July 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 00/21] ASoC: SOF: updates for 5.4
by Pierre-Louis Bossart 23 Jul '19

23 Jul '19

Core updates for runtime-pm, loader extensions for i.MX support, increased timeouts for QEMU tests and state machine fixes to make PulseAudio happy. I included in the same set Intel HDaudio patches, mostly about code refactoring, jack detection and robustness improvements for IPC/DMA handling. If Mark/Takashi prefer separate patchsets for core and Intel stuff I can split the two. This is a rather large set mostly due to Summer vacations and a peak of product validation earlier in Q2, I expect more frequent/smaller updates moving forward. We also have quite a few SoundWire-related patches but they aren't strictly related to SOF. Comments and feedback welcome! Daniel Baluta (3): ASoC: SOF: Introduce snd_sof_dsp_get_bar_index ops ASoC: SOF: loader: Use the BAR provided by FW ASoC: SOF: loader: Don't ignore SRAM block types Fred Oh (1): ASoC: SOF: remove unused state variable in suspend function Guennadi Liakhovetski (1): ASoC: SOF: Intel: hda: add a parameter to disable MSI Janusz Jankowski (1): ASoC: SOF: Intel: ssp: BCLK delay parameter Kai Vehmanen (6): ASoC: SOF: reset DMA state in prepare ASoC: SOF: use common code to send PCM_FREE IPC ASoC: SOF: ipc: use timeout configured at probe ASoC: SOF: core: increase default IPC timeouts ASoC: SOF: Intel: hda: reset link DMA state in prepare ASoC: SOF: Intel: hda: remove misleading error trace from IRQ thread Keyon Jie (1): ASoC: SOF: Intel: hda: correct ROM state mask Pan Xiuli (1): ASoC: SOF: pci: mark last_busy value at runtime PM init Rander Wang (4): ASoC: SOF: Intel: hda: reduce ifdef usage for hda ASoC: SOF: Intel: hda: Enable jack detection ASoC: SOF: Intel: hda: fix link DMA config ASoC: SOF: Intel: hda: fix stream id setting Zhu Yingjiang (3): ASoC: SOF: Intel: hda: set position buffer in init chip ASoC: SOF: Intel: hda: use SOF defined init chip in resume ASoC: SOF: Intel: hda: remove duplicated clear WAKESTS include/sound/sof/dai-intel.h | 3 + include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/tokens.h | 1 + sound/soc/sof/core.c | 4 +- sound/soc/sof/intel/cnl.c | 4 +- sound/soc/sof/intel/hda-codec.c | 44 +++++++++++++- sound/soc/sof/intel/hda-ctrl.c | 21 ++++--- sound/soc/sof/intel/hda-dai.c | 38 +++++++----- sound/soc/sof/intel/hda-dsp.c | 100 +++++++++++--------------------- sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sof/intel/hda.c | 28 +++++---- sound/soc/sof/intel/hda.h | 9 +-- sound/soc/sof/ipc.c | 8 +-- sound/soc/sof/loader.c | 14 ++++- sound/soc/sof/ops.h | 28 +++++++-- sound/soc/sof/pcm.c | 63 ++++++++++---------- sound/soc/sof/pm.c | 6 +- sound/soc/sof/sof-pci-dev.c | 3 + sound/soc/sof/sof-priv.h | 12 ++-- sound/soc/sof/topology.c | 3 + 20 files changed, 229 insertions(+), 166 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: Recover BXT FW on DSP boot timeout error
by Pawel Harlozinski 23 Jul '19

23 Jul '19

When DSP boots with timeout error try to reload firmware to keep best user expeirience. Signed-off-by: Szymon Mielczarek <szymonx.mielczarek(a)intel.com> Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 440bca7afbf1..dd0260557530 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -455,13 +455,19 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) /* If core 1 was turned on for booting core 0, turn it off */ skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1)); if (ret == 0) { - dev_err(ctx->dev, "%s: DSP boot timeout\n", __func__); - dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + dev_warn(ctx->dev, + "DSP boot timeout: Error code=0x%x: FW status=0x%x\n", sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); - dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); - ret = -EIO; - goto err; + dev_warn(ctx->dev,"Reloading FW\n"); + + ret = bxt_sst_init_fw(skl->dev, skl); + if (ret < 0) { + dev_err(ctx->dev, "Reload fw failed: %d\n", ret); + dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); + ret = -EIO; + goto err; + } } } @@ -486,7 +492,6 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) if (core_id == SKL_DSP_CORE0_ID) core_mask |= SKL_DSP_CORE_MASK(1); skl_dsp_disable_core(ctx, core_mask); - return ret; } -- 2.17.1

