September 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 0/9] soundwire: use FIELD_{GET|PREP} in subsystem
by Vinod Koul 04 Sep '20

04 Sep '20

Use the FIELD_{GET|PREP} in soundwire subsytem and remove the local SDW_REG_SHIFT(). This makes code IMO look much neater Tested this on db845c board Changes in v2: - fold in fixes provided by Bard and Pierre - Use GENMAS_ULL for addr - rebase to next and fold in new intel change Vinod Koul (9): soundwire: define and use addr bit masks soundwire: bus: use FIELD_GET() soundwire: slave: use SDW_DISCO_LINK_ID() soundwire: stream: use FIELD_{GET|PREP} soundwire: qcom : use FIELD_{GET|PREP} soundwire: cadence: use FIELD_{GET|PREP} soundwire: intel: use FIELD_{GET|PREP} soundwire: intel_init: use FIELD_{GET|PREP} soundwire: remove SDW_REG_SHIFT() drivers/soundwire/bus.c | 6 +-- drivers/soundwire/cadence_master.c | 61 +++++++++++-------------- drivers/soundwire/intel.c | 52 ++++++++------------- drivers/soundwire/intel_init.c | 2 +- drivers/soundwire/qcom.c | 22 +++------ drivers/soundwire/slave.c | 2 +- drivers/soundwire/stream.c | 13 ++---- include/linux/soundwire/sdw.h | 21 ++++++--- include/linux/soundwire/sdw_registers.h | 7 --- 9 files changed, 75 insertions(+), 111 deletions(-) -- 2.26.2

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[PATCH 0/7] ASoC: soundwire: Move sdw stream operations to
by Bard Liao 04 Sep '20

04 Sep '20

sdw stream operation APIs can be called once per stream. dailink callbacks are good places to call these APIs. Pierre-Louis Bossart (7): ASoC: soc-dai: clarify return value for get_sdw_stream() soundwire: stream: fix NULL/IS_ERR confusion soundwire: intel: fix NULL/ERR_PTR confusion ASOC: Intel: sof_sdw: add dailink .trigger callback ASOC: Intel: sof_sdw: add dailink .prepare and .hw_free callback soundwire: intel: remove .trigger operation soundwire: intel: remove stream handling from .prepare and .hw_free drivers/soundwire/intel.c | 60 ++++------------------- drivers/soundwire/stream.c | 2 +- include/sound/soc-dai.h | 3 +- sound/soc/intel/boards/sof_sdw.c | 81 ++++++++++++++++++++++++++++++++ 4 files changed, 92 insertions(+), 54 deletions(-) -- 2.17.1

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[PATCH v3 0/7] Add H6 I2S support
by Clément Péron 04 Sep '20

04 Sep '20

Hi, This is a sequel of Marcus Cooper serie[0], where remarks made by Maxime have been fixed. I have tested it on my Beelink GS1 board. Thanks, Clement 0: https://lore.kernel.org/patchwork/cover/1139949/ Changes since v2 (thanks Maxime): - Add details in commit log about sign extend sample - Only set FIFO regs as volatile in regmap - Missing a space (detected by checkpatch) Changes since v1: - Fix missing header in set sign extend sample Jernej Skrabec (3): dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S arm64: dts: sun50i-h6: Add HDMI audio to H6 DTSI Marcus Cooper (4): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support ASoC: sun4i-i2s: Adjust regmap settings .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 31 ++ sound/soc/sunxi/sun4i-i2s.c | 265 +++++++++++++++++- 3 files changed, 296 insertions(+), 2 deletions(-) -- 2.20.1

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[PATCH 1/3] ALSA: hda/realtek: Add quirk for System76 Lemur Pro (lemp9)
by Jeremy Soller 03 Sep '20

03 Sep '20

The System76 Lemur Pro (lemp9) requires the ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC quirk for the headset microphone to function. Signed-off-by: Jeremy Soller <jeremy(a)system76.com> Cc: alsa-devel(a)alsa-project.org --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1fa983d2a94..58d5ebb6e196 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7702,6 +7702,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x1401, "System76 Lemur Pro (lemp9)", ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC), SND_PCI_QUIRK(0x1558, 0x8550, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8551, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC),

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[PATCH 1/2] dt-bindings: sound: midas-audio: Correct parsing sound-dai phandles
by Krzysztof Kozlowski 03 Sep '20

