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

[alsa-devel] [PATCH 1/3] ASoC: wm_adsp: free memory when unloaded or closed
by Richard Fitzgerald 27 Apr '16

27 Apr '16

The patch adds a wm_adsp2_remove() function to ensure that memory is freed when the driver is unloaded or shut down. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 20 ++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 1 + 2 files changed, 21 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b6470de..c350a52 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1039,6 +1039,13 @@ static void wm_adsp_ctl_work(struct work_struct *work) kfree(ctl_work); } +static void wm_adsp_free_ctl_blk(struct wm_coeff_ctl *ctl) +{ + kfree(ctl->cache); + kfree(ctl->name); + kfree(ctl); +} + static int wm_adsp_create_control(struct wm_adsp *dsp, const struct wm_adsp_alg_region *alg_region, unsigned int offset, unsigned int len, @@ -2434,6 +2441,19 @@ int wm_adsp2_init(struct wm_adsp *dsp) } EXPORT_SYMBOL_GPL(wm_adsp2_init); +void wm_adsp2_remove(struct wm_adsp *dsp) +{ + struct wm_coeff_ctl *ctl; + + while (!list_empty(&dsp->ctl_list)) { + ctl = list_first_entry(&dsp->ctl_list, struct wm_coeff_ctl, + list); + list_del(&ctl->list); + wm_adsp_free_ctl_blk(ctl); + } +} +EXPORT_SYMBOL_GPL(wm_adsp2_remove); + int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) { struct wm_adsp_compr *compr; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index b61cb57..feb61e2 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -92,6 +92,7 @@ extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); +void wm_adsp2_remove(struct wm_adsp *dsp); int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp1_event(struct snd_soc_dapm_widget *w, -- 1.9.1

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[alsa-devel] [PATCH v4 0/2] ASoC: codecs: add support for TAS5720 digital amplifier
by Andreas Dannenberg 27 Apr '16

27 Apr '16

Minor update to driver for TI's TAS5720L/M digital audio amplifiers incorporating previous feedback (thanks Mark Brown and Andrew F Davis), see below change log for details. Also re-tested different aspects of the driver to make sure it still builds/works. -- Andreas Dannenberg Texas Instruments Inc Changes since v3: - Rework fault condition reporting to use dev_crit() to pay justice to the severity of the issues, also shorten the error string for more clarity/brevity - Fix Kconfig help entry to no longer mention the interrupt signal that was used previously - Simplify hw_params checks as the driver in its current form only supports left-justified audio formats and the codec is tolerant regarding the number of bits that are actually received. - Clarify DT bindings to make the power supply properties required - Remove .owner assignment from I2C driver struct as it is no longer needed (taken care off by the Kernel) Changes since v2: - Switched fault handling from using interrupts to polling mode - Remove interrupt related description from DT bindings doc - Remove unlikely() to simplify an expression - Remove unnesseary tas5720_set_dai_sysclk() function stub Changes since v1: - Simplified DT interrupt documentation (Thanks Rob Herring) - Fixed potential race condition during codec probe where deferred work that's used by the threaded IRQ handler was setup after the IRQ was initialized (would lead to an issue when the TAS5720 was already throwing interrupts at the time of probe) Andreas Dannenberg (2): ASoC: codecs: add TA5720 digital amplifier DT bindings ASoC: codecs: add support for TAS5720 digital amplifier .../devicetree/bindings/sound/tas5720.txt | 25 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5720.c | 620 +++++++++++++++++++++ sound/soc/codecs/tas5720.h | 90 +++ 5 files changed, 745 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas5720.txt create mode 100644 sound/soc/codecs/tas5720.c create mode 100644 sound/soc/codecs/tas5720.h -- 2.6.4

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[alsa-devel] [PATCH 1/2] ASoC: topology: Check failure to create a widget
by mengdong.lin＠linux.intel.com 27 Apr '16

27 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Stop loading topology info if error happens when creating a widget. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index bde79f7..e38664e 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1500,9 +1500,11 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg, for (i = 0; i < count; i++) { widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos; ret = soc_tplg_dapm_widget_create(tplg, widget); - if (ret < 0) + if (ret < 0) { dev_err(tplg->dev, "ASoC: failed to load widget %s\n", widget->name); + return ret; + } } return 0; -- 2.5.0

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[alsa-devel] [PATCH 2/2] ASoC: topology: Check size mismatch of ABI objects before parsing
by mengdong.lin＠linux.intel.com 27 Apr '16

27 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> If size mismatch of manifest, ABI headers or elements is found, stop parsing topology info and return the error. New fields may be append to the tail of ABI objects which will cause object size to increase. If user space and kernel use different versions of ABI, size mismatch will be detected here. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index e38664e..2261585 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1023,6 +1023,11 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; + if (control_hdr->size != sizeof(*control_hdr)) { + dev_err(tplg->dev, "ASoC: invalid control size\n"); + return -EINVAL; + } + switch (control_hdr->ops.info) { case SND_SOC_TPLG_CTL_VOLSW: case SND_SOC_TPLG_CTL_STROBE: @@ -1499,6 +1504,11 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg, for (i = 0; i < count; i++) { widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos; + if (widget->size != sizeof(*widget)) { + dev_err(tplg->dev, "ASoC: invalid widget size\n"); + return -EINVAL; + } + ret = soc_tplg_dapm_widget_create(tplg, widget); if (ret < 0) { dev_err(tplg->dev, "ASoC: failed to load widget %s\n", @@ -1651,8 +1661,6 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, if (tplg->pass != SOC_TPLG_PASS_PCM_DAI) return 0; - pcm = (struct snd_soc_tplg_pcm *)tplg->pos; - if (soc_tplg_check_elem_count(tplg, sizeof(struct snd_soc_tplg_pcm), count, hdr->payload_size, "PCM DAI")) { @@ -1662,7 +1670,13 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, } /* create the FE DAIs and DAI links */ + pcm = (struct snd_soc_tplg_pcm *)tplg->pos; for (i = 0; i < count; i++) { + if (pcm->size != sizeof(*pcm)) { + dev_err(tplg->dev, "ASoC: invalid pcm size\n"); + return -EINVAL; + } + soc_tplg_pcm_create(tplg, pcm); pcm++; } @@ -1682,6 +1696,11 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, return 0; manifest = (struct snd_soc_tplg_manifest *)tplg->pos; + if (manifest->size != sizeof(*manifest)) { + dev_err(tplg->dev, "ASoC: invalid manifest size\n"); + return -EINVAL; + } + tplg->pos += sizeof(struct snd_soc_tplg_manifest); if (tplg->comp && tplg->ops && tplg->ops->manifest) @@ -1698,6 +1717,14 @@ static int soc_valid_header(struct soc_tplg *tplg, if (soc_tplg_get_hdr_offset(tplg) >= tplg->fw->size) return 0; + if (hdr->size != sizeof(*hdr)) { + dev_err(tplg->dev, + "ASoC: invalid header size for type %d at offset 0x%lx size 0x%zx.\n", + hdr->type, soc_tplg_get_hdr_offset(tplg), + tplg->fw->size); + return -EINVAL; + } + /* big endian firmware objects not supported atm */ if (hdr->magic == cpu_to_be32(SND_SOC_TPLG_MAGIC)) { dev_err(tplg->dev, -- 2.5.0

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[alsa-devel] [GIT PULL] ASoC fixes for v4.6
by Mark Brown 27 Apr '16

