *codec, hda_nid_t mix_nid, + hda_nid_t pin, unsigned int *mix_val, + unsigned int *mute_val) +{ + int idx, num_conns; + const hda_nid_t *list; + hda_nid_t nid; + + idx = snd_hda_get_conn_index(codec, mix_nid, pin, true); + if (idx < 0) + return false; + + *mix_val = *mute_val = 0; + if (nid_has_volume(codec, mix_nid, HDA_INPUT)) + *mix_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); + if (nid_has_mute(codec, mix_nid, HDA_INPUT)) + *mute_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); + if (*mix_val && *mute_val) + return true; + + /* check leaf node */ + num_conns = snd_hda_get_conn_list(codec, mix_nid, &list); + if (num_conns < idx) + return false; + nid = list[idx]; + if (!*mix_val && nid_has_volume(codec, nid, HDA_OUTPUT)) + *mix_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + if (!*mute_val && nid_has_mute(codec, nid, HDA_OUTPUT)) + *mute_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + + return *mix_val || *mute_val; +} + /* create input playback/capture controls for the given pin */ static int new_analog_input(struct hda_codec *codec, int input_idx, hda_nid_t pin, const char *ctlname, int ctlidx, @@ -2815,12 +2851,11 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, { struct hda_gen_spec *spec = codec->spec; struct nid_path *path; - unsigned int val; + unsigned int mix_val, mute_val; int err, idx; - if (!nid_has_volume(codec, mix_nid, HDA_INPUT) && - !nid_has_mute(codec, mix_nid, HDA_INPUT)) - return 0; /* no need for analog loopback */ + if (!look_for_mix_leaf_ctls(codec, mix_nid, pin, &mix_val, &mute_val)) + return 0; path = snd_hda_add_new_path(codec, pin, mix_nid, 0); if (!path) @@ -2829,20 +2864,18 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, spec->loopback_paths[input_idx] = snd_hda_get_path_idx(codec, path); idx = path->idx[path->depth - 1]; - if (nid_has_volume(codec, mix_nid, HDA_INPUT)) { - val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); - err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, val); + if (mix_val) { + err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, mix_val); if (err < 0) return err; - path->ctls[NID_PATH_VOL_CTL] = val; + path->ctls[NID_PATH_VOL_CTL] = mix_val; } - if (nid_has_mute(codec, mix_nid, HDA_INPUT)) { - val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); - err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, val); + if (mute_val) { + err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, mute_val); if (err < 0) return err; - path->ctls[NID_PATH_MUTE_CTL] = val; + path->ctls[NID_PATH_MUTE_CTL] = mute_val; } path->active = true; -- 1.8.4.4

1 0

[alsa-devel] [PATCH v7 0/4] Add dual-fifo mode support of i.MX ssi
by Nicolin Chen 28 Nov '13

28 Nov '13

* ! This series of patches has a direct dependency between them. When * ! applying them, we need to apply to one single branch. Otherwise, * ! it would break currect branches. Changelog v7: * Appended missing Acked-by to all four patches. * Sorry that I didn't add them at the first place. v6: * PATCH-1: Use goto err_firmware instead of return directly. * * Nothing changes for the other three ack-ed patches v5: * PATCH-3: Add period size constraint when using dual fifo mode * * Nothing changes for the other three patches v4: * PATCH-3: Drop useless register configuration. * * Nothing changes for the other three patches v3: * PATCH-1: Add comments to indicate the end of v1 and v2 array. * PATCH-3: Use better way to keep watermark as even number. * * Nothing changes for PATCH-2 and PATCH-4 v2: * Instead of adding rogue scripts to current SDMA driver based on firmware * V1, we define the new SDMA firmware as version 2 and bisect the PATCH-1 * to two patches: The first is to add version check code to the SDMA driver; * And the second is to add SSI dual FIFO DMATYPE. * * Nothing changes for the last two patches. v1: * SSI can reduce hardware overrun/underrun possibility when using dual * fifo mode. To support this mode, we need to first update sdma sciprt * list, and then enable dual fifo BIT in SSI driver, and last update DT * bindings of i.MX series. Nicolin Chen (4): dma: imx-sdma: Add sdma firmware version 2 support dma: imx-sdma: Add new dma type for ssi dual fifo script ASoC: fsl_ssi: Add dual fifo mode support ARM: dts: imx: use dual-fifo sdma script for ssi .../devicetree/bindings/dma/fsl-imx-sdma.txt | 1 + arch/arm/boot/dts/imx51.dtsi | 4 ++-- arch/arm/boot/dts/imx53.dtsi | 4 ++-- arch/arm/boot/dts/imx6qdl.dtsi | 12 +++++----- arch/arm/boot/dts/imx6sl.dtsi | 12 +++++----- drivers/dma/imx-sdma.c | 19 ++++++++++++++- include/linux/platform_data/dma-imx-sdma.h | 5 ++++ include/linux/platform_data/dma-imx.h | 1 + sound/soc/fsl/fsl_ssi.c | 27 +++++++++++++++++++++- 9 files changed, 67 insertions(+), 18 deletions(-) -- 1.8.4

3 7

[alsa-devel] Intermittent cut out on the video
by Dolevo Jay 28 Nov '13

28 Nov '13

Hi experts, In my system, I use two headphone jacks to connect my speakers, one is to the jack on the pc itself and the other one is to the jack located on the monitor which I connect to the same pc with hdmi cable. I have two sound cards on my system and I use the following asound.conf to have audio from two jacks simultaneously. http://pastebin.com/2yd6EW8C As you see, I also use dmix because I have multiple of audio sources coming and I mix them. However, I now have the following problem. I connect two speakers to the each jack and start hearing an audio stream. The audio coming from the jack on the pc is cutting out intermittently. It just cuts out for a fraction of a second and continues as normal and after some time it happens again. The frequency of the occurrence increases over time. I have absolutely no problem with the audio from the jack on the display. Could anyone shed some light on this issue? In my opinion, this is a sort of configuration issue since I have no problem at one source while having at the other source. Thanks in advance, Regards, Dolevo

2 2

Re: [alsa-devel] [PATCHv2 3/8] ARM: dts: Enables SAI ALSA SoC DAI device for Vybrid VF610 TOWER board.
by Bill Pringlemeir 28 Nov '13

28 Nov '13

[sorry, I should have copied alsa-devel(a)alsa-project.org previously]. On 19 Nov 2013, Li.Xiubo(a)freescale.com wrote: >>> This patch add and enable SAI device. >>> >>> arch/arm/boot/dts/vf610-twr.dts | 6 ++++++ >>> 1 file changed, 6 insertions(+) >>> >>> diff --git a/arch/arm/boot/dts/vf610-twr.dts >>> b/arch/arm/boot/dts/vf610-twr.dts index 1a58678..e4106dd 100644 >>> --- a/arch/arm/boot/dts/vf610-twr.dts >>> +++ b/arch/arm/boot/dts/vf610-twr.dts >>>>> -57,6 +57,12 @@ >>> status = "okay"; >>> }; >> >> Should we put a comment here that a TWR-AUDIO-SGTL board is needed? Or >> is this better in a menuconfig? I guess TWR-AUDIO-SGTL is needed as this >> codec is not on the VF610-TWR main board? >> > > Yes, the SGTL5000 codec is not on the VF610-TWR main board, and needs an extra > TWR-AUDIO-SGTL board. > > IMO, puting this comment in the menuconfig is much better. I agree. I guess the patch set 6 makes more sense. Also, you might consider adding a printk in the 'fsl_sgtl5000_probe()'. I guess that devm_snd_soc_register_card() will fail if the board is not present? As the board is not 'standard' people may come asking why the audio fails on the 'VF610-TWR'. + ret = devm_snd_soc_register_card(&pdev->dev, &fsl_sgt1500_card); + if (ret) { + dev_err(&pdev->dev, "register soc sound card failed :%d\n" + "TWR-AUDIO-SGTL board required.\n", + ret); + return ret; + } Especially, you can make one kernel that handles both cases w/wo TWR-AUDIO-SGTL board attached. Thanks for your work. Bill Pringlemeir.

2 5

[alsa-devel] alsa-lib-1.0.27.2 not compatible with ppc64le architecture
by Madhu Pavan 28 Nov '13

28 Nov '13

Hi, We are in the process of adding support for new arch (ppc64le) and as part of that we wanted to send patches for config.guess and configure as appropriate. alsa-lib-1.0.27.2 could not be built on ppc64le because it had an outdated configure/config.guess/libtool.m4. Since your pkg uses libtool and the latest tarball of your pkg didn't build on ppc64le due to outdated libtool present on your system. We wanted to request you to upgrade your libtool to the alpha release of libtool which has all the bits and pieces needed for ppc64le support. In that context, I wanted to confirm the below 1) Should we send patches to this mailing list or there is some other mailing list ? 2) New libtool alpha release which supports ppc64le arch is available, you may pick up the alpha release from either of the below : ftp://alpha.gnu.org/gnu/libtool/libtool-2.4.2.418.tar.gz (1.6MB) ftp://alpha.gnu.org/gnu/libtool/libtool-2.4.2.418.tar.xz (920KB) It would be great if you can install this new libtool which supports ppc64le, and spin the pkg and let us know the new tar availability. We can test it on ppc64le architecture for you. Please let me know if you are planning to release or spin newer version. Currently I used http://www.alsa-project.org/main/index.php/Main_Page Which does not have ppc64le support. Your response is highly appreciated. Thanks in advance, Madhu Pavan Kothapally

1 0

[alsa-devel] [PATCHv1] ASoC: SGTL5000: Fix kernel failed while getting regulator consumers
by Xiubo Li 28 Nov '13

28 Nov '13

SGTL5000 codec allows to use the internal LDO instead of VDDD, if the internal LDO is used, this caused the following kernel failed while trying to get the external VDDD regulator consumer. Adding sgtl5000_external_vddd_used() to fix it. And this has been tested on VF610-TWR board. sgtl5000 0-000a: Failed to get supply 'VDDD': -19 Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1f4093f..1a2d35a 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1298,6 +1298,20 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) return 0; } +static int sgtl5000_external_vddd_used(struct snd_soc_codec *codec) +{ + struct regulator *consumer; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + consumer = regulator_get(codec->dev, sgtl5000->supplies[VDDD].supply); + if (IS_ERR(consumer)) { + return 0; + } + regulator_put(consumer); + + return 1; +} + static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { int reg; @@ -1310,9 +1324,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) sgtl5000->supplies[i].supply = supply_names[i]; - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (!ret) + if (sgtl5000_external_vddd_used(codec)) external_vddd = 1; else { ret = sgtl5000_replace_vddd_with_ldo(codec); -- 1.8.4

4 6

[alsa-devel] [PATCH v2 16/18] ASoC: wm8990: Use IS_ENABLED() macro
by Fabio Estevam 27 Nov '13

27 Nov '13

Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Fix Subject and commit log sound/soc/codecs/wm8990.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 253c88b..34f0020 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1376,7 +1376,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .volatile_register = wm8990_volatile_register, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8990_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1423,7 +1423,7 @@ static struct i2c_driver wm8990_i2c_driver = { static int __init wm8990_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8990_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8990 I2C driver: %d\n", @@ -1436,7 +1436,7 @@ module_init(wm8990_modinit); static void __exit wm8990_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8990_i2c_driver); #endif } -- 1.8.1.2

2 5

[alsa-devel] [PATCH 00/18] ASoC: codecs: Use IS_ENABLED() macro
by Fabio Estevam 27 Nov '13

27 Nov '13

Using the IS_ENABLED() macro can make the code shorter and simpler.

2 34

[alsa-devel] [PATCH v3 1/2] ASoC: Add devicetree support for CS42L52
by Brian Austin 27 Nov '13

27 Nov '13

This patch adds device tree support for the CS42L52 Codec v3 Adds better descriptors for platform data. Change mic differential setting to boolean Change chgfreq to chgfreq-divisor Add vendor cirrus to of properties Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 45 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 42 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index e68450e..ed3c20f 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -17,7 +17,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -#include <linux/gpio.h> +#include <linux/of_gpio.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/input.h> @@ -1193,6 +1193,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, int ret; unsigned int devid = 0; unsigned int reg; + u32 val32; cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l52_private), GFP_KERNEL); @@ -1206,9 +1207,39 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); return ret; } - - if (pdata) + if (pdata) { + cs42l52->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs42l52_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + if (of_property_read_bool(i2c_client->dev.of_node, + "cirrus,mica-differential-cfg")) + pdata->mica_diff_cfg = true; + + if (of_property_read_bool(i2c_client->dev.of_node, + "cirrus,micb-differential-cfg")) + pdata->micb_diff_cfg = true; + + if (of_property_read_u32(i2c_client->dev.of_node, + "cirrus,micbias-lvl", &val32) >= 0) + pdata->micbias_lvl = val32; + + if (of_property_read_u32(i2c_client->dev.of_node, + "cirrus,chgfreq-divisor", &val32) >= 0) + pdata->chgfreq_divisor = val32; + + pdata->reset_gpio = + of_get_named_gpio(i2c_client->dev.of_node, + "cirrus,reset-gpio", 0); + } cs42l52->pdata = *pdata; + } if (cs42l52->pdata.reset_gpio) { ret = gpio_request_one(cs42l52->pdata.reset_gpio, @@ -1280,6 +1311,13 @@ static int cs42l52_i2c_remove(struct i2c_client *client) return 0; } +static const struct of_device_id cs42l52_of_match[] = { + { .compatible = "cirrus,cs42l52", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs42l52_of_match); + + static const struct i2c_device_id cs42l52_id[] = { { "cs42l52", 0 }, { } @@ -1290,6 +1328,7 @@ static struct i2c_driver cs42l52_i2c_driver = { .driver = { .name = "cs42l52", .owner = THIS_MODULE, + .of_match_table = cs42l52_of_match, }, .id_table = cs42l52_id, .probe = cs42l52_i2c_probe, -- 1.8.4.3

2 2

[alsa-devel] [PATCH] ASoC: add RT286 CODEC driver
by bardliao＠realtek.com 27 Nov '13

27 Nov '13

From: Bard Liao <bardliao(a)realtek.com> This patch adds the ALC286 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- This is the initial codec driver of ALC586. I tested DMCI/Headset recording and Speaker/Headphone playback. --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1609 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 113 ++++ 4 files changed, 1727 insertions(+) create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..56748d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -313,6 +313,9 @@ config SND_SOC_PCM1792A config SND_SOC_PCM3008 tristate +config SND_SOC_RT286 + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..b31615c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -48,6 +48,7 @@ snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o +snd-soc-rt286-objs := rt286.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -181,6 +182,7 @@ obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 0000000..e95f76c --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1609 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#define DEBUG +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "../../pci/hda/hda_codec.h" +#include "rt286.h" + +static int hp_amp_time = 100; +module_param(hp_amp_time, int, 0644); + +#define RT_SUPPORT_COMBO_JACK 1 +#define VERSION "0.0.1-beta3 alsa 1.0.25" + +struct rt286_priv { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct i2c_client *i2c; +}; + +static unsigned long rt286_read(struct snd_soc_codec *codec, + unsigned int vid, + unsigned int nid, + unsigned int data) +{ + struct i2c_msg xfer[2]; + int ret; + unsigned long verb; + unsigned long buf = 0x0; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + verb = nid << 20 | vid << 8 | data; + verb = cpu_to_be32(verb); + /* Write register */ + xfer[0].addr = rt286->i2c->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)&verb; + + /* Read data */ + xfer[1].addr = rt286->i2c->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(rt286->i2c->adapter, xfer, 2); + if (ret != 2) { + dev_err(&rt286->i2c->dev, "i2c_transfer() returned %d\n", ret); + return 0; + } + + return be32_to_cpu(buf); +} + +static int rt286_write(struct snd_soc_codec *codec, + unsigned int vid, + unsigned int nid, + unsigned long value) +{ + u8 data[4]; + int ret; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if ((vid & 0xff) == 0) { /* 4bit VID */ + vid = vid >> 8; + data[0] = (nid >> 4) & 0xf; + data[1] = ((nid & 0xf) << 4) | vid; + data[2] = (value >> 8) & 0xff; + data[3] = value & 0xff; + } else { /* 12bit VID */ + data[0] = (nid >> 4) & 0xf; + data[1] = ((nid & 0xf) << 4) | ((vid >> 8) & 0xf); + data[2] = vid & 0xff; + data[3] = value & 0xff; + } + + ret = i2c_master_send(rt286->i2c, data, 4); + pr_debug("write 0x%x 0x%x 0x%lx\n", vid, nid, value); + if (ret == 4) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static int rt286_index_write(struct snd_soc_codec *codec, + unsigned int WidgetID, unsigned int index, unsigned int data) +{ + int ret; + + ret = rt286_write(codec, AC_VERB_SET_COEF_INDEX, + WidgetID, index); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + goto err; + } + ret = rt286_write(codec, AC_VERB_SET_PROC_COEF, + WidgetID, data); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + + return 0; +err: + return ret; +} + +static unsigned long rt286_index_read(struct snd_soc_codec *codec, + unsigned int WidgetID, unsigned int index) +{ + int ret; + + ret = rt286_write(codec, AC_VERB_SET_COEF_INDEX, + WidgetID, index); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + return ret; + } + return rt286_read(codec, AC_VERB_GET_PROC_COEF, + WidgetID, index); +} + +static int rt286_index_update_bits(struct snd_soc_codec *codec, + unsigned int WidgetID, unsigned int index, + unsigned int mask, unsigned int data) +{ + unsigned long old, new; + int change, ret; + + ret = rt286_index_read(codec, WidgetID, index); + if (ret < 0) { + dev_err(codec->dev, "Failed to read private reg: %d\n", ret); + goto err; + } + + old = ret; + new = (old & ~mask) | (data & mask); + change = old != new; + + if (change) { + ret = rt286_index_write(codec, WidgetID, index, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_support_power_controls[] = { + NODE_ID_DAC_OUT1, + NODE_ID_DAC_OUT2, + NODE_ID_ADC_IN1, + NODE_ID_ADC_IN2, + NODE_ID_MIC1, + NODE_ID_DMIC1, + NODE_ID_DMIC2, + NODE_ID_SPK_OUT, + NODE_ID_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static const struct reg_default rt286_reg[TOTAL_NODE_ID + 1] = { + { NODE_ID_DAC_OUT1, 0x7f7f }, + { NODE_ID_DAC_OUT2, 0x7f7f }, + { NODE_ID_SPDIF, 0x0000 }, + { NODE_ID_ADC_IN1, 0x4343 }, + { NODE_ID_ADC_IN2, 0x4343 }, + { NODE_ID_MIC1, 0x0002 }, + { NODE_ID_MIXER_IN, 0x000b }, + { NODE_ID_MIXER_OUT1, 0x0002 }, + { NODE_ID_MIXER_OUT2, 0x0000 }, + { NODE_ID_SPK_OUT, 0x0000 }, + { NODE_ID_HP_OUT, 0x0000 }, + { NODE_ID_MIXER_IN1, 0x0005 }, + { NODE_ID_MIXER_IN2, 0x0005 }, +}; + +static bool rt286_volatile_register(struct device *dev, unsigned int reg) +{ + return 0; +} + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NODE_ID_DAC_OUT1: + case NODE_ID_DAC_OUT2: + case NODE_ID_ADC_IN1: + case NODE_ID_ADC_IN2: + case NODE_ID_MIC1: + case NODE_ID_SPDIF: + case NODE_ID_MIXER_IN: + case NODE_ID_MIXER_OUT1: + case NODE_ID_MIXER_OUT2: + case NODE_ID_DMIC1: + case NODE_ID_DMIC2: + case NODE_ID_SPK_OUT: + case NODE_ID_HP_OUT: + case NODE_ID_MIXER_IN1: + case NODE_ID_MIXER_IN2: + return 1; + default: + return 0; + } +} + +int rt286_jack_detect(struct snd_soc_codec *codec, bool *bHP, bool *bMIC) +{ + unsigned int val; + int i; + + *bHP = false; + *bMIC = false; + +#if RT_SUPPORT_COMBO_JACK + *bHP = rt286_read(codec, AC_VERB_GET_PIN_SENSE, NODE_ID_HP_OUT, 0); + if (*bHP) { + /* power on HV,VERF */ + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x03, 0x1001, 0x0000); + /* power LDO1 */ + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x08, 0x0004, 0x0004); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_MIC1, 0x24); + rt286_write(codec, AC_VERB_SET_COEF_INDEX, + 0x20, 0x50); + val = rt286_read(codec, AC_VERB_GET_PROC_COEF, + 0x20, 0x50); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + rt286_write(codec, AC_VERB_SET_COEF_INDEX, + 0x20, 0x50); + val = rt286_read(codec, + AC_VERB_GET_PROC_COEF, 0x20, 0x50); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *bMIC = false; + + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_MIC1, 0x20); + /* power off HV,VERF */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x03, 0x1001, 0x1001); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x0000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *bMIC = true; + } else { + *bMIC = false; + } + + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x20, 0x0060, 0x0000); + } else { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x20, 0x0060, 0x0020); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x8000); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0020); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x8000); + + *bMIC = false; + } + +#else + *bHP = rt286_read(codec, AC_VERB_GET_PIN_SENSE, NODE_ID_HP_OUT, 0); + *bMIC = rt286_read(codec, AC_VERB_GET_PIN_SENSE, NODE_ID_MIC1, 0); +#endif + + + /* Clear IRQ */ + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x33, 0x1, 0x1); + return 0; +} + + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static int rt286_playback_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + + ucontrol->value.integer.value[0] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0xa000) & 0x7f; + ucontrol->value.integer.value[1] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x8000) & 0x7f; + + return 0; +} + +static int rt286_playback_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vall, valr; + + vall = ucontrol->value.integer.value[0]; + valr = ucontrol->value.integer.value[1]; + if (vall == valr) { + vall = vall | 0xb000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + } else { + vall = vall | 0xa000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + valr = valr | 0x9000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + } + + return 0; +} + +static int rt286_capture_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + + ucontrol->value.integer.value[0] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x0) & 0x7f; + ucontrol->value.integer.value[1] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x2000) & 0x7f; + + return 0; +} + +static int rt286_capture_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vall, valr; + + vall = ucontrol->value.integer.value[0]; + valr = ucontrol->value.integer.value[1]; + if (vall == valr) { + vall = vall | 0x7000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + } else { + vall = vall | 0x6000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + valr = valr | 0x5000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + } + + return 0; +} + +static int rt286_mic_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x0) & 0x3; + + return 0; +} + +static int rt286_mic_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int val; + + val = ucontrol->value.integer.value[0]; + val = val | 0x7000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, val); + + return 0; +} + +static int rt286_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + + ucontrol->value.integer.value[0] = + !(rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0xa000) & 0x80); + ucontrol->value.integer.value[1] = + !(rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x8000) & 0x80); + + return 0; +} + +static int rt286_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vall, valr; + + vall = (!ucontrol->value.integer.value[0] << 7); + valr = (!ucontrol->value.integer.value[1] << 7); + + if (vall == valr) { + vall = vall | 0xb000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + } else { + vall = vall | 0xa000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + valr = valr | 0x9000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + } + + return 0; +} + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_EXT_TLV("DAC0 Playback Volume", NODE_ID_DAC_OUT1, + 8, 0, 0x7f, 0, + rt286_playback_vol_get, rt286_playback_vol_put, + out_vol_tlv), + SOC_DOUBLE_EXT_TLV("ADC0 Capture Volume", NODE_ID_ADC_IN1, + 8, 0, 0x7f, 0, + rt286_capture_vol_get, rt286_capture_vol_put, + out_vol_tlv), + SOC_SINGLE_EXT_TLV("AMIC Gain", NODE_ID_MIC1, 0, 0x3, 0, + rt286_mic_gain_get, rt286_mic_gain_put, + mic_vol_tlv), + SOC_DOUBLE_EXT("Headphone Playback Switch", NODE_ID_HP_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), + SOC_DOUBLE_EXT("Speaker Playback Switch", NODE_ID_SPK_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), +}; + +static void hwCodecConfigure(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int i; + + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D3); + for (i = 0; i < RT286_POWER_REG_LEN; i++) { + rt286_write(codec, AC_VERB_SET_POWER_STATE, + rt286_support_power_controls[i], AC_PWRST_D3); + } + + rt286_index_write(codec, NODE_ID_VENDOR_REGISTERS, 0x4F, 0xb029); + rt286_index_write(codec, NODE_ID_VENDOR_REGISTERS, 0x09, 0xc400); + rt286_index_write(codec, NODE_ID_VENDOR_REGISTERS, 0x0A, 0x0120); + rt286_index_write(codec, NODE_ID_VENDOR_REGISTERS, 0x33, 0x020A); + +#if !RT_SUPPORT_COMBO_JACK + rt286_index_write(codec, NODE_ID_VENDOR_REGISTERS, 0x50, 0x0000); + rt286_index_write(codec, NODE_ID_VENDOR_REGISTERS, 0x19, 0x0816); + rt286_index_write(codec, NODE_ID_VENDOR_REGISTERS, 0x20, 0x0000); +#endif + + mdelay(10); + + /* Set configuration default */ + /* MIC1 node 0x18 */ + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_MIC1, 0x00); + + /* DMIC1/2 */ + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_DMIC1, 0x00); + /* DMIC3/4 */ + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_DMIC2, 0x00); + + mdelay(10); + +#if RT_SUPPORT_COMBO_JACK + rt286_index_write(codec, NODE_ID_VENDOR_REGISTERS, 0x4f, 0x5029); +#endif + + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x08, 0x0008, 0x0000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x62, 0x0007, 0x0007); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x0e, 0x00c0, 0x0040); + /* Sync volume */ + regmap_read(rt286->regmap, NODE_ID_ADC_IN1, &val); + val = (val >> 8) & 0x7f; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_ADC_IN1, 0x6000 | val); + regmap_read(rt286->regmap, NODE_ID_ADC_IN1, &val); + val = val & 0x7f; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_ADC_IN1, 0x5000 | val); + regmap_read(rt286->regmap, NODE_ID_ADC_IN2, &val); + val = (val >> 8) & 0x7f; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_ADC_IN2, 0x6000 | val); + regmap_read(rt286->regmap, NODE_ID_ADC_IN2, &val); + val = val & 0x7f; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_ADC_IN2, 0x5000 | val); + regmap_read(rt286->regmap, NODE_ID_DAC_OUT1, &val); + val = (val >> 8) & 0x7f; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_DAC_OUT1, 0xa000 | val); + regmap_read(rt286->regmap, NODE_ID_DAC_OUT1, &val); + val = val & 0x7f; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_DAC_OUT1, 0x9000 | val); + regmap_read(rt286->regmap, NODE_ID_DAC_OUT2, &val); + val = (val >> 8) & 0x7f; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_DAC_OUT2, 0xa000 | val); + regmap_read(rt286->regmap, NODE_ID_DAC_OUT2, &val); + val = val & 0x7f; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_DAC_OUT2, 0x9000 | val); + regmap_read(rt286->regmap, NODE_ID_MIC1, &val); + val = val & 0x3; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_MIC1, 0x7000 | val); + /* Sync the mux selection */ + regmap_read(rt286->regmap, NODE_ID_MIXER_IN1, &val); + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, + NODE_ID_MIXER_IN1, val & RT286_ADC_SEL_MASK); + regmap_read(rt286->regmap, NODE_ID_MIXER_IN2, &val); + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, + NODE_ID_MIXER_IN2, val & RT286_ADC_SEL_MASK); + +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", NODE_ID_MIXER_OUT1, + RT286_M_FRONT_DAC_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", NODE_ID_MIXER_OUT1, + RT286_M_FRONT_REC_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Beep Switch", NODE_ID_MIXER_IN, + RT286_M_REC_BEEP_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", NODE_ID_MIXER_IN, + RT286_M_REC_LINE1_SFT, 1, 1), + SOC_DAPM_SINGLE("Mic1 Switch", NODE_ID_MIXER_IN, + RT286_M_REC_MIC1_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", NODE_ID_MIXER_IN, + RT286_M_REC_I2S_SFT, 1, 1), +}; + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "Dmic" +}; + +static int rt286_adc_values[] = { + 0, 5, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, NODE_ID_MIXER_IN1, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_VALUE_ENUM("ADC 0 source", rt286_adc0_enum); + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, NODE_ID_MIXER_IN2, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_VALUE_ENUM("ADC 1 source", rt286_adc1_enum); + +/* HP-OUT source */ +static const char * const rt286_hpo_src[] = { + "Front", "Surr" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, NODE_ID_HP_OUT, + RT286_HP_SEL_SFT, rt286_hpo_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static const char * const rt286_spo_src[] = { + "Front", "Surr" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_spo_enum, NODE_ID_SPK_OUT, + RT286_SPK_SEL_SFT, rt286_spo_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +/* SPDIF source */ +static const char * const rt286_spdif_src[] = { + "PCM-IN 0", "PCM-IN 1", "SP-OUT", "PP" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_spdif_enum, NODE_ID_SPDIF, + RT286_SPDIF_SEL_SFT, rt286_spdif_src); + +static const struct snd_kcontrol_new rt286_spdif_mux = +SOC_DAPM_ENUM("SPDIF source", rt286_spdif_enum); + + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_SPK_OUT, AC_PWRST_D0); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_SPK_OUT, 0x40); + rt286_write(codec, AC_VERB_SET_EAPD_BTLENABLE, + NODE_ID_SPK_OUT, 0x0002); + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_SPK_OUT, 0xb000); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_SPK_OUT, 0xb080); + rt286_write(codec, AC_VERB_SET_EAPD_BTLENABLE, + NODE_ID_SPK_OUT, 0x0000); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_SPK_OUT, 0x00); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_SPK_OUT, AC_PWRST_D3); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_DMIC1, 0x20); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC1, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC1, AC_PWRST_D3); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_DMIC1, 0x0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC2, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC2, AC_PWRST_D3); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_hp_pow_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_HP_OUT, AC_PWRST_D0); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_HP_OUT, 0x40); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_HP_OUT, 0x00); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_HP_OUT, AC_PWRST_D3); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xb000); + mdelay(hp_amp_time); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xb080); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_dac0_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DAC_OUT1, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DAC_OUT1, AC_PWRST_D3); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_dac1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DAC_OUT2, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DAC_OUT2, AC_PWRST_D3); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_adc0_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: +#if (RT_SUPPORT_COMBO_JACK == 1) + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x8000); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0020); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x8000); +#endif + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0f, 0x0020, 0x0000); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_ADC_IN1, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_ADC_IN1, AC_PWRST_D3); + break; + case SND_SOC_DAPM_POST_REG: + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, + NODE_ID_MIXER_IN1, snd_soc_read(codec, + NODE_ID_MIXER_IN1) & RT286_ADC_SEL_MASK); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_adc1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + +#if (RT_SUPPORT_COMBO_JACK == 1) + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x8000); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0020); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x8000); +#endif + rt286_write(codec, AC_VERB_SET_CHANNEL_STREAMID, + NODE_ID_ADC_IN1, 0x10); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0f, 0x0020, 0x0000); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_ADC_IN2, AC_PWRST_D0); + + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_ADC_IN2, AC_PWRST_D3); + break; + case SND_SOC_DAPM_POST_REG: + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, + NODE_ID_MIXER_IN2, snd_soc_read(codec, + NODE_ID_MIXER_IN2) & RT286_ADC_SEL_MASK); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_spk_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_REG: + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, + NODE_ID_SPK_OUT, snd_soc_read(codec, + NODE_ID_SPK_OUT) & RT286_SPK_SEL_MASK); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_hpo_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_REG: + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, + NODE_ID_HP_OUT, snd_soc_read(codec, + NODE_ID_HP_OUT) & RT286_HP_SEL_MASK); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("DMIC2", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_set_dmic2_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", SND_SOC_NOPM, 0, 0, + &rt286_adc0_mux, rt286_adc0_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", SND_SOC_NOPM, 0, 0, + &rt286_adc1_mux, rt286_adc1_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + + /* Output Side */ + + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC 0", NULL, SND_SOC_NOPM, + 0, 0, rt286_dac0_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC 1", NULL, SND_SOC_NOPM, + 0, 0, rt286_dac1_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Output Mux */ + SND_SOC_DAPM_MUX_E("SPK Mux", SND_SOC_NOPM, 0, 0, + &rt286_spo_mux, + rt286_spk_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_MUX_E("HPO Mux", SND_SOC_NOPM, 0, 0, + &rt286_hpo_mux, + rt286_hpo_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_SUPPLY("HP Power", SND_SOC_NOPM, + 0, 0, rt286_hp_pow_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_MUX("SPDIF Mux", SND_SOC_NOPM, 0, 0, + &rt286_spdif_mux), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", SND_SOC_NOPM, 0, 0, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surr", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_PGA_E("SPO", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("HPO", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surr", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surr", "Surr"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surr", "Surr"}, + + {"SPO", NULL, "SPK Mux"}, + {"HPO", NULL, "HPO Mux"}, + {"HPO", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO"}, +}; + +static int rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned long val; + int d_len_code; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val = rt286_read(codec, AC_VERB_GET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, 0); + else + val = rt286_read(codec, AC_VERB_GET_STREAM_FORMAT, + NODE_ID_ADC_IN1, 0); + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 48000: + val &= ~0x4000; + break; + case 44100: + val |= 0x4000; + break; + default: + pr_err("unsportted sample rate %d\n", params_rate(params)); + return -EINVAL; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val &= ~0xf; + val |= (params_channels(params) - 1); + } else { + pr_err("unsportted channels %d\n", params_channels(params)); + return -EINVAL; + } + + val &= ~0x70; + d_len_code = 0; + switch (params_format(params)) { + /* bit 6:4 Bits per Sample */ + case SNDRV_PCM_FORMAT_S16_LE: + d_len_code = 0; + val |= (0x1 << 4); + break; + case SNDRV_PCM_FORMAT_S32_LE: + d_len_code = 2; + val |= (0x4 << 4); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + d_len_code = 1; + val |= (0x2 << 4); + break; + case SNDRV_PCM_FORMAT_S24_LE: + d_len_code = 2; + val |= (0x3 << 4); + break; + case SNDRV_PCM_FORMAT_S8: + break; + default: + return -EINVAL; + } + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0018, d_len_code << 3); + pr_debug("format val = 0x%lx\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rt286_write(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, val); + else + rt286_write(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_ADC_IN1, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned long val_dac, val_adc; + + val_dac = rt286_read(codec, AC_VERB_GET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, 0); + val_adc = rt286_read(codec, AC_VERB_GET_STREAM_FORMAT, + NODE_ID_ADC_IN1, 0); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + val_dac &= ~0x8000; + val_adc &= ~0x8000; + rt286_write(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, val_dac); + rt286_write(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_ADC_IN1, val_adc); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + + pr_debug("%s freq=%d\n", __func__, freq); + + switch (freq) { + case 19200000: + case 24000000: + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x63, 0x4, 0x4); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x49, 0x20, 0x0); + break; + case 12288000: + case 11289600: + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x0108, 0x0000); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0b, 0xfc1e, 0x0004); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x49, 0x0020, 0x0020); + break; + case 24576000: + case 22579200: + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x0108, 0x0008); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0b, 0xfc1e, 0x5406); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x49, 0x0020, 0x0020); + break; + default: + pr_err("unsported system clock\n"); + return -EINVAL; + } + + return 0; +} + +static int rt286_set_dai_dfs(struct snd_soc_dai *dai, unsigned int fs) +{ + struct snd_soc_codec *codec = dai->codec; + + pr_debug("%s fs=%d\n", __func__, fs); + if (50 == fs) + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x1000, 0x1000); + + + return 0; +} + +static ssize_t rt286_codec_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct i2c_client *client = to_i2c_client(dev); + struct rt286_priv *rt286 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = rt286->codec; + unsigned long val; + int cnt = 0, i; + + cnt += sprintf(buf, "RT286 codec register\n"); + for (i = 0; i <= TOTAL_NODE_ID; i++) { + if (cnt + 31 >= PAGE_SIZE) + break; + if (!rt286_readable_register(dev, i)) + continue; + val = rt286_read(codec, AC_VERB_GET_POWER_STATE, i, 0); + + cnt += snprintf(buf + cnt, 31, + "vid: 0x%x nid: %02x val: %04lx\n", + AC_VERB_GET_POWER_STATE, i, val); + } + for (i = 0; i <= TOTAL_NODE_ID; i++) { + if (cnt + 31 >= PAGE_SIZE) + break; + if (!rt286_readable_register(dev, i)) + continue; + val = rt286_read(codec, AC_VERB_GET_PIN_WIDGET_CONTROL, i, 0); + + cnt += snprintf(buf + cnt, 31, + "vid: 0x%x nid: %02x val: %04lx\n", + AC_VERB_GET_PIN_WIDGET_CONTROL, i, val); + } + for (i = 0; i <= TOTAL_NODE_ID; i++) { + if (cnt + 31 >= PAGE_SIZE) + break; + if (!rt286_readable_register(dev, i)) + continue; + val = rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, i, 0); + + cnt += snprintf(buf + cnt, 31, + "vid: 0x%x nid: %02x val: %04lx\n", + AC_VERB_GET_AMP_GAIN_MUTE, i, val); + } + + val = rt286_index_read(codec, NODE_ID_VENDOR_REGISTERS, 0x09); + cnt += snprintf(buf + cnt, 31, "Index: 0x%x val: %04lx\n", 0x09, val); + + + if (cnt >= PAGE_SIZE) + cnt = PAGE_SIZE - 1; + + return cnt; +} + +static ssize_t rt286_codec_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct i2c_client *client = to_i2c_client(dev); + struct rt286_priv *rt286 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = rt286->codec; + unsigned int nid = 0, vid = 0; + unsigned long data = 0; + int i; + + pr_debug("register \"%s\" count=%ld\n", buf, count); + + for (i = 0; i < count; i++) { + if (*(buf + i) <= '9' && *(buf + i) >= '0') + vid = (vid << 4) | (*(buf + i) - '0'); + else if (*(buf + i) <= 'f' && *(buf + i) >= 'a') + vid = (vid << 4) | ((*(buf + i) - 'a') + 0xa); + else if (*(buf + i) <= 'F' && *(buf + i) >= 'A') + vid = (vid << 4) | ((*(buf + i) - 'A') + 0xa); + else + break; + } + + for (i = i + 1; i < count; i++) { + if (*(buf + i) <= '9' && *(buf + i) >= '0') + nid = (nid << 4) | (*(buf + i) - '0'); + else if (*(buf + i) <= 'f' && *(buf + i) >= 'a') + nid = (nid << 4) | ((*(buf + i) - 'a') + 0xa); + else if (*(buf + i) <= 'F' && *(buf + i) >= 'A') + nid = (nid << 4) | ((*(buf + i) - 'A') + 0xa); + else + break; + } + + for (i = i + 1; i < count; i++) { + if (*(buf + i) <= '9' && *(buf + i) >= '0') + data = (data << 4) | (*(buf + i) - '0'); + else if (*(buf + i) <= 'f' && *(buf + i) >= 'a') + data = (data << 4) | ((*(buf + i) - 'a') + 0xa); + else if (*(buf + i) <= 'F' && *(buf + i) >= 'A') + data = (data << 4) | ((*(buf + i) - 'A') + 0xa); + else + break; + } + + if (*(buf + i + 1) == 'r') { + pr_info("vid=0x%x nid=0x%x data=0x%lx\n", + vid, nid, rt286_read(codec, vid, nid, data)); + } else { + rt286_write(codec, vid, nid, data); + pr_debug("vid=0x%x nid=0x%x data=0x%lx\n", vid, nid, data); + } + + return count; +} + +static DEVICE_ATTR(codec_reg, 0666, rt286_codec_show, rt286_codec_store); + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + pr_debug("In case SND_SOC_BIAS_ON:\n"); + break; + + case SND_SOC_BIAS_PREPARE: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D0); + pr_debug("In case SND_SOC_BIAS_PREPARE:\n"); + break; + + case SND_SOC_BIAS_STANDBY: + pr_debug("In case SND_SOC_BIAS_STANDBY:\n"); + break; + + case SND_SOC_BIAS_OFF: + pr_debug("In case SND_SOC_BIAS_OFF:\n"); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D3); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int ret; + + pr_info("Codec driver version %s\n", VERSION); + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + hwCodecConfigure(codec); + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + rt286->codec = codec; + + ret = device_create_file(codec->dev, &dev_attr_codec_reg); + if (ret != 0) { + dev_err(codec->dev, + "Failed to create codec_reg sysfs files: %d\n", ret); + return ret; + } + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + rt286_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +#ifdef CONFIG_PM +static int rt286_suspend(struct snd_soc_codec *codec) +{ + return 0; +} + +static int rt286_resume(struct snd_soc_codec *codec) +{ + return 0; +} +#else +#define rt286_suspend NULL +#define rt286_resume NULL +#endif + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_dfs = rt286_set_dai_dfs, +}; + +struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .suspend = rt286_suspend, + .resume = rt286_resume, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = TOTAL_NODE_ID + 1, + .volatile_reg = rt286_volatile_register, + .readable_reg = rt286_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt286_reg, + .num_reg_defaults = ARRAY_SIZE(rt286_reg), +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt286", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static struct acpi_device_id rt286_acpi_match[] = { + {"INT343A", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_priv *rt286; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, + sizeof(struct rt286_priv), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->i2c = i2c; + i2c_set_clientdata(i2c, rt286); + + rt286->regmap = devm_regmap_init_i2c(i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regcache_cache_only(rt286->regmap, true); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static void rt286_i2c_shutdown(struct i2c_client *client) +{ + struct rt286_priv *rt286 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = rt286->codec; + + if (codec != NULL) + rt286_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt286", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .shutdown = rt286_i2c_shutdown, + .id_table = rt286_i2c_id, +}; + +static int __init rt286_modinit(void) +{ + return i2c_add_driver(&rt286_i2c_driver); +} + +module_init(rt286_modinit); + +static void __exit rt286_modexit(void) +{ + i2c_del_driver(&rt286_i2c_driver); +} + +module_exit(rt286_modexit); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 0000000..3702d9a --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,113 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT286_H__ +#define __RT286_H__ + +#define NODE_ID_AUDIO_FUNCTION_GROUP 0x01 +#define NODE_ID_DAC_OUT1 0x02 +#define NODE_ID_DAC_OUT2 0x03 +#define NODE_ID_ADC_IN1 0x09 +#define NODE_ID_ADC_IN2 0x08 +#define NODE_ID_MIXER_IN 0x0b +#define NODE_ID_MIXER_OUT1 0x0c +#define NODE_ID_MIXER_OUT2 0x0d +#define NODE_ID_DMIC1 0x12 +#define NODE_ID_DMIC2 0x13 +#define NODE_ID_SPK_OUT 0x14 +#define NODE_ID_MIC1 0x18 +#define NODE_ID_LINE1 0x1a +#define NODE_ID_BEEP 0x1d +#define NODE_ID_SPDIF 0x1e +#define NODE_ID_VENDOR_REGISTERS 0x20 +#define NODE_ID_HP_OUT 0x21 +#define NODE_ID_MIXER_IN1 0x22 +#define NODE_ID_MIXER_IN2 0x23 +#define TOTAL_NODE_ID 0x23 + +#define CONNECTION_INDEX_MIC1 0X0 +#define CONNECTION_INDEX_DMIC 0X5 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +/* System Clock Source */ +enum { + RT286_SCLK_S_MCLK, + RT286_SCLK_S_PLL1, + RT286_SCLK_S_RCCLK, +}; + +enum { + RT286_AIF1, + RT286_AIF2, + RT286_AIF3, + RT286_AIFS, +}; + +enum { + RT286_DMIC_DIS, + RT286_DMIC1, + RT286_DMIC2, +}; + +enum { + RT286_J_IN_EVENT, /* Jack insert */ + RT286_J_OUT_EVENT, /* Jack evulse */ + RT286_BP_EVENT, /* Button Press */ + RT286_BR_EVENT, /* Button Release */ + RT286_UN_EVENT, /* Unknown */ +}; + +struct rt286_pll_code { + bool m_bp; /* Indicates bypass m code or not. */ + int m_code; + int n_code; + int k_code; +}; +#endif /* __RT286_H__ */ -- 1.8.1.1.439.g50a6b54

3 2

[alsa-devel] [PATCH] ASoC: kirkwood: Fix erroneous double output while playing
by Jean-Francois Moine 27 Nov '13

