May 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: hisilicon: fix spelling mistake: "uknown" -> "unknown"
by Colin King 13 May '18

13 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message text Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/hisilicon/hi6210-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 07a57209e055..53344a3b7a60 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -498,7 +498,7 @@ static int hi6210_i2s_trigger(struct snd_pcm_substream *substream, int cmd, hi6210_i2s_txctrl(cpu_dai, 0); break; default: - dev_err(cpu_dai->dev, "uknown cmd\n"); + dev_err(cpu_dai->dev, "unknown cmd\n"); return -EINVAL; } return 0; -- 2.17.0

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[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt5640: Fix compile error
by Hans de Goede 13 May '18

13 May '18

Fix the compile error introduced by: "ASoC: Intel: bytcr_rt5640: Configure PLL1 before using it". Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Mark, feel free to squash this into the original commit --- sound/soc/intel/boards/bytcr_rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 493240194991..828be866fb00 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -210,14 +210,14 @@ static int byt_rt5640_prepare_and_enable_pll1(struct snd_soc_dai *codec_dai, } if (ret < 0) { - dev_err(codec_dai->codec->dev, "can't set pll: %d\n", ret); + dev_err(codec_dai->component->dev, "can't set pll: %d\n", ret); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, rate * 512, SND_SOC_CLOCK_IN); if (ret < 0) { - dev_err(codec_dai->codec->dev, "can't set clock %d\n", ret); + dev_err(codec_dai->component->dev, "can't set clock %d\n", ret); return ret; } -- 2.17.0

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[alsa-devel] What is the meaning of #sound-dai-cells in a device tree?
by sdaau 11 May '18

11 May '18

Hi all, Apologies if this isn't the right forum (my thinking was, is that it is related to sound, and thus ALSA, more) - however, if it isn't, please let me know where I could forward this query (maybe devicetree mailing list ?) I have tried, but I really cannot find any documentation where I can read about, what does conceptually #sound-dai-cells in a device tree mean. Most I've found so far, is: linux_git$ grep -r 'sound-dai-cells' --exclude-dir='.git' arch/arm/boot/dts/sun7i-a20-cubietruck.dts: #sound-dai-cells = <0>; ... arch/arm/boot/dts/r8a7790.dtsi: * #sound-dai-cells is required arch/arm/boot/dts/r8a7790.dtsi: * Single DAI : #sound-dai-cells = <0>; <&rcar_sound>; arch/arm/boot/dts/r8a7790.dtsi: * Multi DAI : #sound-dai-cells = <1>; <&rcar_sound N>; ... Documentation/devicetree/bindings/sound/hisilicon,hi6210-i2s.txt:- #sound-dai-cells: Should be set to 1 (for multi-dai) ... As far as I can tell: * if it is 1, then you can refer to "ports" (?) of a soundcard node, i.e. <&rcar_sound N>; * if it is 0, then you cannot refer to "ports", and you can only refer to <&rcar_sound> Also, it seems to appear only in these kernel source files: drivers/misc/atmel-ssc.c: if (!of_property_read_bool(np, "#sound-dai-cells")) drivers/misc/atmel-ssc.c: if (of_property_read_bool(ssc->pdev->dev.of_node, "#sound-dai-cells")) ... sound/soc/generic/simple-card.c:#define CELL "#sound-dai-cells" ... sound/soc/generic/simple-scu-card.c:#define CELL "#sound-dai-cells" ... sound/soc/soc-core.c: "#sound-dai-cells", 0, &args); sound/soc/soc-core.c: "#sound-dai-cells"); sound/soc/soc-core.c: "#sound-dai-cells", ... so, I guess, it only has meaning in what would be nodes under `target = <&sound>;` in a device tree - is that correct? But then - what is sound-dai-cells actually describing? Number of entries in `static struct snd_soc_dai_driver` of a (codec?) driver, if it is defined as an array? Or a number of DAI links? Could anyone offer a more proper conceptual description of what #sound-dai-cells is supposed to describe? Many thanks in advance for any pointers, s.