27 Apr '16

The following changes since commit 02da2d72174c61988eb4456b53f405e3ebdebce4: Linux 4.6-rc5 (2016-04-24 16:17:05 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.6-rc5 for you to fetch changes up to 8c0f55100478649e3c0133684aa4aebab77ce10f: Merge remote-tracking branches 'asoc/fix/rt5640' and 'asoc/fix/wm8962' into asoc-linus (2016-04-26 19:25:18 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.6 This is a fairly large collection of fixes but almost all driver specific ones, especially to the new Intel drivers which have had a lot of recent development. The one core fix is a change to the debugfs code to avoid crashes in some relatively unusual configurations. ---------------------------------------------------------------- Arnd Bergmann (2): ASoC: cs35l32: avoid uninitialized variable access ASoC: rt5616: add I2C dependency Ben Zhang (1): ASoC: nau8825: Fix jack detection across suspend Charles Keepax (3): ASoC: wm8962: Correct typo when setting DSPCLK rate ASoC: arizona: Free speaker thermal IRQs in CODEC remove ASoC: wm5102: Free compressed IRQ in CODEC remove Dan Carpenter (1): ASoC: Intel: sst: fix a loop timeout in sst_hsw_stream_reset() Jeeja KP (2): ASoC: Intel: Skylake: Fix for unloading module only when it is loaded ASoC: Intel: Skylake: Fix DSP resource de-allocation Mark Brown (5): ASoC: dapm: Make sure we have a card when displaying component widgets Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/cs35l32', 'asoc/fix/hdac', 'asoc/fix/nau8825' and 'asoc/fix/rt5616' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5640' and 'asoc/fix/wm8962' into asoc-linus Subhransu S. Prusty (4): ASoC: hdac_hdmi: Fix to use dev_pm ops instead soc pm ASoC: hdac_hdmi: Fix codec power state in S3 during playback ASoC: Intel: Skylake: Fix to turn OFF codec power when entering S3 ASoC: Intel: Skylake: Fix ibs/obs calc for non-integral sampling rates Sugar Zhang (1): ASoC: rt5640: Correct the digital interface data select Vinod Koul (7): ALSA: hda: use list macro for parsing on cleanup ASoC: Intel: Skylake: free codec objects on removal ASoC: Intel: Skylake: Freeup properly on skl_dsp_free ASoC: Intel: Skylake: Unmap the address last ASoC: Intel: Skylake: Call i915 exit last ASoC: Intel: Skylake: remove call to pci_dev_put ASoC: Intel: Skylake: Fix kbuild dependency sound/hda/ext/hdac_ext_stream.c | 5 +- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/arizona.c | 12 +++ sound/soc/codecs/arizona.h | 2 + sound/soc/codecs/cs35l32.c | 17 +++- sound/soc/codecs/cs47l24.c | 3 + sound/soc/codecs/hdac_hdmi.c | 94 ++++++++++------------ sound/soc/codecs/nau8825.c | 126 +++++++++++++++++------------- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5640.h | 36 ++++----- sound/soc/codecs/wm5102.c | 5 ++ sound/soc/codecs/wm5110.c | 2 + sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8997.c | 2 + sound/soc/codecs/wm8998.c | 2 + sound/soc/intel/Kconfig | 1 - sound/soc/intel/haswell/sst-haswell-ipc.c | 2 +- sound/soc/intel/skylake/skl-sst-dsp.c | 5 ++ sound/soc/intel/skylake/skl-topology.c | 42 ++++++---- sound/soc/intel/skylake/skl-topology.h | 8 +- sound/soc/intel/skylake/skl.c | 32 +++++--- sound/soc/soc-dapm.c | 7 ++ 22 files changed, 245 insertions(+), 163 deletions(-)

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[alsa-devel] [PATCH v7 0/9] ASoC: MT8173 HDMI codec support
by Philipp Zabel 27 Apr '16

27 Apr '16

Since the hdmi-codec patches are now applied [1], I have merged the hdmi-codec interface into the initial MT8173 HDMI [2] driver submission. This series now only enables the HDMI dai-links in the mt8173-rt5650-rt5676 and mt8173-rt5650 machine drivers and adds hdmi-codec ELD support and the still up in the air hdmi notifications between DRM encoder driver and hdmi-codec. Changes since v6: - Dropped "drm/mediatek: hdmi: Add audio interface to the hdmi-codec driver", patch now merged into MT8173 HDMI encoder patch. - Fixed jack detection in case HDMI is already connected at boot and the hdmi-codec driver registers the notifier block after the HDMI_CONNECTED notification is issued. [1] commit 09184118a8ab ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/hdmi The mediatek drm patches apply on top of [2] https://patchwork.kernel.org/patch/8887051/ ("drm/mediatek: Add HDMI support") and need this patch: [3] https://patchwork.kernel.org/patch/8091531/ ("video: hdmi: add helper function for N and CTS") regards Philipp Koro Chen (1): ASoC: mediatek: Add HDMI dai-links in the mt8173-rt5650-rt5676 machine driver PC Liao (1): ASoC: mediatek: Add HDMI dai-links to the mt8173-rt5650 machine driver Philipp Zabel (7): ASoC: hdmi-codec: Add ELD control video: rmk's HDMI notification prototype ASoC: hdmi-codec: Use HDMI notifications to add jack support ASoC: mediatek: Add jack detection support to mt8173-rt5650-rt5676 machine driver ASoC: mediatek: Add jack detection support to the mt8173-rt5650 machine driver drm/mediatek: hdmi: issue notifications drm/mediatek: hdmi: use helper function for N and CTS calculation .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 +- .../devicetree/bindings/sound/mt8173-rt5650.txt | 5 +- drivers/gpu/drm/mediatek/mtk_cec.c | 11 ++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 88 ++++----------- drivers/video/Kconfig | 3 + drivers/video/Makefile | 1 + drivers/video/hdmi-notifier.c | 61 ++++++++++ include/linux/hdmi-notifier.h | 44 ++++++++ include/sound/hdmi-codec.h | 6 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdmi-codec.c | 124 +++++++++++++++++++-- sound/soc/mediatek/Kconfig | 2 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 48 ++++++++ sound/soc/mediatek/mt8173-rt5650.c | 47 ++++++++ 14 files changed, 370 insertions(+), 76 deletions(-) create mode 100644 drivers/video/hdmi-notifier.c create mode 100644 include/linux/hdmi-notifier.h -- 2.8.0.rc3

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[alsa-devel] [PATCH RESEND] ASoC: arizona: Do not create OUT4R widget for CS47L24/WM1831
by Richard Fitzgerald 27 Apr '16

27 Apr '16

The CS47L24 and WM1831 codecs only use the OUT4L widget so we can skip creation of the OUT4R widget. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index da61c62..0cd27e0 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -223,6 +223,8 @@ int arizona_init_spk(struct snd_soc_codec *codec) switch (arizona->type) { case WM8997: + case CS47L24: + case WM1831: break; default: ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1); -- 1.9.1

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[alsa-devel] [PATCH 1/3] ASoC: bcm2835: add 24bit support
by kernel＠martin.sperl.org 26 Apr '16

