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[PATCH] ASoC: fsl_sai: Support multiple data channel enable bits
by Shengjiu Wang 03 Sep '20

03 Sep '20

One data channel is one data line. From imx7ulp, the SAI IP is enhanced to support multiple data channels. If there is only two channels input and slots is 2, then enable one data channel is enough for data transfer. So enable the TCE/RCE and transmit/receive mask register according to the input channels and slots configuration. Move the data channel enablement from startup() to hw_params(). Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 30 ++++++++++++------------------ sound/soc/fsl/fsl_sai.h | 2 +- 2 files changed, 13 insertions(+), 19 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 62c5fdb678fc..38c7bcbb361d 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -443,6 +443,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, u32 slots = (channels == 1) ? 2 : channels; u32 slot_width = word_width; int adir = tx ? RX : TX; + u32 pins; int ret; if (sai->slots) @@ -451,6 +452,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, if (sai->slot_width) slot_width = sai->slot_width; + pins = DIV_ROUND_UP(channels, slots); + if (!sai->is_slave_mode) { if (sai->bclk_ratio) ret = fsl_sai_set_bclk(cpu_dai, tx, @@ -501,13 +504,17 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR5_FBT_MASK, val_cr5); } + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), + FSL_SAI_CR3_TRCE_MASK, + FSL_SAI_CR3_TRCE((1 << pins) - 1)); regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | FSL_SAI_CR5_FBT_MASK, val_cr5); - regmap_write(sai->regmap, FSL_SAI_xMR(tx), ~0UL - ((1 << channels) - 1)); + regmap_write(sai->regmap, FSL_SAI_xMR(tx), + ~0UL - ((1 << min(channels, slots)) - 1)); return 0; } @@ -517,6 +524,10 @@ static int fsl_sai_hw_free(struct snd_pcm_substream *substream, { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int ofs = sai->soc_data->reg_offset; + + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), + FSL_SAI_CR3_TRCE_MASK, 0); if (!sai->is_slave_mode && sai->mclk_streams & BIT(substream->stream)) { @@ -651,14 +662,9 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - unsigned int ofs = sai->soc_data->reg_offset; bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int ret; - regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), - FSL_SAI_CR3_TRCE_MASK, - FSL_SAI_CR3_TRCE); - /* * EDMA controller needs period size to be a multiple of * tx/rx maxburst @@ -675,17 +681,6 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, return ret; } -static void fsl_sai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - unsigned int ofs = sai->soc_data->reg_offset; - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - - regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), - FSL_SAI_CR3_TRCE_MASK, 0); -} - static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .set_bclk_ratio = fsl_sai_set_dai_bclk_ratio, .set_sysclk = fsl_sai_set_dai_sysclk, @@ -695,7 +690,6 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .hw_free = fsl_sai_hw_free, .trigger = fsl_sai_trigger, .startup = fsl_sai_startup, - .shutdown = fsl_sai_shutdown, }; static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 6aba7d28f5f3..5f630be74853 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -109,7 +109,7 @@ #define FSL_SAI_CR2_DIV_MASK 0xff /* SAI Transmit and Receive Configuration 3 Register */ -#define FSL_SAI_CR3_TRCE BIT(16) +#define FSL_SAI_CR3_TRCE(x) ((x) << 16) #define FSL_SAI_CR3_TRCE_MASK GENMASK(23, 16) #define FSL_SAI_CR3_WDFL(x) (x) #define FSL_SAI_CR3_WDFL_MASK 0x1f -- 2.27.0

3 2

[PATCH v2 00/10] soundwire: intel: add multi-link support
by Bard Liao 03 Sep '20

03 Sep '20

This series enables multi-link support for Intel platforms. Changes in v2: - Remove the "soundwire: intel: Only call sdw stream APIs for the first cpu_dai" patch form this series. We will move the sounwdire stream operations to machine driver in another series. - Update the commit message and title of "soundwire: intel: add error log for clock-stop invalid configs" Pierre-Louis Bossart (10): soundwire: intel: disable shim wake on suspend soundwire: intel: ignore software command retries soundwire: intel: add multi-link support soundwire: intel: add missing support for all clock stop modes soundwire: bus: update multi-link definition with hw sync details soundwire: intel: add multi-link hw_synchronization information soundwire: stream: enable hw_sync as needed by hardware soundwire: intel: add error log for clock-stop invalid configs soundwire: intel: pass link_mask information to each master soundwire: intel: don't manage link power individually drivers/soundwire/intel.c | 264 ++++++++++++++++++++++++++++----- drivers/soundwire/intel.h | 2 + drivers/soundwire/intel_init.c | 1 + drivers/soundwire/stream.c | 15 +- include/linux/soundwire/sdw.h | 6 + 5 files changed, 243 insertions(+), 45 deletions(-) -- 2.17.1

2 11

[PATCH v4 0/3] ASoC: soundwire: fix port_ready[] dynamic allocation
by Bard Liao 03 Sep '20