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[alsa-devel] [PATCH v6 00/13] soundwire: Add stream support
by Vinod Koul 11 May '18

11 May '18

This series adds support in SoundWire subsystem for: - Documentation for stream support - stream management - data port management - DAI ops in cadence and Intel drivers - ASoC API to propagate SDW stream Updates in v6: - Fix return error as pointed by Sanyog - Update stream state enum documentation as pointed by Pierre and few other nits Updated in v5: - Add todo for device-device changes in future and some comment updates based on feedback from Pierre Updates in v4: - Add more comments for sdw_release_master_stream() - Make sure ports are freed upon sdw_stream_remove_slave() - Clarify snd_soc_dai_set_sdw_stream() and setting of NULL upon release. Updates in v3: - Remove stream state transition checks and redefine stream states - Update bansk switch routines - Other nitpicks pointed out by Pierre. Updates in v2: - Make ASoC API inlined - Make stream states as states and not action - Update the direction enum - Fix some typos and comment updates *** SUBJECT HERE *** *** BLURB HERE *** Sanyog Kale (7): Documentation: soundwire: Add more documentation soundwire: Add support for SoundWire stream management soundwire: Add support for port management soundwire: Add Master and Slave port programming soundwire: Add helpers for ports operations soundwire: Add bank switch routine soundwire: Add stream configuration APIs Shreyas NC (2): ASoC: Add SoundWire stream programming interface soundwire: Remove cdns_master_ops Vinod Koul (4): soundwire: cdns: Add port routines soundwire: cdns: Add stream routines soundwire: intel: Add stream initialization soundwire: intel: Add audio DAI ops .../driver-api/soundwire/error_handling.rst | 65 + Documentation/driver-api/soundwire/index.rst | 3 + Documentation/driver-api/soundwire/locking.rst | 106 ++ Documentation/driver-api/soundwire/stream.rst | 372 +++++ drivers/soundwire/Kconfig | 2 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 43 + drivers/soundwire/bus.h | 72 + drivers/soundwire/cadence_master.c | 451 +++++- drivers/soundwire/cadence_master.h | 151 ++ drivers/soundwire/intel.c | 525 ++++++- drivers/soundwire/intel.h | 4 + drivers/soundwire/intel_init.c | 3 + drivers/soundwire/stream.c | 1479 ++++++++++++++++++++ include/linux/soundwire/sdw.h | 332 ++++- include/linux/soundwire/sdw_intel.h | 14 + include/sound/soc-dai.h | 23 + 17 files changed, 3633 insertions(+), 14 deletions(-) create mode 100644 Documentation/driver-api/soundwire/error_handling.rst create mode 100644 Documentation/driver-api/soundwire/locking.rst create mode 100644 Documentation/driver-api/soundwire/stream.rst create mode 100644 drivers/soundwire/stream.c -- 2.7.4

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Only disable inactive serializer
by Vishal Thanki 11 May '18

11 May '18

As a side effect of the following commit, the active TX serializer may get disabled which may result in distorted audio output. ASoC: davinci-mcasp: Add support for multichannel playback (2952b27e2e463b28d5c0f04000f96b968137ca42) For example, if a 4 channel I2S playback with two TX serializers is activated. Later on, if a recording of 2 channels, with only 1 RX serializer is started, which will also disable one of the TX serializer because max_active_serializers is only calculated for RX (recording) stream. This patch fixes this issue. Signed-off-by: Vishal Thanki <vishalthanki(a)gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 79fc4aa..1f96c9d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -789,7 +789,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, rx_ser < max_active_serializers) { mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; - } else { + } else if (mcasp->serial_dir[i] == INACTIVE_MODE) { mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), SRMOD_INACTIVE, SRMOD_MASK); } -- 2.7.4

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[alsa-devel] [asoc:for-4.18 139/159] sound/soc/intel/boards/bytcr_rt5640.c:177:20: error: 'struct snd_soc_dai' has no member named 'codec'
by kbuild test robot 11 May '18