27 Nov '13

This patch fixes the setting of the register KIRKWOOD_PLAYCTL which did always streaming on both I2S and SPDIF, ignoring the DAI ID. The bug was introduced by the commit 75b9b65ee5a "ASoC: kirkwood: add S/PDIF support" Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d34d917..b27f826 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -244,15 +244,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl); } - if (dai->id == 0) - ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ - else - ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ - switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_play; + if (dai->id == 0) + ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ + value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_PLAYCTL); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH] ASoC: kirkwood: Fix invalid SPDIF format
by Jean-Francois Moine 27 Nov '13

27 Nov '13

This patch removes the 32 bits format which is not supported by S/PDIF output. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index b27f826..0b18f65 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -33,6 +33,10 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define KIRKWOOD_SPDIF_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + static int kirkwood_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -449,14 +453,14 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, }, @@ -493,7 +497,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .capture = { .channels_min = 1, @@ -501,7 +505,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, }, -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

5 7

[alsa-devel] [PATCH 1/2] ASoC: soc_pcm_init_runtime_hw: Fix rate_max calculation
by Lars-Peter Clausen 27 Nov '13

27 Nov '13

In order to make sure that the sample rate is in the supported range of both components the maximum rate of the card should be the minimum of the maximum rate of each components. There is one special case to consider though, if max_rate is set to 0 this means there is no maximum specified, so use min_not_zero() macro which will give use the desired result. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 170ff96..c4ef880 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -220,7 +220,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw, struct snd_soc_pcm_stream *cpu_stream) { hw->rate_min = max(codec_stream->rate_min, cpu_stream->rate_min); - hw->rate_max = max(codec_stream->rate_max, cpu_stream->rate_max); + hw->rate_max = min_not_zero(codec_stream->rate_max, cpu_stream->rate_max); hw->channels_min = max(codec_stream->channels_min, cpu_stream->channels_min); hw->channels_max = min(codec_stream->channels_max, -- 1.8.0

3 10

[alsa-devel] [PATCH] ASoC: wm5110: Remove output OSR and PGA volume controls
by Charles Keepax 27 Nov '13

27 Nov '13

These are managed automatically in current revisions. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 25 ------------------------- 1 files changed, 0 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 0e63d8c..9eb8517 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -309,19 +309,6 @@ ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE), -SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUT1_OSR_SHIFT, 1, 0), -SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, - ARIZONA_OUT2_OSR_SHIFT, 1, 0), -SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUT3_OSR_SHIFT, 1, 0), -SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, - ARIZONA_OUT4_OSR_SHIFT, 1, 0), -SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, - ARIZONA_OUT5_OSR_SHIFT, 1, 0), -SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L, - ARIZONA_OUT6_OSR_SHIFT, 1, 0), - SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, @@ -354,18 +341,6 @@ SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L, ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUTPUT_PATH_CONFIG_1R, - ARIZONA_OUT1L_PGA_VOL_SHIFT, - 0x34, 0x40, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, - ARIZONA_OUTPUT_PATH_CONFIG_2R, - ARIZONA_OUT2L_PGA_VOL_SHIFT, - 0x34, 0x40, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUTPUT_PATH_CONFIG_3R, - ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), - SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: hda - resume codecs in parallel
by mengdong.lin＠intel.com 27 Nov '13

27 Nov '13

From: Mengdong Lin <mengdong.lin(a)intel.com> To reduce driver resume time, this patch resumes the codecs in parallel if there are multiple codecs on the bus. - The PM workqueue of bus is also used to parallel resuming multiple codecs. - The work item 'pm_work' is renamed to 'suspend_work' to parallel suspending codecs. - Add a work item 'resume_work' to parallel resuming codecs. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 60ff45a..1e04a20 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -104,7 +104,8 @@ static inline void hda_call_pm_notify(struct hda_bus *bus, bool power_up) if (bus->ops.pm_notify) bus->ops.pm_notify(bus, power_up); } -static void hda_pm_work(struct work_struct *work); +static void hda_suspend_work(struct work_struct *work); +static void hda_resume_work(struct work_struct *work); #else #define codec_in_pm(codec) 0 static inline void hda_keep_power_on(struct hda_codec *codec) {} @@ -1447,7 +1448,8 @@ int snd_hda_codec_new(struct hda_bus *bus, #ifdef CONFIG_PM spin_lock_init(&codec->power_lock); INIT_DELAYED_WORK(&codec->power_work, hda_power_work); - INIT_WORK(&codec->pm_work, hda_pm_work); + INIT_WORK(&codec->suspend_work, hda_suspend_work); + INIT_WORK(&codec->resume_work, hda_resume_work); /* snd_hda_codec_new() marks the codec as power-up, and leave it as is. * the caller has to power down appropriatley after initialization * phase. @@ -5058,13 +5060,21 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power); -static void hda_pm_work(struct work_struct *work) +static void hda_suspend_work(struct work_struct *work) { struct hda_codec *codec = - container_of(work, struct hda_codec, pm_work); + container_of(work, struct hda_codec, suspend_work); hda_call_codec_suspend(codec, false); } + +static void hda_resume_work(struct work_struct *work) +{ + struct hda_codec *codec = + container_of(work, struct hda_codec, resume_work); + + hda_call_codec_resume(codec); +} #endif /* @@ -5633,7 +5643,7 @@ int snd_hda_suspend(struct hda_bus *bus) cancel_delayed_work_sync(&codec->jackpoll_work); if (hda_codec_is_power_on(codec)) { if (bus->num_codecs > 1) - queue_work(bus->pm_wq, &codec->pm_work); + queue_work(bus->pm_wq, &codec->suspend_work); else hda_call_codec_suspend(codec, false); } @@ -5659,9 +5669,15 @@ int snd_hda_resume(struct hda_bus *bus) printk("snd_hda_resume, pci device 0x%04x\n", bus->pci->device); list_for_each_entry(codec, &bus->codec_list, list) { - hda_call_codec_resume(codec); + if (bus->num_codecs > 1) + queue_work(bus->pm_wq, &codec->resume_work); + else + hda_call_codec_resume(codec); } + if (bus->num_codecs > 1) + flush_workqueue(bus->pm_wq); + printk("snd_hda_resume ok, pci device 0x%04x\n", bus->pci->device); return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 82807ef..bedb436 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -920,7 +920,9 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; - struct work_struct pm_work; /* task to parallel multi-codec PM */ + /* tasks to parallel multi-codec suspend/resume */ + struct work_struct suspend_work; + struct work_struct resume_work; #endif /* filter the requested power state per nid */ -- 1.8.1.2

2 1

[alsa-devel] [RFC PATCH] ALSA: hda - suspend codecs in parallel
by mengdong.lin＠intel.com 27 Nov '13

27 Nov '13

From: Mengdong Lin <mengdong.lin(a)intel.com> The time to suspend a single codec may be several hundreds of ms. When runtime power saving is not enabled, driver suspend time can be long especially there are more than one codec on the bus. To reduce driver suspend time, this patch creates a work queue for each codec and will suspend the codecs in parallel, if there are multiple codecs on bus. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index cb0c776..c9092a1 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -97,6 +97,7 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset); #ifdef CONFIG_PM #define codec_in_pm(codec) ((codec)->in_pm) static void hda_power_work(struct work_struct *work); +static void hda_pm_work(struct work_struct *work); static void hda_keep_power_on(struct hda_codec *codec); #define hda_codec_is_power_on(codec) ((codec)->power_on) static inline void hda_call_pm_notify(struct hda_bus *bus, bool power_up) @@ -1355,6 +1356,8 @@ static void snd_hda_codec_free(struct hda_codec *codec) #ifdef CONFIG_PM if (!codec->pm_down_notified) /* cancel leftover refcounts */ hda_call_pm_notify(codec->bus, false); + if (codec->pm_wq) + destroy_workqueue(codec->pm_wq); #endif module_put(codec->owner); free_hda_cache(&codec->amp_cache); @@ -1434,6 +1437,13 @@ int snd_hda_codec_new(struct hda_bus *bus, */ hda_keep_power_on(codec); hda_call_pm_notify(bus, true); + + codec->pm_wq = create_workqueue("hda_codec_wq"); + if (!codec->pm_wq) { + snd_hda_codec_free(codec); + return -ENOMEM; + } + INIT_WORK(&codec->pm_work, hda_pm_work); #endif if (codec->bus->modelname) { @@ -1445,6 +1455,7 @@ int snd_hda_codec_new(struct hda_bus *bus, } list_add_tail(&codec->list, &bus->codec_list); + bus->num_codecs++; bus->caddr_tbl[codec_addr] = codec; codec->vendor_id = snd_hda_param_read(codec, AC_NODE_ROOT, @@ -5032,6 +5043,14 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, return 0; } EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power); + +static void hda_pm_work(struct work_struct *work) +{ + struct hda_codec *codec = + container_of(work, struct hda_codec, pm_work); + + hda_call_codec_suspend(codec, false); +} #endif /* @@ -5595,11 +5614,24 @@ EXPORT_SYMBOL_HDA(snd_hda_add_imux_item); int snd_hda_suspend(struct hda_bus *bus) { struct hda_codec *codec; + unsigned int mask = 0; list_for_each_entry(codec, &bus->codec_list, list) { cancel_delayed_work_sync(&codec->jackpoll_work); - if (hda_codec_is_power_on(codec)) - hda_call_codec_suspend(codec, false); + if (hda_codec_is_power_on(codec)) { + if (codec->epss || bus->num_codecs == 1) + hda_call_codec_suspend(codec, false); + else { + mask |= 1 << codec->addr; + queue_work(codec->pm_wq, &codec->pm_work); + } + } + } + + list_for_each_entry(codec, &bus->codec_list, list) { + if (mask & (1 << codec->addr)) + flush_work(&codec->pm_work); + } return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 7aa9870..ffb1866 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -673,6 +673,7 @@ struct hda_bus { /* codec linked list */ struct list_head codec_list; + unsigned int num_codecs; /* link caddr -> codec */ struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1]; @@ -916,6 +917,8 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; + struct workqueue_struct *pm_wq; + struct work_struct pm_work; #endif /* filter the requested power state per nid */ -- 1.8.1.2

3 15

[alsa-devel] [PATCH] ALSA: usb: use multiple packets per urb for Wireless USB inbound audio
by Thomas Pugliese 27 Nov '13

27 Nov '13

For Wireless USB audio devices, use multiple isoc packets per URB for inbound endpoints with a datainterval < 5. This allows the WUSB host controller to take advantage of bursting to service endpoints whose logical polling interval is less than the 4ms minimum polling interval limit in WUSB. Signed-off-by: Thomas Pugliese <thomas.pugliese(a)gmail.com> --- sound/usb/endpoint.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index b9ba0fc..83aabea2 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -636,8 +636,22 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, if (usb_pipein(ep->pipe) || snd_usb_endpoint_implicit_feedback_sink(ep)) { + urb_packs = packs_per_ms; + /* + * Wireless devices can poll at a max rate of once per 4ms. + * For dataintervals less than 5, increase the packet count to + * allow the host controller to use bursting to fill in the + * gaps. + */ + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) { + int interval = ep->datainterval; + while (interval < 5) { + urb_packs <<= 1; + ++interval; + } + } /* make capture URBs <= 1 ms and smaller than a period */ - urb_packs = min(max_packs_per_urb, packs_per_ms); + urb_packs = min(max_packs_per_urb, urb_packs); while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) urb_packs >>= 1; ep->nurbs = MAX_URBS; -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Enable mute/mic-mute LEDs for more Thinkpads with Conexant codec
by Hui Wang 27 Nov '13

27 Nov '13

Most Thinkpad Edge series laptops use conexant codec, so far although the codecs have different minor Vendor Id and minor Subsystem Id, but they all belong to the cxt_5066 family, this change can make the mute/mic-mute LEDs support more generic among cxt_5066 family. Cc: Alex Hung <alex.hung(a)canonical.com> Cc: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_local.h | 22 ++++++++++++++++++++++ sound/pci/hda/patch_conexant.c | 3 +++ sound/pci/hda/patch_realtek.c | 20 -------------------- 3 files changed, 25 insertions(+), 20 deletions(-) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index d398b64..1155958 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -777,4 +777,26 @@ void snd_hdmi_write_eld_info(struct hdmi_eld *eld, #define SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE 80 void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen); +#if IS_ENABLED(CONFIG_THINKPAD_ACPI) +#include <acpi/acpi.h> + +static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, + void **rv) +{ + bool *found = context; + *found = true; + return AE_OK; +} + +static inline bool is_thinkpad(struct hda_codec *codec) +{ + bool found = false; + if (codec->subsystem_id >> 16 != 0x17aa) + return false; + if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) + return true; + found = false; + return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; +} +#endif #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c205bb1..8063ae9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3279,6 +3279,8 @@ static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec, bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { + if (!is_thinkpad(codec)) + return; if (!led_set_func) led_set_func = symbol_request(tpacpi_led_set); if (!led_set_func) { @@ -3494,6 +3496,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), {} diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5e42059..bedf35f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3673,29 +3673,9 @@ static void alc290_fixup_mono_speakers(struct hda_codec *codec, #if IS_ENABLED(CONFIG_THINKPAD_ACPI) #include <linux/thinkpad_acpi.h> -#include <acpi/acpi.h> static int (*led_set_func)(int, bool); -static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, - void **rv) -{ - bool *found = context; - *found = true; - return AE_OK; -} - -static bool is_thinkpad(struct hda_codec *codec) -{ - bool found = false; - if (codec->subsystem_id >> 16 != 0x17aa) - return false; - if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) - return true; - found = false; - return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; -} - static void update_tpacpi_mute_led(void *private_data, int enabled) { if (led_set_func) -- 1.8.1.2

4 7

Re: [alsa-devel] Commit:ce6cfaf1 query for shared control
by Lars-Peter Clausen 27 Nov '13

27 Nov '13

On 11/25/2013 11:57 PM, noman pouigt wrote: > On Sun, Nov 24, 2013 at 11:45 PM, Lars-Peter Clausen <lars(a)metafoo.de> wrote: >> On 11/25/2013 08:12 AM, noman pouigt wrote: >>> Hello Lars, >>> >>> I was wondering if you can enlighten me >>> on below query. >>> >>> Commit:ce6cfaf1 >>> >>> From the commit text "The input and >>> output paths for each widgets are only >>> updated though during the respective run >>> for that widget". > Thanks for replying.Probably my query was > not clear.I wanted to understand the commit > text which i quoted above.And my understanding > of that commit text is below: The text just describes the behavior as it was before the commit, which was separate power-up/power-down sequence for each widget. >>> Does this commit text mean: when mux >>> or mixer_power_update is called then only >>> the widget is powered and it's peer- widget >>> source(path source) and widget sink(path sink)? >> >> Hi, >> >> What the commit does is to make sure that if a control is shared between >> multiple mixers/muxes to apply all the changes in the same update sequence. >> Instead of, as it was done before, to run a update sequence for each mixer/mux. > I understood this as you have clearly mentioned it with a nice > diagram. >> >>> >>> +------+ >>> A1 ------| | >>> | MUX1 |----- C1 >>> B1 ------| | >>> +------+ >>> | >>> control ---+ >>> | >>> +------+ >>> A2 ------| | >>> | MUX2 |----- C2 >>> B2 ------| | >>> +------+ >>> >>> Can I represent the above diagram as below: >>> >>> char *input_text_1[] = { >>> "A1", "B1" >>> }; >>> >>> struct soc_enum input_enum_1 = >>> SOC_ENUM_SINGLE(some_register, some_bit, >>> 2, input_text_1); >>> >>> char *input_text_2[] = { >>> "A2", "B2" >>> }; >>> struct soc_enum input_enum_2 = >>> SOC_ENUM_SINGLE(some_register, some_bit, >>> 2, input_text_2); >>> >>> >>> struct snd_kcontrol_new input_mux = >>> SOC_DAPM_ENUM("C1", input_enum_1); >>> struct snd_kcontrol_new input_mux = >>> SOC_DAPM_ENUM("C2", input_enum_2); >>> >>> dapm_route[] { >>> "A1", "control","C1" >>> "B1", "control","C1" >>> "A2", "control","C2" >>> "B2", "control","C2" >>> }; >> >> The control part needs to have the name of one of the enum items. So this >> would be { "A1", "A1", "C1" } and so on. > Yes makes sense as then only it will know > which widget to connect to as it has multiple > widgets. >> >> - Lars >> >> >> >> >>

1 0

[alsa-devel] [PATCH 0/5] A few HD-audio things for 3.14
by Takashi Iwai 26 Nov '13

26 Nov '13

Hi, here are a few patches that are targeted for 3.14. So they are not urgently needed, mostly clean ups, but also includes some runtime PM changes as we're discussing now. The patches are found in sound-unstable git tree test/hda branch, too. Takashi

2 8

Re: [alsa-devel] [alsa-cvslog] [ALSA GIT]ALSA utilities repository branch master updated. v1.0.27.2-14-gd2d9b6f
by Clemens Ladisch 26 Nov '13

26 Nov '13

GIT server wrote: > - Log ----------------------------------------------------------------- > commit d2d9b6fc8d20fca1e06fc261ac20fd6cfd8a8d1b > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Tue Nov 26 11:44:40 2013 +0100 > > alsactl: Add monitor command > mask = snd_ctl_event_elem_get_mask(event); > if (mask & SND_CTL_EVENT_MASK_VALUE) > printf(" VALUE"); > if (mask & SND_CTL_EVENT_MASK_INFO) > printf(" INFO"); > if (mask & SND_CTL_EVENT_MASK_ADD) > printf(" ADD"); > if (mask & SND_CTL_EVENT_MASK_TLV) > printf(" TLV"); > if (mask == SND_CTL_EVENT_MASK_REMOVE) > printf(" REMOVE"); The documentation of SND_CTL_EVENT_MASK_REMOVE says: | Warning: test this first and if set don't test the other masks Regards, Clemens

2 1

[alsa-devel] HD-Audio: Is it okay to delay resuming codecs when resuming from S3?
by Lin, Mengdong 26 Nov '13

26 Nov '13

Hi Takashi, Is it okay to delay resuming codecs when resuming from S3? i.e. skip snd_hda_resume() in azx_resume()? This would reduce driver resume time from S3. And the codecs will be resumed when being used for the 1st time after S3. Resuming a codec may cost more than 100ms. So even if we parallel resuming codecs in snd_hda_resume(), driver resume time on some platforms is still a bit long. Regards Mengdong

2 3

[alsa-devel] [PATCH] ALSA: hda - Drop bus->avoid_link_reset flag
by Takashi Iwai 26 Nov '13

26 Nov '13

Use bus->power_keep_link_on instead. The controller shouldn't go to D3 when the link isn't reset, so essentially avoiding the link reset means avoiding the runtime PM. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.h | 1 - sound/pci/hda/hda_intel.c | 3 +-- sound/pci/hda/patch_sigmatel.c | 3 ++- 3 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 77db69480c19..7aa9870040c1 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -698,7 +698,6 @@ struct hda_bus { unsigned int in_reset:1; /* during reset operation */ unsigned int power_keep_link_on:1; /* don't power off HDA link */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ - unsigned int avoid_link_reset:1; /* don't reset link at runtime PM */ int primary_dig_out_type; /* primary digital out PCM type */ }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7a09404579a7..c6d230193da6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2994,8 +2994,7 @@ static int azx_runtime_suspend(struct device *dev) STATESTS_INT_MASK); azx_stop_chip(chip); - if (!chip->bus->avoid_link_reset) - azx_enter_link_reset(chip); + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) hda_display_power(false); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d2cc0041d9d3..088a5afbd1b9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2094,7 +2094,8 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ - codec->bus->avoid_link_reset = 1; + /* resetting controller clears GPIO, so we need to keep on */ + codec->bus->power_keep_link_on = 1; } } -- 1.8.4.3

1 0

[alsa-devel] [PATCH v2 0/9] ASoC: fsl-ssi: offline/online configuration and cleanups
by Markus Pargmann 26 Nov '13

26 Nov '13

Hi, This patch series fixes some fsl-ssi code that does not act exactly as it is described in the reference manuals. The reference manuals before imx50 do mention that some register bits, including TDMAE/RDMAE, should not be changed while the SSI unit is enabled (SSIEN). At the same time the SDMA unit has a undefined request handling if there are DMA requests before the SDMA engine and its channel is configured. The reference manual states that SSI DMA requests are sent as soon as the FIFO and the DMAE bits are enabled. It does not mention a dependency to SCR's TE/RE bit. To avoid sending DMA requests before a channel is configured, we have to seperate the fsl-ssi behavior into offline and online configuration. SoCs before imx50 have to use offline configuration while later SoC versions can use online reconfiguration. This series adds support for online configuration and cleans up the configuration register code especially in fsl_ssi_trigger. At the end we have a seperation between enable/disable logic and the actual configuration register values used to enable/disable TX/RX. I tested this series on mx53. The series is based on Mark Brown's topic/fsl branch. Regards, Markus Changes in v2: - Enable fsl-ssi interrupts independent of CONFIG_DEBUG_FS. - Move irq mapping fix into seperate patch (was in "Move sysfs stats to debugfs") - Rebased on v3.13-rc1 topic/fsl branch. Changes in v1: - Not removing imx21-ssi from the compatible array in the DTS patch. - Dropped "ASoC: fsl-ssi: Drop AC97 debug register usage" for the moment as I don't have time to setup AC97 hardware for some tests Markus Pargmann (9): ASoC: fsl-ssi: Move sysfs stats to debugfs ASoC: fsl-ssi: Fix interrupt mapping and release ASoC: fsl-ssi: Add imx50-ssi and of_device_id matching ASoC: fsl-ssi: Add offline_config flag ASoC: fsl-ssi: Add configuration helper functions ASoC: fsl-ssi: Move RX/TX configuration to seperate functions ASoC: fsl-ssi: Drop ac97 specific trigger function ARM: DTS: imx5* imx6*, use imx50-ssi ASoC: fsl-ssi: Drop AC97 debug register usage arch/arm/boot/dts/imx51.dtsi | 10 +- arch/arm/boot/dts/imx53.dtsi | 10 +- arch/arm/boot/dts/imx6qdl.dtsi | 12 +- arch/arm/boot/dts/imx6sl.dtsi | 12 +- sound/soc/fsl/fsl_ssi.c | 521 +++++++++++++++++++++++++++-------------- 5 files changed, 371 insertions(+), 194 deletions(-) -- 1.8.4.2

2 11

[alsa-devel] [PATCH 1/2] ALSA: hda/realtek - Add support of ALC231 codec
by Takashi Iwai 26 Nov '13

26 Nov '13

From: Kailang Yang <kailang(a)realtek.com> It's compatible with ALC269. Signed-off-by: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5e42059f10a1..60973fd6eb82 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5175,6 +5175,7 @@ static int patch_alc680(struct hda_codec *codec) */ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 }, + { .id = 0x10ec0231, .name = "ALC231", .patch = patch_alc269 }, { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 }, { .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 }, { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, -- 1.8.4.3

1 1

[alsa-devel] [PATCH] ALSA: cs46xx: Move DSP blob out of kernel tree
by Takashi Iwai 26 Nov '13

26 Nov '13

... and load from the external firmware files. The firmware binary blobs in cs46xx driver have been in a gray zone regarding the license. It's most likely should be OK, but still unclear. And, the size isn't that small, too. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=10750 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/Kconfig | 1 + sound/pci/cs46xx/cs46xx.h | 5 + sound/pci/cs46xx/cs46xx_image.h | 3468 ------------------------------------ sound/pci/cs46xx/cs46xx_lib.c | 265 ++- sound/pci/cs46xx/imgs/cwc4630.h | 320 ---- sound/pci/cs46xx/imgs/cwcasync.h | 176 -- sound/pci/cs46xx/imgs/cwcbinhack.h | 48 - sound/pci/cs46xx/imgs/cwcdma.asp | 170 -- sound/pci/cs46xx/imgs/cwcdma.h | 68 - sound/pci/cs46xx/imgs/cwcsnoop.h | 46 - 10 files changed, 226 insertions(+), 4341 deletions(-) delete mode 100644 sound/pci/cs46xx/cs46xx_image.h delete mode 100644 sound/pci/cs46xx/imgs/cwc4630.h delete mode 100644 sound/pci/cs46xx/imgs/cwcasync.h delete mode 100644 sound/pci/cs46xx/imgs/cwcbinhack.h delete mode 100644 sound/pci/cs46xx/imgs/cwcdma.asp delete mode 100644 sound/pci/cs46xx/imgs/cwcdma.h delete mode 100644 sound/pci/cs46xx/imgs/cwcsnoop.h diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 46ed9e8ae0fd..3a0504dfeb8a 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -254,6 +254,7 @@ config SND_CS46XX tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x" select SND_RAWMIDI select SND_AC97_CODEC + select FW_LOADER help Say Y here to include support for Cirrus Logic CS4610/CS4612/ CS4614/CS4615/CS4622/CS4624/CS4630/CS4280 chips. diff --git a/sound/pci/cs46xx/cs46xx.h b/sound/pci/cs46xx/cs46xx.h index fc339ef0a0ae..c49a082c378b 100644 --- a/sound/pci/cs46xx/cs46xx.h +++ b/sound/pci/cs46xx/cs46xx.h @@ -1716,9 +1716,14 @@ struct snd_cs46xx { struct snd_pcm *pcm_rear; struct snd_pcm *pcm_center_lfe; struct snd_pcm *pcm_iec958; + +#define CS46XX_DSP_MODULES 5 + struct dsp_module_desc *modules[CS46XX_DSP_MODULES]; #else /* for compatibility */ struct snd_cs46xx_pcm *playback_pcm; unsigned int play_ctl; + + struct ba1_struct *ba1; #endif #ifdef CONFIG_PM_SLEEP diff --git a/sound/pci/cs46xx/cs46xx_image.h b/sound/pci/cs46xx/cs46xx_image.h deleted file mode 100644 index dc93f62db2c2..000000000000 diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 1b66efd9b728..f18e5878f58b 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -54,7 +54,9 @@ #include <linux/gameport.h> #include <linux/mutex.h> #include <linux/export.h> - +#include <linux/module.h> +#include <linux/firmware.h> +#include <linux/vmalloc.h> #include <sound/core.h> #include <sound/control.h> @@ -330,13 +332,146 @@ int snd_cs46xx_download(struct snd_cs46xx *chip, return 0; } +static inline void memcpy_le32(void *dst, const void *src, unsigned int len) +{ +#ifdef __LITTLE_ENDIAN + memcpy(dst, src, len); +#else + u32 *_dst = dst; + const __le32 *_src = src; + len /= 4; + while (len-- > 0) + *_dst++ = le32_to_cpu(*_src++); +#endif +} + #ifdef CONFIG_SND_CS46XX_NEW_DSP -#include "imgs/cwc4630.h" -#include "imgs/cwcasync.h" -#include "imgs/cwcsnoop.h" -#include "imgs/cwcbinhack.h" -#include "imgs/cwcdma.h" +static const char *module_names[CS46XX_DSP_MODULES] = { + "cwc4630", "cwcasync", "cwcsnoop", "cwcbinhack", "cwcdma" +}; + +MODULE_FIRMWARE("cs46xx/cwc4630"); +MODULE_FIRMWARE("cs46xx/cwcasync"); +MODULE_FIRMWARE("cs46xx/cwcsnoop"); +MODULE_FIRMWARE("cs46xx/cwcbinhack"); +MODULE_FIRMWARE("cs46xx/cwcdma"); + +static void free_module_desc(struct dsp_module_desc *module) +{ + if (!module) + return; + kfree(module->module_name); + kfree(module->symbol_table.symbols); + if (module->segments) { + int i; + for (i = 0; i < module->nsegments; i++) + kfree(module->segments[i].data); + kfree(module->segments); + } +} + +/* firmware binary format: + * le32 nsymbols; + * struct { + * le32 address; + * char symbol_name[DSP_MAX_SYMBOL_NAME]; + * le32 symbol_type; + * } symbols[nsymbols]; + * le32 nsegments; + * struct { + * le32 segment_type; + * le32 offset; + * le32 size; + * le32 data[size]; + * } segments[nsegments]; + */ + +static int load_firmware(struct snd_cs46xx *chip, + struct dsp_module_desc **module_ret, + const char *fw_name) +{ + int i, err; + unsigned int nums, fwlen, fwsize; + const __le32 *fwdat; + struct dsp_module_desc *module = NULL; + const struct firmware *fw; + char fw_path[32]; + + sprintf(fw_path, "cs46xx/%s", fw_name); + err = request_firmware(&fw, fw_path, &chip->pci->dev); + if (err < 0) + return err; + fwsize = fw->size / 4; + if (fwsize < 2) { + err = -EINVAL; + goto error; + } + + err = -ENOMEM; + module = kzalloc(sizeof(*module), GFP_KERNEL); + if (!module) + goto error; + module->module_name = kstrdup(fw_name, GFP_KERNEL); + if (!module->module_name) + goto error; + + fwlen = 0; + fwdat = (const __le32 *)fw->data; + nums = module->symbol_table.nsymbols = le32_to_cpu(fwdat[fwlen++]); + if (nums >= 40) + goto error_inval; + module->symbol_table.symbols = + kcalloc(nums, sizeof(struct dsp_symbol_entry), GFP_KERNEL); + if (!module->symbol_table.symbols) + goto error; + for (i = 0; i < nums; i++) { + struct dsp_symbol_entry *entry = + &module->symbol_table.symbols[i]; + if (fwlen + 2 + DSP_MAX_SYMBOL_NAME / 4 > fwsize) + goto error_inval; + entry->address = le32_to_cpu(fwdat[fwlen++]); + memcpy(entry->symbol_name, &fwdat[fwlen], DSP_MAX_SYMBOL_NAME - 1); + fwlen += DSP_MAX_SYMBOL_NAME / 4; + entry->symbol_type = le32_to_cpu(fwdat[fwlen++]); + } + + if (fwlen >= fwsize) + goto error_inval; + nums = module->nsegments = le32_to_cpu(fwdat[fwlen++]); + if (nums > 10) + goto error_inval; + module->segments = + kcalloc(nums, sizeof(struct dsp_segment_desc), GFP_KERNEL); + if (!module->segments) + goto error; + for (i = 0; i < nums; i++) { + struct dsp_segment_desc *entry = &module->segments[i]; + if (fwlen + 3 > fwsize) + goto error_inval; + entry->segment_type = le32_to_cpu(fwdat[fwlen++]); + entry->offset = le32_to_cpu(fwdat[fwlen++]); + entry->size = le32_to_cpu(fwdat[fwlen++]); + if (fwlen + entry->size > fwsize) + goto error_inval; + entry->data = kmalloc(entry->size * 4, GFP_KERNEL); + if (!entry->data) + goto error; + memcpy_le32(entry->data, &fwdat[fwlen], entry->size * 4); + fwlen += entry->size; + } + + *module_ret = module; + release_firmware(fw); + return 0; + + error_inval: + err = -EINVAL; + error: + free_module_desc(module); + release_firmware(fw); + return err; +} int snd_cs46xx_clear_BA1(struct snd_cs46xx *chip, unsigned long offset, @@ -361,20 +496,63 @@ int snd_cs46xx_clear_BA1(struct snd_cs46xx *chip, #else /* old DSP image */ -#include "cs46xx_image.h" +struct ba1_struct { + struct { + u32 offset; + u32 size; + } memory[BA1_MEMORY_COUNT]; + u32 map[BA1_DWORD_SIZE]; +}; + +MODULE_FIRMWARE("cs46xx/ba1"); + +static int load_firmware(struct snd_cs46xx *chip) +{ + const struct firmware *fw; + int i, size, err; + + err = request_firmware(&fw, "cs46xx/ba1", &chip->pci->dev); + if (err < 0) + return err; + if (fw->size != sizeof(*chip->ba1)) { + err = -EINVAL; + goto error; + } + + chip->ba1 = vmalloc(sizeof(*chip->ba1)); + if (!chip->ba1) { + err = -ENOMEM; + goto error; + } + + memcpy_le32(chip->ba1, fw->data, sizeof(*chip->ba1)); + + /* sanity check */ + size = 0; + for (i = 0; i < BA1_MEMORY_COUNT; i++) + size += chip->ba1->memory[i].size; + if (size > BA1_DWORD_SIZE * 4) + err = -EINVAL; + + error: + release_firmware(fw); + return err; +} int snd_cs46xx_download_image(struct snd_cs46xx *chip) { int idx, err; - unsigned long offset = 0; + unsigned int offset = 0; + struct ba1_struct *ba1 = chip->ba1; for (idx = 0; idx < BA1_MEMORY_COUNT; idx++) { - if ((err = snd_cs46xx_download(chip, - &BA1Struct.map[offset], - BA1Struct.memory[idx].offset, - BA1Struct.memory[idx].size)) < 0) + err = snd_cs46xx_download(chip, + &ba1->map[offset], + ba1->memory[idx].offset, + ba1->memory[idx].size); + if (err < 0) return err; - offset += BA1Struct.memory[idx].size >> 2; + offset += ba1->memory[idx].size >> 2; } return 0; } @@ -2798,6 +2976,10 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip) cs46xx_dsp_spos_destroy(chip); chip->dsp_spos_instance = NULL; } + for (idx = 0; idx < CS46XX_DSP_MODULES; idx++) + free_module_desc(chip->modules[idx]); +#else + vfree(chip->ba1); #endif #ifdef CONFIG_PM_SLEEP @@ -3067,6 +3249,11 @@ static void cs46xx_enable_stream_irqs(struct snd_cs46xx *chip) int snd_cs46xx_start_dsp(struct snd_cs46xx *chip) { unsigned int tmp; +#ifdef CONFIG_SND_CS46XX_NEW_DSP + int i; +#endif + int err; + /* * Reset the processor. */ @@ -3075,45 +3262,33 @@ int snd_cs46xx_start_dsp(struct snd_cs46xx *chip) * Download the image to the processor. */ #ifdef CONFIG_SND_CS46XX_NEW_DSP -#if 0 - if (cs46xx_dsp_load_module(chip, &cwcemb80_module) < 0) { - snd_printk(KERN_ERR "image download error\n"); - return -EIO; - } -#endif - - if (cs46xx_dsp_load_module(chip, &cwc4630_module) < 0) { - snd_printk(KERN_ERR "image download error [cwc4630]\n"); - return -EIO; - } - - if (cs46xx_dsp_load_module(chip, &cwcasync_module) < 0) { - snd_printk(KERN_ERR "image download error [cwcasync]\n"); - return -EIO; - } - - if (cs46xx_dsp_load_module(chip, &cwcsnoop_module) < 0) { - snd_printk(KERN_ERR "image download error [cwcsnoop]\n"); - return -EIO; - } - - if (cs46xx_dsp_load_module(chip, &cwcbinhack_module) < 0) { - snd_printk(KERN_ERR "image download error [cwcbinhack]\n"); - return -EIO; - } - - if (cs46xx_dsp_load_module(chip, &cwcdma_module) < 0) { - snd_printk(KERN_ERR "image download error [cwcdma]\n"); - return -EIO; + for (i = 0; i < CS46XX_DSP_MODULES; i++) { + err = load_firmware(chip, &chip->modules[i], module_names[i]); + if (err < 0) { + snd_printk(KERN_ERR "firmware load error [%s]\n", + module_names[i]); + return err; + } + err = cs46xx_dsp_load_module(chip, chip->modules[i]); + if (err < 0) { + snd_printk(KERN_ERR "image download error [%s]\n", + module_names[i]); + return err; + } } if (cs46xx_dsp_scb_and_task_init(chip) < 0) return -EIO; #else + err = load_firmware(chip); + if (err < 0) + return err; + /* old image */ - if (snd_cs46xx_download_image(chip) < 0) { + err = snd_cs46xx_download_image(chip); + if (err < 0) { snd_printk(KERN_ERR "image download error\n"); - return -EIO; + return err; } /* diff --git a/sound/pci/cs46xx/imgs/cwc4630.h b/sound/pci/cs46xx/imgs/cwc4630.h deleted file mode 100644 index 37c4f1318dc5..000000000000 diff --git a/sound/pci/cs46xx/imgs/cwcasync.h b/sound/pci/cs46xx/imgs/cwcasync.h deleted file mode 100644 index 70e63e13c2b3..000000000000 diff --git a/sound/pci/cs46xx/imgs/cwcbinhack.h b/sound/pci/cs46xx/imgs/cwcbinhack.h deleted file mode 100644 index f4d93689cd49..000000000000 diff --git a/sound/pci/cs46xx/imgs/cwcdma.asp b/sound/pci/cs46xx/imgs/cwcdma.asp deleted file mode 100644 index a65e1193c89a..000000000000 diff --git a/sound/pci/cs46xx/imgs/cwcdma.h b/sound/pci/cs46xx/imgs/cwcdma.h deleted file mode 100644 index 7ff0d4587161..000000000000 diff --git a/sound/pci/cs46xx/imgs/cwcsnoop.h b/sound/pci/cs46xx/imgs/cwcsnoop.h deleted file mode 100644 index 6929d0a5a3f3..000000000000 -- 1.8.4.3

1 0

[alsa-devel] [PATCH 1/6] ASoC: Add resource managed snd_dmaengine_pcm_register()
by Lars-Peter Clausen 25 Nov '13

25 Nov '13

For many drivers using the generic dmaengine PCM driver one of the few (or the only) things left to do in the drivers remove function is to unregister the PCM device. This patch adds a resource managed version of snd_dmaengine_pcm_register() which makes it possible to simplify the remove function as well as the error path in the probe function for those drivers. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/dmaengine_pcm.h | 4 ++++ sound/soc/soc-generic-dmaengine-pcm.c | 36 +++++++++++++++++++++++++++++++++++ 2 files changed, 40 insertions(+) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 1501731..4ef986c 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -140,6 +140,10 @@ int snd_dmaengine_pcm_register(struct device *dev, unsigned int flags); void snd_dmaengine_pcm_unregister(struct device *dev); +int devm_snd_dmaengine_pcm_register(struct device *dev, + const struct snd_dmaengine_pcm_config *config, + unsigned int flags); + int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index cbc9c96..20690c0 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -366,4 +366,40 @@ void snd_dmaengine_pcm_unregister(struct device *dev) } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister); +static void devm_dmaengine_pcm_release(struct device *dev, void *res) +{ + snd_dmaengine_pcm_unregister(*(struct device **)res); +} + +/** + * devm_snd_dmaengine_pcm_register - resource managed dmaengine PCM registration + * @dev: The parent device for the PCM device + * @config: Platform specific PCM configuration + * @flags: Platform specific quirks + * + * Register a dmaengine based PCM device with automatic unregistration when the + * device is unregistered. + */ +int devm_snd_dmaengine_pcm_register(struct device *dev, + const struct snd_dmaengine_pcm_config *config, unsigned int flags) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_dmaengine_pcm_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_dmaengine_pcm_register(dev, config, flags); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_dmaengine_pcm_register); + MODULE_LICENSE("GPL"); -- 1.8.0

4 13

[alsa-devel] ASOC: stream widget created two times
by Arnaud POULIQUEN 25 Nov '13

25 Nov '13

Hello, I 'm newbie on asoc dev. I'm trying to develop a driver based on dynamic PCM implementation with backend and frontend. I'm working on 3.10 I have declared a FE : { .name = "MMDL0", .stream_name = "MM DL0 Frontend", .cpu_dai_name = "mm_dl0", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .dynamic=1, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .ignore_pmdown_time = 1, /* Power down without delay */ .init = sti_dummy_init, }, CPU DAI associated: static struct snd_soc_dai_driver sti_frontend_cpu_dais[] = { { .name = "mm_dl0", .playback = { .stream_name = "MM_DL0", .channels_min = 1, .channels_max = 8, .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE }, .probe = sti_frontend_dai_probe, .probe_order = SND_SOC_COMP_ORDER_FIRST, }, } Platform is declared using platform_of_node using DTS. I have also declared a BE and the route that links the FE cpu dai widget with the BE codec dai widget. My concern is that MM_DL0 widget is created 2 times. One time in soc_probe_link_dais and one time in soc_probe_platform. After analysis it seems that this occurs because my cpu dai is registered in platform device. In this case the condition if (dai->dev != platform->dev) in soc_probe_platform is false, so dai widget is created a second time. So I would like to know if this is a normal behavior and I have to create a specific device to register my cpu_dai? Thanks in advance for your answers BR Arnaud