03 Sep '20

The existing code allocates memory for the total number of ports. This only works if the ports are contiguous, but will break if e.g. a Devices uses port0, 1, and 14. The port_ready[] array would contain 3 elements, which would lead to an out-of-bounds access. Conversely in other cases, the wrong port index would be used leading to timeouts on prepare. This can be fixed by allocating for the worst-case of 15 ports (DP0..DP14). In addition since the number is now fixed, we can use an array instead of a dynamic allocation. Changes in v4: - Reorder the patches order to allow bisect. Changes in v3: - Add ASoC tag in the cover letter title. - Edit the title and commit message of the third patch for better understanding. Changes in v2: - Split patches into sdw and asoc patches. Please note that "soundwire: fix port_ready[] dynamic allocation in mipi_disco" and "ASoC: codecs: fix port_ready[] dynamic allocation in ASoC codecs" should be merged at the same time. Pierre-Louis Bossart (3): ASoC: codecs: soundwire: remove port_ready[] usage from codecs. soundwire: add definition for maximum number of ports soundwire: fix port_ready[] dynamic allocation in mipi_disco drivers/soundwire/mipi_disco.c | 18 +----------------- drivers/soundwire/slave.c | 4 ++++ include/linux/soundwire/sdw.h | 5 +++-- sound/soc/codecs/max98373-sdw.c | 15 +-------------- sound/soc/codecs/rt1308-sdw.c | 14 +------------- sound/soc/codecs/rt5682-sdw.c | 15 +-------------- sound/soc/codecs/rt700-sdw.c | 15 +-------------- sound/soc/codecs/rt711-sdw.c | 15 +-------------- sound/soc/codecs/rt715-sdw.c | 33 +-------------------------------- 9 files changed, 14 insertions(+), 120 deletions(-) -- 2.17.1

3 5

a saner API for allocating DMA addressable pages
by Christoph Hellwig 03 Sep '20

03 Sep '20

Hi all, this series replaced the DMA_ATTR_NON_CONSISTENT flag to dma_alloc_attrs with a separate new dma_alloc_pages API, which is available on all platforms. In addition to cleaning up the convoluted code path, this ensures that other drivers that have asked for better support for non-coherent DMA to pages with incurring bounce buffering over can finally be properly supported. I'm still a little unsure about the API naming, as alloc_pages sort of implies a struct page return value, but we return a kernel virtual address. The other alternative would be to name the API dma_alloc_noncoherent, but the whole non-coherent naming seems to put people off. As a follow up I plan to move the implementation of the DMA_ATTR_NO_KERNEL_MAPPING flag over to this framework as well, given that is also is a fundamentally non coherent allocation. The replacement for that flag would then return a struct page, as it is allowed to actually return pages without a kernel mapping as the name suggested (although most of the time they will actually have a kernel mapping..) In addition to the conversions of the existing non-coherent DMA users the last three patches also convert the DMA coherent allocations in the NVMe driver to use this new framework through a dmapool addition. This was both to give me a good testing vehicle, but also because it should speed up the NVMe driver on platforms with non-coherent DMA nicely, without a downside on platforms with cache coherent DMA. A git tree is available here: git://git.infradead.org/users/hch/misc.git dma_alloc_pages Gitweb: http://git.infradead.org/users/hch/misc.git/shortlog/refs/heads/dma_alloc_p… Diffstat: Documentation/core-api/dma-api.rst | 92 ++---- Documentation/core-api/dma-attributes.rst | 8 Documentation/userspace-api/media/v4l/buffer.rst | 17 - Documentation/userspace-api/media/v4l/vidioc-reqbufs.rst | 1 arch/alpha/kernel/pci_iommu.c | 2 arch/arm/include/asm/dma-direct.h | 4 arch/arm/mm/dma-mapping-nommu.c | 2 arch/arm/mm/dma-mapping.c | 4 arch/ia64/Kconfig | 3 arch/ia64/hp/common/sba_iommu.c | 2 arch/ia64/kernel/dma-mapping.c | 14 arch/ia64/mm/init.c | 3 arch/mips/Kconfig | 1 arch/mips/bmips/dma.c | 4 arch/mips/cavium-octeon/dma-octeon.c | 4 arch/mips/include/asm/dma-direct.h | 4 arch/mips/include/asm/jazzdma.h | 2 arch/mips/jazz/jazzdma.c | 102 +------ arch/mips/loongson2ef/fuloong-2e/dma.c | 4 arch/mips/loongson2ef/lemote-2f/dma.c | 4 arch/mips/loongson64/dma.c | 4 arch/mips/mm/dma-noncoherent.c | 48 +-- arch/mips/pci/pci-ar2315.c | 4 arch/mips/pci/pci-xtalk-bridge.c | 4 arch/mips/sgi-ip32/ip32-dma.c | 4 arch/parisc/Kconfig | 1 arch/parisc/kernel/pci-dma.c | 6 arch/powerpc/include/asm/dma-direct.h | 4 arch/powerpc/kernel/dma-iommu.c | 2 arch/powerpc/platforms/ps3/system-bus.c | 4 arch/powerpc/platforms/pseries/vio.c | 2 arch/s390/pci/pci_dma.c | 2 arch/x86/kernel/amd_gart_64.c | 8 drivers/gpu/drm/exynos/exynos_drm_gem.c | 2 drivers/gpu/drm/nouveau/nvkm/subdev/instmem/gk20a.c | 3 drivers/iommu/dma-iommu.c | 2 drivers/iommu/intel/iommu.c | 6 drivers/media/common/videobuf2/videobuf2-core.c | 36 -- drivers/media/common/videobuf2/videobuf2-dma-contig.c | 19 - drivers/media/common/videobuf2/videobuf2-dma-sg.c | 3 drivers/media/common/videobuf2/videobuf2-v4l2.c | 12 drivers/net/ethernet/amd/au1000_eth.c | 15 - drivers/net/ethernet/i825xx/lasi_82596.c | 36 +- drivers/net/ethernet/i825xx/lib82596.c | 148 +++++----- drivers/net/ethernet/i825xx/sni_82596.c | 23 - drivers/net/ethernet/seeq/sgiseeq.c | 24 - drivers/nvme/host/pci.c | 79 ++--- drivers/parisc/ccio-dma.c | 2 drivers/parisc/sba_iommu.c | 2 drivers/scsi/53c700.c | 120 ++++---- drivers/scsi/53c700.h | 9 drivers/scsi/sgiwd93.c | 14 drivers/xen/swiotlb-xen.c | 2 include/linux/dma-direct.h | 55 ++- include/linux/dma-mapping.h | 32 +- include/linux/dma-noncoherent.h | 21 - include/linux/dmapool.h | 23 + include/linux/gfp.h | 6 include/media/videobuf2-core.h | 3 include/uapi/linux/videodev2.h | 2 kernel/dma/Kconfig | 9 kernel/dma/Makefile | 1 kernel/dma/coherent.c | 17 + kernel/dma/direct.c | 112 +++++-- kernel/dma/mapping.c | 104 ++----- kernel/dma/ops_helpers.c | 86 ++++++ kernel/dma/pool.c | 2 kernel/dma/swiotlb.c | 4 kernel/dma/virt.c | 2 mm/dmapool.c | 211 +++++++++------ sound/mips/hal2.c | 58 +--- 71 files changed, 872 insertions(+), 803 deletions(-)