03 Sep '20

The "sound-dai" property has cells therefore phandle-array should be used, even if it is just one phandle. This fixes dtbs_check warnings like: arch/arm/boot/dts/exynos4412-trats2.dt.yaml: sound: cpu:sound-dai:0:1: missing phandle tag in 0 arch/arm/boot/dts/exynos4412-trats2.dt.yaml: sound: cpu:sound-dai:0: [158, 0] is too long Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- .../devicetree/bindings/sound/samsung,midas-audio.yaml | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml index 1c755de686f7..578928e67e5c 100644 --- a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -21,7 +21,8 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle-array + maxItems: 1 description: phandle to the I2S controller required: - sound-dai @@ -30,7 +31,8 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle-array + maxItems: 1 description: phandle to the WM1811 CODEC required: - sound-dai -- 2.17.1

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[PATCH 1/2] control: Improve general control interface documentation.
by Tanjeff-N. Moos 03 Sep '20

03 Sep '20

Signed-off-by: Tanjeff-N. Moos <tanjeff(a)cccmz.de> --- src/control/control.c | 72 +++++++++++++++++++++++++++++++++++++------ 1 file changed, 63 insertions(+), 9 deletions(-) diff --git a/src/control/control.c b/src/control/control.c index 1bcf1ab2..497a5399 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -31,7 +31,7 @@ /*! \page control Control interface <P>Control interface is designed to access primitive controls. There is -also interface notifying about control and structure changes. +also an interface for notifying about control and structure changes. \section control_general_overview General overview @@ -43,20 +43,74 @@ are managed according to below model. Some element sets can be added to a sound card by drivers in kernel and userspace applications. - element - - An element can be identified by userspace applications. Each element has - own identical information. + - A control element might be a master volume control, for example, or a + read-only indicator, such as a sync status. An element has a type (e.g. + INTEGER or BOOLEAN) and - depending on the type - min/max values, a step + size, a set of possible values (for enums), etc. - member - - An element includes some members to have a value. The value of each member - can be changed by both of userspace applications and drivers in kernel. -Each element can be identified by two ways; the numerical number (numid), or the -combination of interface, device, subdevice, name, and index. + - An element includes one or more member(s) to have a value. For + example, a stereo volume control element has two members (for + left/right). The members share the same properties (e.g. both + volume controls have the same min/max values). The value of each + member can be changed by both of userspace applications and + drivers in kernel. -The type of element set is one of integer, integerr64, boolean, enumerators, + +\section identifying_elements Identifying the Elements + +Each element has the following identifying properties: + + - The numid (a numeric identifier, assigned when the sound card is + detected, constant while the sound card is kept connected) + + - The interface type (e.g. MIXER, CARD or PCM) + - The device + - The subdevice + - Its name + - Its index + +An element can be identified either by its numid or by the tuple +(interface type, device, subdevice, name, index). This tuple is always +the same (driver updates can change it, but in practice this is +rare). The numid can change on each boot. In case of an USB sound +card, the numid can also change when it is reconnected. + + +\section element_lists Element Lists + +An element list can be used to obtain a list of all elements of the +sound card. The list contains generic information (e.g. how many +elements the card has), and the identifying properties of the elements +(numid, card, name, ...). See #snd_ctl_elem_list_t to learn more about +element lists. + + +\section working_with_elements Working with Elements + +It is possible to obtain information about an element using the +snd_ctl_elem_info_*() functions. For enums, the allowed values can be +obtained, for integers, the min/max values can be obtained, and so +on. In addition, these functions can report the identifying +properties. E.g. when the element is addressed using its numid, the +functions complements the name, index, etc. + +To access the values of a control, use the snd_ctl_elem_value*() +functions. These allow to get and set the actual values or +settings. It is also possible to get and set the ID values (such as +the numid or the name). + + +\section element_sets Element Sets + +The type of element set is one of integer, integer64, boolean, enumerators, bytes and IEC958 structure. This indicates the type of value for each member in elements included in the element set. -When the value of member is changed, corresponding events are transferred to + +\section events Events + +When the value of a member is changed, corresponding events are transferred to userspace applications. The applications should subscribe any events in advance. \section tlv_blob Supplemental data for elements in an element set -- 2.17.1

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[PATCH v2] 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> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- changes in v2 - update note - add acked-by Nicolin sound/soc/fsl/fsl_sai.c | 10 ++++++++-- sound/soc/fsl/fsl_sai.h | 2 ++ 2 files changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 38c7bcbb361d..b2d65e53dbc4 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -489,6 +489,10 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(slots); + /* Set to output mode to avoid tri-stated data pins */ + 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), @@ -497,7 +501,8 @@ 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 | @@ -508,7 +513,8 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, 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, + 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 5f630be74853..736a437450c8 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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[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

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[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

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[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

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