2 1

[alsa-devel] [PATCHv5] ASoC: Add support for BCM2835
by Florian Meier 25 Nov '13

25 Nov '13

This driver adds support for digital audio (I2S) for the BCM2835 SoC that is used by the Raspberry Pi. External audio codecs can be connected to the Raspberry Pi via P5 header. It relies on cyclic DMA engine support for BCM2835. Signed-off-by: Florian Meier <florian.meier(a)koalo.de> --- - Uses devm_snd_soc_register_component now - Fixes error handling in probe .../devicetree/bindings/sound/bcm2835-i2s.txt | 25 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/bcm/Kconfig | 10 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/bcm2835-i2s.c | 886 ++++++++++++++++++++ 6 files changed, 928 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/bcm2835-i2s.txt create mode 100644 sound/soc/bcm/Kconfig create mode 100644 sound/soc/bcm/Makefile create mode 100644 sound/soc/bcm/bcm2835-i2s.c diff --git a/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt b/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt new file mode 100644 index 0000000..65783de --- /dev/null +++ b/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt @@ -0,0 +1,25 @@ +* Broadcom BCM2835 SoC I2S/PCM module + +Required properties: +- compatible: "brcm,bcm2835-i2s" +- reg: A list of base address and size entries: + * The first entry should cover the PCM registers + * The second entry should cover the PCM clock registers +- dmas: List of DMA controller phandle and DMA request line ordered pairs. +- dma-names: Identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. + + One of the DMA channels will be responsible for transmission (should be + named "tx") and one for reception (should be named "rx"). + +Example: + +bcm2835_i2s: i2s@7e203000 { + compatible = "brcm,bcm2835-i2s"; + reg = <0x7e203000 0x20>, + <0x7e101098 0x02>; + + dmas = <&dma 2>, + <&dma 3>; + dma-names = "tx", "rx"; +}; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5138b84..a5e3a70 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM # All the supported SoCs source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" +source "sound/soc/bcm/Kconfig" source "sound/soc/blackfin/Kconfig" source "sound/soc/cirrus/Kconfig" source "sound/soc/davinci/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 8b9e701..b52d4aa 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -10,6 +10,7 @@ obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ +obj-$(CONFIG_SND_SOC) += bcm/ obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += cirrus/ obj-$(CONFIG_SND_SOC) += davinci/ diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig new file mode 100644 index 0000000..3d82a29 --- /dev/null +++ b/sound/soc/bcm/Kconfig @@ -0,0 +1,10 @@ +config SND_BCM2835_SOC_I2S + tristate "SoC Audio support for the Broadcom BCM2835 I2S module" + depends on ARCH_BCM2835 || COMPILE_TEST + select SND_SOC_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + Say Y or M if you want to add support for codecs attached to + the BCM2835 I2S interface. You will also need + to select the audio interfaces to support below. diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile new file mode 100644 index 0000000..bc816b7 --- /dev/null +++ b/sound/soc/bcm/Makefile @@ -0,0 +1,5 @@ +# BCM2835 Platform Support +snd-soc-bcm2835-i2s-objs := bcm2835-i2s.o + +obj-$(CONFIG_SND_BCM2835_SOC_I2S) += snd-soc-bcm2835-i2s.o + diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c new file mode 100644 index 0000000..f49b007 --- /dev/null +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -0,0 +1,886 @@ +/* + * ALSA SoC I2S Audio Layer for Broadcom BCM2835 SoC + * + * Author: Florian Meier <florian.meier(a)koalo.de> + * Copyright 2013 + * + * Based on + * Raspberry Pi PCM I2S ALSA Driver + * Copyright (c) by Phil Poole 2013 + * + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * Vladimir Barinov, <vbarinov(a)embeddedalley.com> + * Copyright (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> + * + * OMAP ALSA SoC DAI driver using McBSP port + * Copyright (C) 2008 Nokia Corporation + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> + * Peter Ujfalusi <peter.ujfalusi(a)ti.com> + * + * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver + * Author: Timur Tabi <timur(a)freescale.com> + * Copyright 2007-2010 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * 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/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +/* Clock registers */ +#define BCM2835_CLK_PCMCTL_REG 0x00 +#define BCM2835_CLK_PCMDIV_REG 0x04 + +/* Clock register settings */ +#define BCM2835_CLK_PASSWD (0x5a000000) +#define BCM2835_CLK_PASSWD_MASK (0xff000000) +#define BCM2835_CLK_MASH(v) ((v) << 9) +#define BCM2835_CLK_FLIP BIT(8) +#define BCM2835_CLK_BUSY BIT(7) +#define BCM2835_CLK_KILL BIT(5) +#define BCM2835_CLK_ENAB BIT(4) +#define BCM2835_CLK_SRC(v) (v) + +#define BCM2835_CLK_SHIFT (12) +#define BCM2835_CLK_DIVI(v) ((v) << BCM2835_CLK_SHIFT) +#define BCM2835_CLK_DIVF(v) (v) +#define BCM2835_CLK_DIVF_MASK (0xFFF) + +enum { + BCM2835_CLK_MASH_0 = 0, + BCM2835_CLK_MASH_1, + BCM2835_CLK_MASH_2, + BCM2835_CLK_MASH_3, +}; + +enum { + BCM2835_CLK_SRC_GND = 0, + BCM2835_CLK_SRC_OSC, + BCM2835_CLK_SRC_DBG0, + BCM2835_CLK_SRC_DBG1, + BCM2835_CLK_SRC_PLLA, + BCM2835_CLK_SRC_PLLC, + BCM2835_CLK_SRC_PLLD, + BCM2835_CLK_SRC_HDMI, +}; + +/* Most clocks are not useable (freq = 0) */ +static const unsigned int bcm2835_clk_freq[BCM2835_CLK_SRC_HDMI+1] = { + [BCM2835_CLK_SRC_GND] = 0, + [BCM2835_CLK_SRC_OSC] = 19200000, + [BCM2835_CLK_SRC_DBG0] = 0, + [BCM2835_CLK_SRC_DBG1] = 0, + [BCM2835_CLK_SRC_PLLA] = 0, + [BCM2835_CLK_SRC_PLLC] = 0, + [BCM2835_CLK_SRC_PLLD] = 500000000, + [BCM2835_CLK_SRC_HDMI] = 0, +}; + +/* I2S registers */ +#define BCM2835_I2S_CS_A_REG 0x00 +#define BCM2835_I2S_FIFO_A_REG 0x04 +#define BCM2835_I2S_MODE_A_REG 0x08 +#define BCM2835_I2S_RXC_A_REG 0x0c +#define BCM2835_I2S_TXC_A_REG 0x10 +#define BCM2835_I2S_DREQ_A_REG 0x14 +#define BCM2835_I2S_INTEN_A_REG 0x18 +#define BCM2835_I2S_INTSTC_A_REG 0x1c +#define BCM2835_I2S_GRAY_REG 0x20 + +/* I2S register settings */ +#define BCM2835_I2S_STBY BIT(25) +#define BCM2835_I2S_SYNC BIT(24) +#define BCM2835_I2S_RXSEX BIT(23) +#define BCM2835_I2S_RXF BIT(22) +#define BCM2835_I2S_TXE BIT(21) +#define BCM2835_I2S_RXD BIT(20) +#define BCM2835_I2S_TXD BIT(19) +#define BCM2835_I2S_RXR BIT(18) +#define BCM2835_I2S_TXW BIT(17) +#define BCM2835_I2S_CS_RXERR BIT(16) +#define BCM2835_I2S_CS_TXERR BIT(15) +#define BCM2835_I2S_RXSYNC BIT(14) +#define BCM2835_I2S_TXSYNC BIT(13) +#define BCM2835_I2S_DMAEN BIT(9) +#define BCM2835_I2S_RXTHR(v) ((v) << 7) +#define BCM2835_I2S_TXTHR(v) ((v) << 5) +#define BCM2835_I2S_RXCLR BIT(4) +#define BCM2835_I2S_TXCLR BIT(3) +#define BCM2835_I2S_TXON BIT(2) +#define BCM2835_I2S_RXON BIT(1) +#define BCM2835_I2S_EN (1) + +#define BCM2835_I2S_CLKDIS BIT(28) +#define BCM2835_I2S_PDMN BIT(27) +#define BCM2835_I2S_PDME BIT(26) +#define BCM2835_I2S_FRXP BIT(25) +#define BCM2835_I2S_FTXP BIT(24) +#define BCM2835_I2S_CLKM BIT(23) +#define BCM2835_I2S_CLKI BIT(22) +#define BCM2835_I2S_FSM BIT(21) +#define BCM2835_I2S_FSI BIT(20) +#define BCM2835_I2S_FLEN(v) ((v) << 10) +#define BCM2835_I2S_FSLEN(v) (v) + +#define BCM2835_I2S_CHWEX BIT(15) +#define BCM2835_I2S_CHEN BIT(14) +#define BCM2835_I2S_CHPOS(v) ((v) << 4) +#define BCM2835_I2S_CHWID(v) (v) +#define BCM2835_I2S_CH1(v) ((v) << 16) +#define BCM2835_I2S_CH2(v) (v) + +#define BCM2835_I2S_TX_PANIC(v) ((v) << 24) +#define BCM2835_I2S_RX_PANIC(v) ((v) << 16) +#define BCM2835_I2S_TX(v) ((v) << 8) +#define BCM2835_I2S_RX(v) (v) + +#define BCM2835_I2S_INT_RXERR BIT(3) +#define BCM2835_I2S_INT_TXERR BIT(2) +#define BCM2835_I2S_INT_RXR BIT(1) +#define BCM2835_I2S_INT_TXW BIT(0) + +/* I2S DMA interface */ +/* FIXME: Needs IOMMU support */ +#define BCM2835_VCMMU_SHIFT (0x7E000000 - 0x20000000) + +/* General device struct */ +struct bcm2835_i2s_dev { + struct device *dev; + struct snd_dmaengine_dai_dma_data dma_data[2]; + unsigned int fmt; + unsigned int bclk_ratio; + + struct regmap *i2s_regmap; + struct regmap *clk_regmap; +}; + +static void bcm2835_i2s_start_clock(struct bcm2835_i2s_dev *dev) +{ + /* Start the clock if in master mode */ + unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB, + BCM2835_CLK_PASSWD | BCM2835_CLK_ENAB); + break; + default: + break; + } +} + +static void bcm2835_i2s_stop_clock(struct bcm2835_i2s_dev *dev) +{ + uint32_t clkreg; + int timeout = 1000; + + /* Stop clock */ + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB, + BCM2835_CLK_PASSWD); + + /* Wait for the BUSY flag going down */ + while (--timeout) { + regmap_read(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, &clkreg); + if (!(clkreg & BCM2835_CLK_BUSY)) + break; + } + + if (!timeout) { + /* KILL the clock */ + dev_err(dev->dev, "I2S clock didn't stop. Kill the clock!\n"); + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_KILL | BCM2835_CLK_PASSWD_MASK, + BCM2835_CLK_KILL | BCM2835_CLK_PASSWD); + } +} + +static void bcm2835_i2s_clear_fifos(struct bcm2835_i2s_dev *dev, + bool tx, bool rx) +{ + int timeout = 1000; + uint32_t syncval; + uint32_t csreg; + uint32_t i2s_active_state; + uint32_t clkreg; + uint32_t clk_active_state; + uint32_t off; + uint32_t clr; + + off = tx ? BCM2835_I2S_TXON : 0; + off |= rx ? BCM2835_I2S_RXON : 0; + + clr = tx ? BCM2835_I2S_TXCLR : 0; + clr |= rx ? BCM2835_I2S_RXCLR : 0; + + /* Backup the current state */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg); + i2s_active_state = csreg & (BCM2835_I2S_RXON | BCM2835_I2S_TXON); + + regmap_read(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, &clkreg); + clk_active_state = clkreg & BCM2835_CLK_ENAB; + + /* Start clock if not running */ + if (!clk_active_state) { + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB, + BCM2835_CLK_PASSWD | BCM2835_CLK_ENAB); + } + + /* Stop I2S module */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, off, 0); + + /* + * Clear the FIFOs + * Requires at least 2 PCM clock cycles to take effect + */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, clr, clr); + + /* Wait for 2 PCM clock cycles */ + + /* + * Toggle the SYNC flag. After 2 PCM clock cycles it can be read back + * FIXME: This does not seem to work for slave mode! + */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &syncval); + syncval &= BCM2835_I2S_SYNC; + + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_SYNC, ~syncval); + + /* Wait for the SYNC flag changing it's state */ + while (--timeout) { + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg); + if ((csreg & BCM2835_I2S_SYNC) != syncval) + break; + } + + if (!timeout) + dev_err(dev->dev, "I2S SYNC error!\n"); + + /* Stop clock if it was not running before */ + if (!clk_active_state) + bcm2835_i2s_stop_clock(dev); + + /* Restore I2S state */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_RXON | BCM2835_I2S_TXON, i2s_active_state); +} + +static int bcm2835_i2s_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + dev->fmt = fmt; + return 0; +} + +static int bcm2835_i2s_set_dai_bclk_ratio(struct snd_soc_dai *dai, + unsigned int ratio) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + dev->bclk_ratio = ratio; + return 0; +} + +static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + unsigned int sampling_rate = params_rate(params); + unsigned int data_length, data_delay, bclk_ratio; + unsigned int ch1pos, ch2pos, mode, format; + unsigned int mash = BCM2835_CLK_MASH_1; + unsigned int divi, divf, target_frequency; + int clk_src = -1; + unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + bool bit_master = (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBS_CFM); + + bool frame_master = (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBM_CFS); + uint32_t csreg; + + /* + * If a stream is already enabled, + * the registers are already set properly. + */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg); + + if (csreg & (BCM2835_I2S_TXON | BCM2835_I2S_RXON)) + return 0; + + /* + * Adjust the data length according to the format. + * We prefill the half frame length with an integer + * divider of 2400 as explained at the clock settings. + * Maybe it is overwritten there, if the Integer mode + * does not apply. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + data_length = 16; + bclk_ratio = 40; + break; + case SNDRV_PCM_FORMAT_S32_LE: + data_length = 32; + bclk_ratio = 80; + break; + default: + return -EINVAL; + } + + /* If bclk_ratio already set, use that one. */ + if (dev->bclk_ratio) + bclk_ratio = dev->bclk_ratio; + + /* + * Clock Settings + * + * The target frequency of the bit clock is + * sampling rate * frame length + * + * Integer mode: + * Sampling rates that are multiples of 8000 kHz + * can be driven by the oscillator of 19.2 MHz + * with an integer divider as long as the frame length + * is an integer divider of 19200000/8000=2400 as set up above. + * This is no longer possible if the sampling rate + * is too high (e.g. 192 kHz), because the oscillator is too slow. + * + * MASH mode: + * For all other sampling rates, it is not possible to + * have an integer divider. Approximate the clock + * with the MASH module that induces a slight frequency + * variance. To minimize that it is best to have the fastest + * clock here. That is PLLD with 500 MHz. + */ + target_frequency = sampling_rate * bclk_ratio; + clk_src = BCM2835_CLK_SRC_OSC; + mash = BCM2835_CLK_MASH_0; + + if (bcm2835_clk_freq[clk_src] % target_frequency == 0 + && bit_master && frame_master) { + divi = bcm2835_clk_freq[clk_src] / target_frequency; + divf = 0; + } else { + uint64_t dividend; + + if (!dev->bclk_ratio) { + /* + * Overwrite bclk_ratio, because the + * above trick is not needed or can + * not be used. + */ + bclk_ratio = 2 * data_length; + } + + target_frequency = sampling_rate * bclk_ratio; + + clk_src = BCM2835_CLK_SRC_PLLD; + mash = BCM2835_CLK_MASH_1; + + dividend = bcm2835_clk_freq[clk_src]; + dividend <<= BCM2835_CLK_SHIFT; + do_div(dividend, target_frequency); + divi = dividend >> BCM2835_CLK_SHIFT; + divf = dividend & BCM2835_CLK_DIVF_MASK; + } + + /* Set clock divider */ + regmap_write(dev->clk_regmap, BCM2835_CLK_PCMDIV_REG, BCM2835_CLK_PASSWD + | BCM2835_CLK_DIVI(divi) + | BCM2835_CLK_DIVF(divf)); + + /* Setup clock, but don't start it yet */ + regmap_write(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, BCM2835_CLK_PASSWD + | BCM2835_CLK_MASH(mash) + | BCM2835_CLK_SRC(clk_src)); + + /* Setup the frame format */ + format = BCM2835_I2S_CHEN; + + if (data_length > 24) + format |= BCM2835_I2S_CHWEX; + + format |= BCM2835_I2S_CHWID((data_length-8)&0xf); + + switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + data_delay = 1; + break; + default: + /* + * TODO + * Others are possible but are not implemented at the moment. + */ + dev_err(dev->dev, "%s:bad format\n", __func__); + return -EINVAL; + } + + ch1pos = data_delay; + ch2pos = bclk_ratio / 2 + data_delay; + + switch (params_channels(params)) { + case 2: + format = BCM2835_I2S_CH1(format) | BCM2835_I2S_CH2(format); + format |= BCM2835_I2S_CH1(BCM2835_I2S_CHPOS(ch1pos)); + format |= BCM2835_I2S_CH2(BCM2835_I2S_CHPOS(ch2pos)); + break; + default: + return -EINVAL; + } + + /* + * Set format for both streams. + * We cannot set another frame length + * (and therefore word length) anyway, + * so the format will be the same. + */ + regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, format); + regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, format); + + /* Setup the I2S mode */ + mode = 0; + + if (data_length <= 16) { + /* + * Use frame packed mode (2 channels per 32 bit word) + * We cannot set another frame length in the second stream + * (and therefore word length) anyway, + * so the format will be the same. + */ + mode |= BCM2835_I2S_FTXP | BCM2835_I2S_FRXP; + } + + mode |= BCM2835_I2S_FLEN(bclk_ratio - 1); + mode |= BCM2835_I2S_FSLEN(bclk_ratio / 2); + + /* Master or slave? */ + switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* CPU is master */ + break; + case SND_SOC_DAIFMT_CBM_CFS: + /* + * CODEC is bit clock master + * CPU is frame master + */ + mode |= BCM2835_I2S_CLKM; + break; + case SND_SOC_DAIFMT_CBS_CFM: + /* + * CODEC is frame master + * CPU is bit clock master + */ + mode |= BCM2835_I2S_FSM; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* CODEC is master */ + mode |= BCM2835_I2S_CLKM; + mode |= BCM2835_I2S_FSM; + break; + default: + dev_err(dev->dev, "%s:bad master\n", __func__); + return -EINVAL; + } + + /* + * Invert clocks? + * + * The BCM approach seems to be inverted to the classical I2S approach. + */ + switch (dev->fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* None. Therefore, both for BCM */ + mode |= BCM2835_I2S_CLKI; + mode |= BCM2835_I2S_FSI; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Both. Therefore, none for BCM */ + break; + case SND_SOC_DAIFMT_NB_IF: + /* + * Invert only frame sync. Therefore, + * invert only bit clock for BCM + */ + mode |= BCM2835_I2S_CLKI; + break; + case SND_SOC_DAIFMT_IB_NF: + /* + * Invert only bit clock. Therefore, + * invert only frame sync for BCM + */ + mode |= BCM2835_I2S_FSI; + break; + default: + return -EINVAL; + } + + regmap_write(dev->i2s_regmap, BCM2835_I2S_MODE_A_REG, mode); + + /* Setup the DMA parameters */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_RXTHR(1) + | BCM2835_I2S_TXTHR(1) + | BCM2835_I2S_DMAEN, 0xffffffff); + + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_DREQ_A_REG, + BCM2835_I2S_TX_PANIC(0x10) + | BCM2835_I2S_RX_PANIC(0x30) + | BCM2835_I2S_TX(0x30) + | BCM2835_I2S_RX(0x20), 0xffffffff); + + /* Clear FIFOs */ + bcm2835_i2s_clear_fifos(dev, true, true); + + return 0; +} + +static int bcm2835_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + uint32_t cs_reg; + + bcm2835_i2s_start_clock(dev); + + /* + * Clear both FIFOs if the one that should be started + * is not empty at the moment. This should only happen + * after overrun. Otherwise, hw_params would have cleared + * the FIFO. + */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &cs_reg); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK + && !(cs_reg & BCM2835_I2S_TXE)) + bcm2835_i2s_clear_fifos(dev, true, false); + else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE + && (cs_reg & BCM2835_I2S_RXD)) + bcm2835_i2s_clear_fifos(dev, false, true); + + return 0; +} + +static void bcm2835_i2s_stop(struct bcm2835_i2s_dev *dev, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + uint32_t mask; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + mask = BCM2835_I2S_RXON; + else + mask = BCM2835_I2S_TXON; + + regmap_update_bits(dev->i2s_regmap, + BCM2835_I2S_CS_A_REG, mask, 0); + + /* Stop also the clock when not SND_SOC_DAIFMT_CONT */ + if (!dai->active && !(dev->fmt & SND_SOC_DAIFMT_CONT)) + bcm2835_i2s_stop_clock(dev); +} + +static int bcm2835_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + uint32_t mask; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + bcm2835_i2s_start_clock(dev); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + mask = BCM2835_I2S_RXON; + else + mask = BCM2835_I2S_TXON; + + regmap_update_bits(dev->i2s_regmap, + BCM2835_I2S_CS_A_REG, mask, mask); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + bcm2835_i2s_stop(dev, substream, dai); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int bcm2835_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + if (dai->active) + return 0; + + /* Should this still be running stop it */ + bcm2835_i2s_stop_clock(dev); + + /* Enable PCM block */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_EN, BCM2835_I2S_EN); + + /* + * Disable STBY. + * Requires at least 4 PCM clock cycles to take effect. + */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_STBY, BCM2835_I2S_STBY); + + return 0; +} + +static void bcm2835_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + bcm2835_i2s_stop(dev, substream, dai); + + /* If both streams are stopped, disable module and clock */ + if (dai->active) + return; + + /* Disable the module */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_EN, 0); + + /* + * Stopping clock is necessary, because stop does + * not stop the clock when SND_SOC_DAIFMT_CONT + */ + bcm2835_i2s_stop_clock(dev); +} + +static const struct snd_soc_dai_ops bcm2835_i2s_dai_ops = { + .startup = bcm2835_i2s_startup, + .shutdown = bcm2835_i2s_shutdown, + .prepare = bcm2835_i2s_prepare, + .trigger = bcm2835_i2s_trigger, + .hw_params = bcm2835_i2s_hw_params, + .set_fmt = bcm2835_i2s_set_dai_fmt, + .set_bclk_ratio = bcm2835_i2s_set_dai_bclk_ratio +}; + +static int bcm2835_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, + &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK], + &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]); + + return 0; +} + +static struct snd_soc_dai_driver bcm2835_i2s_dai = { + .name = "bcm2835-i2s", + .probe = bcm2835_i2s_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S32_LE + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S32_LE + }, + .ops = &bcm2835_i2s_dai_ops, + .symmetric_rates = 1 +}; + +static bool bcm2835_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case BCM2835_I2S_CS_A_REG: + case BCM2835_I2S_FIFO_A_REG: + case BCM2835_I2S_INTSTC_A_REG: + case BCM2835_I2S_GRAY_REG: + return true; + default: + return false; + }; +} + +static bool bcm2835_i2s_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case BCM2835_I2S_FIFO_A_REG: + return true; + default: + return false; + }; +} + +static bool bcm2835_clk_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case BCM2835_CLK_PCMCTL_REG: + return true; + default: + return false; + }; +} + +static const struct regmap_config bcm2835_regmap_config[] = { + { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = BCM2835_I2S_GRAY_REG, + .precious_reg = bcm2835_i2s_precious_reg, + .volatile_reg = bcm2835_i2s_volatile_reg, + .cache_type = REGCACHE_RBTREE, + }, + { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = BCM2835_CLK_PCMDIV_REG, + .volatile_reg = bcm2835_clk_volatile_reg, + .cache_type = REGCACHE_RBTREE, + }, +}; + +static const struct snd_soc_component_driver bcm2835_i2s_component = { + .name = "bcm2835-i2s-comp", +}; + +static int bcm2835_i2s_probe(struct platform_device *pdev) +{ + struct bcm2835_i2s_dev *dev; + int i; + int ret; + struct regmap *regmap[2]; + struct resource *mem[2]; + + /* Request both ioareas */ + for (i = 0; i <= 1; i++) { + void __iomem *base; + + mem[i] = platform_get_resource(pdev, IORESOURCE_MEM, i); + base = devm_ioremap_resource(&pdev->dev, mem[i]); + if (IS_ERR(base)) + return PTR_ERR(base); + + regmap[i] = devm_regmap_init_mmio(&pdev->dev, base, + &bcm2835_regmap_config[i]); + if (IS_ERR(regmap[i])) + return PTR_ERR(regmap[i]); + } + + dev = devm_kzalloc(&pdev->dev, sizeof(*dev), + GFP_KERNEL); + if (!dev) + return -ENOMEM; + + dev->i2s_regmap = regmap[0]; + dev->clk_regmap = regmap[1]; + + /* Set the DMA address */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = + (dma_addr_t)mem[0]->start + BCM2835_I2S_FIFO_A_REG + + BCM2835_VCMMU_SHIFT; + + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = + (dma_addr_t)mem[0]->start + BCM2835_I2S_FIFO_A_REG + + BCM2835_VCMMU_SHIFT; + + /* Set the bus width */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr_width = + DMA_SLAVE_BUSWIDTH_4_BYTES; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr_width = + DMA_SLAVE_BUSWIDTH_4_BYTES; + + /* Set burst */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].maxburst = 2; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].maxburst = 2; + + /* BCLK ratio - use default */ + dev->bclk_ratio = 0; + + /* Store the pdev */ + dev->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, dev); + + ret = devm_snd_soc_register_component(&pdev->dev, + &bcm2835_i2s_component, &bcm2835_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + return ret; + } + + ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct of_device_id bcm2835_i2s_of_match[] = { + { .compatible = "brcm,bcm2835-i2s", }, + {}, +}; + +static int bcm2835_i2s_remove(struct platform_device *pdev) +{ + snd_dmaengine_pcm_unregister(&pdev->dev); + return 0; +} + +static struct platform_driver bcm2835_i2s_driver = { + .probe = bcm2835_i2s_probe, + .remove = bcm2835_i2s_remove, + .driver = { + .name = "bcm2835-i2s", + .owner = THIS_MODULE, + .of_match_table = bcm2835_i2s_of_match, + }, +}; + +module_platform_driver(bcm2835_i2s_driver); + +MODULE_ALIAS("platform:bcm2835-i2s"); +MODULE_DESCRIPTION("BCM2835 I2S interface"); +MODULE_AUTHOR("Florian Meier <florian.meier(a)koalo.de>"); +MODULE_LICENSE("GPL v2"); -- 1.7.9.5

2 1

[alsa-devel] ASoC fixes for v3.13
by Mark Brown 25 Nov '13

25 Nov '13

The following changes since commit 3d800c6d75b8c92fa928a0bcaf95cd7ac5fd1ce5: ASoC: cs42l52: Correct MIC CTL mask (2013-11-15 10:16:53 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-5 for you to fetch changes up to 9a5c543dd71952719d40429b6ef252a46c9b2246: Merge remote-tracking branch 'asoc/fix/wm8962' into asoc-linus (2013-11-20 15:20:38 +0000) Not sure why the diffstat is still going crazy here... :/ ---------------------------------------------------------------- ASoC: Fixes for v3.13 A bunch of device specific fixes, nothing with a general impact here. ---------------------------------------------------------------- Charles Keepax (1): ASoC: wm5110: Add post SYSCLK register patch for rev D chip Dan Carpenter (1): ASoC: rcar: off by one in rsnd_scu_set_route() Kuninori Morimoto (2): ASoC: rcar: fixup mod access before checking ASoC: rcar: fixup dma_async_issue_pending() timing Lee Jones (1): ASoC: ab8500: Revert to using custom I/O functions Mark Brown (8): Merge remote-tracking branch 'asoc/fix/ab8500' into asoc-linus Merge remote-tracking branch 'asoc/fix/arizona' into asoc-linus Merge remote-tracking branch 'asoc/fix/blackfin' into asoc-linus Merge remote-tracking branch 'asoc/fix/cs42l52' into asoc-linus Merge remote-tracking branch 'asoc/fix/dma' into asoc-linus Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8962' into asoc-linus Richard Fitzgerald (1): ASoC: arizona: Set FLL to free-run before disabling .../devicetree/bindings/sound/cs42l73.txt | 22 + .../bindings/sound/davinci-evm-audio.txt | 42 ++ .../bindings/sound/davinci-mcasp-audio.txt | 41 +- .../devicetree/bindings/sound/tlv320aic3x.txt | 26 + .../devicetree/bindings/sound/tpa6130a2.txt | 27 + Documentation/sound/alsa/soc/DPCM.txt | 380 +++++++++++++ Documentation/sound/alsa/soc/codec.txt | 46 +- Documentation/sound/alsa/soc/dapm.txt | 71 ++- Documentation/sound/alsa/soc/machine.txt | 6 +- Documentation/sound/alsa/soc/platform.txt | 19 +- drivers/base/regmap/internal.h | 8 +- drivers/base/regmap/regcache.c | 3 +- drivers/base/regmap/regmap.c | 294 ++++++++-- drivers/mfd/mc13xxx-core.c | 5 - drivers/mfd/mc13xxx-spi.c | 5 + include/linux/mfd/mc13xxx.h | 7 + include/linux/platform_data/davinci_asp.h | 2 + include/linux/regmap.h | 44 ++ include/sound/cs42l52.h | 2 + include/sound/cs42l73.h | 22 + include/sound/dmaengine_pcm.h | 8 + include/sound/rcar_snd.h | 1 - include/sound/soc-dai.h | 17 + include/sound/soc-dapm.h | 4 + include/sound/soc.h | 100 ++-- include/trace/events/asoc.h | 1 + sound/arm/pxa2xx-ac97-lib.c | 27 +- sound/soc/Makefile | 2 +- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 8 - sound/soc/atmel/sam9g20_wm8731.c | 1 + sound/soc/blackfin/bf5xx-i2s.c | 1 + sound/soc/cirrus/Kconfig | 2 +- sound/soc/cirrus/ep93xx-pcm.c | 13 + sound/soc/codecs/88pm860x-codec.c | 75 +-- sound/soc/codecs/88pm860x-codec.h | 117 ++-- sound/soc/codecs/ab8500-codec.c | 26 +- sound/soc/codecs/adau1373.c | 298 ++++++++-- sound/soc/codecs/adav80x.c | 147 +++-- sound/soc/codecs/ak4104.c | 11 +- sound/soc/codecs/ak4642.c | 4 +- sound/soc/codecs/arizona.c | 27 +- sound/soc/codecs/cq93vc.c | 46 +- sound/soc/codecs/cs4271.c | 1 + sound/soc/codecs/cs42l52.c | 93 +-- sound/soc/codecs/cs42l52.h | 2 +- sound/soc/codecs/cs42l73.c | 114 ++-- sound/soc/codecs/cs42l73.h | 105 ++-- sound/soc/codecs/max98088.c | 624 +++++++++------------ sound/soc/codecs/max98095.c | 466 ++++++--------- sound/soc/codecs/max9850.c | 39 +- sound/soc/codecs/mc13783.c | 135 +++-- sound/soc/codecs/ml26124.c | 2 + sound/soc/codecs/pcm1681.c | 1 + sound/soc/codecs/pcm1792a.c | 1 + sound/soc/codecs/rt5640.c | 31 +- sound/soc/codecs/si476x.c | 64 +-- sound/soc/codecs/sn95031.c | 35 +- sound/soc/codecs/tas5086.c | 171 ++++-- sound/soc/codecs/tlv320aic23.c | 84 ++- sound/soc/codecs/tlv320aic26.c | 139 +---- sound/soc/codecs/tlv320aic26.h | 5 +- sound/soc/codecs/tlv320aic32x4.c | 101 +--- sound/soc/codecs/tlv320aic3x.c | 234 ++++---- sound/soc/codecs/tpa6130a2.c | 32 +- sound/soc/codecs/twl4030.c | 80 ++- sound/soc/codecs/twl6040.c | 26 +- sound/soc/codecs/wm0010.c | 8 +- sound/soc/codecs/wm5110.c | 55 +- sound/soc/codecs/wm8400.c | 95 +--- sound/soc/codecs/wm8962.c | 228 ++++---- sound/soc/codecs/wm8996.c | 2 + sound/soc/codecs/wm_adsp.c | 27 +- sound/soc/davinci/Kconfig | 18 +- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 188 ++++++- sound/soc/davinci/davinci-mcasp.c | 169 ++++-- sound/soc/davinci/davinci-mcasp.h | 12 + sound/soc/fsl/eukrea-tlv320.c | 15 +- sound/soc/fsl/fsl_spdif.c | 22 +- sound/soc/fsl/fsl_ssi.c | 22 +- sound/soc/fsl/imx-audmux.c | 9 +- sound/soc/fsl/imx-mc13783.c | 1 + sound/soc/fsl/imx-pcm-dma.c | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 29 +- sound/soc/fsl/imx-sgtl5000.c | 4 +- sound/soc/fsl/imx-spdif.c | 4 +- sound/soc/fsl/imx-ssi.c | 3 - sound/soc/fsl/imx-wm8962.c | 4 +- sound/soc/generic/simple-card.c | 5 + sound/soc/kirkwood/kirkwood-dma.c | 6 +- sound/soc/kirkwood/kirkwood-i2s.c | 108 +++- sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- sound/soc/kirkwood/kirkwood.h | 4 +- sound/soc/mid-x86/mfld_machine.c | 10 +- sound/soc/mxs/mxs-saif.c | 42 +- sound/soc/mxs/mxs-saif.h | 5 + sound/soc/mxs/mxs-sgtl5000.c | 20 +- sound/soc/omap/omap-mcpdm.c | 12 +- sound/soc/omap/omap-twl4030.c | 5 +- sound/soc/pxa/brownstone.c | 1 + sound/soc/pxa/corgi.c | 1 + sound/soc/pxa/e740_wm9705.c | 1 + sound/soc/pxa/e750_wm9705.c | 1 + sound/soc/pxa/e800_wm9712.c | 1 + sound/soc/pxa/imote2.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 1 + sound/soc/pxa/mmp-sspa.c | 5 +- sound/soc/pxa/palm27x.c | 1 + sound/soc/pxa/poodle.c | 1 + sound/soc/pxa/pxa2xx-ac97.c | 56 +- sound/soc/pxa/tosa.c | 1 + sound/soc/pxa/ttc-dkb.c | 1 + sound/soc/samsung/bells.c | 1 + sound/soc/samsung/i2s.c | 25 +- sound/soc/samsung/smdk_wm8994.c | 14 +- sound/soc/sh/rcar/adg.c | 11 +- sound/soc/sh/rcar/core.c | 94 +--- sound/soc/sh/rcar/gen.c | 261 +++++---- sound/soc/sh/rcar/rsnd.h | 5 +- sound/soc/sh/rcar/scu.c | 14 +- sound/soc/sh/rcar/ssi.c | 52 +- sound/soc/soc-cache.c | 263 ++------- sound/soc/soc-core.c | 343 ++++++----- sound/soc/soc-dapm.c | 120 ++-- sound/soc/soc-devres.c | 86 +++ sound/soc/soc-generic-dmaengine-pcm.c | 105 +++- sound/soc/soc-io.c | 26 - sound/soc/soc-jack.c | 7 +- sound/soc/soc-pcm.c | 60 +- sound/soc/soc-utils.c | 6 +- sound/soc/spear/spdif_in.c | 12 +- sound/soc/spear/spdif_out.c | 14 +- sound/soc/tegra/tegra30_ahub.c | 115 +++- sound/soc/tegra/tegra30_ahub.h | 38 +- sound/soc/tegra/tegra30_i2s.c | 51 +- sound/soc/tegra/tegra30_i2s.h | 7 + sound/soc/tegra/tegra_asoc_utils.c | 2 + sound/soc/tegra/tegra_asoc_utils.h | 1 + sound/soc/tegra/tegra_pcm.c | 1 - 141 files changed, 4424 insertions(+), 3118 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs42l73.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tpa6130a2.txt create mode 100644 Documentation/sound/alsa/soc/DPCM.txt create mode 100644 include/sound/cs42l73.h create mode 100644 sound/soc/soc-devres.c

2 4

[alsa-devel] [PATCH RESEND] ASoC: clk: probe defer when clock not yet ready
by Jean-Francois Moine 25 Nov '13

25 Nov '13

At probe time, a clock device may not be ready when some other device wants to use it. This patch lets the functions clk_get/devm_clk_get return a probe defer when the clock is defined in the DT but not yet available. It also fixes an erroneous call to clk_get_sys() when __clk_get() fails. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- resend: - patch subject change from [PATCH v3 1/2] ASoC: kirkwood: clk: probe defer when clock not yet ready - base kernel 3.13.0-rc1 v2: fix __clk_get() failure (from Russell King) --- drivers/clk/clk.c | 2 +- drivers/clk/clkdev.c | 7 ++++++- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/drivers/clk/clk.c b/drivers/clk/clk.c index 2cf2ea6..41361c6 100644 --- a/drivers/clk/clk.c +++ b/drivers/clk/clk.c @@ -2187,7 +2187,7 @@ EXPORT_SYMBOL_GPL(of_clk_del_provider); struct clk *of_clk_get_from_provider(struct of_phandle_args *clkspec) { struct of_clk_provider *provider; - struct clk *clk = ERR_PTR(-ENOENT); + struct clk *clk = ERR_PTR(-EPROBE_DEFER); /* Check if we have such a provider in our array */ mutex_lock(&of_clk_lock); diff --git a/drivers/clk/clkdev.c b/drivers/clk/clkdev.c index 442a313..1f402b9 100644 --- a/drivers/clk/clkdev.c +++ b/drivers/clk/clkdev.c @@ -157,7 +157,12 @@ struct clk *clk_get(struct device *dev, const char *con_id) if (dev) { clk = of_clk_get_by_name(dev->of_node, con_id); - if (!IS_ERR(clk) && __clk_get(clk)) + if (!IS_ERR(clk)) { + if (!__clk_get(clk)) + clk = ERR_PTR(-ENOENT); + return clk; + } + if (PTR_ERR(clk) == -EPROBE_DEFER) return clk; } -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH 1/5] ASoC: wm8990: Mark the register map as dirty when powering down
by Mark Brown 25 Nov '13

25 Nov '13

From: Mark Brown <broonie(a)linaro.org> Otherwise we'll skip sync on resume. Signed-off-by: Mark Brown <broonie(a)linaro.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/wm8990.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 253c88bb7a4c..4f05fb88bddf 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1259,6 +1259,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ snd_soc_write(codec, WM8990_ANTIPOP2, 0x0); + + codec->cache_sync = 1; break; } -- 1.8.4.4

2 5

[alsa-devel] [PATCH 0/8] ASoC: fsl-ssi: offline/online configuration and cleanups
by Markus Pargmann 25 Nov '13

25 Nov '13

Hi, This patch series fixes some fsl-ssi code that does not act exactly as it is described in the reference manuals. The reference manuals before imx50 do mention that some register bits, including TDMAE/RDMAE, should not be changed while the SSI unit is enabled (SSIEN). At the same time the SDMA unit has a undefined request handling if there are DMA requests before the SDMA engine and its channel is configured. The reference manual states that SSI DMA requests are sent as soon as the FIFO and the DMAE bits are enabled. It does not mention a dependency to SCR's TE/RE bit. To avoid sending DMA requests before a channel is configured, we have to seperate the fsl-ssi behavior into offline and online configuration. SoCs before imx50 have to use offline configuration while later SoC versions can use online reconfiguration. This series adds support for online configuration and cleans up the configuration register code especially in fsl_ssi_trigger. At the end we have a seperation between enable/disable logic and the actual configuration register values used to enable/disable TX/RX. I tested this series on mx53. The series is based on Mark Brown's topic/fsl branch. Regards, Markus Changes in v1: - Not removing imx21-ssi from the compatible array in the DTS patch. - Dropped "ASoC: fsl-ssi: Drop AC97 debug register usage" for the moment as I don't have time to setup AC97 hardware for some tests Markus Pargmann (8): ASoC: fsl-ssi: Move ac97 specific setup to seperate function ASoC: fsl-ssi: Move sysfs stats to debugfs ASoC: fsl-ssi: Add imx50-ssi and of_device_id matching ASoC: fsl-ssi: Add offline_config flag ASoC: fsl-ssi: Add configuration helper functions ASoC: fsl-ssi: Move RX/TX configuration to seperate functions ASoC: fsl-ssi: Drop ac97 specific trigger function ARM: DTS: imx5* imx6*, use imx50-ssi arch/arm/boot/dts/imx51.dtsi | 10 +- arch/arm/boot/dts/imx53.dtsi | 10 +- arch/arm/boot/dts/imx6qdl.dtsi | 12 +- arch/arm/boot/dts/imx6sl.dtsi | 12 +- sound/soc/fsl/fsl_ssi.c | 588 +++++++++++++++++++++++++++-------------- 5 files changed, 417 insertions(+), 215 deletions(-) -- 1.8.4.2

3 15

Re: [alsa-devel] Commit:ce6cfaf1 query for shared control
by Lars-Peter Clausen 25 Nov '13

25 Nov '13

On 11/25/2013 08:12 AM, noman pouigt wrote: > Hello Lars, > > I was wondering if you can enlighten me > on below query. > > Commit:ce6cfaf1 > > From the commit text "The input and > output paths for each widgets are only > updated though during the respective run > for that widget". > > Does this commit text mean: when mux > or mixer_power_update is called then only > the widget is powered and it's peer- widget > source(path source) and widget sink(path sink)? Hi, What the commit does is to make sure that if a control is shared between multiple mixers/muxes to apply all the changes in the same update sequence. Instead of, as it was done before, to run a update sequence for each mixer/mux. > > +------+ > A1 ------| | > | MUX1 |----- C1 > B1 ------| | > +------+ > | > control ---+ > | > +------+ > A2 ------| | > | MUX2 |----- C2 > B2 ------| | > +------+ > > Can I represent the above diagram as below: > > char *input_text_1[] = { > "A1", "B1" > }; > > struct soc_enum input_enum_1 = > SOC_ENUM_SINGLE(some_register, some_bit, > 2, input_text_1); > > char *input_text_2[] = { > "A2", "B2" > }; > struct soc_enum input_enum_2 = > SOC_ENUM_SINGLE(some_register, some_bit, > 2, input_text_2); > > > struct snd_kcontrol_new input_mux = > SOC_DAPM_ENUM("C1", input_enum_1); > struct snd_kcontrol_new input_mux = > SOC_DAPM_ENUM("C2", input_enum_2); > > dapm_route[] { > "A1", "control","C1" > "B1", "control","C1" > "A2", "control","C2" > "B2", "control","C2" > }; The control part needs to have the name of one of the enum items. So this would be { "A1", "A1", "C1" } and so on. - Lars

1 0

[alsa-devel] [PATCH] ASoC: dapm: Use SND_SOC_DAPM_INIT_REG_VAL in SND_SOC_DAPM_MUX
by Stephen Warren 24 Nov '13

24 Nov '13

From: Stephen Warren <swarren(a)nvidia.com> SND_SOC_DAPM_MUX() doesn't currently initialize the .mask field. This results in the mux never affecting HW, since no bits are ever set or cleared. Fix SND_SOC_DAPM_MUX() to use SND_SOC_DAPM_INIT_REG_VAL() to set up the reg, shift, on_val, and off_val fields like almost all other SND_SOC_xxx() macros. It looks like this was a "typo" in the fixed commit linked below. This makes the speakers on the Toshiba AC100 (PAZ00) laptop work again. Fixes: de9ba98b6d26 ("ASoC: dapm: Make widget power register settings more flexible") Cc: <stable(a)vger.kernel.org> # v3.12+ Cc: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- include/sound/soc-dapm.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 2037c45adfe6..56ebdfca6273 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -104,7 +104,8 @@ struct device; SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .kcontrol_news = wcontrols, .num_kcontrols = 1} #define SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols) \ -{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, \ +{ .id = snd_soc_dapm_mux, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .kcontrol_news = wcontrols, .num_kcontrols = 1} #define SND_SOC_DAPM_VIRT_MUX(wname, wreg, wshift, winvert, wcontrols) \ { .id = snd_soc_dapm_virt_mux, .name = wname, \ -- 1.8.1.5

2 4

[alsa-devel] [PATCH] ASoC: Rename mid-x86 directory to intel
by Jarkko Nikula 24 Nov '13

24 Nov '13

We have other Intel platforms coming having the Smart Sound Technology (SST) so rename the mid-x86 directory to intel as originally directory name reflected only Intel MID platform. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/Kconfig | 2 +- sound/soc/Makefile | 2 +- sound/soc/{mid-x86 => intel}/Kconfig | 0 sound/soc/{mid-x86 => intel}/Makefile | 0 sound/soc/{mid-x86 => intel}/mfld_machine.c | 0 sound/soc/{mid-x86 => intel}/sst_dsp.h | 0 sound/soc/{mid-x86 => intel}/sst_platform.c | 0 sound/soc/{mid-x86 => intel}/sst_platform.h | 0 8 files changed, 2 insertions(+), 2 deletions(-) rename sound/soc/{mid-x86 => intel}/Kconfig (100%) rename sound/soc/{mid-x86 => intel}/Makefile (100%) rename sound/soc/{mid-x86 => intel}/mfld_machine.c (100%) rename sound/soc/{mid-x86 => intel}/sst_dsp.h (100%) rename sound/soc/{mid-x86 => intel}/sst_platform.c (100%) rename sound/soc/{mid-x86 => intel}/sst_platform.h (100%) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5138b8493051..463a9e25e04f 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -42,7 +42,7 @@ source "sound/soc/jz4740/Kconfig" source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/kirkwood/Kconfig" -source "sound/soc/mid-x86/Kconfig" +source "sound/soc/intel/Kconfig" source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 8b9e70105dd2..ff291d3e60af 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -16,7 +16,7 @@ obj-$(CONFIG_SND_SOC) += davinci/ obj-$(CONFIG_SND_SOC) += dwc/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += jz4740/ -obj-$(CONFIG_SND_SOC) += mid-x86/ +obj-$(CONFIG_SND_SOC) += intel/ obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/intel/Kconfig similarity index 100% rename from sound/soc/mid-x86/Kconfig rename to sound/soc/intel/Kconfig diff --git a/sound/soc/mid-x86/Makefile b/sound/soc/intel/Makefile similarity index 100% rename from sound/soc/mid-x86/Makefile rename to sound/soc/intel/Makefile diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/intel/mfld_machine.c similarity index 100% rename from sound/soc/mid-x86/mfld_machine.c rename to sound/soc/intel/mfld_machine.c diff --git a/sound/soc/mid-x86/sst_dsp.h b/sound/soc/intel/sst_dsp.h similarity index 100% rename from sound/soc/mid-x86/sst_dsp.h rename to sound/soc/intel/sst_dsp.h diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/intel/sst_platform.c similarity index 100% rename from sound/soc/mid-x86/sst_platform.c rename to sound/soc/intel/sst_platform.c diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/intel/sst_platform.h similarity index 100% rename from sound/soc/mid-x86/sst_platform.h rename to sound/soc/intel/sst_platform.h -- 1.8.4.2

2 2

[alsa-devel] [patch] ASoC: rcar: some dubious one-bit signed bitfields
by Dan Carpenter 24 Nov '13

24 Nov '13

Because these are signed they can either be 0 or -1 instead of 0 and 1 as intended. It doesn't cause a problem from what I can see, but it's dangerous and Sparse complains: sound/soc/sh/rcar/rsnd.h:177:25: error: dubious one-bit signed bitfield Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 9e463e5..b5ac3a2 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -174,11 +174,11 @@ struct rsnd_dai { struct rsnd_dai_stream playback; struct rsnd_dai_stream capture; - int clk_master:1; - int bit_clk_inv:1; - int frm_clk_inv:1; - int sys_delay:1; - int data_alignment:1; + unsigned int clk_master:1; + unsigned int bit_clk_inv:1; + unsigned int frm_clk_inv:1; + unsigned int sys_delay:1; + unsigned int data_alignment:1; }; #define rsnd_dai_nr(priv) ((priv)->dai_nr)