26 Apr '16

From: Matthias Reichl <hias(a)horus.com> This adds 24 bit support to the I2S driver of the BCM2835 Code ported from bcm2708-i2s driver in Raspberry Pi tree. Signed-off-by: Florian Meier <florian.meier(a)koalo.de> Signed-off-by: Matthias Reichl <hias(a)horus.com> Signed-off-by: Martin Sperl <kernel(a)martin.sperl.org> --- sound/soc/bcm/bcm2835-i2s.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 1c1f221..d2663e7 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -259,6 +259,9 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: data_length = 16; break; + case SNDRV_PCM_FORMAT_S24_LE: + data_length = 24; + break; case SNDRV_PCM_FORMAT_S32_LE: data_length = 32; break; @@ -279,7 +282,7 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, /* Setup the frame format */ format = BCM2835_I2S_CHEN; - if (data_length > 24) + if (data_length >= 24) format |= BCM2835_I2S_CHWEX; format |= BCM2835_I2S_CHWID((data_length-8)&0xf); @@ -570,6 +573,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, .capture = { @@ -577,6 +581,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, .ops = &bcm2835_i2s_dai_ops, -- 2.1.4

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[alsa-devel] [PATCH] ASoC: omap-pcm: Initialize DMA configuration
by J.D. Schroeder 26 Apr '16

26 Apr '16

From: Jim Lodes <jim.lodes(a)garmin.com> Initialize the dma_slave_config for PCM DMA transfers, instead of leaving it uninitialized. Keeps previous data on the stack from giving us invalid values in uninitialized members of the config structure. Signed-off-by: Jim Lodes <jim.lodes(a)garmin.com> Signed-off-by: J.D. Schroeder <jay.schroeder(a)garmin.com> --- sound/soc/omap/omap-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 99381a2..a84f677 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -82,6 +82,8 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct dma_chan *chan; int err = 0; + memset(&config, 0x00, sizeof(config)); + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); /* return if this is a bufferless transfer e.g. -- 1.9.1

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[alsa-devel] [PATCH] ASoC: cs42l56: Use IS_ENABLED() instead of checking for built-in or module
by Javier Martinez Canillas 26 Apr '16

26 Apr '16

The IS_ENABLED() macro checks if a Kconfig symbol has been enabled either built-in or as a module, use that macro instead of open coding the same. Signed-off-by: Javier Martinez Canillas <javier(a)osg.samsung.com> --- sound/soc/codecs/cs42l56.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 7cd5f769bb61..eec1ff853b98 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -56,7 +56,7 @@ struct cs42l56_private { u8 iface; u8 iface_fmt; u8 iface_inv; -#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +#if IS_ENABLED(CONFIG_INPUT) struct input_dev *beep; struct work_struct beep_work; int beep_rate; -- 2.5.5

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[alsa-devel] [PATCH] ALSA: intel_hda - Add dock support for ThinkPad X260
by Conrad Kostecki 26 Apr '16

26 Apr '16

Hi! My shiny new ThinkPad X260 is the same way affected, as older ThinkPad (X*40, X*50) generations. That means, I am unable to get sound output, when I am using the Lenovo CES 2013 docking station series (basic, pro, ultra). It can be fixed the same way, as it was already done for X240 and X250, as the X260 uses the same docking connector. I am attaching my patch, which works for me. Cheers Conrad ------------------------------ Fixes audio output on a ThinkPad X260, when using Lenovo CES 2013 docking station series (basic, pro, ultra). Signed-off-by: Conrad Kostecki <ck+linuxkernel(a)bl4ckb0x.de> diff -uprN -X linux-4.6-rc4-vanilla/Documentation/dontdiff linux-4.6-rc4-vanilla/sound/pci/hda/patch_realtek.c linux-4.6-rc4/sound/pci/hda/patch_realtek.c --- linux-4.6-rc4-vanilla/sound/pci/hda/patch_realtek.c 2016-04-24 03:26:36.330983586 +0200 +++ linux-4.6-rc4/sound/pci/hda/patch_realtek.c 2016-04-24 03:27:13.737981843 +0200 @@ -5570,6 +5570,7 @@ static const struct snd_pci_quirk alc269 SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x504a, "ThinkPad X260", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), ------------------------------ -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

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[alsa-devel] [PATCH 1/2] ASoC: rt298: reset AD dilter is there is no MCLK
by Bard Liao 26 Apr '16

26 Apr '16

rt298 need to reset AD filter and the ADC settings will take effort. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 20 ++++++++++++++++++++ sound/soc/codecs/rt298.h | 2 ++ 2 files changed, 22 insertions(+) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index f0e6c06..178b1cc 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -481,6 +481,26 @@ static int rt298_adc_event(struct snd_soc_dapm_widget *w, snd_soc_update_bits(codec, VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), 0x7080, 0x7000); + /* If MCLK doesn't exist, reset AD filter */ + if (!(snd_soc_read(codec, RT298_VAD_CTRL) & 0x200)) { + pr_info("NO MCLK\n"); + switch (nid) { + case RT298_ADC_IN1: + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x2, 0x2); + mdelay(10); + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x2, 0x0); + break; + case RT298_ADC_IN2: + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x4, 0x4); + mdelay(10); + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x4, 0x0); + break; + } + } break; case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(codec, diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h index d66f884..3638f3d 100644 --- a/sound/soc/codecs/rt298.h +++ b/sound/soc/codecs/rt298.h @@ -137,6 +137,7 @@ #define RT298_A_BIAS_CTRL2 0x02 #define RT298_POWER_CTRL1 0x03 #define RT298_A_BIAS_CTRL3 0x04 +#define RT298_D_FILTER_CTRL 0x05 #define RT298_POWER_CTRL2 0x08 #define RT298_I2S_CTRL1 0x09 #define RT298_I2S_CTRL2 0x0a @@ -148,6 +149,7 @@ #define RT298_IRQ_CTRL 0x33 #define RT298_WIND_FILTER_CTRL 0x46 #define RT298_PLL_CTRL1 0x49 +#define RT298_VAD_CTRL 0x4e #define RT298_CBJ_CTRL1 0x4f #define RT298_CBJ_CTRL2 0x50 #define RT298_PLL_CTRL 0x63 -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH] ASoC: mediatek: HDMI audio LR channel swapped
by PC Liao 26 Apr '16

26 Apr '16

Because LRCK of TDM use High to Low as default setting, this patch changes the TDM setting to inverse LRCK. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index f1c58a2..2b5df2e 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -123,6 +123,7 @@ #define AFE_TDM_CON1_WLEN_32BIT (0x2 << 8) #define AFE_TDM_CON1_MSB_ALIGNED (0x1 << 4) #define AFE_TDM_CON1_1_BCK_DELAY (0x1 << 3) +#define AFE_TDM_CON1_LRCK_INV (0x1 << 2) #define AFE_TDM_CON1_BCK_INV (0x1 << 1) #define AFE_TDM_CON1_EN (0x1 << 0) @@ -449,6 +450,7 @@ static int mtk_afe_hdmi_prepare(struct snd_pcm_substream *substream, runtime->rate * runtime->channels * 32); val = AFE_TDM_CON1_BCK_INV | + AFE_TDM_CON1_LRCK_INV | AFE_TDM_CON1_1_BCK_DELAY | AFE_TDM_CON1_MSB_ALIGNED | /* I2S mode */ AFE_TDM_CON1_WLEN_32BIT | -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: intel_hda - Add dock support for ThinkPad T460s
by Conrad Kostecki 26 Apr '16