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[alsa-devel] [PATCH v2] ASoC: madera: Read device tree configuration
by Charles Keepax 23 Jul '19

23 Jul '19

Read the configuration of the Madera ASoC driver from device tree. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes since v1: - Returned to using a helping in the madera driver itself, discussions around adding a generic helper indicated that all the error reporting would still need to be contained in the driver which constitutes the vast majority of the code in the helper anyway. - Updated the code to use the new device_property_count_u32 helper. Thanks, Charles sound/soc/codecs/madera.c | 97 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 97 insertions(+) diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index 1b1be19a2f991..5f1e32a5a8556 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -300,6 +300,100 @@ int madera_free_overheat(struct madera_priv *priv) } EXPORT_SYMBOL_GPL(madera_free_overheat); +static int madera_get_variable_u32_array(struct device *dev, + const char *propname, + u32 *dest, int n_max, + int multiple) +{ + int n, ret; + + n = device_property_count_u32(dev, propname); + if (n < 0) { + if (n == -EINVAL) + return 0; /* missing, ignore */ + + dev_warn(dev, "%s malformed (%d)\n", propname, n); + + return n; + } else if ((n % multiple) != 0) { + dev_warn(dev, "%s not a multiple of %d entries\n", + propname, multiple); + + return -EINVAL; + } + + if (n > n_max) + n = n_max; + + ret = device_property_read_u32_array(dev, propname, dest, n); + if (ret < 0) + return ret; + + return n; +} + +static void madera_prop_get_inmode(struct madera_priv *priv) +{ + struct madera *madera = priv->madera; + struct madera_codec_pdata *pdata = &madera->pdata.codec; + u32 tmp[MADERA_MAX_INPUT * MADERA_MAX_MUXED_CHANNELS]; + int n, i, in_idx, ch_idx; + + BUILD_BUG_ON(ARRAY_SIZE(pdata->inmode) != MADERA_MAX_INPUT); + BUILD_BUG_ON(ARRAY_SIZE(pdata->inmode[0]) != MADERA_MAX_MUXED_CHANNELS); + + n = madera_get_variable_u32_array(madera->dev, "cirrus,inmode", + tmp, ARRAY_SIZE(tmp), + MADERA_MAX_MUXED_CHANNELS); + if (n < 0) + return; + + in_idx = 0; + ch_idx = 0; + for (i = 0; i < n; ++i) { + pdata->inmode[in_idx][ch_idx] = tmp[i]; + + if (++ch_idx == MADERA_MAX_MUXED_CHANNELS) { + ch_idx = 0; + ++in_idx; + } + } +} + +static void madera_prop_get_pdata(struct madera_priv *priv) +{ + struct madera *madera = priv->madera; + struct madera_codec_pdata *pdata = &madera->pdata.codec; + u32 out_mono[ARRAY_SIZE(pdata->out_mono)]; + int i, n; + + madera_prop_get_inmode(priv); + + n = madera_get_variable_u32_array(madera->dev, "cirrus,out-mono", + out_mono, ARRAY_SIZE(out_mono), 1); + if (n > 0) + for (i = 0; i < n; ++i) + pdata->out_mono[i] = !!out_mono[i]; + + madera_get_variable_u32_array(madera->dev, + "cirrus,max-channels-clocked", + pdata->max_channels_clocked, + ARRAY_SIZE(pdata->max_channels_clocked), + 1); + + madera_get_variable_u32_array(madera->dev, "cirrus,pdm-fmt", + pdata->pdm_fmt, + ARRAY_SIZE(pdata->pdm_fmt), 1); + + madera_get_variable_u32_array(madera->dev, "cirrus,pdm-mute", + pdata->pdm_mute, + ARRAY_SIZE(pdata->pdm_mute), 1); + + madera_get_variable_u32_array(madera->dev, "cirrus,dmic-ref", + pdata->dmic_ref, + ARRAY_SIZE(pdata->dmic_ref), 1); +} + int madera_core_init(struct madera_priv *priv) { int i; @@ -308,6 +402,9 @@ int madera_core_init(struct madera_priv *priv) BUILD_BUG_ON(!madera_mixer_texts[MADERA_NUM_MIXER_INPUTS - 1]); BUILD_BUG_ON(!madera_mixer_values[MADERA_NUM_MIXER_INPUTS - 1]); + if (!dev_get_platdata(priv->madera->dev)) + madera_prop_get_pdata(priv); + mutex_init(&priv->rate_lock); for (i = 0; i < MADERA_MAX_HP_OUTPUT; i++) -- 2.11.0