2 1

[alsa-devel] [PATCH][RFC] ASoC: rcar: select REGMAP
by Kuninori Morimoto 24 Nov '13

24 Nov '13

55e5b6fd5af04b6d8b0ac6635edf49476ff298ba (ASoC: rsnd: use regmap instead of original register mapping method) support regmap/regmap_field on Renesas sound driver. It needs CONFIG_REGMAP now. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark I'm not sure this is needed or not, but I tried send it sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 14011d9..ff60e11 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" select SND_SIMPLE_CARD + select REGMAP help This option enables R-Car SUR/SCU/SSIU/SSI sound support -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/4] *** SUBJECT HERE ***
by Charles Keepax 24 Nov '13

24 Nov '13

Hi, Small patch series adding some missing functionality on the WM5110. Given most of the changes are in ASoC I have generated the patches against Mark's tree but let me know if we want to push them through MFD instead and I can regenerate, but I suspect they will cause less conflicts going through ASoC. Thanks, Charles Charles Keepax (3): mfd: wm5110: Make DSP memories readable ASoC: wm5110: Add basic support for ISRCs ASoC: wm5110: Expose input high pass filter controls Mark Brown (1): ASoC: wm5110: Hook up ADSP2 cores drivers/mfd/wm5110-tables.c | 73 +++++++- include/linux/mfd/arizona/registers.h | 37 ++++ sound/soc/codecs/Kconfig | 2 + sound/soc/codecs/arizona.c | 10 + sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 316 ++++++++++++++++++++++++++++++++- 6 files changed, 433 insertions(+), 6 deletions(-) -- 1.7.2.5

3 16

[alsa-devel] [PATCH] ASoC: fsl_ssi: Add missing __iomem annotations
by Lars-Peter Clausen 24 Nov '13

24 Nov '13

There are two places in the fsl_ssi driver where the pointer to the register set is missing the __iomem annotation. Adding it fixes the following warnings from sparse: sound/soc/fsl/fsl_ssi.c:747:45: warning: incorrect type in initializer (different address spaces) sound/soc/fsl/fsl_ssi.c:747:45: expected struct ccsr_ssi *ssi sound/soc/fsl/fsl_ssi.c:747:45: got struct ccsr_ssi [noderef] <asn:2>*ssi sound/soc/fsl/fsl_ssi.c:756:9: warning: incorrect type in argument 2 (different address spaces) sound/soc/fsl/fsl_ssi.c:756:9: expected void volatile [noderef] <asn:2>*addr sound/soc/fsl/fsl_ssi.c:756:9: got unsigned int *<noident> sound/soc/fsl/fsl_ssi.c:759:9: warning: incorrect type in argument 2 (different address spaces) sound/soc/fsl/fsl_ssi.c:759:9: expected void volatile [noderef] <asn:2>*addr sound/soc/fsl/fsl_ssi.c:759:9: got unsigned int *<noident> sound/soc/fsl/fsl_ssi.c:761:25: warning: incorrect type in argument 1 (different address spaces) sound/soc/fsl/fsl_ssi.c:761:25: expected unsigned int [noderef] [usertype] <asn:2>*addr sound/soc/fsl/fsl_ssi.c:761:25: got unsigned int *<noident> sound/soc/fsl/fsl_ssi.c:769:45: warning: incorrect type in initializer (different address spaces) sound/soc/fsl/fsl_ssi.c:769:45: expected struct ccsr_ssi *ssi sound/soc/fsl/fsl_ssi.c:769:45: got struct ccsr_ssi [noderef] <asn:2>*ssi sound/soc/fsl/fsl_ssi.c:775:9: warning: incorrect type in argument 2 (different address spaces) sound/soc/fsl/fsl_ssi.c:775:9: expected void volatile [noderef] <asn:2>*addr sound/soc/fsl/fsl_ssi.c:775:9: got unsigned int *<noident> sound/soc/fsl/fsl_ssi.c:776:25: warning: incorrect type in argument 1 (different address spaces) sound/soc/fsl/fsl_ssi.c:776:25: expected unsigned int [noderef] [usertype] <asn:2>*addr sound/soc/fsl/fsl_ssi.c:776:25: got unsigned int *<noident> sound/soc/fsl/fsl_ssi.c:781:16: warning: incorrect type in argument 1 (different address spaces) sound/soc/fsl/fsl_ssi.c:781:16: expected void const volatile [noderef] <asn:2>*addr sound/soc/fsl/fsl_ssi.c:781:16: got unsigned int *<noident> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/fsl/fsl_ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 3df0318..1a9521a 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -751,7 +751,7 @@ static void fsl_ssi_ac97_init(void) static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - struct ccsr_ssi *ssi = fsl_ac97_data->ssi; + struct ccsr_ssi __iomem *ssi = fsl_ac97_data->ssi; unsigned int lreg; unsigned int lval; @@ -773,7 +773,7 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { - struct ccsr_ssi *ssi = fsl_ac97_data->ssi; + struct ccsr_ssi __iomem *ssi = fsl_ac97_data->ssi; unsigned short val = -1; unsigned int lreg; -- 1.8.0

1 0

[alsa-devel] Zoom R16
by Jason Mancine 24 Nov '13

24 Nov '13

Hello, I am looking for some assistance in getting the Zoom R16 working for playback. Please let me know if this is the right forum for this... The device is recognized, but I just can't figure out which interface is which, and how to configure them in quirks-table.h. Any help would be appreciated. To start, the output of lsusb -v for the device is this: --- Bus 001 Device 010: ID 1686:00dd ZOOM Corporation Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x1686 ZOOM Corporation idProduct 0x00dd bcdDevice 0.02 iManufacturer 1 iProduct 2 R16 iSerial 3 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 259 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 480mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 0b 24 01 00 01 35 00 03 01 02 03 ** UNRECOGNIZED: 0c 24 02 05 01 01 00 02 03 00 00 00 ** UNRECOGNIZED: 09 24 03 08 01 03 00 05 00 ** UNRECOGNIZED: 0c 24 02 09 01 02 00 08 00 00 00 00 ** UNRECOGNIZED: 09 24 03 0c 01 01 00 09 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 05 01 01 00 ** UNRECOGNIZED: 14 24 02 01 02 04 18 04 44 ac 00 80 bb 00 88 58 01 00 77 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x006c 1x 108 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 0c 01 01 00 ** UNRECOGNIZED: 14 24 02 01 08 04 18 04 44 ac 00 80 bb 00 88 58 01 00 77 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x01a0 1x 416 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 00 01 41 00 ** UNRECOGNIZED: 06 24 02 01 01 00 ** UNRECOGNIZED: 06 24 02 02 02 00 ** UNRECOGNIZED: 09 24 03 01 03 01 02 01 00 ** UNRECOGNIZED: 09 24 03 02 04 01 01 01 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered --- This is the quirk that Clemens provided on the linux-audio mail list...the device is recognized, but no capture/playback: { /* ZOOM R16 in USB 2.0 mode */ USB_DEVICE(0x1686, 0x00dd), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 1, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 3, .type = QUIRK_MIDI_STANDARD_INTERFACE }, { .ifnum = .1 }, } } }, Now....despite there being no #4 interface in lsusb, if I append the following to the above quirk I get all 8 channels of capture working. I am more confused than ever. :) { .ifnum = 4, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 8, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x84, .ep_attr = 13, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, Obviously, this is not the correct solution and I can't make playback work by adding additional .ifnums. I am quite certain that interface 1 is capture and 2 is playback but if I add the same info to either of those sections my entire system freezes when I plug in the device. Thanks, jason

2 2

Re: [alsa-devel] [pulseaudio-discuss] acceptable noise level for mic boost
by Oleksij Rempel 23 Nov '13

23 Nov '13

Am 22.11.2013 18:34, schrieb Oleksij Rempel: > Am 22.11.2013 17:48, schrieb Tanu Kaskinen: >> On Thu, 2013-11-21 at 16:42 +0100, Oleksij Rempel wrote: >>> Hello all, >>> >>> i posted this email to ALSA, but probably PA devs have some ideas too. >>> >>> ====================================================================== >>> Hello Takashi, >>> >>> what is acceptable noise level for mic boost? >>> My laptop has three levels of mic boost (+36dB). Level 3 is useless >>> since noise level will be about 0db. With level 1 i get -12dB noise, so >>> there is almost 12dB room for data... but it is not enough for normal >>> speech, which is about 20dB. >>> >>> Are there any ALSA guide for HW designers, which will say some thing >>> like acceptable levels are 90-20dB? >>> There are some Skype and Microsoft prescriptions for certification. >>> For example M$ prescribe 18dB or more for speech to noise ration for >>> build in mics. It sound like good number to start. >>> >>> In my case, i need to remove mic boost completely to guarantee minimum >>> 18dB SpNR. Suddenly i didn't found correct way to do it. >>> I will be thankful for some tip. >>> >>> ===================================================================== >>> >>> So my question to PA devs. Do pulse audio need to know about noise level >>> of microphone? Will it help to make correct decisions? Especial for VoIP >>> and automatic gain control. >> >> We don't currently have any idea of the noise level, so we don't have >> code for adjusting our decisions based on that information either. So, >> I'd say that pulseaudio doesn't *need* to know about the noise level. Of >> course, if someone someday complains that pulseaudio does bad decisions >> because it doesn't take the noise level into account, then we would need >> that information. >> > > Thank you for your response. > I have more questions: > Skype continuously trying to adjust record level. If there is a silence > it will try to push the input volume as close to 0dB (in pulseaudio > terms) as possible. > For my laptop 0dB(in PA) will mean some thing +68dB over base volume > (ALSA). The noise will reach 0dBFS. > Well, it wont kill any one, but it sound bad. > > What 0dB mean in pulseaudio? > Why pulseaudio has capture volume range -130 - +11dB? Is it possible to > fit this gain to 16-bit PCM? to answer this question by my self. If i understand correctly pulseaudio trying to imitate professional mixer settings. The difference is, usual mixer has 0dB settings as point for preconfigured sound. You need to use "Gain" or "Trim" control to configure it to acceptable level. 0dB in PulseAudio looks for me incorrect, if no preconfigurable available. Here is one example: http://www.youtube.com/watch?v=YXSJ1oOQ9tY > Now to my hardware. I can blacklist in the driver 2 "mic boost" steps > without introducing any regressions. Which will guaranty maximal noise > level = -12dBFS. But it is still not optimal. Best experience i had with > limit to -24dB. In this case i need to remove Mic Boost completely but > pulseaudio will fail to make a difference between internal and external mic. > > What is best solution in this case? Fix pulse or leave at least one step > for mic boost? After some more testing, i'm sure it should be fixed in pulseaudio. Blacklisting microfone setting is just quick dirty hack. It will work for internal mic. But fail on external. There is no way, ALSA can predict some thing about it. The best thing ALSA can do here is to provide all possible controls. > The microphone on this laptop is a digital mic. Do it make sense to use > hardware amp? Or is it possible to get better results with software? Theoretically, Skype or Empathy or any other app with auto gain control should us 0dB as target. But since ALSA or PulseAudio can't automatically detect correct point, PuleAudio should provide possibility to override 0dB offset. For build in microphones can be used ALSA quirk list or some kind of database for pulseaudio with per device default values to make better default experience. IMO, current solution with blacklists in ALSA is wrong. -- Regards, Oleksij

1 0

[alsa-devel] [PATCH] ASoC: ak4641: Convert to direct regmap API usage
by Mark Brown 22 Nov '13

22 Nov '13

From: Mark Brown <broonie(a)linaro.org> We're trying to remove the ASoC level I/O functions. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak4641.c | 38 +++++++++++++++++++++++++------------- 1 file changed, 25 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 49cc5f6d6dba..94cbe508dd37 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -17,6 +17,7 @@ #include <linux/gpio.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -30,6 +31,7 @@ /* codec private data */ struct ak4641_priv { + struct regmap *regmap; unsigned int sysclk; int deemph; int playback_fs; @@ -38,12 +40,12 @@ struct ak4641_priv { /* * ak4641 register cache */ -static const u8 ak4641_reg[AK4641_CACHEREGNUM] = { - 0x00, 0x80, 0x00, 0x80, - 0x02, 0x00, 0x11, 0x05, - 0x00, 0x00, 0x36, 0x10, - 0x00, 0x00, 0x57, 0x00, - 0x88, 0x88, 0x08, 0x08 +static const struct reg_default ak4641_reg_defaults[] = { + { 0, 0x00 }, { 1, 0x80 }, { 2, 0x00 }, { 3, 0x80 }, + { 4, 0x02 }, { 5, 0x00 }, { 6, 0x11 }, { 7, 0x05 }, + { 8, 0x00 }, { 9, 0x00 }, { 10, 0x36 }, { 11, 0x10 }, + { 12, 0x00 }, { 13, 0x00 }, { 14, 0x57 }, { 15, 0x00 }, + { 16, 0x88 }, { 17, 0x88 }, { 18, 0x08 }, { 19, 0x08 } }; static const int deemph_settings[] = {44100, 0, 48000, 32000}; @@ -396,6 +398,7 @@ static int ak4641_mute(struct snd_soc_dai *dai, int mute) static int ak4641_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); struct ak4641_platform_data *pdata = codec->dev->platform_data; int ret; @@ -417,7 +420,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec, gpio_set_value(pdata->gpio_npdn, 1); mdelay(1); - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(ak4641->regmap); if (ret) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -433,7 +436,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec, gpio_set_value(pdata->gpio_npdn, 0); if (pdata && gpio_is_valid(pdata->gpio_power)) gpio_set_value(pdata->gpio_power, 0); - codec->cache_sync = 1; + regcache_mark_dirty(ak4641->regmap); break; } codec->dapm.bias_level = level; @@ -518,7 +521,7 @@ static int ak4641_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -550,12 +553,17 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = { .dapm_routes = ak4641_audio_map, .num_dapm_routes = ARRAY_SIZE(ak4641_audio_map), .set_bias_level = ak4641_set_bias_level, - .reg_cache_size = ARRAY_SIZE(ak4641_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = ak4641_reg, - .reg_cache_step = 1, }; +static const struct regmap_config ak4641_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = AK4641_BTIF, + .reg_defaults = ak4641_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ak4641_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; static int ak4641_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -569,6 +577,10 @@ static int ak4641_i2c_probe(struct i2c_client *i2c, if (!ak4641) return -ENOMEM; + ak4641->regmap = devm_regmap_init_i2c(i2c, &ak4641_regmap); + if (IS_ERR(ak4641->regmap)) + return PTR_ERR(ak4641->regmap); + if (pdata) { if (gpio_is_valid(pdata->gpio_power)) { ret = gpio_request_one(pdata->gpio_power, -- 1.8.4.4

1 0

[alsa-devel] [PATCH 1/4] ASoC: wm8991: Convert to table based control and widget init
by Mark Brown 22 Nov '13

22 Nov '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8991.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 5078fc8e10f7..86aa33fd11de 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -764,7 +764,7 @@ static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8991_dapm_routes[] = { /* Make DACs turn on when playing even if not mixed into any outputs */ {"Internal DAC Sink", NULL, "Left DAC"}, {"Internal DAC Sink", NULL, "Right DAC"}, @@ -1278,13 +1278,6 @@ static int wm8991_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8991_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - snd_soc_add_codec_controls(codec, wm8991_snd_controls, - ARRAY_SIZE(wm8991_snd_controls)); - - snd_soc_dapm_new_controls(&codec->dapm, wm8991_dapm_widgets, - ARRAY_SIZE(wm8991_dapm_widgets)); - snd_soc_dapm_add_routes(&codec->dapm, audio_map, - ARRAY_SIZE(audio_map)); return 0; } @@ -1333,6 +1326,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8991 = { .suspend = wm8991_suspend, .resume = wm8991_resume, .set_bias_level = wm8991_set_bias_level, + .controls = wm8991_snd_controls, + .num_controls = ARRAY_SIZE(wm8991_snd_controls), + .dapm_widgets = wm8991_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8991_dapm_widgets), + .dapm_routes = wm8991_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8991_dapm_routes), .reg_cache_size = WM8991_MAX_REGISTER + 1, .reg_word_size = sizeof(u16), .reg_cache_default = wm8991_reg_defs -- 1.8.4.4

2 4

[alsa-devel] firewire-lib: an issue to generate packet with 'no data' in blocking mode
by Takashi Sakamoto 22 Nov '13

22 Nov '13

Clemens, I have a question about generate packet with 'no data' in blocking mode. I think there is out of specification in current firewire-lib. If this is a qurk for some devices, I'll prepare patches to switch generating mode because BeBoB cannot sound with current firewire-lib. If this is a bug, then I want to discuss which is better for firewire-lib. In my understanding of IEC 61883-6, there are two ways: 1. generate 'empty packet' defined in IEC 61883-1 - size of packet is 2 quadlets - FDF = sfc - packet includes just CIP headers 2. generate 'special non-empty packet' defined in IEC 61883-6 - size of packet is following to blocking mode - FDF = 0xff ('NO-DATA' code) - packet includes dummy data But current implementation is a strange combination of them. - size of packet is 2 (way 1) - FDF = 0xff (way 2) I attach two patches. 'empty.patch' is for 1 and 'special.patch' is for 2. Regards Takashi Sakamoto

3 4

[alsa-devel] [PATCHv4] ASoC: Add support for BCM2835
by Florian Meier 22 Nov '13

22 Nov '13

This driver adds support for digital audio (I2S) for the BCM2835 SoC that is used by the Raspberry Pi. External audio codecs can be connected to the Raspberry Pi via P5 header. It relies on cyclic DMA engine support for BCM2835. Signed-off-by: Florian Meier <florian.meier(a)koalo.de> --- Minor style improvements compared to PATCHv3. .../devicetree/bindings/sound/bcm2835-i2s.txt | 25 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/bcm/Kconfig | 10 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/bcm2835-i2s.c | 892 ++++++++++++++++++++ 6 files changed, 934 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/bcm2835-i2s.txt create mode 100644 sound/soc/bcm/Kconfig create mode 100644 sound/soc/bcm/Makefile create mode 100644 sound/soc/bcm/bcm2835-i2s.c diff --git a/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt b/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt new file mode 100644 index 0000000..65783de --- /dev/null +++ b/Documentation/devicetree/bindings/sound/bcm2835-i2s.txt @@ -0,0 +1,25 @@ +* Broadcom BCM2835 SoC I2S/PCM module + +Required properties: +- compatible: "brcm,bcm2835-i2s" +- reg: A list of base address and size entries: + * The first entry should cover the PCM registers + * The second entry should cover the PCM clock registers +- dmas: List of DMA controller phandle and DMA request line ordered pairs. +- dma-names: Identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. + + One of the DMA channels will be responsible for transmission (should be + named "tx") and one for reception (should be named "rx"). + +Example: + +bcm2835_i2s: i2s@7e203000 { + compatible = "brcm,bcm2835-i2s"; + reg = <0x7e203000 0x20>, + <0x7e101098 0x02>; + + dmas = <&dma 2>, + <&dma 3>; + dma-names = "tx", "rx"; +}; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5138b84..a5e3a70 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_GENERIC_DMAENGINE_PCM # All the supported SoCs source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" +source "sound/soc/bcm/Kconfig" source "sound/soc/blackfin/Kconfig" source "sound/soc/cirrus/Kconfig" source "sound/soc/davinci/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 8b9e701..b52d4aa 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -10,6 +10,7 @@ obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ +obj-$(CONFIG_SND_SOC) += bcm/ obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += cirrus/ obj-$(CONFIG_SND_SOC) += davinci/ diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig new file mode 100644 index 0000000..3d82a29 --- /dev/null +++ b/sound/soc/bcm/Kconfig @@ -0,0 +1,10 @@ +config SND_BCM2835_SOC_I2S + tristate "SoC Audio support for the Broadcom BCM2835 I2S module" + depends on ARCH_BCM2835 || COMPILE_TEST + select SND_SOC_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + Say Y or M if you want to add support for codecs attached to + the BCM2835 I2S interface. You will also need + to select the audio interfaces to support below. diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile new file mode 100644 index 0000000..bc816b7 --- /dev/null +++ b/sound/soc/bcm/Makefile @@ -0,0 +1,5 @@ +# BCM2835 Platform Support +snd-soc-bcm2835-i2s-objs := bcm2835-i2s.o + +obj-$(CONFIG_SND_BCM2835_SOC_I2S) += snd-soc-bcm2835-i2s.o + diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c new file mode 100644 index 0000000..7b27c35 --- /dev/null +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -0,0 +1,892 @@ +/* + * ALSA SoC I2S Audio Layer for Broadcom BCM2835 SoC + * + * Author: Florian Meier <florian.meier(a)koalo.de> + * Copyright 2013 + * + * Based on + * Raspberry Pi PCM I2S ALSA Driver + * Copyright (c) by Phil Poole 2013 + * + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * Vladimir Barinov, <vbarinov(a)embeddedalley.com> + * Copyright (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> + * + * OMAP ALSA SoC DAI driver using McBSP port + * Copyright (C) 2008 Nokia Corporation + * Contact: Jarkko Nikula <jarkko.nikula(a)bitmer.com> + * Peter Ujfalusi <peter.ujfalusi(a)ti.com> + * + * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver + * Author: Timur Tabi <timur(a)freescale.com> + * Copyright 2007-2010 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * 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/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +/* Clock registers */ +#define BCM2835_CLK_PCMCTL_REG 0x00 +#define BCM2835_CLK_PCMDIV_REG 0x04 + +/* Clock register settings */ +#define BCM2835_CLK_PASSWD (0x5a000000) +#define BCM2835_CLK_PASSWD_MASK (0xff000000) +#define BCM2835_CLK_MASH(v) ((v) << 9) +#define BCM2835_CLK_FLIP BIT(8) +#define BCM2835_CLK_BUSY BIT(7) +#define BCM2835_CLK_KILL BIT(5) +#define BCM2835_CLK_ENAB BIT(4) +#define BCM2835_CLK_SRC(v) (v) + +#define BCM2835_CLK_SHIFT (12) +#define BCM2835_CLK_DIVI(v) ((v) << BCM2835_CLK_SHIFT) +#define BCM2835_CLK_DIVF(v) (v) +#define BCM2835_CLK_DIVF_MASK (0xFFF) + +enum { + BCM2835_CLK_MASH_0 = 0, + BCM2835_CLK_MASH_1, + BCM2835_CLK_MASH_2, + BCM2835_CLK_MASH_3, +}; + +enum { + BCM2835_CLK_SRC_GND = 0, + BCM2835_CLK_SRC_OSC, + BCM2835_CLK_SRC_DBG0, + BCM2835_CLK_SRC_DBG1, + BCM2835_CLK_SRC_PLLA, + BCM2835_CLK_SRC_PLLC, + BCM2835_CLK_SRC_PLLD, + BCM2835_CLK_SRC_HDMI, +}; + +/* Most clocks are not useable (freq = 0) */ +static const unsigned int bcm2835_clk_freq[BCM2835_CLK_SRC_HDMI+1] = { + [BCM2835_CLK_SRC_GND] = 0, + [BCM2835_CLK_SRC_OSC] = 19200000, + [BCM2835_CLK_SRC_DBG0] = 0, + [BCM2835_CLK_SRC_DBG1] = 0, + [BCM2835_CLK_SRC_PLLA] = 0, + [BCM2835_CLK_SRC_PLLC] = 0, + [BCM2835_CLK_SRC_PLLD] = 500000000, + [BCM2835_CLK_SRC_HDMI] = 0, +}; + +/* I2S registers */ +#define BCM2835_I2S_CS_A_REG 0x00 +#define BCM2835_I2S_FIFO_A_REG 0x04 +#define BCM2835_I2S_MODE_A_REG 0x08 +#define BCM2835_I2S_RXC_A_REG 0x0c +#define BCM2835_I2S_TXC_A_REG 0x10 +#define BCM2835_I2S_DREQ_A_REG 0x14 +#define BCM2835_I2S_INTEN_A_REG 0x18 +#define BCM2835_I2S_INTSTC_A_REG 0x1c +#define BCM2835_I2S_GRAY_REG 0x20 + +/* I2S register settings */ +#define BCM2835_I2S_STBY BIT(25) +#define BCM2835_I2S_SYNC BIT(24) +#define BCM2835_I2S_RXSEX BIT(23) +#define BCM2835_I2S_RXF BIT(22) +#define BCM2835_I2S_TXE BIT(21) +#define BCM2835_I2S_RXD BIT(20) +#define BCM2835_I2S_TXD BIT(19) +#define BCM2835_I2S_RXR BIT(18) +#define BCM2835_I2S_TXW BIT(17) +#define BCM2835_I2S_CS_RXERR BIT(16) +#define BCM2835_I2S_CS_TXERR BIT(15) +#define BCM2835_I2S_RXSYNC BIT(14) +#define BCM2835_I2S_TXSYNC BIT(13) +#define BCM2835_I2S_DMAEN BIT(9) +#define BCM2835_I2S_RXTHR(v) ((v) << 7) +#define BCM2835_I2S_TXTHR(v) ((v) << 5) +#define BCM2835_I2S_RXCLR BIT(4) +#define BCM2835_I2S_TXCLR BIT(3) +#define BCM2835_I2S_TXON BIT(2) +#define BCM2835_I2S_RXON BIT(1) +#define BCM2835_I2S_EN (1) + +#define BCM2835_I2S_CLKDIS BIT(28) +#define BCM2835_I2S_PDMN BIT(27) +#define BCM2835_I2S_PDME BIT(26) +#define BCM2835_I2S_FRXP BIT(25) +#define BCM2835_I2S_FTXP BIT(24) +#define BCM2835_I2S_CLKM BIT(23) +#define BCM2835_I2S_CLKI BIT(22) +#define BCM2835_I2S_FSM BIT(21) +#define BCM2835_I2S_FSI BIT(20) +#define BCM2835_I2S_FLEN(v) ((v) << 10) +#define BCM2835_I2S_FSLEN(v) (v) + +#define BCM2835_I2S_CHWEX BIT(15) +#define BCM2835_I2S_CHEN BIT(14) +#define BCM2835_I2S_CHPOS(v) ((v) << 4) +#define BCM2835_I2S_CHWID(v) (v) +#define BCM2835_I2S_CH1(v) ((v) << 16) +#define BCM2835_I2S_CH2(v) (v) + +#define BCM2835_I2S_TX_PANIC(v) ((v) << 24) +#define BCM2835_I2S_RX_PANIC(v) ((v) << 16) +#define BCM2835_I2S_TX(v) ((v) << 8) +#define BCM2835_I2S_RX(v) (v) + +#define BCM2835_I2S_INT_RXERR BIT(3) +#define BCM2835_I2S_INT_TXERR BIT(2) +#define BCM2835_I2S_INT_RXR BIT(1) +#define BCM2835_I2S_INT_TXW BIT(0) + +/* I2S DMA interface */ +/* FIXME: Needs IOMMU support */ +#define BCM2835_VCMMU_SHIFT (0x7E000000 - 0x20000000) + +/* General device struct */ +struct bcm2835_i2s_dev { + struct device *dev; + struct snd_dmaengine_dai_dma_data dma_data[2]; + unsigned int fmt; + unsigned int bclk_ratio; + + struct regmap *i2s_regmap; + struct regmap *clk_regmap; +}; + +static void bcm2835_i2s_start_clock(struct bcm2835_i2s_dev *dev) +{ + /* Start the clock if in master mode */ + unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + + switch (master) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB, + BCM2835_CLK_PASSWD | BCM2835_CLK_ENAB); + break; + default: + break; + } +} + +static void bcm2835_i2s_stop_clock(struct bcm2835_i2s_dev *dev) +{ + uint32_t clkreg; + int timeout = 1000; + + /* Stop clock */ + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB, + BCM2835_CLK_PASSWD); + + /* Wait for the BUSY flag going down */ + while (--timeout) { + regmap_read(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, &clkreg); + if (!(clkreg & BCM2835_CLK_BUSY)) + break; + } + + if (!timeout) { + /* KILL the clock */ + dev_err(dev->dev, "I2S clock didn't stop. Kill the clock!\n"); + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_KILL | BCM2835_CLK_PASSWD_MASK, + BCM2835_CLK_KILL | BCM2835_CLK_PASSWD); + } +} + +static void bcm2835_i2s_clear_fifos(struct bcm2835_i2s_dev *dev, + bool tx, bool rx) +{ + int timeout = 1000; + uint32_t syncval; + uint32_t csreg; + uint32_t i2s_active_state; + uint32_t clkreg; + uint32_t clk_active_state; + uint32_t off; + uint32_t clr; + + off = tx ? BCM2835_I2S_TXON : 0; + off |= rx ? BCM2835_I2S_RXON : 0; + + clr = tx ? BCM2835_I2S_TXCLR : 0; + clr |= rx ? BCM2835_I2S_RXCLR : 0; + + /* Backup the current state */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg); + i2s_active_state = csreg & (BCM2835_I2S_RXON | BCM2835_I2S_TXON); + + regmap_read(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, &clkreg); + clk_active_state = clkreg & BCM2835_CLK_ENAB; + + /* Start clock if not running */ + if (!clk_active_state) { + regmap_update_bits(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, + BCM2835_CLK_PASSWD_MASK | BCM2835_CLK_ENAB, + BCM2835_CLK_PASSWD | BCM2835_CLK_ENAB); + } + + /* Stop I2S module */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, off, 0); + + /* + * Clear the FIFOs + * Requires at least 2 PCM clock cycles to take effect + */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, clr, clr); + + /* Wait for 2 PCM clock cycles */ + + /* + * Toggle the SYNC flag. After 2 PCM clock cycles it can be read back + * FIXME: This does not seem to work for slave mode! + */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &syncval); + syncval &= BCM2835_I2S_SYNC; + + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_SYNC, ~syncval); + + /* Wait for the SYNC flag changing it's state */ + while (--timeout) { + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg); + if ((csreg & BCM2835_I2S_SYNC) != syncval) + break; + } + + if (!timeout) + dev_err(dev->dev, "I2S SYNC error!\n"); + + /* Stop clock if it was not running before */ + if (!clk_active_state) + bcm2835_i2s_stop_clock(dev); + + /* Restore I2S state */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_RXON | BCM2835_I2S_TXON, i2s_active_state); +} + +static int bcm2835_i2s_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + dev->fmt = fmt; + return 0; +} + +static int bcm2835_i2s_set_dai_bclk_ratio(struct snd_soc_dai *dai, + unsigned int ratio) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + dev->bclk_ratio = ratio; + return 0; +} + +static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + unsigned int sampling_rate = params_rate(params); + unsigned int data_length, data_delay, bclk_ratio; + unsigned int ch1pos, ch2pos, mode, format; + unsigned int mash = BCM2835_CLK_MASH_1; + unsigned int divi, divf, target_frequency; + int clk_src = -1; + unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + bool bit_master = (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBS_CFM); + + bool frame_master = (master == SND_SOC_DAIFMT_CBS_CFS + || master == SND_SOC_DAIFMT_CBM_CFS); + uint32_t csreg; + + /* + * If a stream is already enabled, + * the registers are already set properly. + */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &csreg); + + if (csreg & (BCM2835_I2S_TXON | BCM2835_I2S_RXON)) + return 0; + + /* + * Adjust the data length according to the format. + * We prefill the half frame length with an integer + * divider of 2400 as explained at the clock settings. + * Maybe it is overwritten there, if the Integer mode + * does not apply. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + data_length = 16; + bclk_ratio = 40; + break; + case SNDRV_PCM_FORMAT_S32_LE: + data_length = 32; + bclk_ratio = 80; + break; + default: + return -EINVAL; + } + + /* If bclk_ratio already set, use that one. */ + if (dev->bclk_ratio) + bclk_ratio = dev->bclk_ratio; + + /* + * Clock Settings + * + * The target frequency of the bit clock is + * sampling rate * frame length + * + * Integer mode: + * Sampling rates that are multiples of 8000 kHz + * can be driven by the oscillator of 19.2 MHz + * with an integer divider as long as the frame length + * is an integer divider of 19200000/8000=2400 as set up above. + * This is no longer possible if the sampling rate + * is too high (e.g. 192 kHz), because the oscillator is too slow. + * + * MASH mode: + * For all other sampling rates, it is not possible to + * have an integer divider. Approximate the clock + * with the MASH module that induces a slight frequency + * variance. To minimize that it is best to have the fastest + * clock here. That is PLLD with 500 MHz. + */ + target_frequency = sampling_rate * bclk_ratio; + clk_src = BCM2835_CLK_SRC_OSC; + mash = BCM2835_CLK_MASH_0; + + if (bcm2835_clk_freq[clk_src] % target_frequency == 0 + && bit_master && frame_master) { + divi = bcm2835_clk_freq[clk_src] / target_frequency; + divf = 0; + } else { + uint64_t dividend; + + if (!dev->bclk_ratio) { + /* + * Overwrite bclk_ratio, because the + * above trick is not needed or can + * not be used. + */ + bclk_ratio = 2 * data_length; + } + + target_frequency = sampling_rate * bclk_ratio; + + clk_src = BCM2835_CLK_SRC_PLLD; + mash = BCM2835_CLK_MASH_1; + + dividend = bcm2835_clk_freq[clk_src]; + dividend <<= BCM2835_CLK_SHIFT; + do_div(dividend, target_frequency); + divi = dividend >> BCM2835_CLK_SHIFT; + divf = dividend & BCM2835_CLK_DIVF_MASK; + } + + /* Set clock divider */ + regmap_write(dev->clk_regmap, BCM2835_CLK_PCMDIV_REG, BCM2835_CLK_PASSWD + | BCM2835_CLK_DIVI(divi) + | BCM2835_CLK_DIVF(divf)); + + /* Setup clock, but don't start it yet */ + regmap_write(dev->clk_regmap, BCM2835_CLK_PCMCTL_REG, BCM2835_CLK_PASSWD + | BCM2835_CLK_MASH(mash) + | BCM2835_CLK_SRC(clk_src)); + + /* Setup the frame format */ + format = BCM2835_I2S_CHEN; + + if (data_length > 24) + format |= BCM2835_I2S_CHWEX; + + format |= BCM2835_I2S_CHWID((data_length-8)&0xf); + + switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + data_delay = 1; + break; + default: + /* + * TODO + * Others are possible but are not implemented at the moment. + */ + dev_err(dev->dev, "%s:bad format\n", __func__); + return -EINVAL; + } + + ch1pos = data_delay; + ch2pos = bclk_ratio / 2 + data_delay; + + switch (params_channels(params)) { + case 2: + format = BCM2835_I2S_CH1(format) | BCM2835_I2S_CH2(format); + format |= BCM2835_I2S_CH1(BCM2835_I2S_CHPOS(ch1pos)); + format |= BCM2835_I2S_CH2(BCM2835_I2S_CHPOS(ch2pos)); + break; + default: + return -EINVAL; + } + + /* + * Set format for both streams. + * We cannot set another frame length + * (and therefore word length) anyway, + * so the format will be the same. + */ + regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, format); + regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, format); + + /* Setup the I2S mode */ + mode = 0; + + if (data_length <= 16) { + /* + * Use frame packed mode (2 channels per 32 bit word) + * We cannot set another frame length in the second stream + * (and therefore word length) anyway, + * so the format will be the same. + */ + mode |= BCM2835_I2S_FTXP | BCM2835_I2S_FRXP; + } + + mode |= BCM2835_I2S_FLEN(bclk_ratio - 1); + mode |= BCM2835_I2S_FSLEN(bclk_ratio / 2); + + /* Master or slave? */ + switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* CPU is master */ + break; + case SND_SOC_DAIFMT_CBM_CFS: + /* + * CODEC is bit clock master + * CPU is frame master + */ + mode |= BCM2835_I2S_CLKM; + break; + case SND_SOC_DAIFMT_CBS_CFM: + /* + * CODEC is frame master + * CPU is bit clock master + */ + mode |= BCM2835_I2S_FSM; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* CODEC is master */ + mode |= BCM2835_I2S_CLKM; + mode |= BCM2835_I2S_FSM; + break; + default: + dev_err(dev->dev, "%s:bad master\n", __func__); + return -EINVAL; + } + + /* + * Invert clocks? + * + * The BCM approach seems to be inverted to the classical I2S approach. + */ + switch (dev->fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* None. Therefore, both for BCM */ + mode |= BCM2835_I2S_CLKI; + mode |= BCM2835_I2S_FSI; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Both. Therefore, none for BCM */ + break; + case SND_SOC_DAIFMT_NB_IF: + /* + * Invert only frame sync. Therefore, + * invert only bit clock for BCM + */ + mode |= BCM2835_I2S_CLKI; + break; + case SND_SOC_DAIFMT_IB_NF: + /* + * Invert only bit clock. Therefore, + * invert only frame sync for BCM + */ + mode |= BCM2835_I2S_FSI; + break; + default: + return -EINVAL; + } + + regmap_write(dev->i2s_regmap, BCM2835_I2S_MODE_A_REG, mode); + + /* Setup the DMA parameters */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_RXTHR(1) + | BCM2835_I2S_TXTHR(1) + | BCM2835_I2S_DMAEN, 0xffffffff); + + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_DREQ_A_REG, + BCM2835_I2S_TX_PANIC(0x10) + | BCM2835_I2S_RX_PANIC(0x30) + | BCM2835_I2S_TX(0x30) + | BCM2835_I2S_RX(0x20), 0xffffffff); + + /* Clear FIFOs */ + bcm2835_i2s_clear_fifos(dev, true, true); + + return 0; +} + +static int bcm2835_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + uint32_t cs_reg; + + bcm2835_i2s_start_clock(dev); + + /* + * Clear both FIFOs if the one that should be started + * is not empty at the moment. This should only happen + * after overrun. Otherwise, hw_params would have cleared + * the FIFO. + */ + regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &cs_reg); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK + && !(cs_reg & BCM2835_I2S_TXE)) + bcm2835_i2s_clear_fifos(dev, true, false); + else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE + && (cs_reg & BCM2835_I2S_RXD)) + bcm2835_i2s_clear_fifos(dev, false, true); + + return 0; +} + +static void bcm2835_i2s_stop(struct bcm2835_i2s_dev *dev, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + uint32_t mask; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + mask = BCM2835_I2S_RXON; + else + mask = BCM2835_I2S_TXON; + + regmap_update_bits(dev->i2s_regmap, + BCM2835_I2S_CS_A_REG, mask, 0); + + /* Stop also the clock when not SND_SOC_DAIFMT_CONT */ + if (!dai->active && !(dev->fmt & SND_SOC_DAIFMT_CONT)) + bcm2835_i2s_stop_clock(dev); +} + +static int bcm2835_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + uint32_t mask; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + bcm2835_i2s_start_clock(dev); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + mask = BCM2835_I2S_RXON; + else + mask = BCM2835_I2S_TXON; + + regmap_update_bits(dev->i2s_regmap, + BCM2835_I2S_CS_A_REG, mask, mask); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + bcm2835_i2s_stop(dev, substream, dai); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int bcm2835_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + if (dai->active) + return 0; + + /* Should this still be running stop it */ + bcm2835_i2s_stop_clock(dev); + + /* Enable PCM block */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_EN, BCM2835_I2S_EN); + + /* + * Disable STBY. + * Requires at least 4 PCM clock cycles to take effect. + */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_STBY, BCM2835_I2S_STBY); + + return 0; +} + +static void bcm2835_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + bcm2835_i2s_stop(dev, substream, dai); + + /* If both streams are stopped, disable module and clock */ + if (dai->active) + return; + + /* Disable the module */ + regmap_update_bits(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, + BCM2835_I2S_EN, 0); + + /* + * Stopping clock is necessary, because stop does + * not stop the clock when SND_SOC_DAIFMT_CONT + */ + bcm2835_i2s_stop_clock(dev); +} + +static const struct snd_soc_dai_ops bcm2835_i2s_dai_ops = { + .startup = bcm2835_i2s_startup, + .shutdown = bcm2835_i2s_shutdown, + .prepare = bcm2835_i2s_prepare, + .trigger = bcm2835_i2s_trigger, + .hw_params = bcm2835_i2s_hw_params, + .set_fmt = bcm2835_i2s_set_dai_fmt, + .set_bclk_ratio = bcm2835_i2s_set_dai_bclk_ratio +}; + +static int bcm2835_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, + &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK], + &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]); + + return 0; +} + +static struct snd_soc_dai_driver bcm2835_i2s_dai = { + .name = "bcm2835-i2s", + .probe = bcm2835_i2s_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S32_LE + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S32_LE + }, + .ops = &bcm2835_i2s_dai_ops, + .symmetric_rates = 1 +}; + +static bool bcm2835_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case BCM2835_I2S_CS_A_REG: + case BCM2835_I2S_FIFO_A_REG: + case BCM2835_I2S_INTSTC_A_REG: + case BCM2835_I2S_GRAY_REG: + return true; + default: + return false; + }; +} + +static bool bcm2835_i2s_precious_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case BCM2835_I2S_FIFO_A_REG: + return true; + default: + return false; + }; +} + +static bool bcm2835_clk_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case BCM2835_CLK_PCMCTL_REG: + return true; + default: + return false; + }; +} + +static const struct regmap_config bcm2835_regmap_config[] = { + { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = BCM2835_I2S_GRAY_REG, + .precious_reg = bcm2835_i2s_precious_reg, + .volatile_reg = bcm2835_i2s_volatile_reg, + .cache_type = REGCACHE_RBTREE, + }, + { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = BCM2835_CLK_PCMDIV_REG, + .volatile_reg = bcm2835_clk_volatile_reg, + .cache_type = REGCACHE_RBTREE, + }, +}; + +static const struct snd_soc_component_driver bcm2835_i2s_component = { + .name = "bcm2835-i2s-comp", +}; + +static int bcm2835_i2s_probe(struct platform_device *pdev) +{ + struct bcm2835_i2s_dev *dev; + int i; + int ret; + struct regmap *regmap[2]; + struct resource *mem[2]; + + /* Request both ioareas */ + for (i = 0; i <= 1; i++) { + void __iomem *base; + + mem[i] = platform_get_resource(pdev, IORESOURCE_MEM, i); + base = devm_ioremap_resource(&pdev->dev, mem[i]); + if (IS_ERR(base)) + return PTR_ERR(base); + + regmap[i] = devm_regmap_init_mmio(&pdev->dev, base, + &bcm2835_regmap_config[i]); + if (IS_ERR(regmap[i])) { + dev_err(&pdev->dev, "I2S probe: regmap init failed\n"); + return PTR_ERR(regmap[i]); + } + } + + dev = devm_kzalloc(&pdev->dev, sizeof(*dev), + GFP_KERNEL); + if (IS_ERR(base)) + return PTR_ERR(base); + + dev->i2s_regmap = regmap[0]; + dev->clk_regmap = regmap[1]; + + /* Set the DMA address */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = + (dma_addr_t)mem[0]->start + BCM2835_I2S_FIFO_A_REG + + BCM2835_VCMMU_SHIFT; + + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = + (dma_addr_t)mem[0]->start + BCM2835_I2S_FIFO_A_REG + + BCM2835_VCMMU_SHIFT; + + /* Set the bus width */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr_width = + DMA_SLAVE_BUSWIDTH_4_BYTES; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr_width = + DMA_SLAVE_BUSWIDTH_4_BYTES; + + /* Set burst */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].maxburst = 2; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].maxburst = 2; + + /* BCLK ratio - use default */ + dev->bclk_ratio = 0; + + /* Store the pdev */ + dev->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, dev); + + ret = snd_soc_register_component(&pdev->dev, + &bcm2835_i2s_component, &bcm2835_i2s_dai, 1); + + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + return ret; + } + + ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); + snd_soc_unregister_component(&pdev->dev); + return ret; + } + + return 0; +} + +static const struct of_device_id bcm2835_i2s_of_match[] = { + { .compatible = "brcm,bcm2835-i2s", }, + {}, +}; + +static int bcm2835_i2s_remove(struct platform_device *pdev) +{ + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + return 0; +} + +static struct platform_driver bcm2835_i2s_driver = { + .probe = bcm2835_i2s_probe, + .remove = bcm2835_i2s_remove, + .driver = { + .name = "bcm2835-i2s", + .owner = THIS_MODULE, + .of_match_table = bcm2835_i2s_of_match, + }, +}; + +module_platform_driver(bcm2835_i2s_driver); + +MODULE_ALIAS("platform:bcm2835-i2s"); +MODULE_DESCRIPTION("BCM2835 I2S interface"); +MODULE_AUTHOR("Florian Meier <florian.meier(a)koalo.de>"); +MODULE_LICENSE("GPL v2"); -- 1.7.9.5