26 Apr '16

Hi! The new ThinkPad T460s is the same way affected, as other current ThinkPads (e.G. X260) generations. That means, you won't be able to get sound output, when using the Lenovo CES 2013 docking station series (basic, pro, ultra). It can be fixed the same way, as it was already done for X260, as the T460s uses the same docking connector. I am attaching a patch, which works for me. Cheers Conrad ------------------------------ Fixes audio output on a ThinkPad T460s, when using Lenovo CES 2013 docking station series (basic, pro, ultra). Signed-off-by: Conrad Kostecki <ck+linuxkernel(a)bl4ckb0x.de> --- linux-4.6-rc5-vanilla/sound/pci/hda/patch_realtek.c 2016-04-26 11:32:09.124832372 +0200 +++ linux-4.6-rc5/sound/pci/hda/patch_realtek.c 2016-04-26 11:32:52.123834807 +0200 @@ -5585,6 +5585,7 @@ static const struct snd_pci_quirk alc269 SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x5050, "Thinkpad T460p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), ------------------------------ -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

1 1

[alsa-devel] [PATCH V2] ASoC: bcm2835: Register as custom PCM device to enable 16bit pcm
by kernel＠martin.sperl.org 26 Apr '16

26 Apr '16

From: Matthias Reichl <hias(a)horus.com> Register as a custom pcm device with the correct settings supporting also 16 bit. The automatic pcm discovery does not enable 16 bit modes because of a percived mismatch between the dma controller only supporting 32 bit transfers and the requirement by the HW to only support exactly 2 channels, which effectively results in a minimum size of 32 bits, which matches the dma controller transfer width. Code ported from bcm2708-i2s driver in Raspberry Pi tree. Signed-off-by: Florian Meier <florian.meier(a)koalo.de> Signed-off-by: Matthias Reichl <hias(a)horus.com> Signed-off-by: Martin Sperl <kernel(a)martin.sperl.org> Changelog: V1 -> V2: use of SZ_* defines instead of multiple of page size better description of commit message --- sound/soc/bcm/bcm2835-i2s.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index a0026e2..66680f1 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -632,6 +632,27 @@ static const struct snd_soc_component_driver bcm2835_i2s_component = { .name = "bcm2835-i2s-comp", }; +static const struct snd_pcm_hardware bcm2835_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_JOINT_DUPLEX | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .period_bytes_min = 32, + .period_bytes_max = SZ_256K, + .periods_min = 2, + .periods_max = 255, + .buffer_bytes_max = SZ_512K, +}; + +static const struct snd_dmaengine_pcm_config bcm2835_dmaengine_pcm_config = { + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .pcm_hardware = &bcm2835_pcm_hardware, + .prealloc_buffer_size = SZ_1M, +}; + static int bcm2835_i2s_probe(struct platform_device *pdev) { struct bcm2835_i2s_dev *dev; @@ -704,7 +725,9 @@ static int bcm2835_i2s_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = devm_snd_dmaengine_pcm_register( + &pdev->dev, &bcm2835_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_COMPAT); if (ret) { dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); return ret; -- 2.1.4

1 0

[alsa-devel] [PATCH v2] ALSA: hda - Update BCLK also at hotplug for i915 HSW/BDW
by Takashi Iwai 26 Apr '16

26 Apr '16

The recent bug report suggests that BCLK setup for i915 HSW/BDW needs to be updated at each HDMI hotplug, not only at initialization and resume. That is, we need to update HSW_EM4 and HSW_EM5 registers at ELD notification, too. Otherwise the HDMI audio may be out of sync and played in a wrong pitch. However, the HDA codec driver has no access to the controller registers, and currently the code managing these registers is in hda_intel.c, i.e. local to the controller driver. For allowing the explicit BCLK update from the codec driver, as in this patch, the former haswell_set_bclk() in hda_intel.c is moved to hdac_i915.c and exposed as snd_hdac_i915_set_bclk(). This is called from both the HDA controller driver and intel_pin_eld_notify() in HDMI codec driver. Along with this change, snd_hdac_get_display_clk() gets dropped as it's no longer used. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=91410 Cc: <stable(a)vger.kernel.org> # v4.5+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_i915.h | 5 ++-- sound/hda/hdac_i915.c | 62 ++++++++++++++++++++++++++++++++++++++-------- sound/pci/hda/hda_intel.c | 56 +++-------------------------------------- sound/pci/hda/patch_hdmi.c | 1 + 4 files changed, 58 insertions(+), 66 deletions(-) diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index fa341fcb5829..f5842bcd9c94 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -9,7 +9,7 @@ #ifdef CONFIG_SND_HDA_I915 int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); int snd_hdac_display_power(struct hdac_bus *bus, bool enable); -int snd_hdac_get_display_clk(struct hdac_bus *bus); +void snd_hdac_i915_set_bclk(struct hdac_bus *bus); int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate); int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid, bool *audio_enabled, char *buffer, int max_bytes); @@ -25,9 +25,8 @@ static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable) { return 0; } -static inline int snd_hdac_get_display_clk(struct hdac_bus *bus) +static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { - return 0; } static inline int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 54babe1c0b16..607bbeaebddf 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -20,6 +20,7 @@ #include <sound/core.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> +#include <sound/hda_register.h> static struct i915_audio_component *hdac_acomp; @@ -97,26 +98,65 @@ int snd_hdac_display_power(struct hdac_bus *bus, bool enable) } EXPORT_SYMBOL_GPL(snd_hdac_display_power); +#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ + ((pci)->device == 0x0c0c) || \ + ((pci)->device == 0x0d0c) || \ + ((pci)->device == 0x160c)) + /** - * snd_hdac_get_display_clk - Get CDCLK in kHz + * snd_hdac_i915_set_bclk - Reprogram BCLK for HSW/BDW * @bus: HDA core bus * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. + * Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK + * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) + * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: + * BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled and need to + * be restored to avoid abnormal playback speed. * - * This function queries CDCLK value in kHz from the graphics driver and - * returns the value. A negative code is returned in error. + * Call this function at initializing and changing power well, as well as + * at ELD notifier for the hotplug. */ -int snd_hdac_get_display_clk(struct hdac_bus *bus) +void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { struct i915_audio_component *acomp = bus->audio_component; + struct pci_dev *pci = to_pci_dev(bus->dev); + int cdclk_freq; + unsigned int bclk_m, bclk_n; + + if (!acomp || !acomp->ops || !acomp->ops->get_cdclk_freq) + return; /* only for i915 binding */ + if (!CONTROLLER_IN_GPU(pci)) + return; /* only HSW/BDW */ + + cdclk_freq = acomp->ops->get_cdclk_freq(acomp->dev); + switch (cdclk_freq) { + case 337500: + bclk_m = 16; + bclk_n = 225; + break; + + case 450000: + default: /* default CDCLK 450MHz */ + bclk_m = 4; + bclk_n = 75; + break; + + case 540000: + bclk_m = 4; + bclk_n = 90; + break; + + case 675000: + bclk_m = 8; + bclk_n = 225; + break; + } - if (!acomp || !acomp->ops) - return -ENODEV; - - return acomp->ops->get_cdclk_freq(acomp->dev); + snd_hdac_chip_writew(bus, HSW_EM4, bclk_m); + snd_hdac_chip_writew(bus, HSW_EM5, bclk_n); } -EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk); +EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk); /* There is a fixed mapping between audio pin node and display port * on current Intel platforms: diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 637b8a0e2a91..9a0d1445ca5c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -857,50 +857,6 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ -/* Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK - * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) - * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: - * BCLK = CDCLK * M / N - * The values will be lost when the display power well is disabled and need to - * be restored to avoid abnormal playback speed. - */ -static void haswell_set_bclk(struct hda_intel *hda) -{ - struct azx *chip = &hda->chip; - int cdclk_freq; - unsigned int bclk_m, bclk_n; - - if (!hda->need_i915_power) - return; - - cdclk_freq = snd_hdac_get_display_clk(azx_bus(chip)); - switch (cdclk_freq) { - case 337500: - bclk_m = 16; - bclk_n = 225; - break; - - case 450000: - default: /* default CDCLK 450MHz */ - bclk_m = 4; - bclk_n = 75; - break; - - case 540000: - bclk_m = 4; - bclk_n = 90; - break; - - case 675000: - bclk_m = 8; - bclk_n = 225; - break; - } - - azx_writew(chip, HSW_EM4, bclk_m); - azx_writew(chip, HSW_EM5, bclk_n); -} - #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -958,7 +914,7 @@ static int azx_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) { snd_hdac_display_power(azx_bus(chip), true); - haswell_set_bclk(hda); + snd_hdac_i915_set_bclk(azx_bus(chip)); } if (chip->msi) if (pci_enable_msi(pci) < 0) @@ -1058,7 +1014,7 @@ static int azx_runtime_resume(struct device *dev) bus = azx_bus(chip); if (hda->need_i915_power) { snd_hdac_display_power(bus, true); - haswell_set_bclk(hda); + snd_hdac_i915_set_bclk(bus); } else { /* toggle codec wakeup bit for STATESTS read */ snd_hdac_set_codec_wakeup(bus, true); @@ -1796,12 +1752,8 @@ static int azx_first_init(struct azx *chip) /* initialize chip */ azx_init_pci(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - struct hda_intel *hda; - - hda = container_of(chip, struct hda_intel, chip); - haswell_set_bclk(hda); - } + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + snd_hdac_i915_set_bclk(bus); hda_intel_init_chip(chip, (probe_only[dev] & 2) == 0); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 40933aa33afe..1483f85999ec 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2232,6 +2232,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) if (atomic_read(&(codec)->core.in_pm)) return; + snd_hdac_i915_set_bclk(&codec->bus->core); check_presence_and_report(codec, pin_nid); } -- 2.8.1

