September 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 00/14] SPDIF support
by Russell King - ARM Linux 04 Sep '13

04 Sep '13

This is my latest set of patches to get SPDIF working on the Dove Cubox platform. As per the last time, I've left out the "card" part of this, because that is only useful for non-DT setups, which aren't supported in mainline. The first seven patches have already been applied by Mark, and are included here for completeness by anyone who wishes to test this series. There is very little change between this and the previous set of patches. The HACK patch is gone, as I've found out through much digging and experimentation with DAPM (inspite of the complete lack of assistance from ASoC developers) how to use the stream widgets without the widget overwriting bug getting in the way. The CPU DAI widgets are attached to the CPU DAI context, and their access to the driver private data updated accordingly. Why attaching them to the CPU DAI context previously didn't work, I'm not entirely sure, but that certainly previously failed in testing (it may have been due to the duplicated DAI routes and/or the lack of output pin on the SPDIF tranceiver.) I've added three patches: one to add IEC958 channel status setting, another to prefer the external clock over the internal clock, and removal of the IEC958 subframe support (see the commit message for that patch why this can not be supported.) This is a non-DPCM driver, and is the absolute minimum solution which can provide correct hardware behaviour while still being conformant with single zImage kernels and not causing any known regressions. The same conditions apply as per my previous posting - the DAI link needs to be setup and the associated DAPM routes to tell the CPU DAI which outputs are in use, like this: DAI link: .name = "S/PDIF1", .stream_name = "IEC958 Playback", .platform_name = "mvebu-audio.1", .cpu_dai_name = "mvebu-audio.1", .codec_dai_name = "dit-hifi", .codec_name = "spdif-dit", static const struct snd_soc_dapm_route routes[] = { { "Playback", NULL, "spdifdo" }, }; When DPCM is eventually implemented, the DAPM route remains, the DAI link becomes two, as previously illustrated in other discussions. Full diffstat for this series: arch/arm/mach-dove/common.c | 4 +- arch/arm/mach-kirkwood/common.c | 24 ++--- sound/soc/codecs/spdif_transciever.c | 15 +++- sound/soc/kirkwood/Kconfig | 5 - sound/soc/kirkwood/Makefile | 4 +- sound/soc/kirkwood/kirkwood-dma.c | 114 ++++-------------- sound/soc/kirkwood/kirkwood-i2s.c | 213 +++++++++++++++++++++++++++++----- sound/soc/kirkwood/kirkwood-openrd.c | 12 ++- sound/soc/kirkwood/kirkwood-t5325.c | 8 +- sound/soc/kirkwood/kirkwood.h | 45 +++++++- sound/soc/soc-dapm.c | 2 +- 11 files changed, 295 insertions(+), 151 deletions(-) Diffstat for the patches remaining for mainline: sound/soc/kirkwood/kirkwood-dma.c | 4 +- sound/soc/kirkwood/kirkwood-i2s.c | 170 +++++++++++++++++++++++++++++++--- sound/soc/kirkwood/kirkwood-openrd.c | 8 ++ sound/soc/kirkwood/kirkwood-t5325.c | 4 + sound/soc/kirkwood/kirkwood.h | 34 +++++++ 5 files changed, 205 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH 0/3] ASoC: generic simple sound card DT bindings
by Markus Pargmann 04 Sep '13

04 Sep '13

Hi, The patches were already in the fsl-ssi ac97 series. I moved the seperated simple sound card DT driver into the existing simple card driver. Regards, Markus Markus Pargmann (3): ASoC: generic: simple card, use private data ASoC: generic simple sound card DT bindings ASoC: fsl: Kconfig, visible config items for simple card .../bindings/sound/generic,simple-dt-card.txt | 28 +++ include/sound/simple_card.h | 4 - sound/soc/fsl/Kconfig | 23 ++- sound/soc/generic/Kconfig | 1 + sound/soc/generic/simple-card.c | 225 ++++++++++++++++++--- 5 files changed, 246 insertions(+), 35 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/generic,simple-dt-card.txt

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[alsa-devel] [PATCH] ASoC: atmel: disable error interrupt
by Bo Shen 04 Sep '13