2 2

Re: [alsa-devel] [Report] ALSO: hda - CA0132: Bugs and missing features
by Dylan Reid 22 Nov '13

22 Nov '13

On Tue, Nov 19, 2013 at 8:08 AM, Andreas Reis <andreas.reis(a)gmail.com> wrote: > Correction: The other jack duplicated for the front panel is Mic In > (no mention of the rear panel jack's shared Line In functionality > here), not Rear Speaker Out. > > On Tue, Nov 19, 2013 at 5:04 PM, Andreas Reis <andreas.reis(a)gmail.com> wrote: >> Posting here by Takashi's advice after my alsa-user post remained unanswered. >> >> There are a number of grave issues with the CA0132 codec on my >> Gigabyte G1.Sniper M5 (rev. 1.0) (Intel Z87, Recon3Di onboard, latest >> bios F8c). I've tried kernels (x64) from 3.9 to 3.13-preRC1 (as >> currently in Torvald's git), both with pulse on Ubuntu 13.10 and >> without on Arch Linux. >> Sorry I don't have good news for you. The CA0132 is very dependent on the firmware that is loaded into the DSP that resides in the codec. To get your system to work you'll have to obtain the firmware that was designed for your motherboard, then change the Linux driver to be able to talk to it, without a spec for how it works. This is not impossible, but it would be rather time consuming. -Dylan

2 1

[alsa-devel] [PATCH] ALSA: hda - Set current_headset_type to ALC_HEADSET_TYPE_ENUM (janitorial)
by David Henningsson 22 Nov '13

22 Nov '13

current_headset_type should be of the HEADSET_TYPE enum, not the HEADSET_MODE enum. Since ALC_HEADSET_TYPE_UNKNOWN and ALC_HEADSET_MODE_UNKNOWN are both 0, this patch is just janitorial. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d8d9b3..5e42059 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3414,7 +3414,7 @@ static void alc_update_headset_mode_hook(struct hda_codec *codec, static void alc_update_headset_jack_cb(struct hda_codec *codec, struct hda_jack_tbl *jack) { struct alc_spec *spec = codec->spec; - spec->current_headset_type = ALC_HEADSET_MODE_UNKNOWN; + spec->current_headset_type = ALC_HEADSET_TYPE_UNKNOWN; snd_hda_gen_hp_automute(codec, jack); } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: wm8991: Use a supply to manage input power
by Mark Brown 22 Nov '13

22 Nov '13

From: Mark Brown <broonie(a)linaro.org> Instead of using a fake register and events to manage input power use a supply to do the job, saving code and preparing for regmap conversion of the driver. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8991.c | 53 +++++++++++++++-------------------------------- sound/soc/codecs/wm8991.h | 9 -------- 2 files changed, 17 insertions(+), 45 deletions(-) diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 3a39df7a3829..5078fc8e10f7 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -374,30 +374,6 @@ static const struct snd_kcontrol_new wm8991_snd_controls[] = { /* * _DAPM_ Controls */ -static int inmixer_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - u16 reg, fakepower; - - reg = snd_soc_read(w->codec, WM8991_POWER_MANAGEMENT_2); - fakepower = snd_soc_read(w->codec, WM8991_INTDRIVBITS); - - if (fakepower & ((1 << WM8991_INMIXL_PWR_BIT) | - (1 << WM8991_AINLMUX_PWR_BIT))) - reg |= WM8991_AINL_ENA; - else - reg &= ~WM8991_AINL_ENA; - - if (fakepower & ((1 << WM8991_INMIXR_PWR_BIT) | - (1 << WM8991_AINRMUX_PWR_BIT))) - reg |= WM8991_AINR_ENA; - else - reg &= ~WM8991_AINR_ENA; - - snd_soc_write(w->codec, WM8991_POWER_MANAGEMENT_2, reg); - return 0; -} - static int outmixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -655,6 +631,11 @@ static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RIN2"), SND_SOC_DAPM_INPUT("Internal ADC Source"), + SND_SOC_DAPM_SUPPLY("INL", WM8991_POWER_MANAGEMENT_2, + WM8991_AINL_ENA_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("INR", WM8991_POWER_MANAGEMENT_2, + WM8991_AINR_ENA_BIT, 0, NULL, 0), + /* DACs */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8991_POWER_MANAGEMENT_2, WM8991_ADCL_ENA_BIT, 0), @@ -676,26 +657,22 @@ static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = { ARRAY_SIZE(wm8991_dapm_rin34_pga_controls)), /* INMIXL */ - SND_SOC_DAPM_MIXER_E("INMIXL", WM8991_INTDRIVBITS, WM8991_INMIXL_PWR_BIT, 0, + SND_SOC_DAPM_MIXER("INMIXL", SND_SOC_NOPM, 0, 0, &wm8991_dapm_inmixl_controls[0], - ARRAY_SIZE(wm8991_dapm_inmixl_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8991_dapm_inmixl_controls)), /* AINLMUX */ - SND_SOC_DAPM_MUX_E("AINLMUX", WM8991_INTDRIVBITS, WM8991_AINLMUX_PWR_BIT, 0, - &wm8991_dapm_ainlmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("AINLMUX", SND_SOC_NOPM, 0, 0, + &wm8991_dapm_ainlmux_controls), /* INMIXR */ - SND_SOC_DAPM_MIXER_E("INMIXR", WM8991_INTDRIVBITS, WM8991_INMIXR_PWR_BIT, 0, + SND_SOC_DAPM_MIXER("INMIXR", SND_SOC_NOPM, 0, 0, &wm8991_dapm_inmixr_controls[0], - ARRAY_SIZE(wm8991_dapm_inmixr_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8991_dapm_inmixr_controls)), /* AINRMUX */ - SND_SOC_DAPM_MUX_E("AINRMUX", WM8991_INTDRIVBITS, WM8991_AINRMUX_PWR_BIT, 0, - &wm8991_dapm_ainrmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX("AINRMUX", SND_SOC_NOPM, 0, 0, + &wm8991_dapm_ainrmux_controls), /* Output Side */ /* DACs */ @@ -797,6 +774,10 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Right ADC", NULL, "Internal ADC Source"}, /* Input Side */ + {"INMIXL", NULL, "INL"}, + {"AINLMUX", NULL, "INL"}, + {"INMIXR", NULL, "INR"}, + {"AINRMUX", NULL, "INR"}, /* LIN12 PGA */ {"LIN12 PGA", "LIN1 Switch", "LIN1"}, {"LIN12 PGA", "LIN2 Switch", "LIN2"}, diff --git a/sound/soc/codecs/wm8991.h b/sound/soc/codecs/wm8991.h index 07707d8d7e20..08ed383303c0 100644 --- a/sound/soc/codecs/wm8991.h +++ b/sound/soc/codecs/wm8991.h @@ -76,7 +76,6 @@ #define WM8991_PLL1 0x3C #define WM8991_PLL2 0x3D #define WM8991_PLL3 0x3E -#define WM8991_INTDRIVBITS 0x3F #define WM8991_REGISTER_COUNT 60 #define WM8991_MAX_REGISTER 0x3F @@ -807,14 +806,6 @@ */ #define WM8991_PLLK2_MASK 0x00FF /* PLLK2 - [7:0] */ -/* - * R63 (0x3F) - Internal Driver Bits - */ -#define WM8991_INMIXL_PWR_BIT 0 -#define WM8991_AINLMUX_PWR_BIT 1 -#define WM8991_INMIXR_PWR_BIT 2 -#define WM8991_AINRMUX_PWR_BIT 3 - #define WM8991_MCLK_DIV 0 #define WM8991_DACCLK_DIV 1 #define WM8991_ADCCLK_DIV 2 -- 1.8.4.4

2 1

[alsa-devel] Audio Mini Conference Minutes - Edinburgh 21st October 2013
by Liam Girdwood 22 Nov '13

22 Nov '13

I've pasted below the minutes from this years ALSA developer mini conference in Edinburgh. I'll also add it to the developer Wiki alongside some slides from Eric and Tanu (once I know the correct answer to the anti-spam captcha). Fwiw, there are some interesting "unclaimed" tasks/projects below. Please feel free to volunteer for anything below that may interest you. Btw, if you are replying to start a thread about a certain subject below, please snip the other context and change the email to match the new $SUBJECT. This should make things easier to track. Thanks Liam Audio BOF - Monday October 21st 2013 - Edinburgh ================================================ Attendees :- ----------------------------------------------------------------------- Mark Brown, Liam Girdwood, Pierre Bossart, Patrick Lai, Lars-Peter Clausen, Eric Laurent, David Henningsson, Takashi Iwai, Colin Guthrie, Tanu Kaskinen, Arun Raghavan, Paul Handrigan, Daniel Mack, Dylan Reid, Sven Neuman, Dmimtris Papastamos, Charles Keepax, Nariman Poushin, João Paulo Rechi Vita, Alexander Patrakov, Micheal Wu, Lennart Poettering Topics ======= Pulseaudio route management and UCM ----------------------------------- There was interest for shared policy configuration file format across different systems. No such sharing has been planned. Tanu's initial reaction is that it's not feasible, but he is open to discussing it. What to do if there are multiple paths from A to B? One will be chosen, either arbitrarily or based on connection requirements or prioritization. How to control effect offloading? Tanu's vague idea is that if some kind of filtering is required/preferred, it's expressed as a requirement for a connection, and the node implementation will then enable the details of enabling the effect if such effect is available. Tanu doesn't know how well that would work in practice - and probably unaware of many details that would have to be taken into account. UCM being used by some people but missing good documentation. Liam to write documentation and use examples from Arun as "gold standard" example configurations. More UCM configs to go online in alsa-lib repo. UCM configs sometimes need updated for new kernel releases as Kcontrol strings have changed. Suggestions made for additional UUID for each kcontrol to lessen UCM config name changes (e.g. vendor-part-reg-shift-mask could be used as ASoC kcontrol UUID). Naming changes also effect HDA kcontrols, but no HDA UCM configs yet ? kcontrol UUID harder for DSP and HDA. Arun reported UCM issue with combination of devices. e.g. Enable dev 1 & 2 then disable dev 1 & 2. Maybe virtual device that combines devices ?? #include for UCM - should be easy to add. Simplifies config for family of similar devices like Panda, PandaES etc Device Names in UCM names. There are some standard names in UCM header but need more. Liam to add more names. Arbitration, UCM does not do arbitration but leaves that up to sound server. Controls names need more standardisation. Kcontrol naming mainly based on HW specified within data sheets. Difficult for userspace to cope with great degree of naming variations and provide enough information for userspace to know which volume is master etc. Audio Delay and Timing granularity ---------------------------------- Some DMA drivers very bad at providing position information used to work out buffer positions. Difficult for userspace to render audio and wake up correctly with such HW unless we can provide mechanism for userspace to know the DMA position accuracy. IOCTL suggested, info is in driver so easy to expose. There already is the SNDRV_PCM_INFO_BATCH flag which is supposed to indicated that the granularity is period-size-ish. It makes sense for applications like pulse-audio to check whether this flag is set or not. Hostless Audio -------------- Hostless should now use kcontrol to join the graph links to enable or disable and not hostless PCM device (old Android hack). e.g. FM playback loopback to codec, Modem PCM handling. i.e. CODEC <-> CODEC type. Need mixer to expose set config changes as link config is static e.g. WB/NB configs. Audio Offloading for Android ---------------------------- Avail in KK, supports > 1 compressed devices. No guarantee on no dropped samples when changing compressed to PCM and vice versa (or enabling effects ????) Not available yet when in use with video, no tech reason to stop this. AV sync good. Some effects can be offloaded and some not. Compressed API -------------- Intel to upstream remaining fixes. AR - Liam, Vinod. Stable updates with no users. 3.10 is stable but we can changes core API since there are no users. Android Audio Effects Framework Modifications --------------------------------------------- Composite with HW and SW representations. Supports 3rd party GUI controls but they must use the effect API and not bypass with private IOCTLs. How do we do library, like tiny effects ? API exists in use by Factory, can we extend to alsa ? Nobody using compressed and effects outside of Android atm. 24 Bit HiRes Audio for Android ------------------------------ How do we handle 24 bit audio at fast rates e.g. 192kHz. Being pushed by market. Flac suggested as source media and not PCM. We could use offload, on roadmap for Hires audio on android PCM, maybe float instead of 24 bit. Need for 192kHz, Probably not use effects. Effect Support in ALSA ---------------------- Several proposals for APIs. Control device type, do we need new control or use existing CTL ? Per card or per DSP / CODEC. ? Not to use control API for everything , could confuse some userspace. Get/set could be any control type, not just binary ? Binary blobs used to hide workings of effects. Topology Query and edit - query the graph from HDA/DAPM - used to establish connections and end points. per card based IOCTL. Could be new alsamixer to allow users to control volumes. Media controller API. How do we match to ALSA. Need non LGPL library for android. Missing certain features, someone needs to write code. MC API is Video oriented, library is just wrappers around IOCTLs, 1 man week of simple work to implement non LGPL version. Removal/creation - kernel can provide any type of information to user space key values. Document new keys, generic structure. Channel mapping possible using tuples. Need to support dynamic entity information. Handles static pipeline well, and can also expose dynamic with some complexity (with more entities). Abstract away sink+sources in userspace library, maybe too complex. Trivial to use UUIDs for dynamic entities. Start with prototype and see what happens. Action on Intel/Vinod ? Topology 2 NODES and EFFECT IOCTLS could be replaced with media controller as MediaC provides good match. Sending/Receiving Binary data to drivers ---------------------------------------- Use cases that go above 256/512 ? byte limit for binary controls. Extend for larger ? Do something about it. Action for ????? Tiwai/Mark ???? MMAP for some parameters. Visualizer, peak meters. ? Multiple of 4k pages. Can be used for DMA buffer positions on IPC IRQ based DSPs. Avoid copying large amounts of data. trade off for DSP. Usable for DEBUG too. Prototype and see what happens - Who ?? Expose API for sending large chunks of data e.g. 2MB. Nice for generic code to send this data (no special tooling) Needs new API for large data chunks. What about write(). Patrick to prototype some code. DSP resource management ----------------------- Do we need pre-trigger to work out MIPS/MHz + buffer resources for DSP resources. Tables used atm for most DSPs. Something for policy or UCM to specify ? Could change for different FWs. Peak and average loading gives difficulty. Might not get feedback, latency in the pipe and need to agree on units of measurements. Complex problem. Estimate worst case. OMAP4 example uses DAPM, expose reource usage numbers, expose capability limits. Liam to publish clock resource management logic for Haswell DSP. Can be used as starting point for investigations into feasibility etc. HDMI - Hotplug -------------- Can cause changes to audio path. Do we need to tell userspace card has changed ? Specific for ELD or generic problem ? Suggestion to remove and then insert card not usable Signal for monitor re-pluging, ELD not available right away. How to sync ELD information, ELD time out signal ? DSPs can change capabilities, ELDS can be garbage. PA maybe ask to quickly for ELD info, expose new jack for HDA when ELD is known. Generic event to tell userspace that card has changed. HDA/MIMI specific solution. Generic API to talk between audio and DRM devices. Bespoke atm. Code lives in the wrong place atm. Need to sync. Code is already upstream. David to prototype a solution based on deferring the probe until ELD is ready. Mic/LED hotkey -------------- Few different scenarios - from slides. Multiple cards and multiple mics. Which Mic will LED represent ? card or system or other. We'll not use the LED class because it does not buy us anything: LED class is writable for root only, and PA does not run as root. Nobody seemed to have a firm convincing argument in any direction. As such, keeping the current behaviour (reflect mute state of internal card) will be kept for now. Alexander suggested looking at what NetworkManager does with the wifi-off indicator in the presence of multiple wireless devices with (potentially) their own wifi-off switches. Performance - Use case startup ------------------------------ Discussing optimisation from DPCM - users QC, TI and Intel. DSP can route to multiple endpoints from a single PCM. Startup call back -> PCM -> FE -> BE. Then hw_params(), followed by prepare() and trigger(). Cold start latency 113ms on QDSP since some slow ops on QDSP are serialised. Solution to shift some prepare() work into DAPM. DAPM ops are tun in parallel and a workq. Compressed offload ------------------ Video/Audio offload - pass audio timestamp. Metadata for QDSP stored at front of compressed data by QC. Transport stream already supported by compressed API. Timestamps may differ in DSP implementations. AV sync topic requested by V4Linux for long time. Only discussed for PCM not offload, need TS for offload and support of V4Linux guys. Compressed offload transcoding. - How do we specify this transform. New API for userspace to spec output of decoder. All sorts of routing combinations are possible making implementation very difficult. Query the the BE connected to the FE, we may be able to call existing PCM API's on the BE PCM. Patrick to prototype some code. Firmware based Kcontrols and Graphs ----------------------------------- Patches have been developed to attach Kcontrol and Graph meta data to DSP FW. This allows single DSP driver to be used with multiple FWs without modification. Patch V1 reviewed last year before TI layoff and we should be ready for V2 before end of year. Consists of kernel patch and userspace tool to generate the control and graph data. Current userspace tool is needing work to improve config generation. There is no config file format at present. Paul and some Wolfson guys have volunteered to assist in this effort. Code currently works on OMAP4 platforms. Kernel code is here (top 5 patches) :- https://git.kernel.org/cgit/linux/kernel/git/lrg/asoc.git/log/?h=topic/firm… Userspace tool is here :- git@gitorious.org:omap-audio/asoc-fw.git Clean up ALSA code ------------------ Move plugins to their projects - move ffmpeg plugin to ffmpeg project, etc. Drivers, requests to move to device probe model, away from alsa model. Need to register device files at end of driver creation. Input susbsys uses possible method that could be copied. sysfs files need sync to avoid races for creation. Agreement. Process, How do we work ----------------------- Bug reports. Who looks at bugs. No takers. New mailing lists ? no, just use filters to hide emails that are not of interest. Misc ---- Softvol, add SND_CTL for no softvol in ctl open. Use alsa conf file or ctl. Arun encouraged to revive the email thread on the topic. Possible bug where specially crafted javascript can set PA volume to 100%. Defined pulseaudio behviour (Colin), related to flat volumes. New ASoC drivers must now use generic DMA engine PCM driver. Non atomic trigger requirments from DSP drivers. DSPs mostly sleep when performing IPC and need a trigger call that can sleep. Patrick will investigate. Propagate errors from BEs to FEs. Patrick to investigate. This will also include reporting underruns too. System restart for DSP FW crashes. This requires reloading all FW and trying to recover all active streams. Patrick to investigate. Lennart asked about the support of audio "mute" via systemd session management, like input and drm drivers already do. Takashi took a look at input evdev code, and actually it's revoking instead of muting. The advantage of revoking over muting is that it's simpler and doesn't need the capability check, as far as he read through discussion threads. But Takashi is not sure whether revoking behavior is good for PA at all.

4 6

[alsa-devel] [PATCH] ALSA: hda - Provide missing pin configs for VAIO with ALC260
by Takashi Iwai 22 Nov '13

22 Nov '13

Some models (or maybe depending on BIOS version) of Sony VAIO with ALC260 give no proper pin configurations as default, resulting in the non-working speaker, etc. Just provide the whole pin configurations via a fixup. Reported-by: Matthew Markus <mmarkus(a)hearit.co> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bb653dd0f813..6d8d9b3520d0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1512,6 +1512,7 @@ enum { ALC260_FIXUP_KN1, ALC260_FIXUP_FSC_S7020, ALC260_FIXUP_FSC_S7020_JWSE, + ALC260_FIXUP_VAIO_PINS, }; static void alc260_gpio1_automute(struct hda_codec *codec) @@ -1652,6 +1653,24 @@ static const struct hda_fixup alc260_fixups[] = { .chained = true, .chain_id = ALC260_FIXUP_FSC_S7020, }, + [ALC260_FIXUP_VAIO_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* Pin configs are missing completely on some VAIOs */ + { 0x0f, 0x01211020 }, + { 0x10, 0x0001003f }, + { 0x11, 0x411111f0 }, + { 0x12, 0x01a15930 }, + { 0x13, 0x411111f0 }, + { 0x14, 0x411111f0 }, + { 0x15, 0x411111f0 }, + { 0x16, 0x411111f0 }, + { 0x17, 0x411111f0 }, + { 0x18, 0x411111f0 }, + { 0x19, 0x411111f0 }, + { } + } + }, }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { @@ -1660,6 +1679,7 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900), + SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_FIXUP_VAIO_PINS), SND_PCI_QUIRK(0x104d, 0x81e2, "Sony VAIO TX", ALC260_FIXUP_HP_PIN_0F), SND_PCI_QUIRK(0x10cf, 0x1326, "FSC LifeBook S7020", ALC260_FIXUP_FSC_S7020), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1), -- 1.8.4.3

1 0

[alsa-devel] acceptable noise level for mic boost
by Oleksij Rempel 21 Nov '13

21 Nov '13

Hello Takashi, what is acceptable noise level for mic boost? My laptop has three levels of mic boost (+36dB). Level 3 is useless since noise level will be about 0db. With level 1 i get -12dB noise, so there is almost 12dB room for data... but it is not enough for normal speech, which is about 20dB. Are there any ALSA guide for HW designers, which will say some thing like acceptable levels are 90-20dB? There are some Skype and Microsoft prescriptions for certification. For example M$ prescribe 18dB or more for speech to noise ration for build in mics. It sound like good number to start. In my case, i need to remove mic boost completely to guarantee minimum 18dB SpNR. Suddenly i didn't found correct way to do it. I will be thankful for some tip. -- Regards, Oleksij

2 6

[alsa-devel] [PATCH v2 4/4]: ALSA: oxygen: The new Xonar DG mixer implementation
by Roman Volkov 21 Nov '13

21 Nov '13

The mixer now at the upper level of hierarchy and actually controls the Xonar DG driver. There are also other system structures like model description structure needed for upper-level oxygen driver. Headphone volume control is implemented in the mixer instead of selecting the HP impedance (I have my own Xonar DG card and that's okay). Independent volume and mute control of each capture source. Signed-off-by: Roman I. Volkov <v1ron(a)mail.ru> --- diff -uprN linux-3.12/sound/pci/oxygen/Makefile linux-3.12-my/sound/pci/oxygen/Makefile --- linux-3.12/sound/pci/oxygen/Makefile 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/Makefile 2013-11-20 19:04:54.000000000 +0400 @@ -1,5 +1,5 @@ snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o -snd-oxygen-objs := oxygen.o xonar_dg.o +snd-oxygen-objs := oxygen.o xonar_dg_mixer.o xonar_dg.o snd-virtuoso-objs := virtuoso.o xonar_lib.o \ xonar_pcm179x.o xonar_cs43xx.o xonar_wm87x6.o xonar_hdmi.o diff -uprN linux-3.12/sound/pci/oxygen/xonar_dg_mixer.c linux-3.12-my/sound/pci/oxygen/xonar_dg_mixer.c --- linux-3.12/sound/pci/oxygen/xonar_dg_mixer.c 1970-01-01 03:00:00.000000000 +0300 +++ linux-3.12-my/sound/pci/oxygen/xonar_dg_mixer.c 2013-11-21 14:05:55.000000000 +0400 @@ -0,0 +1,483 @@ +/* + * Mixer controls for the Xonar DG/DGX + * + * Copyright (c) Clemens Ladisch <clemens(a)ladisch.de> + * Copyright (c) Roman Volkov <v1ron(a)mail.ru> + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver 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. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, see <http://www.gnu.org/licenses/>. + */ + +/* + * It is not good that we do not have software volume control + * as the Windows driver does. CS4361 cannot attenuate the volume. + * When we are switching from headphones to the speakers, + * which are controlled by the simple DAC CS4361, our headphones + * in the rear jack can stun us. + */ + +#include <linux/pci.h> +#include <linux/delay.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/pcm.h> +#include <sound/tlv.h> +#include "oxygen.h" +#include "xonar_dg.h" +#include "cs4245.h" + +/* CS4245 Headphone Channels A&B Volume Control */ +static int xonar_dg_dac_stereo_volume_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = 0; + info->value.integer.max = 255; + return 0; +} + +static int xonar_dg_dac_stereo_volume_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + unsigned int tmp; + + mutex_lock(&chip->mutex); + tmp = (~data->cs4245_shadow[CS4245_DAC_A_CTRL]) & 255; + val->value.integer.value[0] = tmp; + tmp = (~data->cs4245_shadow[CS4245_DAC_B_CTRL]) & 255; + val->value.integer.value[1] = tmp; + mutex_unlock(&chip->mutex); + return 0; +} + +static int xonar_dg_dac_stereo_volume_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + if ((val->value.integer.value[0] > 255) || + (val->value.integer.value[0] < 0) || + (val->value.integer.value[1] > 255) || + (val->value.integer.value[1] < 0)) { + return -EINVAL; + } + mutex_lock(&chip->mutex); + data->cs4245_shadow[CS4245_DAC_A_CTRL] = ~val->value.integer.value[0]; + data->cs4245_shadow[CS4245_DAC_B_CTRL] = ~val->value.integer.value[1]; + cs4245_write_spi(chip, CS4245_DAC_A_CTRL); + cs4245_write_spi(chip, CS4245_DAC_B_CTRL); + mutex_unlock(&chip->mutex); + return 0; +} + +/* CS4245 DAC Mute */ +static int xonar_dg_dac_mute_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + val->value.integer.value[0] = ((data->cs4245_shadow[CS4245_DAC_CTRL_1] + & CS4245_MUTE_DAC) == 0); + mutex_unlock(&chip->mutex); + return 0; +} + +static int xonar_dg_dac_mute_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + if (val->value.integer.value[0] > 1) + return -EINVAL; + mutex_lock(&chip->mutex); + data->cs4245_shadow[CS4245_DAC_CTRL_1] &= ~CS4245_MUTE_DAC; + data->cs4245_shadow[CS4245_DAC_CTRL_1] |= + (~val->value.integer.value[0] << 2) & CS4245_MUTE_DAC; + cs4245_write_spi(chip, CS4245_DAC_CTRL_1); + mutex_unlock(&chip->mutex); + return 0; +} + +/* Xonar DG Output Select */ +static int xonar_dg_pcm_route_apply(struct oxygen *chip) +{ + struct dg *data = chip->model_data; + + data->cs4245_shadow[CS4245_SIGNAL_SEL] &= + ~CS4245_A_OUT_SEL_MASK; + if (data->pcm_output == PLAYBACK_DST_HP) { + /* Mute FP (aux output) amplifier, switch rear jack to CS4245 */ + oxygen_set_bits8(chip, OXYGEN_GPIO_DATA, GPIO_HP_REAR); + } else if (data->pcm_output == PLAYBACK_DST_HP_FP) { + /* + * Unmute FP amplifier, switch rear jack to CS4361, + * I2S ports 2,3,4 should be inactive + */ + oxygen_clear_bits8(chip, OXYGEN_GPIO_DATA, GPIO_HP_REAR); + data->cs4245_shadow[CS4245_SIGNAL_SEL] |= CS4245_A_OUT_SEL_DAC; + } else { + /* + * 2.0, 4.0, 5.1: switch to CS4361, mute FP amp., + * and change playback routing / DMA channels + */ + oxygen_clear_bits8(chip, OXYGEN_GPIO_DATA, GPIO_HP_REAR); + } + cs4245_write_spi(chip, CS4245_SIGNAL_SEL); + return 0; +} + +static int xonar_dg_pcm_route_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "Stereo Headphones", + "Stereo Headphones FP", + "Multichannel", + }; + + return snd_ctl_enum_info(info, 1, 3, names); +} + +static int xonar_dg_pcm_route_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + val->value.integer.value[0] = data->pcm_output; + mutex_unlock(&chip->mutex); + + return 0; +} + +static int xonar_dg_pcm_route_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + int ret; + + mutex_lock(&chip->mutex); + data->pcm_output = val->value.integer.value[0]; + ret = xonar_dg_pcm_route_apply(chip); + oxygen_update_dac_routing(chip); + mutex_unlock(&chip->mutex); + + return ret; +} + +/* CS4245 PGA Source */ +static int xonar_dg_pga_mute_reg_toggle(struct oxygen *chip, bool mute); +static int xonar_dg_pga_volume_reg_write(struct oxygen *chip, char left, + char right); + +static int xonar_dg_pga_source_apply(struct oxygen *chip) +{ + struct dg *data = chip->model_data; + + data->cs4245_shadow[CS4245_ANALOG_IN] &= ~CS4245_SEL_MASK; + if (data->pga_source == CAPTURE_SRC_FP_MIC) + data->cs4245_shadow[CS4245_ANALOG_IN] |= CS4245_SEL_INPUT_2; + else if (data->pga_source == CAPTURE_SRC_LINE) + data->cs4245_shadow[CS4245_ANALOG_IN] |= CS4245_SEL_INPUT_4; + else if (data->pga_source != CAPTURE_SRC_MIC) + data->cs4245_shadow[CS4245_ANALOG_IN] |= CS4245_SEL_INPUT_1; + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_INPUT_ROUTE); + return cs4245_write_spi(chip, CS4245_ANALOG_IN); +} + +static int xonar_dg_pga_source_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const ps_names[4] = { + "Microphone", + "Microphone FP", + "Line In", + "Aux" + }; + + return snd_ctl_enum_info(info, 1, 4, ps_names); +} + +static int xonar_dg_pga_source_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + val->value.integer.value[0] = data->pga_source; + mutex_unlock(&chip->mutex); + + return 0; +} + +/* Switch the source and change volume/mute */ +static int xonar_dg_pga_source_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + long route = val->value.integer.value[0]; + int ret; + + if ((route > CAPTURE_SRC_AUX) || (route < 0)) + return -EINVAL; + + mutex_lock(&chip->mutex); + data->pga_source = route; + ret = xonar_dg_pga_source_apply(chip); + if (ret < 0) + goto err; + ret = xonar_dg_pga_mute_reg_toggle(chip, data->cap_mute[route]); + if (ret < 0) + goto err; + ret = xonar_dg_pga_volume_reg_write(chip, + data->cap_vol[route][0], + data->cap_vol[route][1]); + if (ret < 0) + goto err; +err: + mutex_unlock(&chip->mutex); + + return ret; +} + +/* Common routines to access capture volume register - Mic, Line, Aux */ +static int xonar_dg_pga_volume_reg_write(struct oxygen *chip, char left, + char right) +{ + struct dg *data = chip->model_data; + int ret; + + data->cs4245_shadow[CS4245_PGA_A_CTRL] = left; + data->cs4245_shadow[CS4245_PGA_B_CTRL] = right; + ret = cs4245_write_spi(chip, CS4245_PGA_A_CTRL); + if (ret < 0) + return ret; + return cs4245_write_spi(chip, CS4245_PGA_B_CTRL); +} + +static int xonar_dg_pga_volume_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = -24; + info->value.integer.max = 24; + return 0; +} + +static int xonar_dg_pga_volume_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + val->value.integer.value[0] = data->cap_vol[ctl->private_value][0]; + val->value.integer.value[1] = data->cap_vol[ctl->private_value][1]; + mutex_unlock(&chip->mutex); + + return 0; +} + +static int xonar_dg_pga_volume_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + long left = val->value.integer.value[0]; + long right = val->value.integer.value[1]; + int ret = 0; + + if ((left > 24) || (left < -24) || (right > 24) || (right < -24)) + return -EINVAL; + + mutex_lock(&chip->mutex); + data->cap_vol[ctl->private_value][0] = left; + data->cap_vol[ctl->private_value][1] = right; + ret = xonar_dg_pga_volume_reg_write(chip, + data->cap_vol[data->pga_source][0], + data->cap_vol[data->pga_source][1]); + mutex_unlock(&chip->mutex); + + return ret; +} + +/* Common routines to access capture mute register - Mic, Line, Aux */ +static int xonar_dg_pga_mute_reg_toggle(struct oxygen *chip, bool mute) +{ + struct dg *data = chip->model_data; + + data->cs4245_shadow[CS4245_ADC_CTRL] &= ~CS4245_MUTE_ADC; + data->cs4245_shadow[CS4245_ADC_CTRL] |= ((mute == 0) << 2) & + CS4245_MUTE_ADC; + return cs4245_write_spi(chip, CS4245_ADC_CTRL); +} + +static int xonar_dg_pga_mute_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + val->value.integer.value[0] = (long)data->cap_mute[ctl->private_value]; + mutex_unlock(&chip->mutex); + + return 0; +} + +static int xonar_dg_pga_mute_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + long tmp = val->value.integer.value[0]; + int ret = 0; + + if ((tmp > 1) || (tmp < 0)) + return -EINVAL; + + mutex_lock(&chip->mutex); + data->cap_mute[ctl->private_value] = tmp != 0; + ret = xonar_dg_pga_mute_reg_toggle(chip, + data->cap_mute[data->pga_source]); + mutex_unlock(&chip->mutex); + + return ret; +} + +#define CAPTURE_VOLUME(xname, private) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = xonar_dg_pga_volume_info, \ + .get = xonar_dg_pga_volume_get, \ + .put = xonar_dg_pga_volume_put, \ + .tlv = { .p = xonar_dg_pga_db_scale, }, \ + .private_value = private, \ + } +#define CAPTURE_SWITCH(xname, private) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xonar_dg_pga_mute_get, \ + .put = xonar_dg_pga_mute_put, \ + .private_value = private, \ + } + +static const DECLARE_TLV_DB_MINMAX(xonar_dg_hp_db_scale, -12550, 0); +static const DECLARE_TLV_DB_MINMAX(xonar_dg_pga_db_scale, -1200, 1200); +static const struct snd_kcontrol_new xonar_dg_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = xonar_dg_dac_stereo_volume_info, + .get = xonar_dg_dac_stereo_volume_get, + .put = xonar_dg_dac_stereo_volume_put, + .tlv = { .p = xonar_dg_hp_db_scale, }, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = xonar_dg_dac_mute_get, + .put = xonar_dg_dac_mute_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output Playback Enum", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = xonar_dg_pcm_route_info, + .get = xonar_dg_pcm_route_get, + .put = xonar_dg_pcm_route_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Capture Enum", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = xonar_dg_pga_source_info, + .get = xonar_dg_pga_source_get, + .put = xonar_dg_pga_source_put, + }, + CAPTURE_VOLUME("Mic Capture Volume", CAPTURE_SRC_MIC), + CAPTURE_SWITCH("Mic Capture Switch", CAPTURE_SRC_MIC), + CAPTURE_VOLUME("FP Mic Capture Volume", CAPTURE_SRC_FP_MIC), + CAPTURE_SWITCH("FP Mic Capture Switch", CAPTURE_SRC_FP_MIC), + CAPTURE_VOLUME("Line Capture Volume", CAPTURE_SRC_LINE), + CAPTURE_SWITCH("Line Capture Switch", CAPTURE_SRC_LINE), + CAPTURE_VOLUME("Aux Capture Volume", CAPTURE_SRC_AUX), + CAPTURE_SWITCH("Aux Capture Switch", CAPTURE_SRC_AUX), +}; + +int xonar_dg_mixer_init(struct oxygen *chip) +{ + unsigned int i; + int ret; + for (i = 0; i < ARRAY_SIZE(xonar_dg_controls); i++) { + ret = snd_ctl_add(chip->card, + snd_ctl_new1(&xonar_dg_controls[i], chip)); + if (ret < 0) + return ret; + } + return 0; +} + +/* Disable built-in volume control, we have only headphone volume */ +static int xonar_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "Master Playback ", 16)) + return 1; + return 0; +} + +struct oxygen_model model_xonar_dg = { + .longname = "C-Media Oxygen HD Audio", + .chip = "CMI8786", + .init = dg_init, + .control_filter = xonar_control_filter, + .mixer_init = xonar_dg_mixer_init, + .cleanup = dg_cleanup, + .suspend = dg_suspend, + .resume = dg_resume, + .set_dac_params = set_cs4245_dac_params, + .set_adc_params = set_cs4245_adc_params, + .adjust_dac_routing = adjust_dg_dac_routing, + .dump_registers = dump_cs4245_registers, + .model_data_size = sizeof(struct dg), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_1 | + CAPTURE_1_FROM_SPDIF, + .dac_channels_pcm = 6, + .dac_channels_mixer = 0, + .function_flags = OXYGEN_FUNCTION_SPI, + .dac_mclks = OXYGEN_MCLKS(256, 128, 128), + .adc_mclks = OXYGEN_MCLKS(256, 128, 128), + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; -- Kind regards, Roman Volkov, v1ron(a)mail.ru

1 0

[alsa-devel] [PATCH v2 3/4]: ALSA: oxygen: Move the Xonar DG mixer code away from this file and actual driver changes
by Roman Volkov 21 Nov '13