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[alsa-devel] [PATCH] ALSA: hda - Update BCLK also at hotplug for i915 HSW/BDW
by Takashi Iwai 26 Apr '16

26 Apr '16

The recent bug report suggests that BCLK setup for i915 HSW/BDW needs to be updated at each HDMI hotplug, not only at initialization and resume. That is, we need to update HSW_EM4 and HSW_EM5 registers at ELD notification, too. Otherwise the HDMI audio may be out of sync and played in a wrong pitch. However, the HDA codec driver has no access to the controller registers, and currently the code managing these registers is in hda_intel.c, i.e. local to the controller driver. For allowing the explicit BCLK update from the codec driver, as in this patch, the former haswell_set_bclk() in hda_intel.c is moved to hdac_i915.c and exposed as snd_hdac_i915_set_bclk(). This is called from both the HDA controller driver and intel_pin_eld_notify() in HDMI codec driver. Along with this change, snd_hdac_get_display_clk() gets dropped as it's no longer used. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=91410 Cc: <stable(a)vger.kernel.org> # v4.5+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_i915.h | 5 ++-- sound/hda/hdac_i915.c | 62 +++++++++++++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_intel.c | 56 +++-------------------------------------- sound/pci/hda/patch_hdmi.c | 1 + 4 files changed, 68 insertions(+), 56 deletions(-) diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index fa341fcb5829..f5842bcd9c94 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -9,7 +9,7 @@ #ifdef CONFIG_SND_HDA_I915 int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); int snd_hdac_display_power(struct hdac_bus *bus, bool enable); -int snd_hdac_get_display_clk(struct hdac_bus *bus); +void snd_hdac_i915_set_bclk(struct hdac_bus *bus); int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate); int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid, bool *audio_enabled, char *buffer, int max_bytes); @@ -25,9 +25,8 @@ static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable) { return 0; } -static inline int snd_hdac_get_display_clk(struct hdac_bus *bus) +static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { - return 0; } static inline int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 54babe1c0b16..181777b43aa1 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -20,6 +20,7 @@ #include <sound/core.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> +#include <sound/hda_register.h> static struct i915_audio_component *hdac_acomp; @@ -116,7 +117,66 @@ int snd_hdac_get_display_clk(struct hdac_bus *bus) return acomp->ops->get_cdclk_freq(acomp->dev); } -EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk); + +#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ + ((pci)->device == 0x0c0c) || \ + ((pci)->device == 0x0d0c) || \ + ((pci)->device == 0x160c)) + +/** + * snd_hdac_i915_set_bclk - Reprogram BCLK for HSW/BDW + * @bus: HDA core bus + * + * Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK + * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) + * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: + * BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled and need to + * be restored to avoid abnormal playback speed. + * + * Call this function at initializing and changing power well, as well as + * at ELD notifier for the hotplug. + */ +void snd_hdac_i915_set_bclk(struct hdac_bus *bus) +{ + struct i915_audio_component *acomp = bus->audio_component; + struct pci_dev *pci = to_pci_dev(bus->dev); + int cdclk_freq; + unsigned int bclk_m, bclk_n; + + if (!acomp || !acomp->ops || !acomp->ops->get_cdclk_freq) + return; /* only for i915 binding */ + if (!CONTROLLER_IN_GPU(pci)) + return; /* only HSW/BDW */ + + cdclk_freq = acomp->ops->get_cdclk_freq(acomp->dev); + switch (cdclk_freq) { + case 337500: + bclk_m = 16; + bclk_n = 225; + break; + + case 450000: + default: /* default CDCLK 450MHz */ + bclk_m = 4; + bclk_n = 75; + break; + + case 540000: + bclk_m = 4; + bclk_n = 90; + break; + + case 675000: + bclk_m = 8; + bclk_n = 225; + break; + } + + snd_hdac_chip_writew(bus, HSW_EM4, bclk_m); + snd_hdac_chip_writew(bus, HSW_EM5, bclk_n); +} +EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk); /* There is a fixed mapping between audio pin node and display port * on current Intel platforms: diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 637b8a0e2a91..9a0d1445ca5c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -857,50 +857,6 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ -/* Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK - * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) - * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: - * BCLK = CDCLK * M / N - * The values will be lost when the display power well is disabled and need to - * be restored to avoid abnormal playback speed. - */ -static void haswell_set_bclk(struct hda_intel *hda) -{ - struct azx *chip = &hda->chip; - int cdclk_freq; - unsigned int bclk_m, bclk_n; - - if (!hda->need_i915_power) - return; - - cdclk_freq = snd_hdac_get_display_clk(azx_bus(chip)); - switch (cdclk_freq) { - case 337500: - bclk_m = 16; - bclk_n = 225; - break; - - case 450000: - default: /* default CDCLK 450MHz */ - bclk_m = 4; - bclk_n = 75; - break; - - case 540000: - bclk_m = 4; - bclk_n = 90; - break; - - case 675000: - bclk_m = 8; - bclk_n = 225; - break; - } - - azx_writew(chip, HSW_EM4, bclk_m); - azx_writew(chip, HSW_EM5, bclk_n); -} - #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -958,7 +914,7 @@ static int azx_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) { snd_hdac_display_power(azx_bus(chip), true); - haswell_set_bclk(hda); + snd_hdac_i915_set_bclk(azx_bus(chip)); } if (chip->msi) if (pci_enable_msi(pci) < 0) @@ -1058,7 +1014,7 @@ static int azx_runtime_resume(struct device *dev) bus = azx_bus(chip); if (hda->need_i915_power) { snd_hdac_display_power(bus, true); - haswell_set_bclk(hda); + snd_hdac_i915_set_bclk(bus); } else { /* toggle codec wakeup bit for STATESTS read */ snd_hdac_set_codec_wakeup(bus, true); @@ -1796,12 +1752,8 @@ static int azx_first_init(struct azx *chip) /* initialize chip */ azx_init_pci(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - struct hda_intel *hda; - - hda = container_of(chip, struct hda_intel, chip); - haswell_set_bclk(hda); - } + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + snd_hdac_i915_set_bclk(bus); hda_intel_init_chip(chip, (probe_only[dev] & 2) == 0); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 40933aa33afe..1483f85999ec 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2232,6 +2232,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) if (atomic_read(&(codec)->core.in_pm)) return; + snd_hdac_i915_set_bclk(&codec->bus->core); check_presence_and_report(codec, pin_nid); } -- 2.8.1