8 76

[PATCH v2] soundwire: fix double free of dangling pointer
by trix＠redhat.com 03 Sep '20

03 Sep '20

From: Tom Rix <trix(a)redhat.com> clang static analysis flags this problem stream.c:844:9: warning: Use of memory after it is freed kfree(bus->defer_msg.msg->buf); ^~~~~~~~~~~~~~~~~~~~~~~ This happens in an error handler cleaning up memory allocated for elements in a list. list_for_each_entry(m_rt, &stream->master_list, stream_node) { bus = m_rt->bus; kfree(bus->defer_msg.msg->buf); kfree(bus->defer_msg.msg); } And is triggered when the call to sdw_bank_switch() fails. There are a two problems. First, when sdw_bank_switch() fails, though it frees memory it does not clear bus's reference 'defer_msg.msg' to that memory. The second problem is the freeing msg->buf. In some cases msg will be NULL so this will dereference a null pointer. Need to check before freeing. Fixes: 99b8a5d608a6 ("soundwire: Add bank switch routine") Signed-off-by: Tom Rix <trix(a)redhat.com> Reviewed-by: Nick Desaulniers <ndesaulniers(a)google.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2 : change title, was 'soundwire: fix error handling' --- drivers/soundwire/stream.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 37290a799023..6e36deb505b1 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -717,6 +717,7 @@ static int sdw_bank_switch(struct sdw_bus *bus, int m_rt_count) kfree(wbuf); error_1: kfree(wr_msg); + bus->defer_msg.msg = NULL; return ret; } @@ -840,9 +841,10 @@ static int do_bank_switch(struct sdw_stream_runtime *stream) error: list_for_each_entry(m_rt, &stream->master_list, stream_node) { bus = m_rt->bus; - - kfree(bus->defer_msg.msg->buf); - kfree(bus->defer_msg.msg); + if (bus->defer_msg.msg) { + kfree(bus->defer_msg.msg->buf); + kfree(bus->defer_msg.msg); + } } msg_unlock: -- 2.18.1

3 2

[PATCH] ALSA: hda/realtek - Improved routing for Thinkpad X1 7th/8th Gen
by Takashi Iwai 03 Sep '20