04 Sep '13

As once the error interrupt is triggered, it can not be cleared. So, disable it. No side effect found while testing on sama5d3xek and at91sam9x5ek boards. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> Acked-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 0ecf356..bb53dea 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -649,7 +649,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, dma_params = ssc_p->dma_params[dir]; ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); - ssc_writel(ssc_p->ssc->regs, IER, dma_params->mask->ssc_error); + ssc_writel(ssc_p->ssc->regs, IDR, dma_params->mask->ssc_error); pr_debug("%s enabled SSC_SR=0x%08x\n", dir ? "receive" : "transmit", -- 1.7.9.5

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Re: [alsa-devel] Documentation Request
by Carl Canuck 04 Sep '13

04 Sep '13

Sorry, I only sent that last reply to Grant, the reply-to-anybody-but-sender options are a little counter-intuitive in Gmail. On Tue, Sep 3, 2013 at 11:10 PM, Carl Canuck <carl.canuck.official(a)gmail.com > wrote: > Hi Grant, > > I have in fact spent a little time with Jack and even conjured up a > working prototype using Jack audio with ALSA MIDI (that will never see the > light of day). However, you're ignoring the fact that: > > 1. Some people don't like having 20 separate audio applications running > and having to save all of their states individually > 2. If you believe that any existing Jack session manager is a stable, > reliable application that adequately solves problem #1, then it's obvious > that you've not spent much time with Jack session managers > 3. Jack is something like 15 years old now, and they still haven't even > managed to solve #1 and #2 yet, among other long-standing issues? > 4. With points #1, #2 and #3 all being common knowledge, why should I put > my faith in Jack competently wrapping ALSA for me instead of taking matters > into my own hands and going straight to ALSA? > 5. If you visit Windows/Mac audio forums, there are legions of people who > will tell you they tried Linux once, found Jack to be buggy and terrible in > general, never intend to try Linux again. In fact, you can register a user > name on any non-Linux music forum, and ask "i'm thinking of trying Linux > for audio, thoughts?", and people will say exactly the things I described > to you, not because they have any unfounded bias against Linux or Jack, but > because that was truly their experience. > 6. Jack's configuration settings and error logging are atrocious, it is > far too often excessively difficult to understand why Jack won't start and > how to fix it. > 7. Is it really so unreasonable to ask for a basic example of a common > use-case for ALSA without being told that I should just abandon the quest > for that knowledge and put my faith in the Jack developers? Should > deciphering the intended use of any API be an exercise left to the would-be > developer? > > If you like Jack, then by all means use it, but it's not ideal for my > use-case, and I won't be spending any more time evaluating it until the > Jack developers address these long-standing issues. > > Carl > > <http://users.suse.com/~mana/alsa090_howto.html> Also, I have in fact seen the link you provided Grant, but that once again is only bits and pieces of the use-case. I'd like to know that I'm using ALSA as it's creators intended, rather than stringing together bits and pieces and hoping that I'm doing it right. Regards, Carl

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[alsa-devel] [PATCH] ALSA: hda - Re-setup HDMI pin and audio infoframe on stream switches
by Takashi Iwai 04 Sep '13