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[alsa-devel] [PATCH] Asoc: Intel: common: Removed unused sst_acpi_mach fields
by Joel Holdsworth 26 Apr '16

26 Apr '16

The fields in question are currently unused, and are unlikely to be useful in the future: - "board" is a very vague name not related to a concrete matching criteria such as a DMI match or ACPI OEM IDs. - "machine_quirk" lacks the pointer arguments to be able to manipulate any sst-acpi structures. For it to be useful these would need to be added. Furthermore, a quirk function is much more likely to be relevant to a whole chipset, rather than a single board. - Without "machine_quirk", "pdata" serves no purpose and should be removed. --- sound/soc/intel/common/sst-acpi.c | 8 ++++---- sound/soc/intel/common/sst-acpi.h | 5 ----- 2 files changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/common/sst-acpi.c b/sound/soc/intel/common/sst-acpi.c index 2c5eda1..aba6575 100644 --- a/sound/soc/intel/common/sst-acpi.c +++ b/sound/soc/intel/common/sst-acpi.c @@ -180,7 +180,7 @@ static int sst_acpi_remove(struct platform_device *pdev) } static struct sst_acpi_mach haswell_machines[] = { - { "INT33CA", "haswell-audio", "intel/IntcSST1.bin", NULL, NULL, NULL }, + { "INT33CA", "haswell-audio", "intel/IntcSST1.bin" }, {} }; @@ -198,7 +198,7 @@ static struct sst_acpi_desc sst_acpi_haswell_desc = { }; static struct sst_acpi_mach broadwell_machines[] = { - { "INT343A", "broadwell-audio", "intel/IntcSST2.bin", NULL, NULL, NULL }, + { "INT343A", "broadwell-audio", "intel/IntcSST2.bin" }, {} }; @@ -217,8 +217,8 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { #if !IS_ENABLED(CONFIG_SND_SST_IPC_ACPI) static struct sst_acpi_mach baytrail_machines[] = { - { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL }, - { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master", NULL, NULL, NULL }, + { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, + { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, {} }; diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h index 4dcfb7e..9eb8ade 100644 --- a/sound/soc/intel/common/sst-acpi.h +++ b/sound/soc/intel/common/sst-acpi.h @@ -28,9 +28,4 @@ struct sst_acpi_mach { const char *drv_name; /* firmware file name */ const char *fw_filename; - - /* board name */ - const char *board; - void (*machine_quirk)(void); - void *pdata; }; -- 2.7.3

2 2

[alsa-devel] [PATCH v4] ASoC: docs: add clocking examples for DAI formats
by Peter Rosin 25 Apr '16

25 Apr '16

Provide *our* view of what the rules are for the different DAI formats, so that we do not have to trust external interpretations for this crucial bit of interoperability. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- Documentation/sound/alsa/soc/clocking.txt | 195 ++++++++++++++++++++++++++++++ 1 file changed, 195 insertions(+) Use rising edge and falling edge (instead of flanks). Also differentiate how transmitters and receivers behave and explain at what edges transmitters change data and at what edges receivers read data. Don't speak about left and right channels for the DSP formats where there are just a number of channels. Cheers, Peter diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt index b1300162e01c..7c2bea8124b0 100644 --- a/Documentation/sound/alsa/soc/clocking.txt +++ b/Documentation/sound/alsa/soc/clocking.txt @@ -47,5 +47,200 @@ rate, number of channels and word size) to save on power. It is also desirable to use the codec (if possible) to drive (or master) the audio clocks as it usually gives more accurate sample rates than the CPU. +The below diagrams all have BCLK as the first signal, LRC as the second signal +and DATA as the third. Below that is an indication about which DATA bits belong +in what channel. + +A "..." marking as DATA indicates that there may be more bits that are not +shown. Also, all DATA bits marked X may or may not be present. DAI +transmitters must add them should LRC not match the word size exactly and DAI +receivers must be prepared to ignore them. DAI transmitters must insert zeros +for I2S and Left Justified modes, and preferably not drive DATA for the DSP +modes for these extra X bits. + + +I2S +--- + +The LRC master should put the edges of LRC so that the LRC slaves can reliably +sample LRC at the rising edges of BCLK, preferably by changing LRC in response +to falling edges of BCLK. + +Transmitters wait for a falling edge of LRC, then skip a rising edge of BCLK, +and start the left word (output MSB) at the following falling edge of BCLK. +Transmitters continue with the rest of the word by changing DATA at the falling +edges to BCLK. Transmitters must pad words with zeros if LRC is not matching +the word size, so that receivers expecting wider words do not get noise. +Transmission of the right word works the same, but is initiated by a rising +edge of LRC instead. + +Receivers wait for a falling edge of LRC, then skip a rising edge of BCLK and +read the MSB of the left word at the following rising edge of BCLK. Receivers +then continue with the rest of the word at the following rising edges of BCLK. +Receivers must ignore the tail if more bits per word are transmitted than they +can use. Reception of the right word works the same, but is initiated by a +rising edge of LRC instead. + +See the official I2S specification +https://www.sparkfun.com/datasheets/BreakoutBoards/I2SBUS.pdf + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- +---. .-------------------------------. + '-------------------------------' '----- +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Left Justified (aka MSB) +------------------------ + +The LRC master should synchronize the LRC clock edges with the falling edges of +BCLK. + +Transmitters start the left word (output MSB) at the rising edge of LRC and +hold MSB over the first rising edge of BCLK. The transmitters then change DATA +at the falling edges of BCLK. Transmitters must pad words with zeros if LRC is +not matching the word size, so that receivers expecting wider words do not get +noise. Transmission of the right word works the same, but is initiated by a +falling edge of LRC instead. + +Receivers wait for a rising edge of LRC and reads the MSB of the left word at +the first rising edge of BCLK after that. Receivers then continue with the rest +of the word at the following rising edges of BCLK. Receivers must ignore the +tail if more bits per word are transmitted than they can use. Reception of the +right word works the same, but is initiated by a falling edge of LRC instead. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Right Justified (aka LSB) +------------------------- + +The LRC master should synchronize the LRC clock edges with the falling edges of +BCLK. + +Transmitters wait for a rising edge of LRC, and then skip the correct number of +rising edges of BCLK. If zero rising edges are skipped, transmitters start the +left word (output MSB) immediately at the rising edge of LRC. If at least one +rising edge of BCLK is skipped, they start the left word at the following +falling edge of BCLK. The correct number of rising edges of BCLK to skip after +the initial rising edge of LRC is such that the LSB is ending at the following +edge of LRC. After the MSB, transmitters continue with the rest of the word at +the falling edges of BCLK. Transmission of the right word works the same, but +is initiated by a falling edge of LRC instead. + +Receivers wait for a rising edge of LRC, and then skip the same number of rising +edges of BCLK as the transmitters, and read the MSB of the left word on the +first rising edge of BCLK after that. Receivers then continue with the rest of +the word at the following rising edges of BCLK. Reception of the right word +works the same, but is initiated by a falling edge of LRC instead. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | X |...| X |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +DSP mode A +---------- +Transmitters wait for a rising edge of LRC, then skip one rising edge of BCLK +and start the first word (output MSB) at the following falling edge of BCLK. +They then change DATA at the falling edges of BCLK. When the first word is +finished, transmitters follow up immediately with further words. + +Receivers wait for a rising edge of LRC, then skip one rising edge of BCLK and +read the MSB of the first word at the following rising edge of BCLK. Receivers +then continue with the rest of the word at the rising edges of BCLK. Further +words are following immediately. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .- .- + -' -' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + | |MSB| |...| |LSB|MSB| |...| |LSB| X | |...| | X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | 1st channel | 2nd channel | | + +LRC examples for DSP mode A + .---. .---. +---' '-------------------------------------------------------' '----- +-. .---------------------------------------------------------. .--------- + '-' '-' + + +DSP mode B +---------- +Transmitters start the first word (output MSB) at the rising edge of LRC, hold +MSB over the first rising edge of BCLK, and continue with the rest of the word +at the following falling edges of BCLK. When the first word is finished, +transmitters follow up immediately with further words. + +Receivers wait for a rising edge of LRC, then read the MSB of the first word at +the following rising edge of BCLK. Receivers then continue with the rest of the +word at the rising edges of BCLK. Further words are following immediately. + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- + .- .- + -' -' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB|MSB| |...| |LSB| X | |...| | X | | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | 1st channel | 2nd channel | | + +LRC examples for DSP mode B + .---. .---. +---' '-------------------------------------------------------' '----- +-. .---------------------------------------------------------. .--------- + '-' '-' + + +Inverted Clocks +--------------- + +The above diagrams show normal BCLK and LRC clocking where DAI transmitters +change DATA on the falling edge of BCLK and DAI receivers read DATA on the +rising edge. For inverted BCLK it is naturally the other way around. Inverted +LRC is just that. + +Left Justified, inverted BCLK +----------------------------- + +-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .- + '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' + .-------------------------------. .----- +---' '-------------------------------' +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | + + +Left Justified, inverted LRC +---------------------------- + + .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. .-. +-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '-' '- +---. .-------------------------------. + '-------------------------------' '----- +---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.---.- + |MSB| |...| |LSB| X |...| X |MSB| |...| |LSB| X |...| X | | +---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'---'- + | Left channel | Right channel | -- 2.1.4