03 Sep '20

There've been quite a few regression reports about the lowered volume (reduced to ca 65% from the previous level) on Lenovo Thinkpad X1 after the commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen"). Although the commit itself does the right thing from HD-audio POV in order to have a volume control for bass speakers, it seems that the machine has some secret recipe under the hood. Through experiments, Benjamin Poirier found out that the following routing gives the best result: * DAC1 (NID 0x02) -> Speaker pin (NID 0x14) * DAC2 (NID 0x03) -> Shared by both Bass Speaker pin (NID 0x17) & Headphone pin (0x21) * DAC3 (NID 0x06) -> Unused DAC1 seems to have some equalizer internally applied, and you'd get again the output in a bad quality if you connect this to the headphone pin. Hence the headphone is connected to DAC2, which is now shared with the bass speaker pin. DAC3 has no volume amp, hence it's not connected at all. For achieving the routing above, this patch introduced a couple of workarounds: * The connection list of bass speaker pin (NID 0x17) is reduced not to include DAC3 (NID 0x06) * Pass preferred_pairs array to specify the fixed connection Here, both workarounds are needed because the generic parser prefers the individual DAC assignment over others. When the routing above is applied, the generic parser creates the two volume controls "Front" and "Bass Speaker". Since we have only two DACs for three output pins, those are not fully controlling each output individually, and it would confuse PulseAudio. For avoiding the pitfall, in this patch, we rename those volume controls to some unique ones ("DAC1" and "DAC2"). Then PulseAudio ignore them and concentrate only on the still good-working "Master" volume control. If a user still wants to control each DAC volume, they can still change manually via "DAC1" and "DAC2" volume controls. Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Reported-by: Benjamin Poirier <benjamin.poirier(a)gmail.com> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Tested-by: Benjamin Poirier <benjamin.poirier(a)gmail.com> Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207407#c10 BugLink: https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… BugLink: https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… Link: https://lore/kernel.org/r/20200829112746.3118-1-benjamin.poirier@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 42 +++++++++++++++++++++++++++++++++++++++++- 1 file changed, 41 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2ef8b080d84b..c521a1f17096 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5868,6 +5868,39 @@ static void alc275_fixup_gpio4_off(struct hda_codec *codec, } } +/* Quirk for Thinkpad X1 7th and 8th Gen + * The following fixed routing needed + * DAC1 (NID 0x02) -> Speaker (NID 0x14); some eq applied secretly + * DAC2 (NID 0x03) -> Bass (NID 0x17) & Headphone (NID 0x21); sharing a DAC + * DAC3 (NID 0x06) -> Unused, due to the lack of volume amp + */ +static void alc285_fixup_thinkpad_x1_gen7(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02, 0x03 }; /* exclude 0x06 */ + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x02, 0x17, 0x03, 0x21, 0x03, 0 + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + spec->gen.preferred_dacs = preferred_pairs; + break; + case HDA_FIXUP_ACT_BUILD: + /* The generic parser creates somewhat unintuitive volume ctls + * with the fixed routing above, and the shared DAC2 may be + * confusing for PA. + * Rename those to unique names so that PA doesn't touch them + * and use only Master volume. + */ + rename_ctl(codec, "Front Playback Volume", "DAC1 Playback Volume"); + rename_ctl(codec, "Bass Speaker Playback Volume", "DAC2 Playback Volume"); + break; + } +} + static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -6136,6 +6169,7 @@ enum { ALC289_FIXUP_DUAL_SPK, ALC294_FIXUP_SPK2_TO_DAC1, ALC294_FIXUP_ASUS_DUAL_SPK, + ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, @@ -7281,11 +7315,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, + [ALC285_FIXUP_THINKPAD_X1_GEN7] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_thinkpad_x1_gen7, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, [ALC285_FIXUP_THINKPAD_HEADSET_JACK] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_jack, .chained = true, - .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1 + .chain_id = ALC285_FIXUP_THINKPAD_X1_GEN7 }, [ALC294_FIXUP_ASUS_HPE] = { .type = HDA_FIXUP_VERBS, -- 2.16.4

1 0

[PATCH v4 00/10] topology: decode: Various fixes
by Piotr Maziarz 03 Sep '20

03 Sep '20

This series fixes various problems with topology decoding mechanism. Some of the problems were critical like improper memory management or infinite loops that were causing undefined behaviour or program crashes, while other resulted in losing some data during conversion. Bugs found while testing with Intel SST topologies. Signed-off-by: Piotr Maziarz <piotrx.maziarz(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Changelog: v2: -Divide into more patches, critical fixes are in separate patches now -More specific descriptions -Fix a typo UML to UCM -Add error reporting in topology: decode: fix channel map memory allocation -Remove goto again in topology: Make buffer for saving dynamic size for better readability v3: -No functional changes -Changed UCM to more descriptive standard ALSA configuration files -removed Gerrit's Change-Id -Added missing signed-offs v4: -Resend, added Reviewed-by tags Piotr Maziarz (10): topology: decode: Fix channel map memory allocation topology: decode: Fix infinite loop in decoding enum control topology: decode: Remove decoding values for enum control topology: decode: Add enum control texts as separate element topology: decode: Fix printing texts section topology: decode: Change declaration of enum decoding function topology: decode: Fix decoding PCM formats and rates topology: decode: Print sig_bits field in PCM capabilities section topology: decode: Add DAI name printing topology: Make buffer for saving dynamic size src/topology/ctl.c | 51 ++++++++++++++++++++++------------------------- src/topology/dapm.c | 3 +-- src/topology/pcm.c | 11 +++++++--- src/topology/save.c | 34 ++++++++++++++++++++++++++----- src/topology/text.c | 2 +- src/topology/tplg_local.h | 2 +- 6 files changed, 64 insertions(+), 39 deletions(-) -- 2.7.4

2 13

[PATCH 00/18] ALSA: cleanups for sparse/cppcheck warnings
by Pierre-Louis Bossart 03 Sep '20