04 Sep '13

When the transcoder:port mapping on Haswell HDMI/DP audio is changed during the stream playback, the sound gets lost. Typically this problem is seen when the user switches the graphics mode from eDP+DP to DP-only configuration, where CRTC 1 is used for DP in the former while CRTC 0 is used for the latter. The graphics controller notifies the change via the normal ELD update procedure, so we get the intrinsic event. For enabling the sound again, the HDMI audio driver needs to reset the pin and set up the audio infoframe again. This patch achieves it by: - keep the current status of channels and info frame setup in per_pin struct, - check the reconnection in the intrinsic event handler, - reset the pin and the re-invoke hdmi_setup_audio_infoframe() accordingly. The hdmi_setup_audio_infoframe() function has been changed, too, so that it can be invoked without passing the substream instance. The patch is mostly based on the work by Mengdong Lin. Cc: Mengdong Lin <mengdong.lin(a)intel.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 41 +++++++++++++++++++++++++++++++---------- 1 file changed, 31 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b83b14f..22b5089 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -67,6 +67,8 @@ struct hdmi_spec_per_pin { struct delayed_work work; struct snd_kcontrol *eld_ctl; int repoll_count; + bool setup; /* the stream has been set up by prepare callback */ + int channels; /* current number of channels */ bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ @@ -879,18 +881,19 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, return true; } -static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, - bool non_pcm, - struct snd_pcm_substream *substream) +static void hdmi_setup_audio_infoframe(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + bool non_pcm) { - struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); hda_nid_t pin_nid = per_pin->pin_nid; - int channels = substream->runtime->channels; + int channels = per_pin->channels; struct hdmi_eld *eld; int ca; union audio_infoframe ai; + if (!channels) + return; + eld = &per_pin->sink_eld; if (!eld->monitor_present) return; @@ -1341,6 +1344,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) eld_changed = true; } if (update_eld) { + bool old_eld_valid = pin_eld->eld_valid; pin_eld->eld_valid = eld->eld_valid; eld_changed = pin_eld->eld_size != eld->eld_size || memcmp(pin_eld->eld_buffer, eld->eld_buffer, @@ -1350,6 +1354,18 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) eld->eld_size); pin_eld->eld_size = eld->eld_size; pin_eld->info = eld->info; + + /* Haswell-specific workaround: re-setup when the transcoder is + * changed during the stream playback + */ + if (codec->vendor_id == 0x80862807 && + eld->eld_valid && !old_eld_valid && per_pin->setup) { + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + hdmi_setup_audio_infoframe(codec, per_pin, + per_pin->non_pcm); + } } mutex_unlock(&pin_eld->lock); @@ -1522,14 +1538,17 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hda_nid_t cvt_nid = hinfo->nid; struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); - hda_nid_t pin_nid = get_pin(spec, pin_idx)->pin_nid; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + hda_nid_t pin_nid = per_pin->pin_nid; bool non_pcm; non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); + per_pin->channels = substream->runtime->channels; + per_pin->setup = true; hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, pin_idx, non_pcm, substream); + hdmi_setup_audio_infoframe(codec, per_pin, non_pcm); return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } @@ -1569,6 +1588,9 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, snd_hda_spdif_ctls_unassign(codec, pin_idx); per_pin->chmap_set = false; memset(per_pin->chmap, 0, sizeof(per_pin->chmap)); + + per_pin->setup = false; + per_pin->channels = 0; } return 0; @@ -1704,8 +1726,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, per_pin->chmap_set = true; memcpy(per_pin->chmap, chmap, sizeof(chmap)); if (prepared) - hdmi_setup_audio_infoframe(codec, pin_idx, per_pin->non_pcm, - substream); + hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); return 0; } -- 1.8.3.4

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[alsa-devel] [PATCH v2] ASoC: fsl: imx-audmux: Do not call imx_audmux_parse_dt_defaults() on non-dt kernel
by Fabio Estevam 04 Sep '13

04 Sep '13

Booting a mx51babbage board with a non-dt kernel leads to the following crash: Unable to handle kernel NULL pointer dereference at virtual address 0000001c pgd = 80004000 [0000001c] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.11.0-next-20130903 #287 task: 9f860000 ti: 9f862000 task.ti: 9f862000 PC is at of_get_next_available_child+0x5c/0x68 LR is at of_get_next_available_child+0x1c/0x68 pc : [<8043ea58>] lr : [<8043ea18>] psr: 60000193 sp : 9f863d58 ip : 00000000 fp : 9f863d74 r10: 9f89a010 r9 : 9f862000 r8 : 807bb26c r7 : 80615d5c r6 : 00000000 r5 : 60000113 r4 : 00000000 r3 : 00000000 r2 : 808770a4 r1 : 00000011 r0 : 60000113 Flags: nZCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5387d Table: 90004019 DAC: 00000017 Process swapper/0 (pid: 1, stack limit = 0x9f862240) Stack: (0x9f863d58 to 0x9f864000) This is caused by commit 8548a464b9 (ASoC: imx-audmux: Read default configuration from devicetree). In order to fix this, add a check for 'of_id' so that imx_audmux_parse_dt_defaults() only gets called when a dt kernel is running. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Only call imx_audmux_parse_dt_defaults for dt kernel. sound/soc/fsl/imx-audmux.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index ab17381..d3bf71a 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -335,7 +335,8 @@ static int imx_audmux_probe(struct platform_device *pdev) if (audmux_type == IMX31_AUDMUX) audmux_debugfs_init(); - imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); + if (of_id) + imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); return 0; } -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: fsl: imx-audmux: Check if we are running a dt kernel
by Fabio Estevam 03 Sep '13