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Re: [alsa-devel] [PATCH 2/3] ASoC: simple-card: Add support jack detection via codec
by Mark Brown 25 Apr '16

25 Apr '16

On Fri, Apr 22, 2016 at 09:37:39AM -0700, Jeremy McDermond wrote: > components fairly well. I’m just trying to wrap my head around > where this all would fit in. Should the codec driver just register > a bunch of clock components in its initialization somewhere? > I’m thinking that it can’t have its own CCF driver separate from > the codec driver because you won’t have proper access to the > registers over I2C or SPI. I don’t find a lot of examples in the > codec drivers, and there aren’t very many other drivers in the tree > that seem to register clocks with CCF that I can find. If the clock code is worth splitting off into a separate driver that's what drivers/mfd is for.

2 1

[alsa-devel] Intel SST baytrail audio on PIPO W6
by Bharath Ghanta 25 Apr '16

25 Apr '16

Hi, I am trying to get audio working on PIPO W6 tablet. I was following this thread [1] in my efforts to get audio working. Initially i was not getting any interrupts. I changed the interrupt order as described in the thread and was able to get interrupts and successfully load the firmware. I also ran the amixer commands described in [2]. Now I am just getting a clicking sound whenever something plays. Pierre-Louis said something about playing with the DAI links and swapping the codec connections here [3]. Can anybody explain to me how its done ? I tried to enable other switches in alsamixer using the below commands but nothing worked. amixer -c0 sset 'codec_out1 mix 0 pcm0_in' on I mailed the manufacturer asking for the schematics of the audio chip and the android source code for this device but got no response. I also tried the ssp0 firmware but i got no sound instead of the clicking sound i was getting with the default firmware. I have the exact problem as the guy in this thread. [4] Thanks, Bharath [1] http://thread.gmane.org/gmane.linux.alsa.devel/134554 [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/0940 80.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/106 262.html [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/106 178.html

1 0

Re: [alsa-devel] [snd_ali5451] snd_ali5451 0000:00:06.0: BUG: , pos = 16384, buffer size = 16384, period size = 1024
by Takashi Iwai 25 Apr '16

25 Apr '16

On Mon, 25 Apr 2016 16:50:09 +0200, Enrico Mioso wrote: > > Thank you a lot Takashi. > > there are two problems: > 1 - The warning doesn't manifest always: in some cases (instances of mplayer?) this warning manifests itself from time to time, some minutes. Yes, this is likely some XRUN by either hardware boundary condition or a system load. > 2 - I think the audio stops a little bit (less of a second I think), then things go ahead. Still I don't think any message is printed by alsa libraries. > I don't know if this per se indicates there is no overrun, but still my sensation is this. > But this system isn't exactly "extremely fast": so the audio might stop even for other reasons. I'll recompile the kernel to apply this patch and let you know. > The timing will be slow, but I will try. My patch won't improve in this regard, I'm afraid. The PCM core already corrects to the position 0 if such an invalid position is detected, and the patch would do it just implicitly. But, it's still interesting to see whether it has any impact on such behavior. Takashi

1 0

[alsa-devel] [PATCH] ASoC: Make aux_dev more like a generic component
by mengdong.lin＠linux.intel.com 25 Apr '16