03 Sep '20

Various cleanups for ALSA core, compressed, hda, usb, ac97 and Digigram cards. Pierre-Louis Bossart (18): ALSA: core: pcm: simplify locking for timers ALSA: core: memalloc: fix fallthrough position ALSA: core: pcm_memory: dereference pointer after NULL checks ALSA: core: timer: remove redundant assignment ALSA: core: timer: clarify operator precedence ALSA: compress_offload: dereference after checking for NULL pointer ALSA: compress_offload: remove redundant initialization ALSA: core: init: use DECLARE_COMPLETION_ONSTACK() macro ALSA: aoa: i2sbus: use DECLARE_COMPLETION_ONSTACK() macro ALSA: hda: auto_parser: remove shadowed variable declaration ALSA: hda: (cosmetic) align function parameters ALSA: usb: scarless_gen2: fix endianness issue ALSA: ac97: (cosmetic) align argument names ALSA: atmel: ac97: clarify operator precedence ALSA: rawmidi: (cosmetic) align function parameters ALSA: vx: vx_core: clarify operator precedence ALSA: vx: vx_pcm: remove redundant assignment ALSA: vx: vx_pcm: remove redundant assignment sound/ac97/ac97_core.h | 2 +- sound/aoa/soundbus/i2sbus/pcm.c | 3 +-- sound/atmel/ac97c.c | 22 +++++++++++----------- sound/core/compress_offload.c | 5 +++-- sound/core/init.c | 3 +-- sound/core/memalloc.c | 2 +- sound/core/pcm.c | 8 +++++--- sound/core/pcm_memory.c | 3 ++- sound/core/rawmidi.c | 2 +- sound/core/timer.c | 6 +++--- sound/drivers/vx/vx_core.c | 4 ++-- sound/drivers/vx/vx_pcm.c | 2 -- sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/hda_jack.h | 2 +- sound/pci/hda/hda_local.h | 8 ++++---- sound/usb/mixer_scarlett_gen2.c | 2 +- 16 files changed, 38 insertions(+), 38 deletions(-) -- 2.25.1

3 21

[PATCH] ALSA: hda/realtek - Add control fixup for Lenovo Thinkpad X1 Carbon 7th
by Benjamin Poirier 03 Sep '20