03 Sep '13

Booting a mx51babbage board with a non-dt kernel leads to the following crash: Unable to handle kernel NULL pointer dereference at virtual address 0000001c pgd = 80004000 [0000001c] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.11.0-next-20130903 #287 task: 9f860000 ti: 9f862000 task.ti: 9f862000 PC is at of_get_next_available_child+0x5c/0x68 LR is at of_get_next_available_child+0x1c/0x68 pc : [<8043ea58>] lr : [<8043ea18>] psr: 60000193 sp : 9f863d58 ip : 00000000 fp : 9f863d74 r10: 9f89a010 r9 : 9f862000 r8 : 807bb26c r7 : 80615d5c r6 : 00000000 r5 : 60000113 r4 : 00000000 r3 : 00000000 r2 : 808770a4 r1 : 00000011 r0 : 60000113 Flags: nZCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5387d Table: 90004019 DAC: 00000017 Process swapper/0 (pid: 1, stack limit = 0x9f862240) Stack: (0x9f863d58 to 0x9f864000) In order to fix it, return an error when 'of_id' is NULL, which happens for the non-dt case. Also checked that audio is still functional when running the dt kernel for this board after applying this patch. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index ab17381..2b29b48 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -317,6 +317,9 @@ static int imx_audmux_probe(struct platform_device *pdev) const struct of_device_id *of_id = of_match_device(imx_audmux_dt_ids, &pdev->dev); + if (!of_id) + return -EINVAL; + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); audmux_base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(audmux_base)) -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: mxs: mxs-sgtl5000: Simplify probe function
by Fabio Estevam 03 Sep '13

03 Sep '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> mxs is a device tree only platform, which allows us to simplify a bit mxs_sgtl5000_probe(), because there is no need to check whether device tree is supported or not. Remove mxs_sgtl5000_probe_dt() and place its content inside mxs_sgtl5000_probe() for making the code simpler. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-sgtl5000.c | 20 +++----------------- 1 file changed, 3 insertions(+), 17 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 4bb2737..61822cc 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -122,14 +122,12 @@ static struct snd_soc_card mxs_sgtl5000 = { .num_links = ARRAY_SIZE(mxs_sgtl5000_dai), }; -static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) +static int mxs_sgtl5000_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &mxs_sgtl5000; + int ret, i; struct device_node *np = pdev->dev.of_node; struct device_node *saif_np[2], *codec_np; - int i; - - if (!np) - return 1; /* no device tree */ saif_np[0] = of_parse_phandle(np, "saif-controllers", 0); saif_np[1] = of_parse_phandle(np, "saif-controllers", 1); @@ -152,18 +150,6 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) of_node_put(saif_np[0]); of_node_put(saif_np[1]); - return 0; -} - -static int mxs_sgtl5000_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &mxs_sgtl5000; - int ret; - - ret = mxs_sgtl5000_probe_dt(pdev); - if (ret < 0) - return ret; - /* * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w). * The Sgtl5000 sysclk is derived from saif0 mclk and it's range -- 1.8.1.2

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[alsa-devel] ASoC updates for v3.12
by Mark Brown 02 Sep '13