11 May '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.18 head: 89027d9ebb5948392dd5a4ccc72d5a4eaa865537 commit: bcd9a325f0b0f407c4559779a94e802977c67274 [139/159] ASoC: Intel: bytcr_rt5640: Configure PLL1 before using it config: x86_64-federa-25 (attached as .config) compiler: gcc-7 (Debian 7.3.0-16) 7.3.0 reproduce: git checkout bcd9a325f0b0f407c4559779a94e802977c67274 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/soc/intel/boards/bytcr_rt5640.c: In function 'byt_rt5640_prepare_and_enable_pll1': >> sound/soc/intel/boards/bytcr_rt5640.c:177:20: error: 'struct snd_soc_dai' has no member named 'codec' dev_err(codec_dai->codec->dev, "can't set pll: %d\n", ret); ^~ sound/soc/intel/boards/bytcr_rt5640.c:184:20: error: 'struct snd_soc_dai' has no member named 'codec' dev_err(codec_dai->codec->dev, "can't set clock %d\n", ret); ^~ vim +177 sound/soc/intel/boards/bytcr_rt5640.c 143 144 static int byt_rt5640_prepare_and_enable_pll1(struct snd_soc_dai *codec_dai, 145 int rate) 146 { 147 int ret; 148 149 /* Configure the PLL before selecting it */ 150 if (!(byt_rt5640_quirk & BYT_RT5640_MCLK_EN)) { 151 /* use bitclock as PLL input */ 152 if ((byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) || 153 (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { 154 /* 2x16 bit slots on SSP0 */ 155 ret = snd_soc_dai_set_pll(codec_dai, 0, 156 RT5640_PLL1_S_BCLK1, 157 rate * 32, rate * 512); 158 } else { 159 /* 2x15 bit slots on SSP2 */ 160 ret = snd_soc_dai_set_pll(codec_dai, 0, 161 RT5640_PLL1_S_BCLK1, 162 rate * 50, rate * 512); 163 } 164 } else { 165 if (byt_rt5640_quirk & BYT_RT5640_MCLK_25MHZ) { 166 ret = snd_soc_dai_set_pll(codec_dai, 0, 167 RT5640_PLL1_S_MCLK, 168 25000000, rate * 512); 169 } else { 170 ret = snd_soc_dai_set_pll(codec_dai, 0, 171 RT5640_PLL1_S_MCLK, 172 19200000, rate * 512); 173 } 174 } 175 176 if (ret < 0) { > 177 dev_err(codec_dai->codec->dev, "can't set pll: %d\n", ret); 178 return ret; 179 } 180 181 ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, 182 rate * 512, SND_SOC_CLOCK_IN); 183 if (ret < 0) { 184 dev_err(codec_dai->codec->dev, "can't set clock %d\n", ret); 185 return ret; 186 } 187 188 return 0; 189 } 190 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 1/4] ASoC: compress: Only assign compr->ops->copy once
by Charles Keepax 11 May '18

11 May '18

There are only one set of ops on the compressed stream so no need to reassign the copy callback repeatedly, stop after copy is seen to be necessary. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-compress.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index ba56f87f96d4c..62875c6a93a14 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -973,6 +973,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) continue; compr->ops->copy = soc_compr_copy; + break; } -- 2.11.0

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[alsa-devel] [PATCH v7 00/24] ASoC: qcom: Add support to QDSP based Audio
by Srinivas Kandagatla 11 May '18