25 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> aux_dev is mainly used by the machine driver to specify analog devices, which are registered as codecs. Making it more like a generic component can help the machine driver to use it to specify any component with topology info by name. Details: - Remove the stub 'rtd_aux' array from the soc card. - Add a list 'aux_comp_list' to store the components of aux_devs. And add a list head 'list_aux' to struct snd_soc_component, for adding such components to the above list. - Add a 'init' ops to a component for machine specific init. soc_bind_aux_dev() will set it to be aux_dev's init. And it will be called when probing the component. - soc_bind_aux_dev() will also search components by name of an aux_dev, since it may not be a codec. - Move probing of aux_devs before checking new DAI links brought by topology. - Move removal of aux_devs later than removal of links. Because topology of aux components may register DAIs and the DAI drivers will go with removal of the aux components, we want soc_remove_link_dais() to remove the DAIs at first. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 49011c3..c03e07a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -798,6 +798,7 @@ struct snd_soc_component { unsigned int registered_as_component:1; struct list_head list; + struct list_head list_aux; /* for auxiliary component of the card */ struct snd_soc_dai_driver *dai_drv; int num_dai; @@ -841,6 +842,9 @@ struct snd_soc_component { int (*probe)(struct snd_soc_component *); void (*remove)(struct snd_soc_component *); + /* machine specific init */ + int (*init)(struct snd_soc_component *component); + #ifdef CONFIG_DEBUG_FS void (*init_debugfs)(struct snd_soc_component *component); const char *debugfs_prefix; @@ -1144,8 +1148,7 @@ struct snd_soc_card { */ struct snd_soc_aux_dev *aux_dev; int num_aux_devs; - struct snd_soc_pcm_runtime *rtd_aux; - int num_aux_rtd; + struct list_head aux_comp_list; const struct snd_kcontrol_new *controls; int num_controls; @@ -1553,6 +1556,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); + INIT_LIST_HEAD(&card->aux_comp_list); } static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e03e82..790ee2b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1413,6 +1413,16 @@ static int soc_probe_component(struct snd_soc_card *card, component->name); } + /* machine specific init */ + if (component->init) { + ret = component->init(component); + if (ret < 0) { + dev_err(component->dev, + "Failed to do machine specific init %d\n", ret); + goto err_probe; + } + } + if (component->controls) snd_soc_add_component_controls(component, component->controls, component->num_controls); @@ -1657,65 +1667,81 @@ static int soc_probe_link_dais(struct snd_soc_card *card, static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - const char *name = aux_dev->codec_name; - - rtd->component = soc_find_component(aux_dev->codec_of_node, name); - if (!rtd->component) { - if (aux_dev->codec_of_node) - name = of_node_full_name(aux_dev->codec_of_node); - - dev_err(card->dev, "ASoC: %s not registered\n", name); - return -EPROBE_DEFER; + struct snd_soc_component *component; + const char *name; + struct device_node *codec_of_node; + + if (aux_dev->codec_of_node || aux_dev->codec_name) { + /* codecs, usually analog devices */ + name = aux_dev->codec_name; + codec_of_node = aux_dev->codec_of_node; + component = soc_find_component(codec_of_node, name); + if (!component) { + if (codec_of_node) + name = of_node_full_name(codec_of_node); + goto err_defer; + } + } else if (aux_dev->name) { + /* generic components */ + name = aux_dev->name; + component = soc_find_component(NULL, name); + if (!component) + goto err_defer; + } else { + dev_err(card->dev, "ASoC: Invalid auxiliary device\n"); + return -EINVAL; } - /* - * Some places still reference rtd->codec, so we have to keep that - * initialized if the component is a CODEC. Once all those references - * have been removed, this code can be removed as well. - */ - rtd->codec = rtd->component->codec; - + component->init = aux_dev->init; + list_add(&component->list_aux, &card->aux_comp_list); return 0; + +err_defer: + dev_err(card->dev, "ASoC: %s not registered\n", name); + return -EPROBE_DEFER; } -static int soc_probe_aux_dev(struct snd_soc_card *card, int num) +static int soc_probe_aux_devices(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; + struct snd_soc_component *comp; + int order; int ret; - ret = soc_probe_component(card, rtd->component); - if (ret < 0) - return ret; - - /* do machine specific initialization */ - if (aux_dev->init) { - ret = aux_dev->init(rtd->component); - if (ret < 0) { - dev_err(card->dev, "ASoC: failed to init %s: %d\n", - aux_dev->name, ret); - return ret; + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + list_for_each_entry(comp, &card->aux_comp_list, list_aux) { + if (comp->driver->probe_order == order) { + ret = soc_probe_component(card, comp); + if (ret < 0) { + dev_err(card->dev, + "ASoC: failed to probe aux component %s %d\n", + comp->name, ret); + return ret; + } + } } } - return soc_post_component_init(rtd, aux_dev->name); + return 0; } -static void soc_remove_aux_dev(struct snd_soc_card *card, int num) +static void soc_remove_aux_devices(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - struct snd_soc_component *component = rtd->component; + struct snd_soc_component *comp, *_comp; + int order; - /* unregister the rtd device */ - if (rtd->dev_registered) { - device_unregister(rtd->dev); - rtd->dev_registered = 0; + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + list_for_each_entry_safe(comp, _comp, + &card->aux_comp_list, list_aux) { + if (comp->driver->remove_order == order) { + soc_remove_component(comp); + /* remove it from the card's aux_comp_list */ + list_del(&comp->list_aux); + } + } } - - if (component) - soc_remove_component(component); } static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) @@ -1894,6 +1920,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } } + /* probe auxiliary components */ + ret = soc_probe_aux_devices(card); + if (ret < 0) + goto probe_dai_err; + /* Find new DAI links added during probing components and bind them. * Components with topology may bring new DAIs and DAI links. */ @@ -1923,16 +1954,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } } - for (i = 0; i < card->num_aux_devs; i++) { - ret = soc_probe_aux_dev(card, i); - if (ret < 0) { - dev_err(card->dev, - "ASoC: failed to add auxiliary devices %d\n", - ret); - goto probe_aux_dev_err; - } - } - snd_soc_dapm_link_dai_widgets(card); snd_soc_dapm_connect_dai_link_widgets(card); @@ -1992,8 +2013,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) return 0; probe_aux_dev_err: - for (i = 0; i < card->num_aux_devs; i++) - soc_remove_aux_dev(card, i); + soc_remove_aux_devices(card); probe_dai_err: soc_remove_dai_links(card); @@ -2039,20 +2059,18 @@ static int soc_probe(struct platform_device *pdev) static int soc_cleanup_card_resources(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; - int i; /* make sure any delayed work runs */ list_for_each_entry(rtd, &card->rtd_list, list) flush_delayed_work(&rtd->delayed_work); - /* remove auxiliary devices */ - for (i = 0; i < card->num_aux_devs; i++) - soc_remove_aux_dev(card, i); - /* remove and free each DAI */ soc_remove_dai_links(card); soc_remove_pcm_runtimes(card); + /* remove auxiliary devices */ + soc_remove_aux_devices(card); + soc_cleanup_card_debugfs(card); /* remove the card */ @@ -2608,16 +2626,6 @@ int snd_soc_register_card(struct snd_soc_card *card) INIT_LIST_HEAD(&card->rtd_list); card->num_rtd = 0; - card->rtd_aux = devm_kzalloc(card->dev, - sizeof(struct snd_soc_pcm_runtime) * - card->num_aux_devs, - GFP_KERNEL); - if (card->rtd_aux == NULL) - return -ENOMEM; - - for (i = 0; i < card->num_aux_devs; i++) - card->rtd_aux[i].card = card; - INIT_LIST_HEAD(&card->dapm_dirty); INIT_LIST_HEAD(&card->dobj_list); card->instantiated = 0; -- 2.5.0

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[alsa-devel] Broken IRQs on BayTrail devices
by Joel Holdsworth 25 Apr '16

25 Apr '16

Dear All, I've been working on a kernel patch for a firmware IRQ quirk issue that affects multiple Intel Atom Bay-Trail boards: https://bugzilla.kernel.org/show_bug.cgi?id=117141 If you have a Bay Trail device with non-functioning sound, can you attach the DSDT and dmidecode output to the bug so I can investigate whether the quirk affect your device. Best Regards Joel Holdsworth

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[alsa-devel] [PATCH 0/3] More robust i915 / audio binding
by Takashi Iwai 25 Apr '16

25 Apr '16

Hi, here is a patchset to make the i915 / audio binding more robustly. It's based on Imre's work to implement the stub component ops for notifying the i915 disablement. Takashi === Imre Deak (1): drm/i915/hda: Add audio component stub Takashi Iwai (2): ALSA: hda - Immediately fail if the i915 audio component is disabled ALSA: hda - Support deferred i915 audio component binding drivers/gpu/drm/i915/i915_drv.c | 34 +++++++++++++-------- drivers/gpu/drm/i915/intel_audio.c | 61 ++++++++++++++++++++++++++++++++++++++ drivers/gpu/drm/i915/intel_drv.h | 2 ++ include/drm/i915_component.h | 5 ++++ sound/hda/hdac_i915.c | 28 ++++++++++++----- 5 files changed, 110 insertions(+), 20 deletions(-) -- 2.8.1

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