03 Sep '20

As a result of commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen"), the sound output level on my machine, an X1 Carbon 7th gen, was reduced to ~65% of its previous level when playing certain sounds. [1] Internally, this laptop model has three outputs (PCM-OUT1, connection 0x02; PCM-OUT2, connection 0x03; SP-OUT PCM, connection 0x06) which can be routed to two sets of stereo speakers: Front (tweeters, node 0x14) and Bass (woofers, node 0x17, aka Rear in some contexts) and one headphone output (node 0x21). The tweeters are noticeably less powerful than the woofers. [2] Before commit d2cd795c4ece, the bass speakers were connected to SP-OUT PCM. SP-OUT PCM is meant for s/pdif output and does not have volume control. This connection made volume control commonly ineffective (using the Master slider in alsa or pulseaudio apparently had little effect or alternated between mute or max with nothing in between). commit d2cd795c4ece added quirk ALC285_FIXUP_SPEAKER2_TO_DAC1 which resulted in assigning both sets of speakers to PCM-OUT1, bringing the two sets of speakers under one effective volume control but also lowering the output volume noticeably. Fix this by connecting PCM-OUT1 to Front speakers and PCM-OUT2 to Rear speakers. The max output volume is restored to what it was before commit d2cd795c4ece. This is done by setting the connection of node 0x17 to 0x03. However, when we do this, the HDA auto config automatically changes the connection of node 0x21 to 0x02. This output, meant for the front speakers, has some "secret" equalizer which changes the output volume according to the level of what's being played, after some delay[3]. This is undesirable with headphones. Therefore, this patch manually limits the connection of node 0x21 to 0x03. Even though there are three speaker/line outputs (nodes 0x14, 0x17, 0x21), there are only two amps for volume control (nodes 0x02, 0x03). When setting node 0x17 to connection 0x03 and node 0x21 to connection 0x03, amp 0x03 is shared by the Woofer and Headphone outputs. Therefore, they cannot have independent volume control. This patch uses the Speaker control element to regulate nodes 0x02 (acts on tweeters) and 0x03 (acts on woofers and headphones). A virtual control element is created for the Headphone mixer which mirrors the Speaker mixer. These changes are only for volume controls; there are 3 hardware mute controls. The current fixup chain applied to this machine is ALC285_FIXUP_THINKPAD_HEADSET_JACK ALC285_FIXUP_SPEAKER2_TO_DAC1 ALC269_FIXUP_THINKPAD_ACPI ALC269_FIXUP_SKU_IGNORE The fixup added by this patch is an alternative to ALC285_FIXUP_SPEAKER2_TO_DAC1 which must be selected explicitly using the module parameter snd_sof_intel_hda_common.hda_model=alc285-tpx1-dual-speakers when using the new SOF driver architecture, or snd_hda_intel.model=alc285-tpx1-dual-speakers when using the snd_hda_intel driver in legacy mode (see the snd_intel_dspcfg.dsp_driver module parameter). In order to be able to apply ALC285_FIXUP_THINKPAD_HEADSET_JACK without changing current behavior, a new variant of that fixup is introduced. When using the model added by this patch, the fixup chain is: ALC285_FIXUP_TPX1_DUAL_SPEAKERS ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI ALC269_FIXUP_THINKPAD_ACPI ALC269_FIXUP_SKU_IGNORE [1] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… [2] https://bugzilla.kernel.org/show_bug.cgi?id=207407#c10 [3] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Link: https://lore.kernel.org/alsa-devel/20200210025249.GA2700@f3/ Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Kailang Yang <kailang(a)realtek.com> Tested-by: Even Brenden <evenbrenden(a)gmail.com> Tested-by: Vincent Bernat <vincent(a)bernat.ch> Signed-off-by: Benjamin Poirier <benjamin.poirier(a)gmail.com> --- This is an updated version of the patchset (now a single patch) to fix the low output volume on the Lenovo ThinkPad X1 Carbon 7th gen. With this patch, the volume output with the Linux driver matches the one with the Windows driver. Volume control is effective and headphone output is unaffected. Multiple users of this machine have reported a low output volume [1] when using a kernel with commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen"). Many have tried these patches or a similar workaround using `hda-verb` and prefer this configuration[2]. [1] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… ... [2] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… ... Changes: v1 == https://lore.kernel.org/alsa-devel/20200211055651.4405-1-benjamin.poirier@g… v2 == https://lore.kernel.org/alsa-devel/20200703080005.8942-1-benjamin.poirier@g… * patch 1 Shortened log * patch 2 Change log to replace experimental observations by information from diagram posted on kernel.org bugzilla. Select exactly the desired connection for node 0x17 (bass speakers) instead of reusing alc295_fixup_disable_dac3(). Control node 0x21 (headphones) connection to avoid the volume "wobble" effect described in the log. Include SET_CONNECT_SEL command in fixup since it is not done by the hda core. Rename one control to reflect its new function and to use as a condition in the ucm config. v3 == * Remove patch 1 which was merged separately as commit 9774dc218bb6 ("ALSA: hda/realtek - Fix Lenovo Thinkpad X1 Carbon 7th quirk subdevice id") * Make fixup activation manual using "alc285-tpx1-dual-speakers" model instead of automatic based on codec id. Suggested by Jaroslav. * Control both tweeter and woofer volume from "Speaker Playback Volume" control. This way a separate ucm change is not needed, unlike in v2 of the patchset. Suggested by Jaroslav. sound/pci/hda/patch_realtek.c | 355 +++++++++++++++++++++++++++++++++- 1 file changed, 351 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1fa983d2a94..8012c4aaf282 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5821,6 +5821,338 @@ static void alc285_fixup_speaker2_to_dac1(struct hda_codec *codec, } } +static struct snd_kcontrol *find_ctl_verbose(struct hda_codec *codec, + const char *name) +{ + struct snd_kcontrol *kctl; + + kctl = snd_hda_find_mixer_ctl(codec, name); + if (!kctl) + codec_warn(codec, "Did not find control \"%s\"\n", name); + return kctl; +} + +struct tpx1_dual_speaker { + struct hda_codec *codec; + struct snd_kcontrol underlying, hp_vol; +}; + +static int tpx1_dual_speaker_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + return speaker_priv->underlying.info(&speaker_priv->underlying, uinfo); +} + +static int tpx1_dual_speaker_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + return speaker_priv->underlying.get(&speaker_priv->underlying, + ucontrol); +} + +static int tpx1_dual_speaker_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + int err; + + /* Control tweeter volume */ + err = speaker_priv->underlying.put(&speaker_priv->underlying, + ucontrol); + if (err < 0) + return err; + + /* Control woofer volume (shared with headphone) */ + err = speaker_priv->hp_vol.put(&speaker_priv->hp_vol, ucontrol); + if (err < 0) + return err; + + snd_ctl_notify(speaker_priv->codec->card, SNDRV_CTL_EVENT_MASK_VALUE, + &speaker_priv->hp_vol.id); + return err; +} + +static int tpx1_dual_speaker_vol_tlv(struct snd_kcontrol *kcontrol, + int op_flag, unsigned int size, + unsigned int __user *tlv) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + return speaker_priv->underlying.tlv.c(&speaker_priv->underlying, + op_flag, size, tlv); +} + +static void tpx1_dual_speaker_vol_free(struct snd_kcontrol *kcontrol) +{ + struct tpx1_dual_speaker *speaker_priv = snd_kcontrol_chip(kcontrol); + + if (speaker_priv->underlying.private_free) + speaker_priv->underlying.private_free( + &speaker_priv->underlying); + kfree(speaker_priv); +} + +static int tpx1_dual_override_speaker_vol(struct hda_codec *codec, + struct snd_kcontrol *speaker_vol, + struct snd_kcontrol *hp_vol) +{ + struct tpx1_dual_speaker *speaker_priv; + + speaker_priv = kmalloc(sizeof(struct tpx1_dual_speaker), GFP_KERNEL); + if (!speaker_priv) + return -ENOMEM; + speaker_priv->codec = codec; + memcpy(&speaker_priv->underlying, speaker_vol, + sizeof(struct snd_kcontrol)); + memcpy(&speaker_priv->hp_vol, hp_vol, sizeof(struct snd_kcontrol)); + + speaker_vol->info = &tpx1_dual_speaker_vol_info; + speaker_vol->get = &tpx1_dual_speaker_vol_get; + speaker_vol->put = &tpx1_dual_speaker_vol_put; + speaker_vol->tlv.c = &tpx1_dual_speaker_vol_tlv; + speaker_vol->private_data = speaker_priv; + speaker_vol->private_free = &tpx1_dual_speaker_vol_free; + + return 0; +} + +struct tpx1_dual_hp { + struct hda_codec *codec; + struct snd_kcontrol underlying, *speaker_vol; +}; + +static int tpx1_dual_hp_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + return hp_priv->underlying.info(&hp_priv->underlying, uinfo); +} + +static int tpx1_dual_hp_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + return hp_priv->speaker_vol->get(hp_priv->speaker_vol, ucontrol); +} + +static int tpx1_dual_hp_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + int err; + + err = hp_priv->speaker_vol->put(hp_priv->speaker_vol, ucontrol); + if (err < 0) + return err; + snd_ctl_notify(hp_priv->codec->card, SNDRV_CTL_EVENT_MASK_VALUE, + &hp_priv->speaker_vol->id); + + return 0; +} + +static int tpx1_dual_hp_vol_tlv(struct snd_kcontrol *kcontrol, + int op_flag, unsigned int size, + unsigned int __user *tlv) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + return hp_priv->underlying.tlv.c(&hp_priv->underlying, op_flag, size, + tlv); +} + +static void tpx1_dual_hp_vol_free(struct snd_kcontrol *kcontrol) +{ + struct tpx1_dual_hp *hp_priv = snd_kcontrol_chip(kcontrol); + + if (hp_priv->underlying.private_free) + hp_priv->underlying.private_free(&hp_priv->underlying); + kfree(hp_priv); +} + +static int tpx1_dual_override_hp_vol(struct hda_codec *codec, + struct snd_kcontrol *speaker_vol, + struct snd_kcontrol *hp_vol) +{ + struct tpx1_dual_hp *hp_priv; + + hp_priv = kmalloc(sizeof(struct tpx1_dual_hp), GFP_KERNEL); + if (!hp_priv) + return -ENOMEM; + hp_priv->codec = codec; + memcpy(&hp_priv->underlying, hp_vol, sizeof(struct snd_kcontrol)); + hp_priv->speaker_vol = speaker_vol; + + hp_vol->info = &tpx1_dual_hp_vol_info; + hp_vol->get = &tpx1_dual_hp_vol_get; + hp_vol->put = &tpx1_dual_hp_vol_put; + hp_vol->tlv.c = &tpx1_dual_hp_vol_tlv; + hp_vol->private_data = hp_priv; + hp_vol->private_free = &tpx1_dual_hp_vol_free; + + return 0; +} + +/* We cannot use snd_hda_add_vmaster() because it expects the controls to be + * HDA (see init_follower_0dB()). This is not the case here. + */ +static int tpx1_dual_create_main_volume_ctl(struct hda_codec *codec) +{ + struct snd_kcontrol *main_vol, *speaker_vol; + struct snd_ctl_elem_value *ucontrol = NULL; + struct hda_gen_spec *spec = codec->spec; + struct snd_ctl_elem_info *uinfo; + int ch, err; + + speaker_vol = find_ctl_verbose(codec, "Speaker Playback Volume"); + if (!speaker_vol) + return -ENOENT; + + main_vol = snd_ctl_make_virtual_master("Master Playback Volume", + spec->vmaster_tlv); + if (!main_vol) + return -ENOMEM; + err = snd_hda_ctl_add(codec, 0, main_vol); + if (err < 0) + return err; + err = snd_ctl_add_follower(main_vol, speaker_vol); + if (err < 0) + return err; + + /* Initialize values */ + uinfo = kmalloc(sizeof(struct snd_ctl_elem_info), GFP_KERNEL); + if (!uinfo) + return -ENOMEM; + + uinfo->id = speaker_vol->id; + err = speaker_vol->info(speaker_vol, uinfo); + if (err) + goto out; + + ucontrol = kmalloc(sizeof(struct snd_ctl_elem_value), GFP_KERNEL); + if (!ucontrol) { + err = -ENOMEM; + goto out; + } + + ucontrol->id = speaker_vol->id; + for (ch = 0; ch < uinfo->count; ch++) { + ucontrol->value.integer.value[ch] = + uinfo->value.integer.max; + } + err = speaker_vol->put(speaker_vol, ucontrol); + if (err < 0) + goto out; + else + err = 0; + +out: + kfree(ucontrol); + kfree(uinfo); + return err; +} + +static void alc285_fixup_tpx1_dual_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + static const hda_nid_t conn[] = { 0x03 }; + + /* For NID 0x17 (bass speakers), the connection list is {0x02, + * 0x03, 0x06}. Disable SP-OUT PCM (0x06) selection since it + * has no volume control, disable PCM1 (0x02) selection since + * it is for front speakers. This leaves PCM2 (0x03). + */ + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), + conn); + /* For NID 0x21 (headphone out), the connection list is {0x02, + * 0x03}. Disable LOUT1 (0x02) selection since its volume + * fluctuates according to input level. This leaves LOUT2 + * (0x03). + */ + snd_hda_override_conn_list(codec, 0x21, ARRAY_SIZE(conn), + conn); + + /* "Master" controls will be created manually after + * subordinate controls have been changed in the + * HDA_FIXUP_ACT_BUILD phase. + */ + spec->gen.suppress_vmaster = 1; + } else if (action == HDA_FIXUP_ACT_INIT) { + /* Because the overridden connection lists contain only a + * single node, __parse_nid_path() does not label the output + * as "multi". This leads snd_hda_activate_path() to skip the + * AC_VERB_SET_CONNECT_SEL even though it might be needed. Do + * it here instead. + * Note that when doing AC_VERB_SET_CONNECT_SEL, the + * connection is specified by index instead of nid. + */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + 0x1); + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_CONNECT_SEL, + 0x1); + } else if (action == HDA_FIXUP_ACT_BUILD) { + struct snd_kcontrol *speaker_vol, *hp_vol; + int err; + + speaker_vol = find_ctl_verbose(codec, "Speaker Playback Volume"); + if (!speaker_vol) + return; + + hp_vol = find_ctl_verbose(codec, "Headphone Playback Volume"); + if (!hp_vol) + return; + + err = tpx1_dual_override_speaker_vol(codec, speaker_vol, hp_vol); + if (err) { + codec_warn(codec, "Failed to override speaker volume control, err %d\n", + err); + return; + } + + err = tpx1_dual_override_hp_vol(codec, speaker_vol, hp_vol); + if (err) { + codec_warn(codec, "Failed to override headphone volume control, err %d\n", + err); + return; + } + + err = tpx1_dual_create_main_volume_ctl(codec); + if (err) { + codec_warn(codec, "Failed to create main volume control, err %d\n", + err); + return; + } + + err = __snd_hda_add_vmaster( + codec, "Master Playback Switch", NULL, + ((const char * const[]){ + "Speaker", "Bass Speaker", "Headphone", NULL + }), "Playback Switch", true, + &spec->gen.vmaster_mute.sw_kctl); + if (err) { + codec_warn(codec, "Failed to create main mute control, err %d\n", + err); + return; + } + + if (spec->gen.vmaster_mute.hook) { + snd_hda_add_vmaster_hook(codec, + &spec->gen.vmaster_mute, + spec->gen.vmaster_mute_enum); + snd_hda_sync_vmaster_hook(&spec->gen.vmaster_mute); + } + } +} + /* Hook to update amp GPIO4 for automute */ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) @@ -6083,6 +6415,7 @@ enum { ALC225_FIXUP_DISABLE_MIC_VREF, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, + ALC285_FIXUP_TPX1_DUAL_SPEAKERS, ALC285_FIXUP_SPEAKER2_TO_DAC1, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, @@ -6135,7 +6468,8 @@ enum { ALC289_FIXUP_DUAL_SPK, ALC294_FIXUP_SPK2_TO_DAC1, ALC294_FIXUP_ASUS_DUAL_SPK, - ALC285_FIXUP_THINKPAD_HEADSET_JACK, + ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK, + ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, ALC285_FIXUP_HP_GPIO_LED, @@ -6905,6 +7239,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_disable_dac3, }, + [ALC285_FIXUP_TPX1_DUAL_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_tpx1_dual_speakers, + .chained = true, + .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI + }, [ALC285_FIXUP_SPEAKER2_TO_DAC1] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_speaker2_to_dac1, @@ -7280,12 +7620,18 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, - [ALC285_FIXUP_THINKPAD_HEADSET_JACK] = { + [ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_jack, .chained = true, .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1 }, + [ALC285_FIXUP_THINKPAD_HEADSET_JACK_ACPI] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_jack, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, [ALC294_FIXUP_ASUS_HPE] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -7741,8 +8087,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), - SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK), + SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK_SPK), 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), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), @@ -7934,6 +8280,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"}, {.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc225-dell1"}, {.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"}, + {.id = ALC285_FIXUP_TPX1_DUAL_SPEAKERS, .name = "alc285-tpx1-dual-speakers"}, {.id = ALC285_FIXUP_SPEAKER2_TO_DAC1, .name = "alc285-speaker2-to-dac1"}, {.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"}, {.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"}, -- 2.28.0