21 Nov '13

Move the mixer controls and other system structures to a separate file. Also there are actual driver changes: Multichannel routing, CS4245 initialization, CS4245 SPI reading and writing, DAC/ADC settings, suspend/resume logic. Signed-off-by: Roman I. Volkov <v1ron(a)mail.ru> --- I can confirm, the driver also works with OXYGEN_FUNCTION_ENABLE_SPI_4_5 removed. diff -uprN linux-3.12/sound/pci/oxygen/oxygen_mixer.c linux-3.12-my/sound/pci/oxygen/oxygen_mixer.c --- linux-3.12/sound/pci/oxygen/oxygen_mixer.c 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/oxygen_mixer.c 2013-11-20 19:13:09.000000000 +0400 @@ -190,6 +190,7 @@ void oxygen_update_dac_routing(struct ox if (chip->model.update_center_lfe_mix) chip->model.update_center_lfe_mix(chip, chip->dac_routing > 2); } +EXPORT_SYMBOL(oxygen_update_dac_routing); static int upmix_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { diff -uprN linux-3.12/sound/pci/oxygen/cs4245.h linux-3.12-my/sound/pci/oxygen/cs4245.h --- linux-3.12/sound/pci/oxygen/cs4245.h 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/cs4245.h 2013-11-20 12:08:07.000000000 +0400 @@ -103,5 +103,6 @@ #define CS4245_ADC_UNDRFL 0x01 -#define CS4245_SPI_ADDRESS (0x9e << 16) -#define CS4245_SPI_WRITE (0 << 16) +#define CS4245_SPI_ADDRESS 0x9e +#define CS4245_SPI_WRITE 0 +#define CS4245_SPI_READ 1 diff -uprN linux-3.12/sound/pci/oxygen/xonar_dg.h linux-3.12-my/sound/pci/oxygen/xonar_dg.h --- linux-3.12/sound/pci/oxygen/xonar_dg.h 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/xonar_dg.h 2013-11-21 13:41:44.000000000 +0400 @@ -3,6 +3,51 @@ #include "oxygen.h" +#define GPIO_MAGIC 0x0008 +#define GPIO_HP_DETECT 0x0010 +#define GPIO_INPUT_ROUTE 0x0060 +#define GPIO_HP_REAR 0x0080 +#define GPIO_OUTPUT_ENABLE 0x0100 + +#define CAPTURE_SRC_MIC 0 +#define CAPTURE_SRC_FP_MIC 1 +#define CAPTURE_SRC_LINE 2 +#define CAPTURE_SRC_AUX 3 + +#define PLAYBACK_DST_HP 0 +#define PLAYBACK_DST_HP_FP 1 +#define PLAYBACK_DST_MULTICH 2 + extern struct oxygen_model model_xonar_dg; +/* Xonar DG Specific Data */ +struct dg { + unsigned char cs4245_shadow[16]; + /* Output select: headphone/speakers */ + unsigned char pcm_output; + /* Input select: Mic, FP Mic, Line, Aux */ + unsigned char pga_source; + /* Independent capture volume */ + char cap_vol[4][2]; + bool cap_mute[4]; +}; + +/* Xonar DG control routines */ +int cs4245_write_spi(struct oxygen *chip, u8 reg); +int cs4245_read_spi(struct oxygen *chip, u8 reg); +int cs4245_read_shadow(struct oxygen *chip); +int cs4245_write_shadow(struct oxygen *chip); +void dg_init(struct oxygen *chip); +void set_cs4245_dac_params(struct oxygen *chip, + struct snd_pcm_hw_params *params); +void set_cs4245_adc_params(struct oxygen *chip, + struct snd_pcm_hw_params *params); +unsigned int adjust_dg_dac_routing(struct oxygen *chip, + unsigned int play_routing); +void dump_cs4245_registers(struct oxygen *chip, + struct snd_info_buffer *buffer); +void dg_suspend(struct oxygen *chip); +void dg_resume(struct oxygen *chip); +void dg_cleanup(struct oxygen *chip); + #endif diff -uprN linux-3.12/sound/pci/oxygen/xonar_dg.c linux-3.12-my/sound/pci/oxygen/xonar_dg.c --- linux-3.12/sound/pci/oxygen/xonar_dg.c 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/xonar_dg.c 2013-11-21 16:35:02.000000000 +0400 @@ -1,8 +1,8 @@ /* - * card driver for the Xonar DG/DGX + * Card driver for the Xonar DG/DGX * * Copyright (c) Clemens Ladisch <clemens(a)ladisch.de> - * + * Copyright (c) Roman Volkov <v1ron(a)mail.ru> * * This driver is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License, version 2. @@ -20,29 +20,37 @@ * Xonar DG/DGX * ------------ * + * CS4245/CS4361 both will mute all outputs if any clock ratio of + * any I²S channel is invalid. + * * CMI8788: * * SPI 0 -> CS4245 * + * Playback: * I²S 1 -> CS4245 * I²S 2 -> CS4361 (center/LFE) * I²S 3 -> CS4361 (surround) * I²S 4 -> CS4361 (front) + * Capture: + * I²S ADC 1 <- CS4245 * * GPIO 3 <- ? * GPIO 4 <- headphone detect - * GPIO 5 -> route input jack to line-in (0) or mic-in (1) - * GPIO 6 -> route input jack to line-in (0) or mic-in (1) - * GPIO 7 -> enable rear headphone amp - * GPIO 8 -> enable output to speakers + * GPIO 5 -> ADC analog circuit power supply control for left channel (?) + * GPIO 6 -> ADC analog circuit power supply control for right channel (?) + * GPIO 7 -> switches green rear output jack between CS4245 / CS4361 first + * channel (mechanical relay) + * GPIO 8 -> DAC analog circuit power supply control (?) * * CS4245: * - * input 1 <- aux - * input 2 <- front mic - * input 4 <- line/mic + * input 0 <- Microphone + * input 1 <- Aux + * input 2 <- Front Panel Mic + * input 4 <- Line In * DAC out -> headphones - * aux out -> front panel headphones + * AUX out -> front panel headphones */ #include <linux/pci.h> @@ -56,553 +64,214 @@ #include "xonar_dg.h" #include "cs4245.h" -#define GPIO_MAGIC 0x0008 -#define GPIO_HP_DETECT 0x0010 -#define GPIO_INPUT_ROUTE 0x0060 -#define GPIO_HP_REAR 0x0080 -#define GPIO_OUTPUT_ENABLE 0x0100 - -struct dg { - unsigned int output_sel; - s8 input_vol[4][2]; - unsigned int input_sel; - u8 hp_vol_att; - u8 cs4245_regs[0x11]; -}; - -static void cs4245_write(struct oxygen *chip, unsigned int reg, u8 value) -{ - struct dg *data = chip->model_data; - - oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | - OXYGEN_SPI_DATA_LENGTH_3 | - OXYGEN_SPI_CLOCK_1280 | - (0 << OXYGEN_SPI_CODEC_SHIFT) | - OXYGEN_SPI_CEN_LATCH_CLOCK_HI, - CS4245_SPI_ADDRESS | - CS4245_SPI_WRITE | - (reg << 8) | value); - data->cs4245_regs[reg] = value; -} - -static void cs4245_write_cached(struct oxygen *chip, unsigned int reg, u8 value) -{ - struct dg *data = chip->model_data; - - if (value != data->cs4245_regs[reg]) - cs4245_write(chip, reg, value); -} - -static void cs4245_registers_init(struct oxygen *chip) -{ - struct dg *data = chip->model_data; - - cs4245_write(chip, CS4245_POWER_CTRL, CS4245_PDN); - cs4245_write(chip, CS4245_DAC_CTRL_1, - data->cs4245_regs[CS4245_DAC_CTRL_1]); - cs4245_write(chip, CS4245_ADC_CTRL, - data->cs4245_regs[CS4245_ADC_CTRL]); - cs4245_write(chip, CS4245_SIGNAL_SEL, - data->cs4245_regs[CS4245_SIGNAL_SEL]); - cs4245_write(chip, CS4245_PGA_B_CTRL, - data->cs4245_regs[CS4245_PGA_B_CTRL]); - cs4245_write(chip, CS4245_PGA_A_CTRL, - data->cs4245_regs[CS4245_PGA_A_CTRL]); - cs4245_write(chip, CS4245_ANALOG_IN, - data->cs4245_regs[CS4245_ANALOG_IN]); - cs4245_write(chip, CS4245_DAC_A_CTRL, - data->cs4245_regs[CS4245_DAC_A_CTRL]); - cs4245_write(chip, CS4245_DAC_B_CTRL, - data->cs4245_regs[CS4245_DAC_B_CTRL]); - cs4245_write(chip, CS4245_DAC_CTRL_2, - CS4245_DAC_SOFT | CS4245_DAC_ZERO | CS4245_INVERT_DAC); - cs4245_write(chip, CS4245_INT_MASK, 0); - cs4245_write(chip, CS4245_POWER_CTRL, 0); -} - -static void cs4245_init(struct oxygen *chip) +int cs4245_write_spi(struct oxygen *chip, u8 reg) { struct dg *data = chip->model_data; - data->cs4245_regs[CS4245_DAC_CTRL_1] = - CS4245_DAC_FM_SINGLE | CS4245_DAC_DIF_LJUST; - data->cs4245_regs[CS4245_ADC_CTRL] = - CS4245_ADC_FM_SINGLE | CS4245_ADC_DIF_LJUST; - data->cs4245_regs[CS4245_SIGNAL_SEL] = - CS4245_A_OUT_SEL_HIZ | CS4245_ASYNCH; - data->cs4245_regs[CS4245_PGA_B_CTRL] = 0; - data->cs4245_regs[CS4245_PGA_A_CTRL] = 0; - data->cs4245_regs[CS4245_ANALOG_IN] = - CS4245_PGA_SOFT | CS4245_PGA_ZERO | CS4245_SEL_INPUT_4; - data->cs4245_regs[CS4245_DAC_A_CTRL] = 0; - data->cs4245_regs[CS4245_DAC_B_CTRL] = 0; - cs4245_registers_init(chip); - snd_component_add(chip->card, "CS4245"); -} - -static void dg_output_enable(struct oxygen *chip) -{ - msleep(2500); - oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); + return oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_3 | OXYGEN_SPI_CLOCK_1280 | + (0 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_HI, + (((u32)CS4245_SPI_ADDRESS) << 16) | (((u32)reg) << 8) | + ((u32)data->cs4245_shadow[reg])); } -static void dg_init(struct oxygen *chip) +int cs4245_read_spi(struct oxygen *chip, u8 reg) { struct dg *data = chip->model_data; + int ret; - data->output_sel = 0; - data->input_sel = 3; - data->hp_vol_att = 2 * 16; + ret = oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CEN_LATCH_CLOCK_HI | + OXYGEN_SPI_CLOCK_1280 | (0 << OXYGEN_SPI_CODEC_SHIFT), + (CS4245_SPI_ADDRESS << 8) | reg); + if (ret < 0) + return ret; - cs4245_init(chip); - - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_MAGIC | GPIO_HP_DETECT); - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_INPUT_ROUTE | GPIO_HP_REAR | GPIO_OUTPUT_ENABLE); - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, - GPIO_INPUT_ROUTE | GPIO_HP_REAR); - dg_output_enable(chip); -} - -static void dg_cleanup(struct oxygen *chip) -{ - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); -} - -static void dg_suspend(struct oxygen *chip) -{ - dg_cleanup(chip); -} - -static void dg_resume(struct oxygen *chip) -{ - cs4245_registers_init(chip); - dg_output_enable(chip); -} - -static void set_cs4245_dac_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ - struct dg *data = chip->model_data; - u8 value; - - value = data->cs4245_regs[CS4245_DAC_CTRL_1] & ~CS4245_DAC_FM_MASK; - if (params_rate(params) <= 50000) - value |= CS4245_DAC_FM_SINGLE; - else if (params_rate(params) <= 100000) - value |= CS4245_DAC_FM_DOUBLE; - else - value |= CS4245_DAC_FM_QUAD; - cs4245_write_cached(chip, CS4245_DAC_CTRL_1, value); -} - -static void set_cs4245_adc_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ - struct dg *data = chip->model_data; - u8 value; - - value = data->cs4245_regs[CS4245_ADC_CTRL] & ~CS4245_ADC_FM_MASK; - if (params_rate(params) <= 50000) - value |= CS4245_ADC_FM_SINGLE; - else if (params_rate(params) <= 100000) - value |= CS4245_ADC_FM_DOUBLE; - else - value |= CS4245_ADC_FM_QUAD; - cs4245_write_cached(chip, CS4245_ADC_CTRL, value); -} - -static inline unsigned int shift_bits(unsigned int value, - unsigned int shift_from, - unsigned int shift_to, - unsigned int mask) -{ - if (shift_from < shift_to) - return (value << (shift_to - shift_from)) & mask; - else - return (value >> (shift_from - shift_to)) & mask; -} - -static unsigned int adjust_dg_dac_routing(struct oxygen *chip, - unsigned int play_routing) -{ - return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) | - shift_bits(play_routing, - OXYGEN_PLAY_DAC2_SOURCE_SHIFT, - OXYGEN_PLAY_DAC1_SOURCE_SHIFT, - OXYGEN_PLAY_DAC1_SOURCE_MASK) | - shift_bits(play_routing, - OXYGEN_PLAY_DAC1_SOURCE_SHIFT, - OXYGEN_PLAY_DAC2_SOURCE_SHIFT, - OXYGEN_PLAY_DAC2_SOURCE_MASK) | - shift_bits(play_routing, - OXYGEN_PLAY_DAC0_SOURCE_SHIFT, - OXYGEN_PLAY_DAC3_SOURCE_SHIFT, - OXYGEN_PLAY_DAC3_SOURCE_MASK); -} - -static int output_switch_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *info) -{ - static const char *const names[3] = { - "Speakers", "Headphones", "FP Headphones" - }; - - return snd_ctl_enum_info(info, 1, 3, names); -} + ret = oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CEN_LATCH_CLOCK_HI | + OXYGEN_SPI_CLOCK_1280 | (0 << OXYGEN_SPI_CODEC_SHIFT), + (CS4245_SPI_ADDRESS | CS4245_SPI_READ) << 8); + if (ret < 0) + return ret; -static int output_switch_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - struct dg *data = chip->model_data; + data->cs4245_shadow[reg] = oxygen_read8(chip, OXYGEN_SPI_DATA1); - mutex_lock(&chip->mutex); - value->value.enumerated.item[0] = data->output_sel; - mutex_unlock(&chip->mutex); return 0; } -static int output_switch_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) +int cs4245_read_shadow(struct oxygen *chip) { - struct oxygen *chip = ctl->private_data; - struct dg *data = chip->model_data; - u8 reg; - int changed; - - if (value->value.enumerated.item[0] > 2) - return -EINVAL; + unsigned char address; + int ret; - mutex_lock(&chip->mutex); - changed = value->value.enumerated.item[0] != data->output_sel; - if (changed) { - data->output_sel = value->value.enumerated.item[0]; - - reg = data->cs4245_regs[CS4245_SIGNAL_SEL] & - ~CS4245_A_OUT_SEL_MASK; - reg |= data->output_sel == 2 ? - CS4245_A_OUT_SEL_DAC : CS4245_A_OUT_SEL_HIZ; - cs4245_write_cached(chip, CS4245_SIGNAL_SEL, reg); - - cs4245_write_cached(chip, CS4245_DAC_A_CTRL, - data->output_sel ? data->hp_vol_att : 0); - cs4245_write_cached(chip, CS4245_DAC_B_CTRL, - data->output_sel ? data->hp_vol_att : 0); - - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, - data->output_sel == 1 ? GPIO_HP_REAR : 0, - GPIO_HP_REAR); + for (address = 0; address <= 0x10; address++) { + ret = cs4245_read_spi(chip, address); + if (ret < 0) + return ret; } - mutex_unlock(&chip->mutex); - return changed; -} - -static int hp_volume_offset_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *info) -{ - static const char *const names[3] = { - "< 64 ohms", "64-150 ohms", "150-300 ohms" - }; - - return snd_ctl_enum_info(info, 1, 3, names); -} - -static int hp_volume_offset_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - struct dg *data = chip->model_data; - - mutex_lock(&chip->mutex); - if (data->hp_vol_att > 2 * 7) - value->value.enumerated.item[0] = 0; - else if (data->hp_vol_att > 0) - value->value.enumerated.item[0] = 1; - else - value->value.enumerated.item[0] = 2; - mutex_unlock(&chip->mutex); return 0; } -static int hp_volume_offset_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) +int cs4245_write_shadow(struct oxygen *chip) { - static const s8 atts[3] = { 2 * 16, 2 * 7, 0 }; - struct oxygen *chip = ctl->private_data; - struct dg *data = chip->model_data; - s8 att; - int changed; + unsigned char address; + int ret; - if (value->value.enumerated.item[0] > 2) - return -EINVAL; - att = atts[value->value.enumerated.item[0]]; - mutex_lock(&chip->mutex); - changed = att != data->hp_vol_att; - if (changed) { - data->hp_vol_att = att; - if (data->output_sel) { - cs4245_write_cached(chip, CS4245_DAC_A_CTRL, att); - cs4245_write_cached(chip, CS4245_DAC_B_CTRL, att); - } + for (address = 0; address <= 0x10; address++) { + ret = cs4245_write_spi(chip, address); + if (ret < 0) + return ret; } - mutex_unlock(&chip->mutex); - return changed; -} - -static int input_vol_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *info) -{ - info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - info->count = 2; - info->value.integer.min = 2 * -12; - info->value.integer.max = 2 * 12; - return 0; -} - -static int input_vol_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - struct dg *data = chip->model_data; - unsigned int idx = ctl->private_value; - - mutex_lock(&chip->mutex); - value->value.integer.value[0] = data->input_vol[idx][0]; - value->value.integer.value[1] = data->input_vol[idx][1]; - mutex_unlock(&chip->mutex); return 0; } -static int input_vol_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) +static void cs4245_init(struct oxygen *chip) { - struct oxygen *chip = ctl->private_data; struct dg *data = chip->model_data; - unsigned int idx = ctl->private_value; - int changed = 0; - if (value->value.integer.value[0] < 2 * -12 || - value->value.integer.value[0] > 2 * 12 || - value->value.integer.value[1] < 2 * -12 || - value->value.integer.value[1] > 2 * 12) - return -EINVAL; - mutex_lock(&chip->mutex); - changed = data->input_vol[idx][0] != value->value.integer.value[0] || - data->input_vol[idx][1] != value->value.integer.value[1]; - if (changed) { - data->input_vol[idx][0] = value->value.integer.value[0]; - data->input_vol[idx][1] = value->value.integer.value[1]; - if (idx == data->input_sel) { - cs4245_write_cached(chip, CS4245_PGA_A_CTRL, - data->input_vol[idx][0]); - cs4245_write_cached(chip, CS4245_PGA_B_CTRL, - data->input_vol[idx][1]); - } - } - mutex_unlock(&chip->mutex); - return changed; -} + /* save the initial state */ + cs4245_read_shadow(chip); -static DECLARE_TLV_DB_SCALE(cs4245_pga_db_scale, -1200, 50, 0); + /* + * Power up the CODEC internals, enable soft ramp & zero cross, work + * in async. mode, + * Enable AUX output from DAC. DAC output in the Windows driver is + * inverted. + */ + data->cs4245_shadow[CS4245_POWER_CTRL] = 0; + data->cs4245_shadow[CS4245_SIGNAL_SEL] = CS4245_A_OUT_SEL_DAC | + CS4245_ASYNCH; + data->cs4245_shadow[CS4245_DAC_CTRL_1] = 0; + data->cs4245_shadow[CS4245_DAC_CTRL_2] = CS4245_DAC_SOFT | + CS4245_DAC_ZERO | CS4245_INVERT_DAC; + data->cs4245_shadow[CS4245_ADC_CTRL] = 0; + data->cs4245_shadow[CS4245_ANALOG_IN] = CS4245_PGA_SOFT | + CS4245_PGA_ZERO; -static int input_sel_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *info) -{ - static const char *const names[4] = { - "Mic", "Aux", "Front Mic", "Line" - }; + cs4245_write_shadow(chip); - return snd_ctl_enum_info(info, 1, 4, names); + snd_component_add(chip->card, "CS4245"); } -static int input_sel_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) +void dg_init(struct oxygen *chip) { - struct oxygen *chip = ctl->private_data; - struct dg *data = chip->model_data; + /* + * The pin XGPIO1 as XGPIO1, gpios 8,7,6,5,1,0 as outputs. + */ - mutex_lock(&chip->mutex); - value->value.enumerated.item[0] = data->input_sel; - mutex_unlock(&chip->mutex); - return 0; + cs4245_init(chip); + oxygen_write16(chip, OXYGEN_GPIO_CONTROL, GPIO_OUTPUT_ENABLE | + GPIO_HP_REAR | GPIO_INPUT_ROUTE); + oxygen_write16(chip, OXYGEN_GPIO_DATA, GPIO_HP_REAR | GPIO_INPUT_ROUTE); + /* Anti-pop delay */ + msleep(2500); + oxygen_write16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE | + GPIO_HP_REAR | GPIO_INPUT_ROUTE); } -static int input_sel_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) +void set_cs4245_dac_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) { - static const u8 sel_values[4] = { - CS4245_SEL_MIC, - CS4245_SEL_INPUT_1, - CS4245_SEL_INPUT_2, - CS4245_SEL_INPUT_4 - }; - struct oxygen *chip = ctl->private_data; struct dg *data = chip->model_data; - int changed; - - if (value->value.enumerated.item[0] > 3) - return -EINVAL; + unsigned char dac_ctrl; + unsigned char mclk_freq; - mutex_lock(&chip->mutex); - changed = value->value.enumerated.item[0] != data->input_sel; - if (changed) { - data->input_sel = value->value.enumerated.item[0]; - - cs4245_write(chip, CS4245_ANALOG_IN, - (data->cs4245_regs[CS4245_ANALOG_IN] & - ~CS4245_SEL_MASK) | - sel_values[data->input_sel]); - - cs4245_write_cached(chip, CS4245_PGA_A_CTRL, - data->input_vol[data->input_sel][0]); - cs4245_write_cached(chip, CS4245_PGA_B_CTRL, - data->input_vol[data->input_sel][1]); - - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, - data->input_sel ? 0 : GPIO_INPUT_ROUTE, - GPIO_INPUT_ROUTE); + dac_ctrl = data->cs4245_shadow[CS4245_DAC_CTRL_1] & ~CS4245_DAC_FM_MASK; + mclk_freq = data->cs4245_shadow[CS4245_MCLK_FREQ] & ~CS4245_MCLK1_MASK; + if (params_rate(params) <= 50000) { + dac_ctrl |= CS4245_DAC_FM_SINGLE; + mclk_freq |= CS4245_MCLK_1 << CS4245_MCLK1_SHIFT; + } else if (params_rate(params) <= 100000) { + dac_ctrl |= CS4245_DAC_FM_DOUBLE; + mclk_freq |= CS4245_MCLK_1 << CS4245_MCLK1_SHIFT; + } else { + dac_ctrl |= CS4245_DAC_FM_QUAD; + mclk_freq |= CS4245_MCLK_2 << CS4245_MCLK1_SHIFT; } - mutex_unlock(&chip->mutex); - return changed; + data->cs4245_shadow[CS4245_DAC_CTRL_1] = dac_ctrl; + data->cs4245_shadow[CS4245_MCLK_FREQ] = mclk_freq; + cs4245_write_spi(chip, CS4245_DAC_CTRL_1); + cs4245_write_spi(chip, CS4245_MCLK_FREQ); } -static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +void set_cs4245_adc_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) { - static const char *const names[2] = { "Active", "Frozen" }; + struct dg *data = chip->model_data; + unsigned char adc_ctrl; + unsigned char mclk_freq; - return snd_ctl_enum_info(info, 1, 2, names); + adc_ctrl = data->cs4245_shadow[CS4245_ADC_CTRL] & ~CS4245_ADC_FM_MASK; + mclk_freq = data->cs4245_shadow[CS4245_MCLK_FREQ] & ~CS4245_MCLK2_MASK; + if (params_rate(params) <= 50000) { + adc_ctrl |= CS4245_ADC_FM_SINGLE; + mclk_freq |= CS4245_MCLK_1 << CS4245_MCLK2_SHIFT; + } else if (params_rate(params) <= 100000) { + adc_ctrl |= CS4245_ADC_FM_DOUBLE; + mclk_freq |= CS4245_MCLK_1 << CS4245_MCLK2_SHIFT; + } else { + adc_ctrl |= CS4245_ADC_FM_QUAD; + mclk_freq |= CS4245_MCLK_2 << CS4245_MCLK2_SHIFT; + } + data->cs4245_shadow[CS4245_ADC_CTRL] = adc_ctrl; + data->cs4245_shadow[CS4245_MCLK_FREQ] = mclk_freq; + cs4245_write_spi(chip, CS4245_ADC_CTRL); + cs4245_write_spi(chip, CS4245_MCLK_FREQ); } -static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +unsigned int adjust_dg_dac_routing(struct oxygen *chip, + unsigned int play_routing) { - struct oxygen *chip = ctl->private_data; struct dg *data = chip->model_data; + unsigned int routing = 0; - value->value.enumerated.item[0] = - !!(data->cs4245_regs[CS4245_ADC_CTRL] & CS4245_HPF_FREEZE); - return 0; + switch (data->pcm_output) { + case PLAYBACK_DST_HP: + case PLAYBACK_DST_HP_FP: + oxygen_write8_masked(chip, OXYGEN_PLAY_ROUTING, + OXYGEN_PLAY_MUTE23 | OXYGEN_PLAY_MUTE45 | + OXYGEN_PLAY_MUTE67, OXYGEN_PLAY_MUTE_MASK); + break; + case PLAYBACK_DST_MULTICH: + routing = (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (2 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (1 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT); + oxygen_write8_masked(chip, OXYGEN_PLAY_ROUTING, + OXYGEN_PLAY_MUTE01, OXYGEN_PLAY_MUTE_MASK); + break; + } + return routing; } -static int hpf_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +void dump_cs4245_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) { - struct oxygen *chip = ctl->private_data; struct dg *data = chip->model_data; - u8 reg; - int changed; + unsigned int reg; - mutex_lock(&chip->mutex); - reg = data->cs4245_regs[CS4245_ADC_CTRL] & ~CS4245_HPF_FREEZE; - if (value->value.enumerated.item[0]) - reg |= CS4245_HPF_FREEZE; - changed = reg != data->cs4245_regs[CS4245_ADC_CTRL]; - if (changed) - cs4245_write(chip, CS4245_ADC_CTRL, reg); - mutex_unlock(&chip->mutex); - return changed; -} - -#define INPUT_VOLUME(xname, index) { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .info = input_vol_info, \ - .get = input_vol_get, \ - .put = input_vol_put, \ - .tlv = { .p = cs4245_pga_db_scale }, \ - .private_value = index, \ -} -static const struct snd_kcontrol_new dg_controls[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Output Playback Enum", - .info = output_switch_info, - .get = output_switch_get, - .put = output_switch_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Headphones Impedance Playback Enum", - .info = hp_volume_offset_info, - .get = hp_volume_offset_get, - .put = hp_volume_offset_put, - }, - INPUT_VOLUME("Mic Capture Volume", 0), - INPUT_VOLUME("Aux Capture Volume", 1), - INPUT_VOLUME("Front Mic Capture Volume", 2), - INPUT_VOLUME("Line Capture Volume", 3), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = input_sel_info, - .get = input_sel_get, - .put = input_sel_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "ADC High-pass Filter Capture Enum", - .info = hpf_info, - .get = hpf_get, - .put = hpf_put, - }, -}; + snd_iprintf(buffer, "\nCS4245:"); + cs4245_read_spi(chip, CS4245_INT_STATUS); + for (reg = 0; reg < ARRAY_SIZE(data->cs4245_shadow); ++reg) + snd_iprintf(buffer, " %02x", data->cs4245_shadow[reg]); + snd_iprintf(buffer, "\n"); +} -static int dg_control_filter(struct snd_kcontrol_new *template) +void dg_suspend(struct oxygen *chip) { - if (!strncmp(template->name, "Master Playback ", 16)) - return 1; - return 0; + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); } -static int dg_mixer_init(struct oxygen *chip) +void dg_resume(struct oxygen *chip) { - unsigned int i; - int err; - - for (i = 0; i < ARRAY_SIZE(dg_controls); ++i) { - err = snd_ctl_add(chip->card, - snd_ctl_new1(&dg_controls[i], chip)); - if (err < 0) - return err; - } - return 0; + cs4245_write_shadow(chip); + msleep(2500); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); } -static void dump_cs4245_registers(struct oxygen *chip, - struct snd_info_buffer *buffer) +void dg_cleanup(struct oxygen *chip) { - struct dg *data = chip->model_data; - unsigned int i; - - snd_iprintf(buffer, "\nCS4245:"); - for (i = 1; i <= 0x10; ++i) - snd_iprintf(buffer, " %02x", data->cs4245_regs[i]); - snd_iprintf(buffer, "\n"); + dg_suspend(chip); } - -struct oxygen_model model_xonar_dg = { - .longname = "C-Media Oxygen HD Audio", - .chip = "CMI8786", - .init = dg_init, - .control_filter = dg_control_filter, - .mixer_init = dg_mixer_init, - .cleanup = dg_cleanup, - .suspend = dg_suspend, - .resume = dg_resume, - .set_dac_params = set_cs4245_dac_params, - .set_adc_params = set_cs4245_adc_params, - .adjust_dac_routing = adjust_dg_dac_routing, - .dump_registers = dump_cs4245_registers, - .model_data_size = sizeof(struct dg), - .device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2 | - CAPTURE_1_FROM_SPDIF, - .dac_channels_pcm = 6, - .dac_channels_mixer = 0, - .function_flags = OXYGEN_FUNCTION_SPI, - .dac_mclks = OXYGEN_MCLKS(256, 128, 128), - .adc_mclks = OXYGEN_MCLKS(256, 128, 128), - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, - .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, -}; -- Kind regards, Roman Volkov, v1ron(a)mail.ru

1 0

[alsa-devel] [PATCH v2 2/4]: ALSA: oxygen: Add mute mask to the header file
by Roman Volkov 21 Nov '13

21 Nov '13

I need this mask to mute unnecessary channels Signed-off-by: Roman I. Volkov <v1ron(a)mail.ru> --- diff -uprN linux-3.12/sound/pci/oxygen/oxygen_regs.h linux-3.12-my/sound/pci/oxygen/oxygen_regs.h --- linux-3.12/sound/pci/oxygen/oxygen_regs.h 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/oxygen_regs.h 2013-11-20 12:08:07.000000000 +0400 @@ -318,6 +318,7 @@ #define OXYGEN_PLAY_MUTE23 0x0002 #define OXYGEN_PLAY_MUTE45 0x0004 #define OXYGEN_PLAY_MUTE67 0x0008 +#define OXYGEN_PLAY_MUTE_MASK 0x000f #define OXYGEN_PLAY_MULTICH_MASK 0x0010 #define OXYGEN_PLAY_MULTICH_I2S_DAC 0x0000 #define OXYGEN_PLAY_MULTICH_AC97 0x0010 -- Kind regards, Roman Volkov, v1ron(a)mail.ru

1 0

[alsa-devel] [PATCH v2 1/4] ALSA: oxygen: Change SPI writing/waiting logic
by Roman Volkov 21 Nov '13

21 Nov '13

Move waiting logic into separate function, wait after triggering the transaction (not before), increase the wait timeout, error handling. Should not affect other card models. Signed-off-by: Roman I. Volkov <v1ron(a)mail.ru> --- diff -uprN linux-3.12/sound/pci/oxygen/oxygen.h linux-3.12-my/sound/pci/oxygen/oxygen.h --- linux-3.12/sound/pci/oxygen/oxygen.h 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/oxygen.h 2013-11-20 22:20:44.000000000 +0400 @@ -198,7 +198,7 @@ void oxygen_write_ac97(struct oxygen *ch void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec, unsigned int index, u16 data, u16 mask); -void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data); +int oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data); void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data); void oxygen_reset_uart(struct oxygen *chip); diff -uprN linux-3.12/sound/pci/oxygen/oxygen_io.c linux-3.12-my/sound/pci/oxygen/oxygen_io.c --- linux-3.12/sound/pci/oxygen/oxygen_io.c 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/oxygen_io.c 2013-11-20 15:34:14.000000000 +0400 @@ -194,23 +194,34 @@ void oxygen_write_ac97_masked(struct oxy } EXPORT_SYMBOL(oxygen_write_ac97_masked); -void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) +static int oxygen_wait_spi(struct oxygen *chip) { - unsigned int count; + /* Higher interval to be sure - 400 microseconds */ + unsigned int count = 100; - /* should not need more than 30.72 us (24 * 1.28 us) */ - count = 10; - while ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & OXYGEN_SPI_BUSY) - && count > 0) { + for (;;) { + if ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & + OXYGEN_SPI_BUSY) == 0) + break; + if (count == 0) { + snd_printk(KERN_ERR "oxygen: spi wait timeout\n"); + return -EIO; + } udelay(4); - --count; + count--; } + return 0; +} +int oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) +{ oxygen_write8(chip, OXYGEN_SPI_DATA1, data); oxygen_write8(chip, OXYGEN_SPI_DATA2, data >> 8); if (control & OXYGEN_SPI_DATA_LENGTH_3) oxygen_write8(chip, OXYGEN_SPI_DATA3, data >> 16); oxygen_write8(chip, OXYGEN_SPI_CONTROL, control); + + return oxygen_wait_spi(chip); } EXPORT_SYMBOL(oxygen_write_spi); -- Kind regards, Roman Volkov, v1ron(a)mail.ru

1 0

Re: [alsa-devel] question about aplay fails on arm 64
by Clemens Ladisch 21 Nov '13

21 Nov '13

Please don't top-post. 曹能华 wrote: > Hi, Clemens Don't drop the mailing list from replies. > Do you mean we need to change ALSA's mmap code to adapt to ARM64. Maybe. > Do you know how to port it? No. Regards, Clemens

2 1

[alsa-devel] [PATCH] ALSA: hda - Add headset quirk for Dell Inspiron 3135
by David Henningsson 21 Nov '13

21 Nov '13

Cc: stable(a)vger.kernel.org (3.10+) BugLink: https://bugs.launchpad.net/bugs/1253636 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a90bfab..c08f435 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4152,6 +4152,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] wm8994: avoid to enable dcs done irq before init completion dcs_done
by wangbiao 21 Nov '13

21 Nov '13

From: wang, biao <biao.wang(a)intel.com> Date: Wed, 20 Nov 2013 17:53:28 +0800 Subject: [PATCH] wm8994: avoid to enable dcs done irq before init completion dcs_done once code return from request_threaded_irq, irq was setup enabled by default, but completion var dcs_done not got initialized yet, if then a dcs done irq was raised, system will got hung as the sync mechanism is invalid now. so this patch try not enable irq when request it until we got things ready. Signed-off-by: wang, biao <biao.wang(a)intel.com> Signed-off-by: Zhang, Di <di.zhang(a)intel.com> --- sound/soc/codecs/wm8994.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 86426a1..d7e59dd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3981,6 +3981,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) struct wm8994 *control = dev_get_drvdata(codec->dev->parent); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; + unsigned int dcs_done_irq; unsigned int reg; int ret, i; @@ -4077,6 +4078,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, wm8994_temp_shut, "Thermal shutdown", codec); + dcs_done_irq = regmap_irq_get_virq(wm8994->wm8994->irq_data, + WM8994_IRQ_DCS_DONE); + irq_set_status_flags(dcs_done_irq, IRQ_NOAUTOEN); ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_DCS_DONE, wm_hubs_dcs_done, "DC servo done", &wm8994->hubs); @@ -4313,6 +4317,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } wm_hubs_add_analogue_routes(codec, 0, 0); + enable_irq(dcs_done_irq); snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); switch (control->type) { -- 1.7.0.4

2 1

[alsa-devel] Bug? Sound support for at91sam9x5-wm8731 based boards
by Zhong Li 21 Nov '13

21 Nov '13

When playing stereo sound on the Atmel at91sam9x5-ek board the left and right channel would get swapped about 1 in 6 tries.

2 6

[alsa-devel] [PATCH] ALSA: hda - Fix the headphone jack detection on Sony VAIO TX
by Takashi Iwai 21 Nov '13

21 Nov '13

BIOS sets MISC_NO_PRESENCE bit wrongly to the pin config on NID 0x0f. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cc1dfdc1ff74..1c6f00802719 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1660,6 +1660,7 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900), + SND_PCI_QUIRK(0x104d, 0x81e2, "Sony VAIO TX", ALC260_FIXUP_HP_PIN_0F), SND_PCI_QUIRK(0x10cf, 0x1326, "FSC LifeBook S7020", ALC260_FIXUP_FSC_S7020), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x152d, 0x0729, "Quanta KN1", ALC260_FIXUP_KN1), -- 1.8.4.3

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix missing bass speaker on ASUS N550
by Takashi Iwai 21 Nov '13

21 Nov '13

The laptop has a built-in speaker on NID 0x1a. It's an LFE only on the right channel, so we need to provide an explicit chmap, too. There might be other surround speakers, but they can fixed in addition at later point, so let's fix the easier bass speaker at first. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65091 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a90bfab8530f..cc1dfdc1ff74 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4723,6 +4723,8 @@ enum { ALC668_FIXUP_DELL_MIC_NO_PRESENCE, ALC668_FIXUP_HEADSET_MODE, ALC662_FIXUP_BASS_CHMAP, + ALC662_FIXUP_BASS_1A, + ALC662_FIXUP_BASS_1A_CHMAP, }; static const struct hda_fixup alc662_fixups[] = { @@ -4903,6 +4905,19 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_ASUS_MODE4 }, + [ALC662_FIXUP_BASS_1A] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x1a, 0x80106111}, /* bass speaker */ + {} + }, + }, + [ALC662_FIXUP_BASS_1A_CHMAP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc662_fixup_bass_chmap, + .chained = true, + .chain_id = ALC662_FIXUP_BASS_1A, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -4918,6 +4933,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), -- 1.8.4.3

1 0

[alsa-devel] [PATCH][topic/symmetry] ASoC: soc-pcm: move DAIs parameters cleaning into hw_free()
by Nicolin Chen 20 Nov '13

20 Nov '13

We're now applying soc_hw_params_symmetry() to reject unmatched parameters while we clear parameters in soc_pcm_close(). So here's a use case might be broken by this mechanism: aplay -Dhw:0 44100.wav 48000.wav 32000.wav In this case, we call soc_pcm_open()->soc_pcm_hw_params()->soc_pcm_hw_free() ->soc_pcm_hw_params()->soc_pcm_hw_free()->soc_pcm_close() in order. As we only clear parameters in soc_pcm_close(). The parameters would be remained in the system even if the playback of 44100.wav is finished. Thus, this patch is trying to move parameters cleaning into hw_free() so that the system can continue to serve this kind of use case. Also, since we set them in hw_params(), it should be better to clear them in hw_free() for symmetry. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/soc-pcm.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 60d9cfb..d519757 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -443,19 +443,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; - /* clear the corresponding DAIs rate when inactive */ - if (!cpu_dai->active) { - cpu_dai->rate = 0; - cpu_dai->channels = 0; - cpu_dai->sample_bits = 0; - } - - if (!codec_dai->active) { - codec_dai->rate = 0; - codec_dai->channels = 0; - codec_dai->sample_bits = 0; - } - /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ @@ -671,6 +658,19 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + /* clear the corresponding DAIs parameters when going to be inactive */ + if (cpu_dai->active == 1) { + cpu_dai->rate = 0; + cpu_dai->channels = 0; + cpu_dai->sample_bits = 0; + } + + if (codec_dai->active == 1) { + codec_dai->rate = 0; + codec_dai->channels = 0; + codec_dai->sample_bits = 0; + } + /* apply codec digital mute */ if (!codec->active) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); -- 1.8.4

2 1

[alsa-devel] [PATCH 1/1] ASoC: arizona: Set FLL to free-run before disabling
by Richard Fitzgerald 20 Nov '13

20 Nov '13

Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6f05b17..fea9910 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1529,6 +1529,8 @@ static void arizona_enable_fll(struct arizona_fll *fll, try_wait_for_completion(&fll->ok); regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, 0); + regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (use_sync) regmap_update_bits(arizona->regmap, fll->base + 0x11, @@ -1546,6 +1548,8 @@ static void arizona_disable_fll(struct arizona_fll *fll) struct arizona *arizona = fll->arizona; bool change; + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); regmap_update_bits_check(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, 0, &change); regmap_update_bits(arizona->regmap, fll->base + 0x11, -- 1.7.2.5

2 3

[alsa-devel] [PATCH 1/1] ASoC: wm5110: Add extra AIF2 channels
by Richard Fitzgerald 20 Nov '13

20 Nov '13

Signed-off-by: D.J. Barrow <dbarrow(a)wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- include/linux/mfd/arizona/registers.h | 121 +++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.c | 8 ++ sound/soc/codecs/arizona.h | 2 +- sound/soc/codecs/wm5110.c | 48 +++++++++++++- 4 files changed, 176 insertions(+), 3 deletions(-) diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 4706d3d..8f4c9d7 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -511,6 +511,38 @@ #define ARIZONA_AIF2TX2MIX_INPUT_3_VOLUME 0x74D #define ARIZONA_AIF2TX2MIX_INPUT_4_SOURCE 0x74E #define ARIZONA_AIF2TX2MIX_INPUT_4_VOLUME 0x74F +#define ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE 0x750 +#define ARIZONA_AIF2TX3MIX_INPUT_1_VOLUME 0x751 +#define ARIZONA_AIF2TX3MIX_INPUT_2_SOURCE 0x752 +#define ARIZONA_AIF2TX3MIX_INPUT_2_VOLUME 0x753 +#define ARIZONA_AIF2TX3MIX_INPUT_3_SOURCE 0x754 +#define ARIZONA_AIF2TX3MIX_INPUT_3_VOLUME 0x755 +#define ARIZONA_AIF2TX3MIX_INPUT_4_SOURCE 0x756 +#define ARIZONA_AIF2TX3MIX_INPUT_4_VOLUME 0x757 +#define ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE 0x758 +#define ARIZONA_AIF2TX4MIX_INPUT_1_VOLUME 0x759 +#define ARIZONA_AIF2TX4MIX_INPUT_2_SOURCE 0x75A +#define ARIZONA_AIF2TX4MIX_INPUT_2_VOLUME 0x75B +#define ARIZONA_AIF2TX4MIX_INPUT_3_SOURCE 0x75C +#define ARIZONA_AIF2TX4MIX_INPUT_3_VOLUME 0x75D +#define ARIZONA_AIF2TX4MIX_INPUT_4_SOURCE 0x75E +#define ARIZONA_AIF2TX4MIX_INPUT_4_VOLUME 0x75F +#define ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE 0x760 +#define ARIZONA_AIF2TX5MIX_INPUT_1_VOLUME 0x761 +#define ARIZONA_AIF2TX5MIX_INPUT_2_SOURCE 0x762 +#define ARIZONA_AIF2TX5MIX_INPUT_2_VOLUME 0x763 +#define ARIZONA_AIF2TX5MIX_INPUT_3_SOURCE 0x764 +#define ARIZONA_AIF2TX5MIX_INPUT_3_VOLUME 0x765 +#define ARIZONA_AIF2TX5MIX_INPUT_4_SOURCE 0x766 +#define ARIZONA_AIF2TX5MIX_INPUT_4_VOLUME 0x767 +#define ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE 0x768 +#define ARIZONA_AIF2TX6MIX_INPUT_1_VOLUME 0x769 +#define ARIZONA_AIF2TX6MIX_INPUT_2_SOURCE 0x76A +#define ARIZONA_AIF2TX6MIX_INPUT_2_VOLUME 0x76B +#define ARIZONA_AIF2TX6MIX_INPUT_3_SOURCE 0x76C +#define ARIZONA_AIF2TX6MIX_INPUT_3_VOLUME 0x76D +#define ARIZONA_AIF2TX6MIX_INPUT_4_SOURCE 0x76E +#define ARIZONA_AIF2TX6MIX_INPUT_4_VOLUME 0x76F #define ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE 0x780 #define ARIZONA_AIF3TX1MIX_INPUT_1_VOLUME 0x781 #define ARIZONA_AIF3TX1MIX_INPUT_2_SOURCE 0x782 @@ -3726,6 +3758,35 @@ #define ARIZONA_AIF2TX2_SLOT_WIDTH 6 /* AIF2TX2_SLOT - [5:0] */ /* + * R1355 (0x54B) - AIF2 Frame Ctrl 5 + */ +#define ARIZONA_AIF2TX3_SLOT_MASK 0x003F /* AIF2TX3_SLOT - [5:0] */ +#define ARIZONA_AIF2TX3_SLOT_SHIFT 0 /* AIF2TX3_SLOT - [5:0] */ +#define ARIZONA_AIF2TX3_SLOT_WIDTH 6 /* AIF2TX3_SLOT - [5:0] */ + +/* + * R1356 (0x54C) - AIF2 Frame Ctrl 6 + */ +#define ARIZONA_AIF2TX4_SLOT_MASK 0x003F /* AIF2TX4_SLOT - [5:0] */ +#define ARIZONA_AIF2TX4_SLOT_SHIFT 0 /* AIF2TX4_SLOT - [5:0] */ +#define ARIZONA_AIF2TX4_SLOT_WIDTH 6 /* AIF2TX4_SLOT - [5:0] */ + + +/* + * R1357 (0x54D) - AIF2 Frame Ctrl 7 + */ +#define ARIZONA_AIF2TX5_SLOT_MASK 0x003F /* AIF2TX5_SLOT - [5:0] */ +#define ARIZONA_AIF2TX5_SLOT_SHIFT 0 /* AIF2TX5_SLOT - [5:0] */ +#define ARIZONA_AIF2TX5_SLOT_WIDTH 6 /* AIF2TX5_SLOT - [5:0] */ + +/* + * R1358 (0x54E) - AIF2 Frame Ctrl 8 + */ +#define ARIZONA_AIF2TX6_SLOT_MASK 0x003F /* AIF2TX6_SLOT - [5:0] */ +#define ARIZONA_AIF2TX6_SLOT_SHIFT 0 /* AIF2TX6_SLOT - [5:0] */ +#define ARIZONA_AIF2TX6_SLOT_WIDTH 6 /* AIF2TX6_SLOT - [5:0] */ + +/* * R1361 (0x551) - AIF2 Frame Ctrl 11 */ #define ARIZONA_AIF2RX1_SLOT_MASK 0x003F /* AIF2RX1_SLOT - [5:0] */ @@ -3740,8 +3801,52 @@ #define ARIZONA_AIF2RX2_SLOT_WIDTH 6 /* AIF2RX2_SLOT - [5:0] */ /* + * R1363 (0x553) - AIF2 Frame Ctrl 13 + */ +#define ARIZONA_AIF2RX3_SLOT_MASK 0x003F /* AIF2RX3_SLOT - [5:0] */ +#define ARIZONA_AIF2RX3_SLOT_SHIFT 0 /* AIF2RX3_SLOT - [5:0] */ +#define ARIZONA_AIF2RX3_SLOT_WIDTH 6 /* AIF2RX3_SLOT - [5:0] */ + +/* + * R1364 (0x554) - AIF2 Frame Ctrl 14 + */ +#define ARIZONA_AIF2RX4_SLOT_MASK 0x003F /* AIF2RX4_SLOT - [5:0] */ +#define ARIZONA_AIF2RX4_SLOT_SHIFT 0 /* AIF2RX4_SLOT - [5:0] */ +#define ARIZONA_AIF2RX4_SLOT_WIDTH 6 /* AIF2RX4_SLOT - [5:0] */ + +/* + * R1365 (0x555) - AIF2 Frame Ctrl 15 + */ +#define ARIZONA_AIF2RX5_SLOT_MASK 0x003F /* AIF2RX5_SLOT - [5:0] */ +#define ARIZONA_AIF2RX5_SLOT_SHIFT 0 /* AIF2RX5_SLOT - [5:0] */ +#define ARIZONA_AIF2RX5_SLOT_WIDTH 6 /* AIF2RX5_SLOT - [5:0] */ + +/* + * R1366 (0x556) - AIF2 Frame Ctrl 16 + */ +#define ARIZONA_AIF2RX6_SLOT_MASK 0x003F /* AIF2RX6_SLOT - [5:0] */ +#define ARIZONA_AIF2RX6_SLOT_SHIFT 0 /* AIF2RX6_SLOT - [5:0] */ +#define ARIZONA_AIF2RX6_SLOT_WIDTH 6 /* AIF2RX6_SLOT - [5:0] */ + +/* * R1369 (0x559) - AIF2 Tx Enables */ +#define ARIZONA_AIF2TX6_ENA 0x0020 /* AIF2TX6_ENA */ +#define ARIZONA_AIF2TX6_ENA_MASK 0x0020 /* AIF2TX6_ENA */ +#define ARIZONA_AIF2TX6_ENA_SHIFT 5 /* AIF2TX6_ENA */ +#define ARIZONA_AIF2TX6_ENA_WIDTH 1 /* AIF2TX6_ENA */ +#define ARIZONA_AIF2TX5_ENA 0x0010 /* AIF2TX5_ENA */ +#define ARIZONA_AIF2TX5_ENA_MASK 0x0010 /* AIF2TX5_ENA */ +#define ARIZONA_AIF2TX5_ENA_SHIFT 4 /* AIF2TX5_ENA */ +#define ARIZONA_AIF2TX5_ENA_WIDTH 1 /* AIF2TX5_ENA */ +#define ARIZONA_AIF2TX4_ENA 0x0008 /* AIF2TX4_ENA */ +#define ARIZONA_AIF2TX4_ENA_MASK 0x0008 /* AIF2TX4_ENA */ +#define ARIZONA_AIF2TX4_ENA_SHIFT 3 /* AIF2TX4_ENA */ +#define ARIZONA_AIF2TX4_ENA_WIDTH 1 /* AIF2TX4_ENA */ +#define ARIZONA_AIF2TX3_ENA 0x0004 /* AIF2TX3_ENA */ +#define ARIZONA_AIF2TX3_ENA_MASK 0x0004 /* AIF2TX3_ENA */ +#define ARIZONA_AIF2TX3_ENA_SHIFT 2 /* AIF2TX3_ENA */ +#define ARIZONA_AIF2TX3_ENA_WIDTH 1 /* AIF2TX3_ENA */ #define ARIZONA_AIF2TX2_ENA 0x0002 /* AIF2TX2_ENA */ #define ARIZONA_AIF2TX2_ENA_MASK 0x0002 /* AIF2TX2_ENA */ #define ARIZONA_AIF2TX2_ENA_SHIFT 1 /* AIF2TX2_ENA */ @@ -3754,6 +3859,22 @@ /* * R1370 (0x55A) - AIF2 Rx Enables */ +#define ARIZONA_AIF2RX6_ENA 0x0020 /* AIF2RX6_ENA */ +#define ARIZONA_AIF2RX6_ENA_MASK 0x0020 /* AIF2RX6_ENA */ +#define ARIZONA_AIF2RX6_ENA_SHIFT 5 /* AIF2RX6_ENA */ +#define ARIZONA_AIF2RX6_ENA_WIDTH 1 /* AIF2RX6_ENA */ +#define ARIZONA_AIF2RX5_ENA 0x0010 /* AIF2RX5_ENA */ +#define ARIZONA_AIF2RX5_ENA_MASK 0x0010 /* AIF2RX5_ENA */ +#define ARIZONA_AIF2RX5_ENA_SHIFT 4 /* AIF2RX5_ENA */ +#define ARIZONA_AIF2RX5_ENA_WIDTH 1 /* AIF2RX5_ENA */ +#define ARIZONA_AIF2RX4_ENA 0x0008 /* AIF2RX4_ENA */ +#define ARIZONA_AIF2RX4_ENA_MASK 0x0008 /* AIF2RX4_ENA */ +#define ARIZONA_AIF2RX4_ENA_SHIFT 3 /* AIF2RX4_ENA */ +#define ARIZONA_AIF2RX4_ENA_WIDTH 1 /* AIF2RX4_ENA */ +#define ARIZONA_AIF2RX3_ENA 0x0004 /* AIF2RX3_ENA */ +#define ARIZONA_AIF2RX3_ENA_MASK 0x0004 /* AIF2RX3_ENA */ +#define ARIZONA_AIF2RX3_ENA_SHIFT 2 /* AIF2RX3_ENA */ +#define ARIZONA_AIF2RX3_ENA_WIDTH 1 /* AIF2RX3_ENA */ #define ARIZONA_AIF2RX2_ENA 0x0002 /* AIF2RX2_ENA */ #define ARIZONA_AIF2RX2_ENA_MASK 0x0002 /* AIF2RX2_ENA */ #define ARIZONA_AIF2RX2_ENA_SHIFT 1 /* AIF2RX2_ENA */ diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index fea9910..6188b05 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -292,6 +292,10 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "AIF1RX8", "AIF2RX1", "AIF2RX2", + "AIF2RX3", + "AIF2RX4", + "AIF2RX5", + "AIF2RX6", "AIF3RX1", "AIF3RX2", "SLIMRX1", @@ -395,6 +399,10 @@ int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { 0x27, 0x28, /* AIF2RX1 */ 0x29, + 0x2a, + 0x2b, + 0x2c, + 0x2d, 0x30, /* AIF3RX1 */ 0x31, 0x38, /* SLIMRX1 */ diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 9e81b63..1f96672 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -81,7 +81,7 @@ struct arizona_priv { unsigned int spk_ena_pending:1; }; -#define ARIZONA_NUM_MIXER_INPUTS 99 +#define ARIZONA_NUM_MIXER_INPUTS 103 extern const unsigned int arizona_mixer_tlv[]; extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index f2d1094..1a73a02 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -302,6 +302,10 @@ ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX3", ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX4", ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX5", ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), @@ -361,6 +365,10 @@ ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX3, ARIZONA_AIF2TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX4, ARIZONA_AIF2TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX5, ARIZONA_AIF2TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX6, ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); @@ -561,11 +569,27 @@ SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX3", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX4", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX5", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX6", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX6_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX3", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX4", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX5", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX6", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX6_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, @@ -703,6 +727,10 @@ ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), +ARIZONA_MIXER_WIDGETS(AIF2TX3, "AIF2TX3"), +ARIZONA_MIXER_WIDGETS(AIF2TX4, "AIF2TX4"), +ARIZONA_MIXER_WIDGETS(AIF2TX5, "AIF2TX5"), +ARIZONA_MIXER_WIDGETS(AIF2TX6, "AIF2TX6"), ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), @@ -764,6 +792,10 @@ SND_SOC_DAPM_OUTPUT("MICSUPP"), { name, "AIF1RX8", "AIF1RX8" }, \ { name, "AIF2RX1", "AIF2RX1" }, \ { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF2RX3", "AIF2RX3" }, \ + { name, "AIF2RX4", "AIF2RX4" }, \ + { name, "AIF2RX5", "AIF2RX5" }, \ + { name, "AIF2RX6", "AIF2RX6" }, \ { name, "AIF3RX1", "AIF3RX1" }, \ { name, "AIF3RX2", "AIF3RX2" }, \ { name, "SLIMRX1", "SLIMRX1" }, \ @@ -861,9 +893,17 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AIF2 Capture", NULL, "AIF2TX1" }, { "AIF2 Capture", NULL, "AIF2TX2" }, + { "AIF2 Capture", NULL, "AIF2TX3" }, + { "AIF2 Capture", NULL, "AIF2TX4" }, + { "AIF2 Capture", NULL, "AIF2TX5" }, + { "AIF2 Capture", NULL, "AIF2TX6" }, { "AIF2RX1", NULL, "AIF2 Playback" }, { "AIF2RX2", NULL, "AIF2 Playback" }, + { "AIF2RX3", NULL, "AIF2 Playback" }, + { "AIF2RX4", NULL, "AIF2 Playback" }, + { "AIF2RX5", NULL, "AIF2 Playback" }, + { "AIF2RX6", NULL, "AIF2 Playback" }, { "AIF3 Capture", NULL, "AIF3TX1" }, { "AIF3 Capture", NULL, "AIF3TX2" }, @@ -947,6 +987,10 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + ARIZONA_MIXER_ROUTES("AIF2TX3", "AIF2TX3"), + ARIZONA_MIXER_ROUTES("AIF2TX4", "AIF2TX4"), + ARIZONA_MIXER_ROUTES("AIF2TX5", "AIF2TX5"), + ARIZONA_MIXER_ROUTES("AIF2TX6", "AIF2TX6"), ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), @@ -1079,14 +1123,14 @@ static struct snd_soc_dai_driver wm5110_dai[] = { .playback = { .stream_name = "AIF2 Playback", .channels_min = 1, - .channels_max = 2, + .channels_max = 6, .rates = WM5110_RATES, .formats = WM5110_FORMATS, }, .capture = { .stream_name = "AIF2 Capture", .channels_min = 1, - .channels_max = 2, + .channels_max = 6, .rates = WM5110_RATES, .formats = WM5110_FORMATS, }, -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Disable runtime PM on LynxPoint(-LP) controllers
by Takashi Iwai 20 Nov '13