11 May '18

Thankyou everyone for providing feedback and testing v6 patchset. This patchset aims to provide a basic version of QCOM DSP based audio support which is available in downstream andriod kernels. This patchset support audio playback on HDMI-RX, MI2S, SLIMBus and will add support to other features as we move on. QDSP has both static and dynamic modules. static modules like AFE (Audio FrontEnd), ADM (Audio Device Manager), ASM(Audio Stream Manager) and CORE to provide this audio services. All these services use APR (Asynchronous Packet Router) protocol via smd/glink transport to communicate with Application processor. More details on each module is availble in there respective patch. This patchset is tested on DB820c, with HDMI audio playback, MI2S on DB410c on top of mainline, Also tested SLIMBus analog audio using wcd9355 with an additional patches. Patches are also tested on SDM845 by Rohit. Here is my test branch incase someone want to try these patches https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=v4.17-qd… Here is block diagram to give a quick overview of the components +---------+ +---------+ +---------+ | q6asm | |q6routing| | q6afe | | dais | <------> | mixers | <-----> | dais | +---------+ +---------+ +---------+ ^ ^ ^ | | | | +------------------+----------------+ | | | | | | v v v v v +---------+ +---------+ +---------+ | q6ASM | | q6ADM | | q6AFE | +---------+ +---------+ +---------+ ^ ^ ^ ^ | | | CPU Side | ------+---------------------+-------------------+-------- | | | | |APR(smd/glink) | | | | | +------------------+----------------+ | | | | | | +-----+--+-----------------------------------+--+------- | | | | | QDSP Side | v v v v v v +---------+ +---------+ +---------+ | ASM | <------> | ADM | <-----> | AFE | +---------+ +---------+ +---------+ ^ | +-------------------+ | ---------------------------+-------------------------- | Audio I/O | v v +--------------------------------------------------+ | Audio devices | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | +--------------------------------------------------+ Changes since v6 (https://lkml.org/lkml/2018/4/26/264) - Added relevant acks and reviewed-by tags. - Update apr_pkt_send and callbacks to take structures instead of void * suggested by Bjorn - Updated q6afe bindings to explain clearly on grandchild dai node bindings, suggested by Rob H. - Fixed few typos in APR commit msg spotted by Mark B. - rebased patchset on top of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/?h=for-ne… Srinivas Kandagatla (24): soc: qcom dt-bindings: Add APR bus bindings soc: qcom: Add APR bus driver ASoC: qdsp6: dt-bindings: Add q6core dt bindings ASoC: qdsp6: dt-bindings: Add q6afe dt bindings ASoC: qdsp6: dt-bindings: Add q6adm dt bindings ASoC: qdsp6: dt-bindings: Add q6asm dt bindings ASoC: qdsp6: q6common: Add qdsp6 helper functions ASoC: qdsp6: q6core: Add q6core driver ASoC: qdsp6: q6afe: Add q6afe driver ASoC: qdsp6: qdafe: Add SLIMBus port Support ASoC: qdsp6: q6afe: Add support to MI2S ports ASoC: qdsp6: q6afe: Add support to MI2S sysclks ASoC: qdsp6: q6adm: Add q6adm driver ASoC: qdsp6: q6asm: Add q6asm driver ASoC: qdsp6: q6asm: Add support to memory map and unmap ASoC: qdsp6: q6asm: Add support to audio stream apis ASoC: qdsp6: q6routing: Add q6routing driver ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers ASoC: qdsp6: q6routing: Add support to MI2S Mixers ASoC: qdsp6: q6afe: Add q6afe dai driver ASoC: qdsp6: q6asm: Add q6asm dai driver ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings ASoC: qcom: apq8096: Add db820c machine driver MAINTAINERS: Add myself as co-maintainer of qcom audio .../devicetree/bindings/soc/qcom/qcom,apr.txt | 84 ++ .../devicetree/bindings/sound/qcom,apq8096.txt | 109 ++ .../devicetree/bindings/sound/qcom,q6adm.txt | 33 + .../devicetree/bindings/sound/qcom,q6afe.txt | 104 ++ .../devicetree/bindings/sound/qcom,q6asm.txt | 33 + .../devicetree/bindings/sound/qcom,q6core.txt | 21 + MAINTAINERS | 1 + drivers/soc/qcom/Kconfig | 9 + drivers/soc/qcom/Makefile | 1 + drivers/soc/qcom/apr.c | 378 ++++++ include/dt-bindings/soc/qcom,apr.h | 28 + include/dt-bindings/sound/qcom,q6afe.h | 31 + include/dt-bindings/sound/qcom,q6asm.h | 22 + include/linux/mod_devicetable.h | 11 + include/linux/soc/qcom/apr.h | 128 ++ sound/soc/qcom/Kconfig | 50 + sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8096.c | 238 ++++ sound/soc/qcom/qdsp6/Makefile | 8 + sound/soc/qcom/qdsp6/q6adm.c | 636 +++++++++ sound/soc/qcom/qdsp6/q6adm.h | 25 + sound/soc/qcom/qdsp6/q6afe-dai.c | 752 +++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 1046 +++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 193 +++ sound/soc/qcom/qdsp6/q6asm-dai.c | 628 +++++++++ sound/soc/qcom/qdsp6/q6asm.c | 1401 ++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 69 + sound/soc/qcom/qdsp6/q6core.c | 380 ++++++ sound/soc/qcom/qdsp6/q6core.h | 15 + sound/soc/qcom/qdsp6/q6dsp-common.c | 66 + sound/soc/qcom/qdsp6/q6dsp-common.h | 24 + sound/soc/qcom/qdsp6/q6dsp-errno.h | 51 + sound/soc/qcom/qdsp6/q6routing.c | 983 ++++++++++++++ sound/soc/qcom/qdsp6/q6routing.h | 9 + 34 files changed, 7572 insertions(+) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6adm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6afe.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6core.txt create mode 100644 drivers/soc/qcom/apr.c create mode 100644 include/dt-bindings/soc/qcom,apr.h create mode 100644 include/dt-bindings/sound/qcom,q6afe.h create mode 100644 include/dt-bindings/sound/qcom,q6asm.h create mode 100644 include/linux/soc/qcom/apr.h create mode 100644 sound/soc/qcom/apq8096.c create mode 100644 sound/soc/qcom/qdsp6/Makefile create mode 100644 sound/soc/qcom/qdsp6/q6adm.c create mode 100644 sound/soc/qcom/qdsp6/q6adm.h create mode 100644 sound/soc/qcom/qdsp6/q6afe-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.h create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.c create mode 100644 sound/soc/qcom/qdsp6/q6asm.h create mode 100644 sound/soc/qcom/qdsp6/q6core.c create mode 100644 sound/soc/qcom/qdsp6/q6core.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-errno.h create mode 100644 sound/soc/qcom/qdsp6/q6routing.c create mode 100644 sound/soc/qcom/qdsp6/q6routing.h -- 2.16.2