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[PATCH] ASoC: fsl_sai: Set SAI Channel Mode to Output Mode
by Shengjiu Wang 03 Sep '20

03 Sep '20

Transmit data pins will output zero when slots are masked or channels are disabled. In CHMOD TDM mode, transmit data pins are tri-stated when slots are masked or channels are disabled. When data pins are tri-stated, there is noise on some channels when FS clock value is high and data is read while fsclk is transitioning from high to low. Signed-off-by: Cosmin-Gabriel Samoila <cosmin.samoila(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 12 ++++++++++-- sound/soc/fsl/fsl_sai.h | 2 ++ 2 files changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 62c5fdb678fc..33b194a5c1dc 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -486,6 +486,12 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(slots); + /* Output Mode - data pins transmit 0 when slots are masked + * or channels are disabled + */ + if (tx) + val_cr4 |= FSL_SAI_CR4_CHMOD; + /* * For SAI master mode, when Tx(Rx) sync with Rx(Tx) clock, Rx(Tx) will * generate bclk and frame clock for Tx(Rx), we should set RCR4(TCR4), @@ -494,7 +500,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, if (!sai->is_slave_mode && fsl_sai_dir_is_synced(sai, adir)) { regmap_update_bits(sai->regmap, FSL_SAI_xCR4(!tx, ofs), - FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | + FSL_SAI_CR4_CHMOD_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(!tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | @@ -502,7 +509,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, } regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), - FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | + FSL_SAI_CR4_CHMOD_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 6aba7d28f5f3..19cd4e1bbff9 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) #define FSL_SAI_CR4_SYWD_MASK (0x1f << 8) +#define FSL_SAI_CR4_CHMOD BIT(5) +#define FSL_SAI_CR4_CHMOD_MASK BIT(5) #define FSL_SAI_CR4_MF BIT(4) #define FSL_SAI_CR4_FSE BIT(3) #define FSL_SAI_CR4_FSP BIT(1) -- 2.27.0

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