20 Nov '13

We got bug reports of the stalled HD-audio, typically after S3 or S4, and it turned out that they seemed triggered by runtime PM on Lynx Point and Lynx Point-LP controllers. As there is no way to recover properly from the stalled controller, it's safer to disable the runtime PM support on these chips for now. Further notes: I actually could reproduce this on a few HP laptops here. Go to S3 after runtime suspend, then the next playback fails, resulting in either a codec stall or repeated sounds. The problem seems lying in a deeper level. The complete stall could be avoided by disabling the call of azx_stop_chip() in azx_runtime_suspend(). More specifically, it's the disablement of CORB/RIRB in azx_free_cmd_io(). After removing this call, the sound is resumed. However, even with that workaround, the first playback after resume stalls due to the missing RIRB interrupts (so you get "switch to polling mode" kernel warning). Interestingly, the codec communication in the resume procedure does work. The system goes to runtime suspend immediately after resume, then something gets broken at that point. This missing interrupt problem happens even if you do nothing in runtime suspend/resume callback with empty callbacks. This implies that it's an issue in the underlying layer. So, the only feasible "fix" in the sound driver side to suppress the runtime PM, so far. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Yet another note: the patch is based on v3.12, not on linux-next, so that it can be backported cleanly for 3.12 and earlier kernels. sound/pci/hda/hda_intel.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6e61a019aa5e..27fc33e54a50 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3973,7 +3973,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Wellsburg */ { PCI_DEVICE(0x8086, 0x8d20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, @@ -3981,10 +3981,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Lynx Point-LP */ { PCI_DEVICE(0x8086, 0x9c20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Lynx Point-LP */ { PCI_DEVICE(0x8086, 0x9c21), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH | -- 1.8.4.3

2 4

Re: [alsa-devel] [PATCH 2/2] snd-oxygen: Fixes for the Xonar DG card
by Roman Volkov 20 Nov '13

20 Nov '13

В Wed, 20 Nov 2013 13:29:18 +0100 Clemens Ladisch <clemens(a)ladisch.de> пишет: > Roman Volkov wrote: > > Clemens Ladisch <clemens(a)ladisch.de> пишет: > >> Roman Volkov wrote: > >>> + * 128kHz doesn't work for CS4361 (multichannel playback) > >> > >> Why do you expect 128 kHz to work? > > > > 128kHz will not work because ... > > ... the CMI8788 does not support 128 kHz in the first place. > > > And perhaps my anti-pop delay of 1 second is too long. > > The Windows driver uses 2.5 seconds. > > >> Is the reset pin actually connected? Does the Windows driver do > >> this? > > > > Should be, otherwise this is bad design. > > This driver is not for a theoretical, optimally designed card; it is > for the actual card that Asus builds. If the Windows driver does not > touch the reset pin, we should not do this either (because it might be > connected to the self-destruct trigger). > > > sometimes (rarely) I get completely distorted sound when I power up > > the Windows machine. > > After a cold boot? > > This also happened for some other Xonar card when the Linux driver did > things differently and did not reset these settings when shutting > down. > > >> Use mdelay() only when sleeping is not possible; use msleep() > >> instead. > > > > msleep() is not recommended for small delays (1ms-20ms). > > Only because it isn't very accurate. But using mdelay() is even worse > because it busy-loops. > > >>> + case PLAYBACK_DST_20_CH: > >>> + case PLAYBACK_DST_40_CH: > >>> + case PLAYBACK_DST_51_CH: > >> > >> Why is it necessary to differentiate between those? There should > >> be no mixer control; the driver can automatically detect what > >> format is used. And this should not make any difference because > >> the unused channels will be silent anyway. > > > > Just as in the windows driver. > > This is not necessarily a reason to implement the same in the Linux > driver, especially if it's not required by the actual hardware, and > reduces usability. > > > Should I change this to "HP\HP FP\Speakers"? > > Yes. > > >>> +/* Put the codec into low power mode, disable audio output, save > >>> registers */ > >> > >> Does the Windows driver implement this? > > > > I don't know. Linux docs says I should put my hardware into low > > power mode and I do that. > > As far as I know, none of the Xonar cards have any real power-saving > functionality; they rely on the entire PCI slot being powered down. > > For practical purposes, suspend/resume just means that you have to > restore the registers. > > >>> - .function_flags = OXYGEN_FUNCTION_SPI, > >>> + .function_flags = OXYGEN_FUNCTION_SPI | > >>> OXYGEN_FUNCTION_ENABLE_SPI_4_5, > >> > >> Why? > > > > Because this just works and as in the Windows driver. > > If nothing is connected to those pins, it doesn't matter. This code > in the Windows driver probably was just copy/pasted from the D2 code. > > >>> +++ linux-3.12-my/sound/pci/oxygen/xonar_dg.h > >> > >> This file was intended as the interface between xonar_dg.c and the > >> generic oxygen.c driver. > >> > >> That you have to add so much stuff indicates that xonar_dg.c and > >> xonar_dg_mixer.c are not very independent and maybe should have > >> stayed a single file. > > > > But moving mixer to another file so simplifies things... > > Your choice. But it makes it hard to review changes when you move the > code around in the same patch. > > > Regards, > Clemens > Okay, I won't argue. If the driver will work with the changes, I don't care. How should I name the subject? Should I mark this with "RESEND" or "UPDATE" or leave the subject named like "[PATCH 2/2] snd-oxygen: Fixes for the Xonar DG card"? -- Kind regards, Roman Volkov, v1ron(a)mail.ru

1 0

[alsa-devel] question about aplay fails on arm 64
by 曹能华 20 Nov '13

20 Nov '13

Hello, alsa maintainer: Recently, i encounter aplay failed on arm 64 bits core. My test environment is: Linux kernel use arm 64 bits, and alsa-lib/aplay is 32 bit version. So the ioctl interface should use snd_pcm_ioctl_compat. but when i use aplay to playback music, i encounter "aplay" segment fault. Do we need some special configuration for alsa-lib/aplay to support 64 bit kernel? Any suggestion will be appreciated. Thanks! [\u@\h: \W]\#aplay windows.wav [ 85.237519] 823 (aplay) snd_pcm_playback_open+++ [ 85.247876] 823 (aplay) snd_pcm_aio_write--- [ 85.254708] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.263597] 823 (aplay) ======snd_pcm_ioctl_compat3333 [ 85.271850] 823 (aplay) snd_pcm_playback_ioctl1+++ [ 85.279019] 823 (aplay) snd_pcm_common_ioctl1+++ [ 85.285318] 823 (aplay) snd_pcm_common_ioctl122 [ 85.292253] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.300186] 823 (aplay) ======snd_pcm_ioctl_compat3333 [ 85.307569] 823 (aplay) snd_pcm_playback_ioctl1+++ [ 85.314021] 823 (aplay) snd_pcm_common_ioctl1+++ [ 85.320697] 823 (aplay) snd_pcm_common_ioctl111 [ 85.328240] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.336255] 823 (aplay) ======snd_pcm_ioctl_compat3333 [ 85.343185] 823 (aplay) snd_pcm_playback_ioctl1+++ [ 85.350090] 823 (aplay) snd_pcm_common_ioctl1+++ [ 85.356800] 823 (aplay) snd_pcm_common_ioctl144 [ 85.366441] 823 (aplay) =====snd_pcm_mmap+++ [ 85.372840] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.380709] 823 (aplay) ======snd_pcm_ioctl_compat9999 [ 85.388074] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat++ [ 85.395184] 823 (aplay) state:0x0 [ 85.400355] 823 (aplay) hw_ptr:0x0 [ 85.405080] 823 (aplay) tstamp:0x0 [ 85.410326] 823 (aplay) suspend_stat:0x0 [ 85.416216] 823 (aplay) tstamp:0x0 [ 85.421039] 823 (aplay) appl_ptr:0x0 [ 85.426489] 823 (aplay) avail_min:0x1 [ 85.431579] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat-- [ 85.443771] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.451883] 823 (aplay) ======snd_pcm_ioctl_compat3333 [ 85.459321] 823 (aplay) snd_pcm_playback_ioctl1+++ [ 85.466280] 823 (aplay) snd_pcm_common_ioctl1+++ [ 85.472539] 823 (aplay) snd_pcm_common_ioctl122 Playing WAVE 'windows.wav' : Signed 16 bit Little Endian, Rate 22050 Hz, Stereo [ 85.500612] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.508756] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.519975] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.528101] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.536112] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.543496] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.552271] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.560207] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.594120] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.602221] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.639453] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.647597] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.656416] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.663826] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.693293] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.701231] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.730988] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.739043] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.748850] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.756855] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.765543] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.772949] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.802272] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.810266] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.839909] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.847899] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.857606] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.865619] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.873814] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.881784] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.891149] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.899062] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.911247] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.919318] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.928068] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.935994] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.945446] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.952828] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.962969] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.971003] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.980633] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 85.988651] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 85.997912] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.005880] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.019755] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.027851] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.037648] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.045668] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.055326] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.062700] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.071957] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.079852] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.093442] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.101428] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.110880] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.118767] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.128022] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.136031] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.145312] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.152642] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.166234] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.174252] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.185470] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.192864] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.202191] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.210222] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.219452] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.227353] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.241140] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.249167] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.258633] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.266583] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.275809] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.283209] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.292507] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.300462] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.312362] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.320295] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.329874] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.337825] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.347005] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.354979] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.364272] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.371671] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.386035] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.394010] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.404122] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.411491] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.420685] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.428587] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.437805] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.445723] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.458333] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.466412] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.476696] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.484743] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.494008] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.501393] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.510721] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.518682] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.528949] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.536991] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.546180] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.554122] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.563543] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.570903] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.583664] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.591141] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.600602] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.608548] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.617702] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.625703] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.634924] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.642317] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.653731] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.661075] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.670427] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.678383] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.687594] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.695585] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.709565] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.717652] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.734313] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.741774] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.755420] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.763633] 823 (aplay) ======snd_pcm_ioctl_compat5555 [ 86.773070] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.780448] 823 (aplay) ======snd_pcm_ioctl_compat66666 [ 86.789495] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.797508] 823 (aplay) ======snd_pcm_ioctl_compat9999 [ 86.804939] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat++ [ 86.812086] 823 (aplay) state:0x1 [ 86.817225] 823 (aplay) hw_ptr:0x0 [ 86.821974] 823 (aplay) tstamp:0x0 [ 86.827171] 823 (aplay) suspend_stat:0x0 [ 86.833049] 823 (aplay) tstamp:0x0 [ 86.837795] 823 (aplay) appl_ptr:0x0 [ 86.843255] 823 (aplay) avail_min:0x1 [ 86.848303] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat-- [ 86.857505] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.865637] 823 (aplay) ======snd_pcm_ioctl_compat77777 [ 86.874082] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.881454] 823 (aplay) ======snd_pcm_ioctl_compat1010 [ 86.889143] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.896981] 823 (aplay) ======snd_pcm_ioctl_compat1010 [ 86.904723] 823 (aplay) =====snd_pcm_mmap+++ [ 86.910536] 823 (aplay) =====snd_pcm_mmap33 [ 86.919994] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.928068] 823 (aplay) ======snd_pcm_ioctl_compat77777 [ 86.937550] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 86.945512] 823 (aplay) ======snd_pcm_ioctl_compat9999 [ 86.952892] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat++ [ 86.959975] 823 (aplay) state:0x1 [ 86.965065] 823 (aplay) hw_ptr:0x0 [ 86.969876] 823 (aplay) tstamp:0x0 [ 86.975111] 823 (aplay) suspend_stat:0x0 [ 86.980497] 823 (aplay) tstamp:0x0 [ 86.985722] 823 (aplay) appl_ptr:0x0 [ 86.990719] 823 (aplay) avail_min:0x300 [ 86.996397] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat-- [ 87.004605] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 87.012676] 823 (aplay) ======snd_pcm_ioctl_compat3333 [ 87.019501] 823 (aplay) snd_pcm_playback_ioctl1+++ [ 87.026451] 823 (aplay) snd_pcm_common_ioctl1+++ [ 87.033169] 823 (aplay) snd_pcm_common_ioctl11111 [ 87.048203] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 87.056227] 823 (aplay) ======snd_pcm_ioctl_compat9999 [ 87.063579] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat++ [ 87.070579] 823 (aplay) state:0x2 [ 87.075679] 823 (aplay) hw_ptr:0x0 [ 87.080477] 823 (aplay) tstamp:0x0 [ 87.085657] 823 (aplay) suspend_stat:0x0 [ 87.091028] 823 (aplay) tstamp:0x0 [ 87.096229] 823 (aplay) appl_ptr:0x0 [ 87.101165] 823 (aplay) avail_min:0x300 [ 87.106813] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat-- [ 87.116559] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 87.124489] 823 (aplay) ======snd_pcm_ioctl_compat77777 [ 87.134835] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 87.142950] 823 (aplay) ======snd_pcm_ioctl_compat9999 [ 87.149868] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat++ [ 87.157350] 823 (aplay) state:0x2 [ 87.162475] 823 (aplay) hw_ptr:0x0 [ 87.167254] 823 (aplay) tstamp:0x0 [ 87.172485] 823 (aplay) suspend_stat:0x0 [ 87.177886] 823 (aplay) tstamp:0x0 [ 87.183131] 823 (aplay) appl_ptr:0x0 [ 87.188110] 823 (aplay) avail_min:0x300 [ 87.193880] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat-- [ 87.202114] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 87.209559] 823 (aplay) ======snd_pcm_ioctl_compat9999 [ 87.217028] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat++ [ 87.224520] 823 (aplay) state:0x2 [ 87.229146] 823 (aplay) hw_ptr:0x0 [ 87.234352] 823 (aplay) tstamp:0x0 [ 87.239101] 823 (aplay) suspend_stat:0x0 [ 87.244924] 823 (aplay) tstamp:0x0 [ 87.249708] 823 (aplay) appl_ptr:0x0 [ 87.255147] 823 (aplay) avail_min:0x300 [ 87.260432] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat-- [ 87.268036] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 87.275875] 823 (aplay) ======snd_pcm_ioctl_compat9999 [ 87.283164] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat++ [ 87.290225] 823 (aplay) state:0x2 [ 87.295283] 823 (aplay) hw_ptr:0x0 [ 87.300041] 823 (aplay) tstamp:0x0 [ 87.305174] 823 (aplay) suspend_stat:0x0 [ 87.310583] 823 (aplay) tstamp:0x0 [ 87.315771] 823 (aplay) appl_ptr:0x0 [ 87.320771] 823 (aplay) avail_min:0x300 [ 87.326468] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat-- [ 87.335868] 823 (aplay) ======snd_pcm_ioctl_compat++++++++ [ 87.343945] 823 (aplay) ======snd_pcm_ioctl_compat9999 [ 87.350825] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat++ [ 87.358387] 823 (aplay) state:0x2 [ 87.363532] 823 (aplay) hw_ptr:0x0 [ 87.368252] 823 (aplay) tstamp:0x0 [ 87.373510] 823 (aplay) suspend_stat:0x0 [ 87.378904] 823 (aplay) tstamp:0x0 [ 87.384100] 823 (aplay) appl_ptr:0x0 [ 87.389050] 823 (aplay) avail_min:0x300 [ 87.394764] 823 (aplay) snd_pcm_ioctl_sync_ptr_compat-- [ 87.403146] 823 (aplay) Bad mode in Error handler detected, code 0xbf000000 [ 87.412595] 823 (aplay) CPU: 0 PID: 823 Comm: aplay Not tainted 3.10.0+ #35 [ 87.421956] 823 (aplay) task: ffffffc019add7c0 ti: ffffffc019a50000 task.ti: ffffffc019a50000 [ 87.433106] 823 (aplay) PC is at 0xf72fd970 [ 87.438849] 823 (aplay) LR is at 0x0 [ 87.443931] 823 (aplay) pc : [<00000000f72fd970>] lr : [<0000000000000000>] pstate: 80000010 [ 87.454906] 823 (aplay) sp : ffffffc019a51ff0 [ 87.460837] x29: 0000000000000000 x28: 0000000000000000 [ 87.468175] x27: 0000000000000000 x26: 0000000000000000 [ 87.475415] x25: 0000000000000000 x24: 0000000000000000 [ 87.482645] x23: 0000000000000000 x22: 0000000000000000 [ 87.489961] x21: 0000000000000000 x20: 0000000000000000 [ 87.497217] x19: 0000000000000000 x18: 0000000000000000 [ 87.504365] x17: 0000000000000000 x16: 0000000000000000 [ 87.511611] x15: 0000000000000000 x14: 00000000f72fd9ec [ 87.518832] x13: 00000000ffa0ee98 x12: 00000000ffa0ee98 [ 87.526061] x11: 00000000ffa0eeec x10: 000000000001e32c [ 87.533332] x9 : 000000000001e31c x8 : 00000000ffa0f168 [ 87.540473] x7 : 00000000ffa0f168 x6 : 00000000ffa0f180 [ 87.547681] x5 : 00000000f738f000 x4 : 000000000000000f [ 87.554817] x3 : 0000000000000300 x2 : 0000000000000001 [ 87.561998] x1 : 0000000000000000 x0 : 0000000000000000 [ 87.569196] 823 (aplay) [ 87.573201] 823 (aplay) Bad mode in Synchronous Abort handler detected, code 0x86000007 [ 87.583794] 823 (aplay) CPU: 0 PID: 823 Comm: aplay Not tainted 3.10.0+ #35 [ 87.593059] 823 (aplay) task: ffffffc019add7c0 ti: ffffffc019a50000 task.ti: ffffffc019a50000 [ 87.603993] 823 (aplay) PC is at 0x0 [ 87.608968] 823 (aplay) LR is at 0x0 [ 87.614003] 823 (aplay) pc : [<0000000000000000>] lr : [<0000000000000000>] pstate: 800003c5 [ 87.624868] 823 (aplay) sp : ffffffc019a51ed0 [ 87.630832] x29: 0000000000000000 x28: 0000000000000000 [ 87.638120] x27: 0000000000000000 x26: 0000000000000000 [ 87.645395] x25: 0000000000000000 x24: 0000000000000000 [ 87.652731] x23: 0000000080000010 x22: 00000000f72fd970 [ 87.659999] x21: ffffffc019a51ff0 x20: 0000000000000000 [ 87.667198] x19: 0000000000000000 x18: 0000000000000000 [ 87.674377] x17: 0000000000000000 x16: 0000000000000000 [ 87.681570] x15: 0000000000000002 x14: 0000000000000082 [ 87.688706] x13: 0000000000000000 x12: 0000000000001db0 [ 87.695986] x11: ffffffc0012376f8 x10: 0000000000000050 [ 87.703184] x9 : ffffffc019a51c60 x8 : ffffffc000590228 [ 87.710415] x7 : 0000000000000003 x6 : 000000000000027d [ 87.717643] x5 : ffffffc000081380 x4 : 0000000000000000 [ 87.724914] x3 : 0000000000000001 x2 : 0000000000000001 [ 87.732107] x1 : ffffffc019a51de0 x0 : 0000000000000000 [ 87.739282] 823 (aplay) [ 87.743126] 823 (aplay) Internal error: Oops - bad mode: 0 [#1] PREEMPT [ 87.752089] 823 (aplay) CPU: 0 PID: 823 Comm: aplay Not tainted 3.10.0+ #35 [ 87.761348] 823 (aplay) task: ffffffc019add7c0 ti: ffffffc019a50000 task.ti: ffffffc019a50000 [ 87.772375] 823 (aplay) PC is at 0x0 [ 87.777318] 823 (aplay) LR is at 0x0 [ 87.782386] 823 (aplay) pc : [<0000000000000000>] lr : [<0000000000000000>] pstate: 800003c5 [ 87.793320] 823 (aplay) sp : ffffffc019a51ed0 [ 87.799289] x29: 0000000000000000 x28: 0000000000000000 [ 87.806491] x27: 0000000000000000 x26: 0000000000000000 [ 87.813665] x25: 0000000000000000 x24: 0000000000000000 [ 87.820915] x23: 0000000080000010 x22: 00000000f72fd970 [ 87.828152] x21: ffffffc019a51ff0 x20: 0000000000000000 [ 87.835382] x19: 0000000000000000 x18: 0000000000000000 [ 87.842585] x17: 0000000000000000 x16: 0000000000000000 [ 87.849845] x15: 0000000000000002 x14: 0000000000000082 [ 87.857020] x13: 0000000000000000 x12: 0000000000001db0 [ 87.864266] x11: ffffffc0012376f8 x10: 0000000000000050 [ 87.871505] x9 : ffffffc019a51c60 x8 : ffffffc000590228 [ 87.878720] x7 : 0000000000000003 x6 : 000000000000027d [ 87.885914] x5 : ffffffc000081380 x4 : 0000000000000000 [ 87.893121] x3 : 0000000000000001 x2 : 0000000000000001 [ 87.900321] x1 : ffffffc019a51de0 x0 : 0000000000000000 [ 87.907443] 823 (aplay) [ 87.911323] 823 (aplay) Process aplay (pid: 823, stack limit = 0xffffffc019a50058) [ 87.921331] 823 (aplay) Stack: (0xffffffc019a51ed0 to 0xffffffc019a52000) [ 87.930547] 823 (aplay) 1ec0: 00000000 00000000 00000000 00000000 [ 87.942875] 823 (aplay) 1ee0: 00000001 00000000 00000300 00000000 0000000f 00000000 f738f000 00000000 [ 87.955305] 823 (aplay) 1f00: ffa0f180 00000000 ffa0f168 00000000 ffa0f168 00000000 0001e31c 00000000 [ 87.967718] 823 (aplay) 1f20: 0001e32c 00000000 ffa0eeec 00000000 ffa0ee98 00000000 ffa0ee98 00000000 [ 87.979993] 823 (aplay) 1f40: f72fd9ec 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 87.992343] 823 (aplay) 1f60: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 88.004660] 823 (aplay) 1f80: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 88.016989] 823 (aplay) 1fa0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 88.029331] 823 (aplay) 1fc0: 00000000 00000000 19a51ff0 ffffffc0 f72fd970 00000000 80000010 00000000 [ 88.041739] 823 (aplay) 1fe0: 00000004 00000000 ffffffff ffffffff 00400080 a201ca14 00400020 200130de [ 88.053478] 823 (aplay) Call trace: [ 88.058456] 823 (aplay) Code: bad PC value [ 88.065099] 823 (aplay) ---[ end trace c58f84f42ebfb68b ]--- [ 88.091636] 823 (aplay) snd_pcm_release++ [ 88.098134] 823 (aplay) snd_pcm_release-- Segmentation fault [\u@\h: \W]\# BR cosmo

2 1

[alsa-devel] [PATCH] ALSA: hda - A casual Dell Headset quirk
by David Henningsson 20 Nov '13

20 Nov '13

You're looking at a casual headset patch, for a specific hardware it will match, and suddenly, the headset jack will work, so please apply this simple quirk! BugLink: https://bugs.launchpad.net/bugs/1253038 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8593d4e..a90bfab 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4915,6 +4915,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] Maya44+ Support in ALSA
by Simone Camporeale 20 Nov '13

20 Nov '13

I found some problems on Maya44+ USB support in linux. Using ubuntu 13.04 , Ubuntu 13.10 with alsa driver alsa-driver-1.0.25+dfsg i found that this card isn't supported. Only 2 of 4 channels work and they are mixed . This card mounts TI TUSB3200AC chip. This chip is also used in Re Loop devices and in Herlcues DJ Console. To fix this problem i found on google a patch for Reloop device. I chanaged VID and PID for MAYA44+ and now it works . This patch involves in edititing ./sound/usb/quirks-table.h according to attached patch Also i provide an .asoundrc. With this .asoundrc you are able to use each input and output channel maya44+ patch --- a/sound/usb/quirks-table.h 2013-11-12 12:43:24.380251603 +0100 +++ b/sound/usb/quirks-table.h 2013-11-06 12:28:20.000000000 +0100 @@ -320,6 +320,93 @@ #undef YAMAHA_DEVICE #undef YAMAHA_INTERFACE + +/* Reloop Play */ +{ + USB_DEVICE(0x200c, 0x100b), + .bInterfaceClass = USB_CLASS_PER_INTERFACE, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_SYNC_ADAPTIVE, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = -1 + } + } + } +}, + +/* Maya44 USB+ Play */ +{ + USB_DEVICE(0x2573, 0x0008), + .bInterfaceClass = USB_CLASS_PER_INTERFACE, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_SYNC_ADAPTIVE, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = -1 + } + } + } +}, + + + + + /* * Roland/RolandED/Edirol/BOSS devices */ .asoundrc example: pcm.dshare { type dmix ipc_key 2048 slave { pcm "hw:1" rate 44100 period_time 0 period_size 1024 buffer_size 8192 channels 4 } bindings { 0 0 1 1 2 2 3 3 } } pcm.dsnooped { type dsnoop ipc_key 2048 slave { pcm "hw:1" rate 48000 periods 0 period_time 0 period_size 1024 buffer_size 4096 channels 4 } } pcm.mayaAin1 { type plug slave { pcm "dsnooped" channels 4 } ttable.0.0 1 } pcm.mayaAin2 { type plug slave { pcm "dsnooped" channels 4 } ttable.0.1 1 } pcm.mayaAin3 { type plug slave { pcm "dsnooped" channels 4 } ttable.0.2 1 } pcm.mayaAin4 { type plug slave { pcm "dsnooped" channels 4 } ttable.0.3 1 } pcm.mayaAout1 { type plug slave { pcm "dshare" channels 4 } ttable.0.0 1 } pcm.mayaAout2 { type plug slave { pcm "dshare" channels 4 } ttable.0.1 1 } pcm.mayaAout3 { type plug slave { pcm "dshare" channels 4 } ttable.0.2 1 } pcm.mayaAout4 { type plug slave { pcm "dshare" channels 4 } ttable.0.3 1 }

2 4

[alsa-devel] [PATCH 2/2] snd-oxygen: Fixes for the Xonar DG card
by Roman Volkov 20 Nov '13