02 Sep '13

The following changes since commit 3fd8a8041b3775840c8f85b6338b61beb5057160: Merge tag 'asoc-v3.12-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next (2013-08-30 12:35:49 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.12-3 for you to fetch changes up to 5df498a27ab385294fdb7cad19c623d4da94a780: Merge remote-tracking branch 'asoc/topic/ssm2602' into tmp (2013-09-01 21:15:59 +0100) These haven't all been in a -next build yet, they work for me though. ---------------------------------------------------------------- ASoC: Final updates for v3.12 A few final updates for v3.12 - some cleanups, a bug fix for ssm2602, pop removal for rt5640 and fixes for the reporting of unidirectional links in the MXS SGTL5000 driver. ---------------------------------------------------------------- Bard Liao (1): ASoC: rt5640: change widget sequence for depop Fabio Estevam (2): ASoC: soc-pcm: Allow to specify unidirectional dai_link ASoC: mxs-sgtl5000: Configure the dai_links as unidirectional Lars-Peter Clausen (5): ASoC: dapm: Fix auto-disable for inverted controls ASoC: Remove unused control_type field from snd_soc_codec struct ASoC: Remove unused debugfs_dapm field from snd_soc_{platform,codec} struct ASoC: Remove unused sysfs_registered field from snd_soc_codec struct ASoC: ssm2602: Fix cache sync Mark Brown (6): Merge remote-tracking branch 'asoc/topic/rt5640' into asoc-dapm ASoC: fsl_spdif: Staticse non-exported symbols Merge remote-tracking branch 'asoc/topic/core' into tmp Merge remote-tracking branch 'asoc/topic/dapm' into tmp Merge remote-tracking branch 'asoc/topic/fsl' into tmp Merge remote-tracking branch 'asoc/topic/ssm2602' into tmp Nicolin Chen (2): ASoC: fsl: Add wrapping for dev_dbg() in fsl_spdif.c ASoC: fsl: Add one blank space after ':=' in Makefile include/sound/soc.h | 8 +- sound/soc/codecs/rt5640.c | 217 ++++++++++++++++++++++++++++++++----------- sound/soc/codecs/rt5640.h | 12 +++ sound/soc/codecs/ssm2602.c | 3 +- sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_spdif.c | 12 +-- sound/soc/mxs/mxs-sgtl5000.c | 2 + sound/soc/soc-dapm.c | 2 + sound/soc/soc-pcm.c | 10 ++ 9 files changed, 202 insertions(+), 66 deletions(-)

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[alsa-devel] [PATCH 1/5] ASoC: Remove support for reg_access_defaults
by Lars-Peter Clausen 02 Sep '13

02 Sep '13

No users of reg_access_defaults are left and new drivers are going to use regmap for this, so support for it can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 24 -------------------- sound/soc/soc-cache.c | 63 --------------------------------------------------- sound/soc/soc-core.c | 9 -------- 3 files changed, 96 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7184805..ae6efc6d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -403,12 +403,6 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value); -int snd_soc_default_volatile_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_readable_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_writable_register(struct snd_soc_codec *codec, - unsigned int reg); int snd_soc_platform_read(struct snd_soc_platform *platform, unsigned int reg); int snd_soc_platform_write(struct snd_soc_platform *platform, @@ -542,22 +536,6 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /** - * struct snd_soc_reg_access - Describes whether a given register is - * readable, writable or volatile. - * - * @reg: the register number - * @read: whether this register is readable - * @write: whether this register is writable - * @vol: whether this register is volatile - */ -struct snd_soc_reg_access { - u16 reg; - u16 read; - u16 write; - u16 vol; -}; - -/** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * * @pin: name of the pin to update @@ -760,8 +738,6 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; - short reg_access_size; - const struct snd_soc_reg_access *reg_access_default; enum snd_soc_compress_type compress_type; /* codec bias level */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index e72f554..eaa898f 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -275,66 +275,3 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return ret; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync); - -static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec, - unsigned int reg) -{ - const struct snd_soc_codec_driver *codec_drv; - unsigned int min, max, index; - - codec_drv = codec->driver; - min = 0; - max = codec_drv->reg_access_size - 1; - do { - index = (min + max) / 2; - if (codec_drv->reg_access_default[index].reg == reg) - return index; - if (codec_drv->reg_access_default[index].reg < reg) - min = index + 1; - else - max = index; - } while (min <= max); - return -1; -} - -int snd_soc_default_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].vol; -} -EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register); - -int snd_soc_default_readable_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].read; -} -EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); - -int snd_soc_default_writable_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].write; -} -EXPORT_SYMBOL_GPL(snd_soc_default_writable_register); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7d13c4f..b78fa93 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4222,15 +4222,6 @@ int snd_soc_register_codec(struct device *dev, } } - if (codec_drv->reg_access_size && codec_drv->reg_access_default) { - if (!codec->volatile_register) - codec->volatile_register = snd_soc_default_volatile_register; - if (!codec->readable_register) - codec->readable_register = snd_soc_default_readable_register; - if (!codec->writable_register) - codec->writable_register = snd_soc_default_writable_register; - } - for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); fixup_codec_formats(&dai_drv[i].capture); -- 1.8.0

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