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[alsa-devel] [PATCH 0/2] wm8904: adapt driver for use with audio-graph-card
by Michał Mirosław 23 Jul '19

23 Jul '19

This series allows to use WM8904 codec as audio-graph-card component. It is a rewrite of an earlier approach from https://patchwork.kernel.org/patch/10738097/ Series based on tiwai/sound/for-next tree. You can also pull from: https://rere.qmqm.pl/git/linux branch: wm8904 Example DTS: / { sound { compatible = "audio-graph-card"; widgets = "Speaker", "Loudspeaker", "Headphone", "Headphone Jack", "Microphone", "Internal Mic", routing = "Headphone Jack", "HPOUTL", "Headphone Jack", "HPOUTR", "Amplifier", "LINEOUTL", "Amplifier", "LINEOUTR", "Loudspeaker", "Amplifier", "IN1L", "MICBIAS", "IN1L", "Internal Mic"; dais = <&ssc0_port>; }; }; &i2c1 { status = "okay"; wm8904: acodec@1a { compatible = "wlf,wm8904"; reg = <0x1a>; clocks = <&rtc_32k>; clock-names = "mclk"; interrupts-extended = <&pioA PIN_PB3 IRQ_TYPE_LEVEL_HIGH>; #sound-dai-cells = <0>; codec_port: port { codec_ep: endpoint { remote-endpoint = <&ssc0_ep>; dai-format = "left_j"; bitclock-master; frame-master; mclk-fs = <256>; }; }; }; }; &ssc0 { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_ssc0_default>; status = "okay"; atmel,shared-fs-pin; #sound-dai-cells = <0>; ssc0_port: port { ssc0_ep: endpoint { remote-endpoint = <&codec_ep>; }; }; }; Michał Mirosław (2): ASoC: wm8904: rework FLL handling ASoC: wm8904: automatically choose clock source .../devicetree/bindings/sound/wm8904.txt | 4 + sound/soc/atmel/atmel_wm8904.c | 11 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8904.c | 494 ++++++------------ sound/soc/codecs/wm8904.h | 5 - sound/soc/codecs/wm_fll.c | 339 ++++++++++++ sound/soc/codecs/wm_fll.h | 26 + 8 files changed, 531 insertions(+), 354 deletions(-) create mode 100644 sound/soc/codecs/wm_fll.c create mode 100644 sound/soc/codecs/wm_fll.h -- 2.20.1

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[alsa-devel] [PATCH] ALSA: isa: gus: Fix a possible null-pointer dereference in snd_gf1_mem_xfree()
by Jia-Ju Bai 23 Jul '19

23 Jul '19

In snd_gf1_mem_xfree(), there is an if statement on line 72 and line 74 to check whether block->next is NULL: if (block->next) When block->next is NULL, block->next is used on line 84: block->next->prev = block->prev; Thus, a possible null-pointer dereference may occur in this case. To fix this possible bug, block->next is checked before using it. This bug is found by a static analysis tool STCheck written by us. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/isa/gus/gus_mem.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index cb02d18dde60..ed6205b88057 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -81,7 +81,8 @@ int snd_gf1_mem_xfree(struct snd_gf1_mem * alloc, struct snd_gf1_mem_block * blo if (block->prev) block->prev->next = NULL; } else { - block->next->prev = block->prev; + if (block->next) + block->next->prev = block->prev; if (block->prev) block->prev->next = block->next; } -- 2.17.0

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[alsa-devel] [PATCH] ALSA: ac97: Fix double free of ac97_codec_device
by Ding Xiang 23 Jul '19

23 Jul '19

put_device will call ac97_codec_release to free ac97_codec_device and other resources, so remove the kfree and other redundant code. Signed-off-by: Ding Xiang <dingxiang(a)cmss.chinamobile.com> --- sound/ac97/bus.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 7b977b7..7985dd8 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -122,17 +122,12 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx, vendor_id); ret = device_add(&codec->dev); - if (ret) - goto err_free_codec; + if (ret) { + put_device(&codec->dev); + return ret; + } return 0; -err_free_codec: - of_node_put(codec->dev.of_node); - put_device(&codec->dev); - kfree(codec); - ac97_ctrl->codecs[idx] = NULL; - - return ret; } unsigned int snd_ac97_bus_scan_one(struct ac97_controller *adrv, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: pcm3168a: Fix a typo in the name of a constant
by Christophe JAILLET 23 Jul '19