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[alsa-devel] Applied "ASoC: zx-i2s: fix spelling mistake: "timeing" -> "timing"" to the asoc tree
by Mark Brown 11 May '18

11 May '18

The patch ASoC: zx-i2s: fix spelling mistake: "timeing" -> "timing" has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d165b5a86fcad715aea1f164186224fe0c2f57f3 Mon Sep 17 00:00:00 2001 From: Colin Ian King <colin.king(a)canonical.com> Date: Thu, 10 May 2018 15:58:38 +0100 Subject: [PATCH] ASoC: zx-i2s: fix spelling mistake: "timeing" -> "timing" Trivial fix to spelling mistake in dev_err message text Signed-off-by: Colin Ian King <colin.king(a)canonical.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/zte/zx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c index 9a0565937d1f..b6b056c607fa 100644 --- a/sound/soc/zte/zx-i2s.c +++ b/sound/soc/zte/zx-i2s.c @@ -197,7 +197,7 @@ static int zx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) val |= (ZX_I2S_TIMING_I2S | ZX_I2S_TIMING_LSB_JUSTIF); break; default: - dev_err(cpu_dai->dev, "Unknown i2s timeing\n"); + dev_err(cpu_dai->dev, "Unknown i2s timing\n"); return -EINVAL; } -- 2.17.0

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[alsa-devel] Applied "ASoC: rt5640: Remove is_sys_clk_from_pll, it has ordering issues" to the asoc tree
by Mark Brown 11 May '18