20 Nov '13

Fixes for the Xonar DG model of the Oxygen driver: CS4245 CODEC now fully supported with the headphone amplifier. Both front panel and rear jacks work. Full support of capturing: from the microphone, line in, and aux input. Multichannel playback routing corrected for 2.0/4.0/5.1 speakers New mixer controls, hardware volume control supported for headphones only. Signed-off-by: Roman I Volkov <v1ron(a)mail.ru> --- Depends on: [PATCH 1/2] snd-oxygen: Changes for oxygen driver to fix the Xonar DG card diff -uprN linux-3.12/sound/pci/oxygen/Makefile linux-3.12-my/sound/pci/oxygen/Makefile --- linux-3.12/sound/pci/oxygen/Makefile 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/Makefile 2013-11-15 16:55:55.000000000 +0400 @@ -1,5 +1,5 @@ snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o -snd-oxygen-objs := oxygen.o xonar_dg.o +snd-oxygen-objs := oxygen.o xonar_dg.o xonar_dg_mixer.o snd-virtuoso-objs := virtuoso.o xonar_lib.o \ xonar_pcm179x.o xonar_cs43xx.o xonar_wm87x6.o xonar_hdmi.o diff -uprN linux-3.12/sound/pci/oxygen/xonar_dg.c linux-3.12-my/sound/pci/oxygen/xonar_dg.c --- linux-3.12/sound/pci/oxygen/xonar_dg.c 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/xonar_dg.c 2013-11-19 01:01:07.000000000 +0400 @@ -2,7 +2,7 @@ * card driver for the Xonar DG/DGX * * Copyright (c) Clemens Ladisch <clemens(a)ladisch.de> - * + * Copyright (c) Roman Volkov <v1ron(a)mail.ru> * * This driver is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License, version 2. @@ -17,32 +17,48 @@ */ /* - * Xonar DG/DGX + * TODOs: + * DGX card: not tested. + * Suspend/resume: not tested + * SPDIF: not tested + * 128kHz doesn't work for CS4361 (multichannel playback) + * * ------------ * + * Pecularities: + * CS4245/CS4361 both will mute outputs if MCLK\BCLK\LRCK rates are invalid. + * CS4245 mutes DAC even if ADC clocks are invalid. + * CS4245 Auxiliary Output is connected to the front panel headphones jack + * through amplifier. + * * CMI8788: * * SPI 0 -> CS4245 * + * Playback: * I²S 1 -> CS4245 * I²S 2 -> CS4361 (center/LFE) * I²S 3 -> CS4361 (surround) * I²S 4 -> CS4361 (front) + * Capture: + * I²S ADC 1 <- CS4245 * - * GPIO 3 <- ? + * GPIO 3 <- unused (?) * GPIO 4 <- headphone detect - * GPIO 5 -> route input jack to line-in (0) or mic-in (1) - * GPIO 6 -> route input jack to line-in (0) or mic-in (1) - * GPIO 7 -> enable rear headphone amp - * GPIO 8 -> enable output to speakers + * GPIO 5 -> ADC analog circuit power supply control for left channel (?) + * GPIO 6 -> ADC analog circuit power supply control for right channel (?) + * GPIO 7 -> switches green rear output jack between CS4245 / CS4361 first + * channel + * GPIO 8 -> DAC analog circuit power supply control (?) * * CS4245: * - * input 1 <- aux - * input 2 <- front mic - * input 4 <- line/mic + * input 0 <- Microphone + * input 1 <- Aux + * input 2 <- Front Panel Mic + * input 4 <- Line In * DAC out -> headphones - * aux out -> front panel headphones + * AUX out -> front panel headphones */ #include <linux/pci.h> @@ -56,536 +72,263 @@ #include "xonar_dg.h" #include "cs4245.h" -#define GPIO_MAGIC 0x0008 -#define GPIO_HP_DETECT 0x0010 -#define GPIO_INPUT_ROUTE 0x0060 -#define GPIO_HP_REAR 0x0080 -#define GPIO_OUTPUT_ENABLE 0x0100 - -struct dg { - unsigned int output_sel; - s8 input_vol[4][2]; - unsigned int input_sel; - u8 hp_vol_att; - u8 cs4245_regs[0x11]; -}; - -static void cs4245_write(struct oxygen *chip, unsigned int reg, u8 value) +int cs4245_write_spi(struct oxygen *chip, u8 reg) { struct dg *data = chip->model_data; - oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | - OXYGEN_SPI_DATA_LENGTH_3 | - OXYGEN_SPI_CLOCK_1280 | - (0 << OXYGEN_SPI_CODEC_SHIFT) | - OXYGEN_SPI_CEN_LATCH_CLOCK_HI, - CS4245_SPI_ADDRESS | - CS4245_SPI_WRITE | - (reg << 8) | value); - data->cs4245_regs[reg] = value; + return oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_3 | OXYGEN_SPI_CLOCK_1280 | + (0 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_HI, + (((u32)CS4245_SPI_ADDRESS) << 16) | (((u32)reg) << 8) | + ((u32)data->cs4245_shadow[reg])); } -static void cs4245_write_cached(struct oxygen *chip, unsigned int reg, u8 value) +int cs4245_read_spi(struct oxygen *chip, u8 reg) { struct dg *data = chip->model_data; + int ret; - if (value != data->cs4245_regs[reg]) - cs4245_write(chip, reg, value); -} - -static void cs4245_registers_init(struct oxygen *chip) -{ - struct dg *data = chip->model_data; + ret = oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CEN_LATCH_CLOCK_HI | + OXYGEN_SPI_CLOCK_1280 | (0 << OXYGEN_SPI_CODEC_SHIFT), + (CS4245_SPI_ADDRESS << 8) | reg); + if (ret < 0) + return ret; - cs4245_write(chip, CS4245_POWER_CTRL, CS4245_PDN); - cs4245_write(chip, CS4245_DAC_CTRL_1, - data->cs4245_regs[CS4245_DAC_CTRL_1]); - cs4245_write(chip, CS4245_ADC_CTRL, - data->cs4245_regs[CS4245_ADC_CTRL]); - cs4245_write(chip, CS4245_SIGNAL_SEL, - data->cs4245_regs[CS4245_SIGNAL_SEL]); - cs4245_write(chip, CS4245_PGA_B_CTRL, - data->cs4245_regs[CS4245_PGA_B_CTRL]); - cs4245_write(chip, CS4245_PGA_A_CTRL, - data->cs4245_regs[CS4245_PGA_A_CTRL]); - cs4245_write(chip, CS4245_ANALOG_IN, - data->cs4245_regs[CS4245_ANALOG_IN]); - cs4245_write(chip, CS4245_DAC_A_CTRL, - data->cs4245_regs[CS4245_DAC_A_CTRL]); - cs4245_write(chip, CS4245_DAC_B_CTRL, - data->cs4245_regs[CS4245_DAC_B_CTRL]); - cs4245_write(chip, CS4245_DAC_CTRL_2, - CS4245_DAC_SOFT | CS4245_DAC_ZERO | CS4245_INVERT_DAC); - cs4245_write(chip, CS4245_INT_MASK, 0); - cs4245_write(chip, CS4245_POWER_CTRL, 0); -} + ret = oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CEN_LATCH_CLOCK_HI | + OXYGEN_SPI_CLOCK_1280 | (0 << OXYGEN_SPI_CODEC_SHIFT), + (CS4245_SPI_ADDRESS | CS4245_SPI_READ) << 8); + if (ret < 0) + return ret; + ret = oxygen_wait_spi(chip); + if (ret < 0) + return ret; -static void cs4245_init(struct oxygen *chip) -{ - struct dg *data = chip->model_data; + data->cs4245_shadow[reg] = oxygen_read8(chip, OXYGEN_SPI_DATA1); - data->cs4245_regs[CS4245_DAC_CTRL_1] = - CS4245_DAC_FM_SINGLE | CS4245_DAC_DIF_LJUST; - data->cs4245_regs[CS4245_ADC_CTRL] = - CS4245_ADC_FM_SINGLE | CS4245_ADC_DIF_LJUST; - data->cs4245_regs[CS4245_SIGNAL_SEL] = - CS4245_A_OUT_SEL_HIZ | CS4245_ASYNCH; - data->cs4245_regs[CS4245_PGA_B_CTRL] = 0; - data->cs4245_regs[CS4245_PGA_A_CTRL] = 0; - data->cs4245_regs[CS4245_ANALOG_IN] = - CS4245_PGA_SOFT | CS4245_PGA_ZERO | CS4245_SEL_INPUT_4; - data->cs4245_regs[CS4245_DAC_A_CTRL] = 0; - data->cs4245_regs[CS4245_DAC_B_CTRL] = 0; - cs4245_registers_init(chip); - snd_component_add(chip->card, "CS4245"); + return 0; } -static void dg_output_enable(struct oxygen *chip) +static int cs4245_dump_regs(struct oxygen *chip, bool bRead) { - msleep(2500); - oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); + u8 address; + + for (address = 0; address <= 0x10; address++) { + return bRead ? cs4245_read_spi(chip, address) : + cs4245_write_spi(chip, address); + } + return 0; } -static void dg_init(struct oxygen *chip) +static void cs4245_init(struct oxygen *chip) { struct dg *data = chip->model_data; - data->output_sel = 0; - data->input_sel = 3; - data->hp_vol_att = 2 * 16; - - cs4245_init(chip); + /* save the initial state */ + cs4245_dump_regs(chip, true); - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_MAGIC | GPIO_HP_DETECT); - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_INPUT_ROUTE | GPIO_HP_REAR | GPIO_OUTPUT_ENABLE); - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, - GPIO_INPUT_ROUTE | GPIO_HP_REAR); - dg_output_enable(chip); -} + /* + * Power up the CODEC internals, enable soft ramp & zero cross, work + * in async. mode, + * Enable aux output from DAC. DAC output in the Windows driver is + * inverted. + */ + data->cs4245_shadow[CS4245_POWER_CTRL] = 0; + data->cs4245_shadow[CS4245_SIGNAL_SEL] = CS4245_A_OUT_SEL_DAC | + CS4245_ASYNCH; + data->cs4245_shadow[CS4245_DAC_CTRL_1] = 0; + data->cs4245_shadow[CS4245_DAC_CTRL_2] = CS4245_DAC_SOFT | + CS4245_DAC_ZERO | CS4245_INVERT_DAC; + data->cs4245_shadow[CS4245_ADC_CTRL] = 0; + data->cs4245_shadow[CS4245_ANALOG_IN] = CS4245_PGA_SOFT | + CS4245_PGA_ZERO; + data->cs4245_shadow[CS4245_MCLK_FREQ] = (CS4245_MCLK_1 << + CS4245_MCLK1_SHIFT) | (CS4245_MCLK_1 << CS4245_MCLK2_SHIFT); + + cs4245_write_spi(chip, CS4245_SIGNAL_SEL); + cs4245_write_spi(chip, CS4245_DAC_CTRL_1); + cs4245_write_spi(chip, CS4245_DAC_CTRL_2); + cs4245_write_spi(chip, CS4245_ADC_CTRL); + cs4245_write_spi(chip, CS4245_ANALOG_IN); + cs4245_write_spi(chip, CS4245_MCLK_FREQ); + cs4245_write_spi(chip, CS4245_POWER_CTRL); -static void dg_cleanup(struct oxygen *chip) -{ - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE); + snd_component_add(chip->card, "CS4245"); } -static void dg_suspend(struct oxygen *chip) +static void dg_init(struct oxygen *chip) { - dg_cleanup(chip); -} + /* + * Recommended sequence from the datasheet: reset the codec and then + * initialize it. + * The pin XGPIO1 as XGPIO1, gpios 8,7,6,5,1,0 as outputs. + */ + oxygen_write8_masked(chip, OXYGEN_FUNCTION, 0, + OXYGEN_FUNCTION_RESET_CODEC); + mdelay(1); + oxygen_write8_masked(chip, OXYGEN_FUNCTION, 0xff, + OXYGEN_FUNCTION_RESET_CODEC); + mdelay(1); -static void dg_resume(struct oxygen *chip) -{ - cs4245_registers_init(chip); - dg_output_enable(chip); + cs4245_init(chip); + oxygen_write16(chip, OXYGEN_GPIO_CONTROL, GPIO_OUTPUT_ENABLE | + GPIO_HP_REAR | GPIO_INPUT_ROUTE); + oxygen_write16(chip, OXYGEN_GPIO_DATA, GPIO_HP_REAR | GPIO_INPUT_ROUTE); + /* Anti-pop delay */ + msleep_interruptible(1000); + oxygen_write16(chip, OXYGEN_GPIO_DATA, GPIO_OUTPUT_ENABLE | + GPIO_HP_REAR | GPIO_INPUT_ROUTE); } static void set_cs4245_dac_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { struct dg *data = chip->model_data; - u8 value; + unsigned char tmp; + unsigned char tmp2; - value = data->cs4245_regs[CS4245_DAC_CTRL_1] & ~CS4245_DAC_FM_MASK; - if (params_rate(params) <= 50000) - value |= CS4245_DAC_FM_SINGLE; - else if (params_rate(params) <= 100000) - value |= CS4245_DAC_FM_DOUBLE; - else - value |= CS4245_DAC_FM_QUAD; - cs4245_write_cached(chip, CS4245_DAC_CTRL_1, value); + tmp = data->cs4245_shadow[CS4245_DAC_CTRL_1] & ~CS4245_DAC_FM_MASK; + tmp2 = data->cs4245_shadow[CS4245_MCLK_FREQ] & ~CS4245_MCLK1_MASK; + if (params_rate(params) <= 50000) { + tmp |= CS4245_DAC_FM_SINGLE; + tmp2 |= CS4245_MCLK_1 << CS4245_MCLK1_SHIFT; + } else if (params_rate(params) <= 100000) { + tmp |= CS4245_DAC_FM_DOUBLE; + tmp2 |= CS4245_MCLK_1 << CS4245_MCLK1_SHIFT; + } else { + tmp |= CS4245_DAC_FM_QUAD; + tmp2 |= CS4245_MCLK_2 << CS4245_MCLK1_SHIFT; + } + data->cs4245_shadow[CS4245_DAC_CTRL_1] = tmp; + data->cs4245_shadow[CS4245_MCLK_FREQ] = tmp2; + cs4245_write_spi(chip, CS4245_DAC_CTRL_1); + cs4245_write_spi(chip, CS4245_MCLK_FREQ); } static void set_cs4245_adc_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { struct dg *data = chip->model_data; - u8 value; + unsigned char tmp; + unsigned char tmp2; - value = data->cs4245_regs[CS4245_ADC_CTRL] & ~CS4245_ADC_FM_MASK; - if (params_rate(params) <= 50000) - value |= CS4245_ADC_FM_SINGLE; - else if (params_rate(params) <= 100000) - value |= CS4245_ADC_FM_DOUBLE; - else - value |= CS4245_ADC_FM_QUAD; - cs4245_write_cached(chip, CS4245_ADC_CTRL, value); -} - -static inline unsigned int shift_bits(unsigned int value, - unsigned int shift_from, - unsigned int shift_to, - unsigned int mask) -{ - if (shift_from < shift_to) - return (value << (shift_to - shift_from)) & mask; - else - return (value >> (shift_from - shift_to)) & mask; + tmp = data->cs4245_shadow[CS4245_ADC_CTRL] & ~CS4245_ADC_FM_MASK; + tmp2 = data->cs4245_shadow[CS4245_MCLK_FREQ] & ~CS4245_MCLK2_MASK; + if (params_rate(params) <= 50000) { + tmp |= CS4245_ADC_FM_SINGLE; + tmp2 |= CS4245_MCLK_1 << CS4245_MCLK2_SHIFT; + } else if (params_rate(params) <= 100000) { + tmp |= CS4245_ADC_FM_DOUBLE; + tmp2 |= CS4245_MCLK_1 << CS4245_MCLK2_SHIFT; + } else { + tmp |= CS4245_ADC_FM_QUAD; + tmp2 |= CS4245_MCLK_2 << CS4245_MCLK2_SHIFT; + } + data->cs4245_shadow[CS4245_ADC_CTRL] = tmp; + data->cs4245_shadow[CS4245_MCLK_FREQ] = tmp2; + cs4245_write_spi(chip, CS4245_ADC_CTRL); + cs4245_write_spi(chip, CS4245_MCLK_FREQ); } static unsigned int adjust_dg_dac_routing(struct oxygen *chip, unsigned int play_routing) { - return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) | - shift_bits(play_routing, - OXYGEN_PLAY_DAC2_SOURCE_SHIFT, - OXYGEN_PLAY_DAC1_SOURCE_SHIFT, - OXYGEN_PLAY_DAC1_SOURCE_MASK) | - shift_bits(play_routing, - OXYGEN_PLAY_DAC1_SOURCE_SHIFT, - OXYGEN_PLAY_DAC2_SOURCE_SHIFT, - OXYGEN_PLAY_DAC2_SOURCE_MASK) | - shift_bits(play_routing, - OXYGEN_PLAY_DAC0_SOURCE_SHIFT, - OXYGEN_PLAY_DAC3_SOURCE_SHIFT, - OXYGEN_PLAY_DAC3_SOURCE_MASK); -} - -static int output_switch_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *info) -{ - static const char *const names[3] = { - "Speakers", "Headphones", "FP Headphones" - }; - - return snd_ctl_enum_info(info, 1, 3, names); -} - -static int output_switch_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - struct dg *data = chip->model_data; - - mutex_lock(&chip->mutex); - value->value.enumerated.item[0] = data->output_sel; - mutex_unlock(&chip->mutex); - return 0; -} - -static int output_switch_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - struct dg *data = chip->model_data; - u8 reg; - int changed; - - if (value->value.enumerated.item[0] > 2) - return -EINVAL; - - mutex_lock(&chip->mutex); - changed = value->value.enumerated.item[0] != data->output_sel; - if (changed) { - data->output_sel = value->value.enumerated.item[0]; - - reg = data->cs4245_regs[CS4245_SIGNAL_SEL] & - ~CS4245_A_OUT_SEL_MASK; - reg |= data->output_sel == 2 ? - CS4245_A_OUT_SEL_DAC : CS4245_A_OUT_SEL_HIZ; - cs4245_write_cached(chip, CS4245_SIGNAL_SEL, reg); - - cs4245_write_cached(chip, CS4245_DAC_A_CTRL, - data->output_sel ? data->hp_vol_att : 0); - cs4245_write_cached(chip, CS4245_DAC_B_CTRL, - data->output_sel ? data->hp_vol_att : 0); - - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, - data->output_sel == 1 ? GPIO_HP_REAR : 0, - GPIO_HP_REAR); - } - mutex_unlock(&chip->mutex); - return changed; -} - -static int hp_volume_offset_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *info) -{ - static const char *const names[3] = { - "< 64 ohms", "64-150 ohms", "150-300 ohms" - }; - - return snd_ctl_enum_info(info, 1, 3, names); -} - -static int hp_volume_offset_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - struct dg *data = chip->model_data; - - mutex_lock(&chip->mutex); - if (data->hp_vol_att > 2 * 7) - value->value.enumerated.item[0] = 0; - else if (data->hp_vol_att > 0) - value->value.enumerated.item[0] = 1; - else - value->value.enumerated.item[0] = 2; - mutex_unlock(&chip->mutex); - return 0; -} - -static int hp_volume_offset_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - static const s8 atts[3] = { 2 * 16, 2 * 7, 0 }; - struct oxygen *chip = ctl->private_data; struct dg *data = chip->model_data; - s8 att; - int changed; + unsigned int routing = 0; - if (value->value.enumerated.item[0] > 2) - return -EINVAL; - att = atts[value->value.enumerated.item[0]]; - mutex_lock(&chip->mutex); - changed = att != data->hp_vol_att; - if (changed) { - data->hp_vol_att = att; - if (data->output_sel) { - cs4245_write_cached(chip, CS4245_DAC_A_CTRL, att); - cs4245_write_cached(chip, CS4245_DAC_B_CTRL, att); - } + switch (data->pcm_output) { + case PLAYBACK_DST_HP: + case PLAYBACK_DST_HP_FP: + oxygen_write8_masked(chip, OXYGEN_PLAY_ROUTING, + OXYGEN_PLAY_MUTE23 | OXYGEN_PLAY_MUTE45 | + OXYGEN_PLAY_MUTE67, OXYGEN_PLAY_MUTE_MASK); + break; + case PLAYBACK_DST_20_CH: + routing = (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (2 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (1 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT); + oxygen_write8_masked(chip, OXYGEN_PLAY_ROUTING, + OXYGEN_PLAY_MUTE01 | OXYGEN_PLAY_MUTE23 | + OXYGEN_PLAY_MUTE45, OXYGEN_PLAY_MUTE_MASK); + break; + case PLAYBACK_DST_40_CH: + routing = (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (2 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (1 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT); + oxygen_write8_masked(chip, OXYGEN_PLAY_ROUTING, + OXYGEN_PLAY_MUTE01 | + OXYGEN_PLAY_MUTE23, OXYGEN_PLAY_MUTE_MASK); + break; + case PLAYBACK_DST_51_CH: + routing = (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (2 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (1 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT); + oxygen_write8_masked(chip, OXYGEN_PLAY_ROUTING, + OXYGEN_PLAY_MUTE01, OXYGEN_PLAY_MUTE_MASK); + break; } - mutex_unlock(&chip->mutex); - return changed; -} - -static int input_vol_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *info) -{ - info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - info->count = 2; - info->value.integer.min = 2 * -12; - info->value.integer.max = 2 * 12; - return 0; -} - -static int input_vol_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - struct dg *data = chip->model_data; - unsigned int idx = ctl->private_value; - - mutex_lock(&chip->mutex); - value->value.integer.value[0] = data->input_vol[idx][0]; - value->value.integer.value[1] = data->input_vol[idx][1]; - mutex_unlock(&chip->mutex); - return 0; + return routing; } -static int input_vol_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) +static void dump_cs4245_registers(struct oxygen *chip, + struct snd_info_buffer *buffer) { - struct oxygen *chip = ctl->private_data; struct dg *data = chip->model_data; - unsigned int idx = ctl->private_value; - int changed = 0; - - if (value->value.integer.value[0] < 2 * -12 || - value->value.integer.value[0] > 2 * 12 || - value->value.integer.value[1] < 2 * -12 || - value->value.integer.value[1] > 2 * 12) - return -EINVAL; - mutex_lock(&chip->mutex); - changed = data->input_vol[idx][0] != value->value.integer.value[0] || - data->input_vol[idx][1] != value->value.integer.value[1]; - if (changed) { - data->input_vol[idx][0] = value->value.integer.value[0]; - data->input_vol[idx][1] = value->value.integer.value[1]; - if (idx == data->input_sel) { - cs4245_write_cached(chip, CS4245_PGA_A_CTRL, - data->input_vol[idx][0]); - cs4245_write_cached(chip, CS4245_PGA_B_CTRL, - data->input_vol[idx][1]); - } - } - mutex_unlock(&chip->mutex); - return changed; -} - -static DECLARE_TLV_DB_SCALE(cs4245_pga_db_scale, -1200, 50, 0); - -static int input_sel_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *info) -{ - static const char *const names[4] = { - "Mic", "Aux", "Front Mic", "Line" - }; + unsigned int reg; - return snd_ctl_enum_info(info, 1, 4, names); + snd_iprintf(buffer, "\nCS4245:"); + cs4245_read_spi(chip, CS4245_INT_STATUS); + for (reg = 0; reg < ARRAY_SIZE(data->cs4245_shadow); ++reg) + snd_iprintf(buffer, " %02x", data->cs4245_shadow[reg]); + snd_iprintf(buffer, "\n"); } -static int input_sel_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) +/* Put the codec into low power mode, disable audio output, save registers */ +static void dg_suspend(struct oxygen *chip) { - struct oxygen *chip = ctl->private_data; struct dg *data = chip->model_data; - - mutex_lock(&chip->mutex); - value->value.enumerated.item[0] = data->input_sel; - mutex_unlock(&chip->mutex); - return 0; + data->cs4245_shadow[CS4245_POWER_CTRL] = CS4245_PDN_MIC | + CS4245_PDN_ADC | CS4245_PDN_DAC | CS4245_PDN; + cs4245_write_spi(chip, CS4245_POWER_CTRL); + cs4245_dump_regs(chip, true); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, 0, GPIO_OUTPUT_ENABLE); } -static int input_sel_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) +static void dg_resume(struct oxygen *chip) { - static const u8 sel_values[4] = { - CS4245_SEL_MIC, - CS4245_SEL_INPUT_1, - CS4245_SEL_INPUT_2, - CS4245_SEL_INPUT_4 - }; - struct oxygen *chip = ctl->private_data; struct dg *data = chip->model_data; - int changed; - - if (value->value.enumerated.item[0] > 3) - return -EINVAL; - - mutex_lock(&chip->mutex); - changed = value->value.enumerated.item[0] != data->input_sel; - if (changed) { - data->input_sel = value->value.enumerated.item[0]; - - cs4245_write(chip, CS4245_ANALOG_IN, - (data->cs4245_regs[CS4245_ANALOG_IN] & - ~CS4245_SEL_MASK) | - sel_values[data->input_sel]); - - cs4245_write_cached(chip, CS4245_PGA_A_CTRL, - data->input_vol[data->input_sel][0]); - cs4245_write_cached(chip, CS4245_PGA_B_CTRL, - data->input_vol[data->input_sel][1]); - - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, - data->input_sel ? 0 : GPIO_INPUT_ROUTE, - GPIO_INPUT_ROUTE); - } - mutex_unlock(&chip->mutex); - return changed; + data->cs4245_shadow[CS4245_POWER_CTRL] = 0; + cs4245_dump_regs(chip, false); + msleep_interruptible(1000); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, 0xffff, + GPIO_OUTPUT_ENABLE); } -static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) -{ - static const char *const names[2] = { "Active", "Frozen" }; - - return snd_ctl_enum_info(info, 1, 2, names); -} - -static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +static void dg_cleanup(struct oxygen *chip) { - struct oxygen *chip = ctl->private_data; - struct dg *data = chip->model_data; - - value->value.enumerated.item[0] = - !!(data->cs4245_regs[CS4245_ADC_CTRL] & CS4245_HPF_FREEZE); - return 0; + dg_suspend(chip); } -static int hpf_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - struct dg *data = chip->model_data; - u8 reg; - int changed; - - mutex_lock(&chip->mutex); - reg = data->cs4245_regs[CS4245_ADC_CTRL] & ~CS4245_HPF_FREEZE; - if (value->value.enumerated.item[0]) - reg |= CS4245_HPF_FREEZE; - changed = reg != data->cs4245_regs[CS4245_ADC_CTRL]; - if (changed) - cs4245_write(chip, CS4245_ADC_CTRL, reg); - mutex_unlock(&chip->mutex); - return changed; -} - -#define INPUT_VOLUME(xname, index) { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .info = input_vol_info, \ - .get = input_vol_get, \ - .put = input_vol_put, \ - .tlv = { .p = cs4245_pga_db_scale }, \ - .private_value = index, \ -} -static const struct snd_kcontrol_new dg_controls[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Output Playback Enum", - .info = output_switch_info, - .get = output_switch_get, - .put = output_switch_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Headphones Impedance Playback Enum", - .info = hp_volume_offset_info, - .get = hp_volume_offset_get, - .put = hp_volume_offset_put, - }, - INPUT_VOLUME("Mic Capture Volume", 0), - INPUT_VOLUME("Aux Capture Volume", 1), - INPUT_VOLUME("Front Mic Capture Volume", 2), - INPUT_VOLUME("Line Capture Volume", 3), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = input_sel_info, - .get = input_sel_get, - .put = input_sel_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "ADC High-pass Filter Capture Enum", - .info = hpf_info, - .get = hpf_get, - .put = hpf_put, - }, -}; - -static int dg_control_filter(struct snd_kcontrol_new *template) +/* Disable built-in volume control, we have only headphone volume */ +static int xonar_control_filter(struct snd_kcontrol_new *template) { if (!strncmp(template->name, "Master Playback ", 16)) return 1; return 0; } -static int dg_mixer_init(struct oxygen *chip) -{ - unsigned int i; - int err; - - for (i = 0; i < ARRAY_SIZE(dg_controls); ++i) { - err = snd_ctl_add(chip->card, - snd_ctl_new1(&dg_controls[i], chip)); - if (err < 0) - return err; - } - return 0; -} - -static void dump_cs4245_registers(struct oxygen *chip, - struct snd_info_buffer *buffer) -{ - struct dg *data = chip->model_data; - unsigned int i; - - snd_iprintf(buffer, "\nCS4245:"); - for (i = 1; i <= 0x10; ++i) - snd_iprintf(buffer, " %02x", data->cs4245_regs[i]); - snd_iprintf(buffer, "\n"); -} - struct oxygen_model model_xonar_dg = { .longname = "C-Media Oxygen HD Audio", .chip = "CMI8786", .init = dg_init, - .control_filter = dg_control_filter, - .mixer_init = dg_mixer_init, + .control_filter = xonar_control_filter, + .mixer_init = xonar_dg_mixer_init, .cleanup = dg_cleanup, .suspend = dg_suspend, .resume = dg_resume, @@ -596,11 +339,11 @@ struct oxygen_model model_xonar_dg = { .model_data_size = sizeof(struct dg), .device_config = PLAYBACK_0_TO_I2S | PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2 | + CAPTURE_0_FROM_I2S_1 | CAPTURE_1_FROM_SPDIF, .dac_channels_pcm = 6, .dac_channels_mixer = 0, - .function_flags = OXYGEN_FUNCTION_SPI, + .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, .dac_mclks = OXYGEN_MCLKS(256, 128, 128), .adc_mclks = OXYGEN_MCLKS(256, 128, 128), .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, diff -uprN linux-3.12/sound/pci/oxygen/xonar_dg.h linux-3.12-my/sound/pci/oxygen/xonar_dg.h --- linux-3.12/sound/pci/oxygen/xonar_dg.h 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/xonar_dg.h 2013-11-19 01:06:12.000000000 +0400 @@ -3,6 +3,48 @@ #include "oxygen.h" +#define GPIO_MAGIC 0x0008 +#define GPIO_HP_DETECT 0x0010 +#define GPIO_INPUT_ROUTE 0x0060 +#define GPIO_HP_REAR 0x0080 +#define GPIO_OUTPUT_ENABLE 0x0100 + +#define CAPTURE_SRC_MIC 0 +#define CAPTURE_SRC_FP_MIC 1 +#define CAPTURE_SRC_LINE 2 +#define CAPTURE_SRC_AUX 3 + +#define PLAYBACK_DST_HP 0 +#define PLAYBACK_DST_HP_FP 1 +#define PLAYBACK_DST_20_CH 2 +#define PLAYBACK_DST_40_CH 3 +#define PLAYBACK_DST_51_CH 4 + extern struct oxygen_model model_xonar_dg; +/* Xonar DG Specific Data */ +struct dg { + unsigned char cs4245_shadow[16]; + /* Output select: headphone/speakers */ + unsigned char pcm_output; + /* Input select: MIC, Line, Aux */ + unsigned char pga_source; + /* Independend capture volumes */ + char mic_vol_l; + char mic_vol_r; + bool mic_m; + char line_vol_l; + char line_vol_r; + bool line_m; + char aux_vol_l; + char aux_vol_r; + bool aux_m; +}; + +/* Xonar DG SPI/I2C control routines */ +int cs4245_write_spi(struct oxygen *chip, u8 reg); + +/* Xonar DG Mixer */ +int xonar_dg_mixer_init(struct oxygen *chip); + #endif diff -uprN linux-3.12/sound/pci/oxygen/xonar_dg_mixer.c linux-3.12-my/sound/pci/oxygen/xonar_dg_mixer.c --- linux-3.12/sound/pci/oxygen/xonar_dg_mixer.c 1970-01-01 03:00:00.000000000 +0300 +++ linux-3.12-my/sound/pci/oxygen/xonar_dg_mixer.c 2013-11-19 01:04:16.000000000 +0400 @@ -0,0 +1,536 @@ +/* + * Mixer controls for the Xonar DG/DGX + * + * Copyright (c) Roman Volkov <v1ron(a)mail.ru> + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver 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. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, see <http://www.gnu.org/licenses/>. + */ + +/* + * It is not good that we do not have software volume control + * as the Windows driver does. CS4361 cannot attenuate the volume. + * When we are switching from headphones to the speakers, + * which are controlled by the simple DAC CS4361, our headphones + * in the rear jack can stun us. + */ + +#include <linux/pci.h> +#include <linux/delay.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/pcm.h> +#include <sound/tlv.h> +#include "oxygen.h" +#include "xonar_dg.h" +#include "cs4245.h" + +/* CS4245 Headphone Channels A&B Volume Control */ +static int xonar_dg_dac_stereo_volume_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = 0; + info->value.integer.max = 255; + return 0; +} + +static int xonar_dg_dac_stereo_volume_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + unsigned int tmp; + + mutex_lock(&chip->mutex); + tmp = (~data->cs4245_shadow[CS4245_DAC_A_CTRL]) & 255; + val->value.integer.value[0] = tmp; + tmp = (~data->cs4245_shadow[CS4245_DAC_B_CTRL]) & 255; + val->value.integer.value[1] = tmp; + mutex_unlock(&chip->mutex); + return 0; +} + +static int xonar_dg_dac_stereo_volume_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + if ((val->value.integer.value[0] > 255) || + (val->value.integer.value[0] < 0) || + (val->value.integer.value[1] > 255) || + (val->value.integer.value[1] < 0)) { + return -EINVAL; + } + mutex_lock(&chip->mutex); + data->cs4245_shadow[CS4245_DAC_A_CTRL] = ~val->value.integer.value[0]; + data->cs4245_shadow[CS4245_DAC_B_CTRL] = ~val->value.integer.value[1]; + cs4245_write_spi(chip, CS4245_DAC_A_CTRL); + cs4245_write_spi(chip, CS4245_DAC_B_CTRL); + mutex_unlock(&chip->mutex); + return 0; +} + +/* CS4245 DAC Mute */ +static int xonar_dg_dac_mute_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + val->value.integer.value[0] = ((data->cs4245_shadow[CS4245_DAC_CTRL_1] + & CS4245_MUTE_DAC) == 0); + mutex_unlock(&chip->mutex); + return 0; +} + +static int xonar_dg_dac_mute_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + if (val->value.integer.value[0] > 1) + return -EINVAL; + mutex_lock(&chip->mutex); + data->cs4245_shadow[CS4245_DAC_CTRL_1] &= ~CS4245_MUTE_DAC; + data->cs4245_shadow[CS4245_DAC_CTRL_1] |= + (~val->value.integer.value[0] << 2) & CS4245_MUTE_DAC; + cs4245_write_spi(chip, CS4245_DAC_CTRL_1); + mutex_unlock(&chip->mutex); + return 0; +} + +/* Xonar DG Output Select */ +static int xonar_dg_pcm_route_apply(struct oxygen *chip) +{ + struct dg *data = chip->model_data; + + if (data->pcm_output == PLAYBACK_DST_HP) { + /* Mute FP (aux output) amplifier, switch rear jack to CS4245 */ + oxygen_write8_masked(chip, OXYGEN_GPIO_DATA, 0xff, + GPIO_HP_REAR); + data->cs4245_shadow[CS4245_SIGNAL_SEL] &= + ~CS4245_A_OUT_SEL_MASK; + } else if (data->pcm_output == PLAYBACK_DST_HP_FP) { + /* + * Unmute FP amplifier, switch rear jack to CS4361, + * I2S ports 2,3,4 should be inactive + */ + oxygen_write8_masked(chip, OXYGEN_GPIO_DATA, 0x00, 0x80); + data->cs4245_shadow[CS4245_SIGNAL_SEL] &= + ~CS4245_A_OUT_SEL_MASK; + data->cs4245_shadow[CS4245_SIGNAL_SEL] |= CS4245_A_OUT_SEL_DAC; + } else { + /* + * 2.0, 4.0, 5.1: switch to CS4361, mute FP amp., + * and just change playback routing / DMA channels + */ + oxygen_write8_masked(chip, OXYGEN_GPIO_DATA, 0x00, 0x80); + data->cs4245_shadow[CS4245_SIGNAL_SEL] &= + ~CS4245_A_OUT_SEL_MASK; + } + cs4245_write_spi(chip, CS4245_SIGNAL_SEL); + return 0; +} + +static int xonar_dg_pcm_route_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[5] = { + "Stereo Headphones", + "Stereo Headphones FP", + "Stereo Speakers", + "4 Speakers", + "5.1 Speakers", + }; + + return snd_ctl_enum_info(info, 1, 5, names); +} + +static int xonar_dg_pcm_route_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + val->value.integer.value[0] = data->pcm_output; + mutex_unlock(&chip->mutex); + + return 0; +} + +static int xonar_dg_pcm_route_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + int ret; + + mutex_lock(&chip->mutex); + data->pcm_output = val->value.integer.value[0]; + ret = xonar_dg_pcm_route_apply(chip); + oxygen_update_dac_routing(chip); + mutex_unlock(&chip->mutex); + + return ret; +} + +/* CS4245 PGA Source */ +static int xonar_dg_pga_mute_reg_toggle(struct oxygen *chip, bool mute); +static int xonar_dg_pga_volume_reg_write(struct oxygen *chip, char left, + char right); + +static int xonar_dg_pga_source_apply(struct oxygen *chip) +{ + struct dg *data = chip->model_data; + + data->cs4245_shadow[CS4245_ANALOG_IN] &= ~CS4245_SEL_MASK; + if (data->pga_source == CAPTURE_SRC_FP_MIC) + data->cs4245_shadow[CS4245_ANALOG_IN] |= CS4245_SEL_INPUT_2; + else if (data->pga_source == CAPTURE_SRC_LINE) + data->cs4245_shadow[CS4245_ANALOG_IN] |= CS4245_SEL_INPUT_4; + else if (data->pga_source != CAPTURE_SRC_MIC) + data->cs4245_shadow[CS4245_ANALOG_IN] |= CS4245_SEL_INPUT_1; + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, 0xffff, GPIO_INPUT_ROUTE); + return cs4245_write_spi(chip, CS4245_ANALOG_IN); +} + +static int xonar_dg_pga_source_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const ps_names[4] = { + "Microphone", + "Microphone FP", + "Line In", + "Aux" + }; + + return snd_ctl_enum_info(info, 1, 4, ps_names); +} + +static int xonar_dg_pga_source_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + val->value.integer.value[0] = data->pga_source; + mutex_unlock(&chip->mutex); + + return 0; +} + +static int xonar_dg_pga_source_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + bool mute = false; + long route = val->value.integer.value[0]; + char left = 0; + char right = 0; + int ret; + + if (route > CAPTURE_SRC_AUX) + return -EINVAL; + + mutex_lock(&chip->mutex); + data->pga_source = route; + ret = xonar_dg_pga_source_apply(chip); + if (ret < 0) + goto err; + switch (route) { + case CAPTURE_SRC_MIC: + case CAPTURE_SRC_FP_MIC: + mute = data->mic_m; + left = data->mic_vol_l; + right = data->mic_vol_r; + break; + case CAPTURE_SRC_LINE: + mute = data->line_m; + left = data->line_vol_l; + right = data->line_vol_r; + break; + case CAPTURE_SRC_AUX: + mute = data->aux_m; + left = data->aux_vol_l; + right = data->aux_vol_r; + break; + } + ret = xonar_dg_pga_mute_reg_toggle(chip, mute); + if (ret < 0) + goto err; + ret = xonar_dg_pga_volume_reg_write(chip, left, right); + if (ret < 0) + goto err; +err: + mutex_unlock(&chip->mutex); + + return ret; +} + +/* Common routines to access capture volume register - Mic, Line, Aux */ +static int xonar_dg_pga_volume_reg_write(struct oxygen *chip, char left, + char right) +{ + struct dg *data = chip->model_data; + int ret; + + data->cs4245_shadow[CS4245_PGA_A_CTRL] = left; + data->cs4245_shadow[CS4245_PGA_B_CTRL] = right; + ret = cs4245_write_spi(chip, CS4245_PGA_A_CTRL); + if (ret < 0) + return ret; + return cs4245_write_spi(chip, CS4245_PGA_B_CTRL); +} + +static int xonar_dg_pga_volume_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = -24; + info->value.integer.max = 24; + return 0; +} + +static int xonar_dg_pga_volume_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + switch (ctl->private_value) { + case CAPTURE_SRC_MIC: + val->value.integer.value[0] = data->mic_vol_l; + val->value.integer.value[1] = data->mic_vol_r; + break; + case CAPTURE_SRC_LINE: + val->value.integer.value[0] = data->line_vol_l; + val->value.integer.value[1] = data->line_vol_r; + break; + case CAPTURE_SRC_AUX: + val->value.integer.value[0] = data->aux_vol_l; + val->value.integer.value[1] = data->aux_vol_r; + break; + } + mutex_unlock(&chip->mutex); + + return 0; +} + +static int xonar_dg_pga_volume_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + long left = val->value.integer.value[0]; + long right = val->value.integer.value[1]; + int ret = 0; + + if ((left > 24) || (left < -24) || (right > 24) || (right < -24)) + return -EINVAL; + + mutex_lock(&chip->mutex); + switch (ctl->private_value) { + case CAPTURE_SRC_MIC: + data->mic_vol_l = left; + data->mic_vol_r = right; + break; + case CAPTURE_SRC_LINE: + data->line_vol_l = left; + data->line_vol_r = right; + break; + case CAPTURE_SRC_AUX: + data->aux_vol_l = left; + data->aux_vol_r = right; + break; + } + switch (data->pga_source) { + case CAPTURE_SRC_MIC: + case CAPTURE_SRC_FP_MIC: + ret = xonar_dg_pga_volume_reg_write(chip, data->mic_vol_l, + data->mic_vol_r); + break; + case CAPTURE_SRC_LINE: + ret = xonar_dg_pga_volume_reg_write(chip, data->line_vol_l, + data->line_vol_r); + break; + case CAPTURE_SRC_AUX: + ret = xonar_dg_pga_volume_reg_write(chip, data->aux_vol_l, + data->aux_vol_r); + break; + } + mutex_unlock(&chip->mutex); + + return ret; +} + +/* Common routines to access capture mute register - Mic, Line, Aux */ +static int xonar_dg_pga_mute_reg_toggle(struct oxygen *chip, bool mute) +{ + struct dg *data = chip->model_data; + + data->cs4245_shadow[CS4245_ADC_CTRL] &= ~CS4245_MUTE_ADC; + data->cs4245_shadow[CS4245_ADC_CTRL] |= ((mute == 0) << 2) & + CS4245_MUTE_ADC; + return cs4245_write_spi(chip, CS4245_ADC_CTRL); +} + +static int xonar_dg_pga_mute_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + + mutex_lock(&chip->mutex); + switch (ctl->private_value) { + case CAPTURE_SRC_MIC: + val->value.integer.value[0] = data->mic_m != false ? 1 : 0; + break; + case CAPTURE_SRC_LINE: + val->value.integer.value[0] = data->mic_m != false ? 1 : 0; + break; + case CAPTURE_SRC_AUX: + val->value.integer.value[0] = data->mic_m != false ? 1 : 0; + break; + } + mutex_unlock(&chip->mutex); + + return 0; +} + +static int xonar_dg_pga_mute_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *val) +{ + struct oxygen *chip = ctl->private_data; + struct dg *data = chip->model_data; + long tmp = val->value.integer.value[0]; + int ret = 0; + + if ((tmp > 1) || (tmp < 0)) + return -EINVAL; + + mutex_lock(&chip->mutex); + switch (ctl->private_value) { + case CAPTURE_SRC_MIC: + data->mic_m = tmp != 0; + break; + case CAPTURE_SRC_LINE: + data->line_m = tmp != 0; + break; + case CAPTURE_SRC_AUX: + data->aux_m = tmp != 0; + break; + } + switch (data->pga_source) { + case CAPTURE_SRC_MIC: + case CAPTURE_SRC_FP_MIC: + ret = xonar_dg_pga_mute_reg_toggle(chip, data->mic_m); + break; + case CAPTURE_SRC_LINE: + ret = xonar_dg_pga_mute_reg_toggle(chip, data->line_m); + break; + case CAPTURE_SRC_AUX: + ret = xonar_dg_pga_mute_reg_toggle(chip, data->aux_m); + break; + } + mutex_unlock(&chip->mutex); + + return ret; +} + +#define CAPTURE_VOLUME(xname, private) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = xonar_dg_pga_volume_info, \ + .get = xonar_dg_pga_volume_get, \ + .put = xonar_dg_pga_volume_put, \ + .tlv = { .p = xonar_dg_pga_db_scale, }, \ + .private_value = private, \ + } +#define CAPTURE_SWITCH(xname, private) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xonar_dg_pga_mute_get, \ + .put = xonar_dg_pga_mute_put, \ + .private_value = private, \ + } + +static const DECLARE_TLV_DB_MINMAX(xonar_dg_hp_db_scale, -12550, 0); +static const DECLARE_TLV_DB_MINMAX(xonar_dg_pga_db_scale, -1200, 1200); +static const struct snd_kcontrol_new xonar_dg_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = xonar_dg_dac_stereo_volume_info, + .get = xonar_dg_dac_stereo_volume_get, + .put = xonar_dg_dac_stereo_volume_put, + .tlv = { .p = xonar_dg_hp_db_scale, }, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = xonar_dg_dac_mute_get, + .put = xonar_dg_dac_mute_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output Playback Enum", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = xonar_dg_pcm_route_info, + .get = xonar_dg_pcm_route_get, + .put = xonar_dg_pcm_route_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Capture Enum", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = xonar_dg_pga_source_info, + .get = xonar_dg_pga_source_get, + .put = xonar_dg_pga_source_put, + }, + CAPTURE_VOLUME("Mic Capture Volume", CAPTURE_SRC_MIC), + CAPTURE_SWITCH("Mic Capture Switch", CAPTURE_SRC_MIC), + CAPTURE_VOLUME("Line Capture Volume", CAPTURE_SRC_LINE), + CAPTURE_SWITCH("Line Capture Switch", CAPTURE_SRC_LINE), + CAPTURE_VOLUME("Aux Capture Volume", CAPTURE_SRC_AUX), + CAPTURE_SWITCH("Aux Capture Switch", CAPTURE_SRC_AUX), +}; + +int xonar_dg_mixer_init(struct oxygen *chip) +{ + unsigned int i; + int ret; + for (i = 0; i < ARRAY_SIZE(xonar_dg_controls); i++) { + ret = snd_ctl_add(chip->card, + snd_ctl_new1(&xonar_dg_controls[i], chip)); + if (ret < 0) + return ret; + } + return 0; +}

2 2

[alsa-devel] [PATCH 07/12] ASoC: rcar: rename GEN2_SRU to GEN2_SCU
by Kuninori Morimoto 20 Nov '13

20 Nov '13

Gen2 has SCU. SRU is for Gen1 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/rcar_snd.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index 12afab1..a818ff7 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -18,7 +18,7 @@ #define RSND_GEN1_ADG 1 #define RSND_GEN1_SSI 2 -#define RSND_GEN2_SRU 0 +#define RSND_GEN2_SCU 0 #define RSND_GEN2_ADG 1 #define RSND_GEN2_SSIU 2 #define RSND_GEN2_SSI 3 -- 1.7.9.5

2 3

[alsa-devel] [PATCH 1/2] snd-oxygen: Changes for oxygen driver to fix the Xonar DG card
by Roman Volkov 20 Nov '13

20 Nov '13

This patch does some preparation of existing Oxygen driver. Some constants added to the header files. To perform the SPI reading, there is a new function to wait when the SPI finishes a transaction. Channel routing must be updated when the user selects the audio output, and there are some extra symbols exported from the snd-oxygen-lib module. Signed-off-by: Roman I Volkov <v1ron(a)mail.ru> --- Should not affect other models of Oxygen driver, but who knows. diff -uprN linux-3.12/sound/pci/oxygen/cs4245.h linux-3.12-my/sound/pci/oxygen/cs4245.h --- linux-3.12/sound/pci/oxygen/cs4245.h 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/cs4245.h 2013-11-15 15:46:40.000000000 +0400 @@ -103,5 +103,6 @@ #define CS4245_ADC_UNDRFL 0x01 -#define CS4245_SPI_ADDRESS (0x9e << 16) -#define CS4245_SPI_WRITE (0 << 16) +#define CS4245_SPI_ADDRESS 0x9e +#define CS4245_SPI_WRITE 0 +#define CS4245_SPI_READ 1 diff -uprN linux-3.12/sound/pci/oxygen/oxygen.h linux-3.12-my/sound/pci/oxygen/oxygen.h --- linux-3.12/sound/pci/oxygen/oxygen.h 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/oxygen.h 2013-11-15 15:47:38.000000000 +0400 @@ -198,7 +198,8 @@ void oxygen_write_ac97(struct oxygen *ch void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec, unsigned int index, u16 data, u16 mask); -void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data); +int oxygen_wait_spi(struct oxygen *chip); +int oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data); void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data); void oxygen_reset_uart(struct oxygen *chip); diff -uprN linux-3.12/sound/pci/oxygen/oxygen_io.c linux-3.12-my/sound/pci/oxygen/oxygen_io.c --- linux-3.12/sound/pci/oxygen/oxygen_io.c 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/oxygen_io.c 2013-11-19 00:56:12.000000000 +0400 @@ -105,8 +105,8 @@ static int oxygen_ac97_wait(struct oxyge * Reading the status register also clears the bits, so we have to save * the read bits in status. */ - wait_event_timeout(chip->ac97_waitqueue, - ({ status |= oxygen_read8(chip, OXYGEN_AC97_INTERRUPT_STATUS); + wait_event_timeout(chip->ac97_waitqueue, ({ status |= oxygen_read8(chip, + OXYGEN_AC97_INTERRUPT_STATUS); status & mask; }), msecs_to_jiffies(1) + 1); /* @@ -194,23 +194,35 @@ void oxygen_write_ac97_masked(struct oxy } EXPORT_SYMBOL(oxygen_write_ac97_masked); -void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) +int oxygen_wait_spi(struct oxygen *chip) { - unsigned int count; + /* Higher interval to be sure - 100 microseconds */ + unsigned int count = 100; - /* should not need more than 30.72 us (24 * 1.28 us) */ - count = 10; - while ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & OXYGEN_SPI_BUSY) - && count > 0) { - udelay(4); - --count; + for (;;) { + if ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & + OXYGEN_SPI_BUSY) == 0) + break; + if (count == 0) { + snd_printk(KERN_ERR "oxygen: spi wait timeout\n"); + return -EIO; + } + udelay(1); + count--; } + return 0; +} +EXPORT_SYMBOL(oxygen_wait_spi); +int oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) +{ oxygen_write8(chip, OXYGEN_SPI_DATA1, data); oxygen_write8(chip, OXYGEN_SPI_DATA2, data >> 8); if (control & OXYGEN_SPI_DATA_LENGTH_3) oxygen_write8(chip, OXYGEN_SPI_DATA3, data >> 16); oxygen_write8(chip, OXYGEN_SPI_CONTROL, control); + + return oxygen_wait_spi(chip); } EXPORT_SYMBOL(oxygen_write_spi); diff -uprN linux-3.12/sound/pci/oxygen/oxygen_mixer.c linux-3.12-my/sound/pci/oxygen/oxygen_mixer.c --- linux-3.12/sound/pci/oxygen/oxygen_mixer.c 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/oxygen_mixer.c 2013-11-18 14:03:54.000000000 +0400 @@ -190,6 +190,7 @@ void oxygen_update_dac_routing(struct ox if (chip->model.update_center_lfe_mix) chip->model.update_center_lfe_mix(chip, chip->dac_routing > 2); } +EXPORT_SYMBOL(oxygen_update_dac_routing); static int upmix_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { diff -uprN linux-3.12/sound/pci/oxygen/oxygen_regs.h linux-3.12-my/sound/pci/oxygen/oxygen_regs.h --- linux-3.12/sound/pci/oxygen/oxygen_regs.h 2013-11-04 03:41:51.000000000 +0400 +++ linux-3.12-my/sound/pci/oxygen/oxygen_regs.h 2013-11-18 13:58:11.000000000 +0400 @@ -318,6 +318,7 @@ #define OXYGEN_PLAY_MUTE23 0x0002 #define OXYGEN_PLAY_MUTE45 0x0004 #define OXYGEN_PLAY_MUTE67 0x0008 +#define OXYGEN_PLAY_MUTE_MASK 0x000f #define OXYGEN_PLAY_MULTICH_MASK 0x0010 #define OXYGEN_PLAY_MULTICH_I2S_DAC 0x0000 #define OXYGEN_PLAY_MULTICH_AC97 0x0010

2 2

[alsa-devel] (no subject)
by Xiubo Li 20 Nov '13

20 Nov '13

Hello, This patch series is mostly Freescale's SAI SoC Digital Audio Interface driver implementation. And the implementation is only compatible with device tree definition. This patch series is based on linux-next and has been tested on Vybrid VF610 Tower board using device tree. Changed in v2: - Use default settings for the generic dmaengine PCM driver. - Separate receive and transmit setting in most functions, but some couldn't for the HW limitation. - Drop some not reduntant code. - Use devm_snd_soc_register_component() instead of snd_soc_register_component(). - Use devm_snd_soc_register_card() instead of devm_snd_soc_register_card(). - Adjust the code sentences sequence. - Make the namespacing consistent. - Rename CONFIG_SND_SOC_FSL_SGTL5000 to CONFIG_SND_SOC_FSL_SGTL5000_VF610. - Drop some meaningless lines. - Rename the binding document file. Added in v1: - Add SAI SoC Digital Audio Interface driver. - Add Freescale SAI ALSA SoC Digital Audio Interface node for VF610. - Enables SAI ALSA SoC DAI device for Vybrid VF610 TOWER board. - Add device tree bindings for Freescale SAI. - Revise the bugs about the sgt15000 codec. - Add SGT15000 based audio machine driver. - Enable SGT15000 codec based audio driver node for VF610. - Add device tree bindings for Freescale VF610 sound.

8 41

[alsa-devel] [PATCH] ASoC: soc-pcm: fix incorrect channels and sample_bits wiping off
by Nicolin Chen 20 Nov '13

20 Nov '13

This patch wipes off the corresponding channels and sample_bits for codec_dai. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- @Mark Brown Sir, I find the previous symmetry patch hasn't been merged into for-next. So are you willing to apply this one or let me refine the previous PATCH and resend that one? sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e1c5a00..60d9cfb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -452,8 +452,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) if (!codec_dai->active) { codec_dai->rate = 0; - cpu_dai->channels = 0; - cpu_dai->sample_bits = 0; + codec_dai->channels = 0; + codec_dai->sample_bits = 0; } /* Muting the DAC suppresses artifacts caused during digital -- 1.8.4

2 2

[alsa-devel] [PATCH 08/10] ASoC: ux500_pcm: Differentiate between pdata and DT initialisation
by Lee Jones 19 Nov '13

19 Nov '13

If booting with full DT support (i.e. DMA too, the last piece of the puzzle), then we don't need to use the compatible request channel call back and, due to the work we laid down earlier in this patch-set, we can use core function calls to populate the DMA slave_config. We also require slightly different flags to inform the core that we 'are' booting with DT. Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_pcm.c | 28 +++++++++++++++++++++++----- 1 file changed, 23 insertions(+), 5 deletions(-) diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index ce554de..acfec98 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -139,15 +139,33 @@ static const struct snd_dmaengine_pcm_config ux500_dmaengine_pcm_config = { .prepare_slave_config = ux500_pcm_prepare_slave_config, }; +static const struct snd_dmaengine_pcm_config ux500_dmaengine_of_pcm_config = { + .pcm_hardware = &ux500_pcm_hw, + .prealloc_buffer_size = 128 * 1024, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, +}; + int ux500_pcm_register_platform(struct platform_device *pdev) { + const struct snd_dmaengine_pcm_config *pcm_config; + struct device_node *np = pdev->dev.of_node; + unsigned int pcm_flags; int ret; - ret = snd_dmaengine_pcm_register(&pdev->dev, - &ux500_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_COMPAT | - SND_DMAENGINE_PCM_FLAG_NO_DT); + if (np) { + pcm_config = &ux500_dmaengine_of_pcm_config; + + pcm_flags = SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_COMPAT; + } else { + pcm_config = &ux500_dmaengine_pcm_config; + + pcm_flags = SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_COMPAT; + } + + ret = snd_dmaengine_pcm_register(&pdev->dev, pcm_config, pcm_flags); if (ret < 0) { dev_err(&pdev->dev, "%s: ERROR: Failed to register platform '%s' (%d)!\n", -- 1.8.1.2

3 4

[alsa-devel] [PATCH v2 1/2] ASOC: tegra: fix matching of AC97 components
by Lucas Stach 19 Nov '13

19 Nov '13

Matching works completely based on the cpu of_node. Signed-off-by: Lucas Stach <dev(a)lynxeye.de> Acked-by: Stephen Warren <swarren(a)nvidia.com> --- v2: no changes CC: <alsa-devel(a)alsa-project.org> CC: Mark Brown <broonie(a)linaro.org> CC: Stephen Warren <swarren(a)wwwdotorg.org> --- sound/soc/tegra/tegra_wm9712.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 5e11963..45b5789 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -55,7 +55,6 @@ static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link tegra_wm9712_dai = { .name = "AC97 HiFi", .stream_name = "AC97 HiFi", - .cpu_dai_name = "tegra20-ac97", .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", .init = tegra_wm9712_init, -- 1.8.3.1

1 1