23 Jul '19

There is a typo in PCM1368A_MAX_SYSCLK, it should be PCM3168A_MAX_SYSCLK (1 and 3 switched in 3168) Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/pcm3168a.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index ed0b4317c9ae..844602710eee 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -263,7 +263,7 @@ static unsigned int pcm3168a_scki_ratios[] = { #define PCM3168A_NUM_SCKI_RATIOS_DAC ARRAY_SIZE(pcm3168a_scki_ratios) #define PCM3168A_NUM_SCKI_RATIOS_ADC (ARRAY_SIZE(pcm3168a_scki_ratios) - 2) -#define PCM1368A_MAX_SYSCLK 36864000 +#define PCM3168A_MAX_SYSCLK 36864000 static int pcm3168a_reset(struct pcm3168a_priv *pcm3168a) { @@ -296,7 +296,7 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai, struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(dai->component); int ret; - if (freq > PCM1368A_MAX_SYSCLK) + if (freq > PCM3168A_MAX_SYSCLK) return -EINVAL; ret = clk_set_rate(pcm3168a->scki, freq); -- 2.20.1

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[alsa-devel] [PATCH] ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI links
by Stephan Gerhold 23 Jul '19

23 Jul '19

apq8016_sbc_parse_of() sets up multiple DAI links, depending on the number of nodes in the device tree. However, at the moment CPU and platform components are only allocated for the first link. This causes an oops when more than one link is defined: Internal error: Oops: 96000044 [#1] SMP CPU: 0 PID: 1015 Comm: kworker/0:2 Not tainted 5.3.0-rc1 #4 Call trace: apq8016_sbc_platform_probe+0x1a8/0x3f0 platform_drv_probe+0x50/0xa0 ... Move the allocation inside the loop to ensure that each link is properly initialized. Fixes: 98b232ca9e0e ("ASoC: qcom: apq8016_sbc: use modern dai_link style") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/qcom/apq8016_sbc.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index f60a71990f66..ac75838bbfab 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -150,17 +150,17 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) link = data->dai_link; - dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); - if (!dlc) - return ERR_PTR(-ENOMEM); + for_each_child_of_node(node, np) { + dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); + if (!dlc) + return ERR_PTR(-ENOMEM); - link->cpus = &dlc[0]; - link->platforms = &dlc[1]; + link->cpus = &dlc[0]; + link->platforms = &dlc[1]; - link->num_cpus = 1; - link->num_platforms = 1; + link->num_cpus = 1; + link->num_platforms = 1; - for_each_child_of_node(node, np) { cpu = of_get_child_by_name(np, "cpu"); codec = of_get_child_by_name(np, "codec"); -- 2.22.0

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[alsa-devel] [PATCH V4 0/2] recover the channel swap after xrun
by shengjiu.wang＠nxp.com 23 Jul '19

23 Jul '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> recover the channel swap after xrun Shengjiu Wang (2): ASoC: fsl_esai: Wrap some operations to be functions ASoC: fsl_esai: recover the channel swap after xrun sound/soc/fsl/fsl_esai.c | 266 ++++++++++++++++++++++++++++----------- 1 file changed, 193 insertions(+), 73 deletions(-) Changes in v4 -add more comments and remove read saisr Changes in V3 - update code sytle. Changes in v2 - add one patch for wrap operations to functions. - fix some coding style issue -- 2.21.0

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[alsa-devel] [PATCH v2] ASoC: rt1308: Convert headers to SPDX
by shumingf＠realtek.com 23 Jul '19

23 Jul '19

From: Shuming Fan <shumingf(a)realtek.com> Convert include and source files to SPDX. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1308.c | 17 +++++++---------- sound/soc/codecs/rt1308.h | 6 ++---- 2 files changed, 9 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index cf3d72f99705..6f2ee6809dbb 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -1,13 +1,10 @@ -/* - * rt1308.c -- RT1308 ALSA SoC amplifier component driver - * - * Copyright 2019 Realtek Semiconductor Corp. - * Author: Derek Fang <derek.fang(a)realtek.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// rt1308.c -- RT1308 ALSA SoC amplifier component driver +// +// Copyright 2019 Realtek Semiconductor Corp. +// Author: Derek Fang <derek.fang(a)realtek.com> +// #include <linux/module.h> #include <linux/moduleparam.h> diff --git a/sound/soc/codecs/rt1308.h b/sound/soc/codecs/rt1308.h index c330aae1d527..ff7c423e879e 100644 --- a/sound/soc/codecs/rt1308.h +++ b/sound/soc/codecs/rt1308.h @@ -1,12 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* - * RT1308.h -- RT1308 ALSA SoC amplifier component driver + * rt1308.h -- RT1308 ALSA SoC amplifier component driver * * Copyright 2019 Realtek Semiconductor Corp. * Author: Derek Fang <derek.fang(a)realtek.com> * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef _RT1308_H_ -- 2.22.0

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