11 May '18

The patch ASoC: rt5640: Remove is_sys_clk_from_pll, it has ordering issues has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8e7a1f1f177fef28cbd5edc663044b8946d08ab2 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Tue, 8 May 2018 17:35:46 +0200 Subject: [PATCH] ASoC: rt5640: Remove is_sys_clk_from_pll, it has ordering issues is_sys_clk_from_pll() is used as a snd_soc_dapm_route.connected callback, checking RT5640_GBL_CLK to determine if the sys-clk is PLL1 and thus the PWR_PLL bit in reg PWR_ANLG2 must be set. RT5640_GBL_CLK is changed by rt5640_set_dai_sysclk(), which gets called by the pre_pmu / post_pmd functions of the "Platform Clock" dapm-supply. This creates an ordering issue, during a dapm transition first all connected() callbacks are called to build a list of supplies to enable and then the complete list is walked to enable the supplies. Since the connected() check happens before enabling any supplies, is_sys_clk_from_pll() ends up deciding if the PWR_PLL bit should be set based on the state the "Platform Clock" supply had *before* the transition. This sometimes results in PWR_PLL being off, even though *after* the transition PLL1 is configured as sys-clk. This commit removes is_sys_clk_from_pll() instead simply setting / clearing PWR_PLL in rt5640_set_dai_sysclk() based on the selected sys-clk, which fixes this and as a bonus results in a nice cleanup. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5640.c | 29 ++++------------------------- 1 file changed, 4 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 05567426f211..140bad07429e 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -476,20 +476,6 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, return idx; } -static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) -{ - struct snd_soc_component *component = snd_soc_dapm_to_component(source->dapm); - unsigned int val; - - val = snd_soc_component_read32(component, RT5640_GLB_CLK); - val &= RT5640_SCLK_SRC_MASK; - if (val == RT5640_SCLK_SRC_PLL1) - return 1; - else - return 0; -} - static int is_using_asrc(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -1071,9 +1057,6 @@ static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, } static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { - SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, - RT5640_PWR_PLL_BIT, 0, NULL, 0), - /* ASRC */ SND_SOC_DAPM_SUPPLY_S("Stereo Filter ASRC", 1, RT5640_ASRC_1, 15, 0, NULL, 0), @@ -1427,22 +1410,18 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Stereo ADC MIXL", "ADC1 Switch", "Stereo ADC L1 Mux"}, {"Stereo ADC MIXL", "ADC2 Switch", "Stereo ADC L2 Mux"}, {"Stereo ADC MIXL", NULL, "Stereo Filter"}, - {"Stereo Filter", NULL, "PLL1", is_sys_clk_from_pll}, {"Stereo ADC MIXR", "ADC1 Switch", "Stereo ADC R1 Mux"}, {"Stereo ADC MIXR", "ADC2 Switch", "Stereo ADC R2 Mux"}, {"Stereo ADC MIXR", NULL, "Stereo Filter"}, - {"Stereo Filter", NULL, "PLL1", is_sys_clk_from_pll}, {"Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux"}, {"Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux"}, {"Mono ADC MIXL", NULL, "Mono Left Filter"}, - {"Mono Left Filter", NULL, "PLL1", is_sys_clk_from_pll}, {"Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux"}, {"Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux"}, {"Mono ADC MIXR", NULL, "Mono Right Filter"}, - {"Mono Right Filter", NULL, "PLL1", is_sys_clk_from_pll}, {"IF2 ADC L", NULL, "Mono ADC MIXL"}, {"IF2 ADC R", NULL, "Mono ADC MIXR"}, @@ -1512,10 +1491,8 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"DIG MIXR", "DAC R1 Switch", "DAC MIXR"}, {"DAC L1", NULL, "Stereo DAC MIXL"}, - {"DAC L1", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC L1", NULL, "DAC L1 Power"}, {"DAC R1", NULL, "Stereo DAC MIXR"}, - {"DAC R1", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC R1", NULL, "DAC R1 Power"}, {"SPK MIXL", "REC MIXL Switch", "RECMIXL"}, @@ -1622,10 +1599,8 @@ static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, {"DAC L2", NULL, "Mono DAC MIXL"}, - {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC L2", NULL, "DAC L2 Power"}, {"DAC R2", NULL, "Mono DAC MIXR"}, - {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC R2", NULL, "DAC R2 Power"}, {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, @@ -1861,6 +1836,7 @@ static int rt5640_set_dai_sysclk(struct snd_soc_dai *dai, struct snd_soc_component *component = dai->component; struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); unsigned int reg_val = 0; + unsigned int pll_bit = 0; if (freq == rt5640->sysclk && clk_id == rt5640->sysclk_src) return 0; @@ -1871,6 +1847,7 @@ static int rt5640_set_dai_sysclk(struct snd_soc_dai *dai, break; case RT5640_SCLK_S_PLL1: reg_val |= RT5640_SCLK_SRC_PLL1; + pll_bit |= RT5640_PWR_PLL; break; case RT5640_SCLK_S_RCCLK: reg_val |= RT5640_SCLK_SRC_RCCLK; @@ -1879,6 +1856,8 @@ static int rt5640_set_dai_sysclk(struct snd_soc_dai *dai, dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); return -EINVAL; } + snd_soc_component_update_bits(component, RT5640_PWR_ANLG2, + RT5640_PWR_PLL, pll_bit); snd_soc_component_update_bits(component, RT5640_GLB_CLK, RT5640_SCLK_SRC_MASK, reg_val); rt5640->sysclk = freq; -- 2.17.0

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