May 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: amd: renoir: Remove redundant assignment to pdm_ctrl and pdm_enable and pdm_dma_enable
by Jiapeng Chong 11 May '21

11 May '21

Variable pdm_ctrl and pdm_enable and pdm_dma_enable are set to '0x00', but they are overwritten later on, so these are redundant assignments that can be removed. Clean up the following clang-analyzer warning: sound/soc/amd/renoir/acp3x-pdm-dma.c:148:2: warning: Value stored to 'pdm_dma_enable' is never read [clang-analyzer-deadcode.DeadStores]. sound/soc/amd/renoir/acp3x-pdm-dma.c:147:2: warning: Value stored to 'pdm_enable' is never read [clang-analyzer-deadcode.DeadStores]. sound/soc/amd/renoir/acp3x-pdm-dma.c:80:2: warning: Value stored to 'pdm_ctrl' is never read [clang-analyzer-deadcode.DeadStores]. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/amd/renoir/acp3x-pdm-dma.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 4c2810e..bd20622b 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -77,7 +77,6 @@ static void enable_pdm_clock(void __iomem *acp_base) u32 pdm_clk_enable, pdm_ctrl; pdm_clk_enable = ACP_PDM_CLK_FREQ_MASK; - pdm_ctrl = 0x00; rn_writel(pdm_clk_enable, acp_base + ACP_WOV_CLK_CTRL); pdm_ctrl = rn_readl(acp_base + ACP_WOV_MISC_CTRL); @@ -144,9 +143,6 @@ static int stop_pdm_dma(void __iomem *acp_base) u32 pdm_enable, pdm_dma_enable; int timeout; - pdm_enable = 0x00; - pdm_dma_enable = 0x00; - pdm_enable = rn_readl(acp_base + ACP_WOV_PDM_ENABLE); pdm_dma_enable = rn_readl(acp_base + ACP_WOV_PDM_DMA_ENABLE); if (pdm_dma_enable & 0x01) { -- 1.8.3.1

2 1

[PATCH] ASoC: rt1019: Add non_legacy_dai_naming config
by derek.fang＠realtek.com 11 May '21

11 May '21

From: Derek Fang <derek.fang(a)realtek.com> Register the codec dai name as 'rt1019-aif' by adding non_legacy_dai_naming configuration. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c index 10656a5927f1..2687ae60fd36 100644 --- a/sound/soc/codecs/rt1019.c +++ b/sound/soc/codecs/rt1019.c @@ -522,6 +522,7 @@ static const struct snd_soc_component_driver soc_component_dev_rt1019 = { .num_dapm_widgets = ARRAY_SIZE(rt1019_dapm_widgets), .dapm_routes = rt1019_dapm_routes, .num_dapm_routes = ARRAY_SIZE(rt1019_dapm_routes), + .non_legacy_dai_naming = 1, }; static const struct regmap_config rt1019_regmap = { -- 2.17.1

2 1

[PATCH] ASoC: jz4740-i2s: fix function name
by H. Nikolaus Schaller 11 May '21

11 May '21

This driver is not related to I2C protocol. s/_i2c_/_i2s_/ Signed-off-by: H. Nikolaus Schaller <hns(a)goldelico.com> --- sound/soc/jz4740/jz4740-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 47d955c0bb6a1..fe5b3a2b239c6 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -372,7 +372,7 @@ static int jz4740_i2s_resume(struct snd_soc_component *component) return 0; } -static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) +static void jz4740_i2s_init_pcm_config(struct jz4740_i2s *i2s) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -397,7 +397,7 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) if (ret) return ret; - jz4740_i2c_init_pcm_config(i2s); + jz4740_i2s_init_pcm_config(i2s); snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, &i2s->capture_dma_data); -- 2.26.2

3 2

[PATCH 0/4] ASoC: cppcheck fixes of the week
by Pierre-Louis Bossart 11 May '21

11 May '21

While running some checks on a rebased branch, I realized I missed a couple of trivial cases on newer code. Pierre-Louis Bossart (4): ASoC: codecs: mt6359-accdet: remove useless initialization ASoc: codecs: mt6359: remove useless initializations ASoC: codecs: rt1019: clarify expression ASoC: fsl: imx-pcm-rpmsg: remove useless initialization sound/soc/codecs/mt6359-accdet.c | 2 +- sound/soc/codecs/mt6359.c | 2 +- sound/soc/codecs/rt1019.c | 4 ++-- sound/soc/fsl/imx-pcm-rpmsg.c | 2 +- 4 files changed, 5 insertions(+), 5 deletions(-) -- 2.25.1

2 5

[PATCH] ASoC: rt711-sdca: fix the function number of SDCA control for feature unit 0x1E
by shumingf＠realtek.com 11 May '21

11 May '21

From: Shuming Fan <shumingf(a)realtek.com> The function number should be FUNC_NUM_MIC_ARRAY(0x2) for the feature unit 0x1E. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index cc36739f7fcf..24a084e0b48a 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -683,13 +683,13 @@ static int rt711_sdca_set_fu1e_capture_ctl(struct rt711_sdca_priv *rt711) ch_r = (rt711->fu1e_dapm_mute || rt711->fu1e_mixer_r_mute) ? 0x01 : 0x00; err = regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_L), ch_l); if (err < 0) return err; err = regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_R), ch_r); if (err < 0) return err; -- 2.29.0

3 2

[PATCH 1/3] ASoC: cs42l42: Add support for set_jack calls
by Lucas Tanure 11 May '21

11 May '21

Replace the internal jack creation by set_jack call, so users can map buttons in their machine driver Also only enable jack detection IRQ after set_jack call Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 43 ++++++++++++++++++++++---------------- sound/soc/codecs/cs42l42.h | 2 +- 2 files changed, 26 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index bf982e145e94..2143957b95e6 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -518,26 +518,33 @@ static const struct snd_soc_dapm_route cs42l42_audio_map[] = { { "SDOUT2", NULL, "ASP TX EN" }, }; +static int cs42l42_set_jack(struct snd_soc_component *component, struct snd_soc_jack *jk, void *d) +{ + struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); + + cs42l42->jack = jk; + + regmap_update_bits(cs42l42->regmap, CS42L42_TSRS_PLUG_INT_MASK, + CS42L42_RS_PLUG_MASK | CS42L42_RS_UNPLUG_MASK | + CS42L42_TS_PLUG_MASK | CS42L42_TS_UNPLUG_MASK, + (1 << CS42L42_RS_PLUG_SHIFT) | (1 << CS42L42_RS_UNPLUG_SHIFT) | + (0 << CS42L42_TS_PLUG_SHIFT) | (0 << CS42L42_TS_UNPLUG_SHIFT)); + + return 0; +} + static int cs42l42_component_probe(struct snd_soc_component *component) { - struct cs42l42_private *cs42l42 = - (struct cs42l42_private *)snd_soc_component_get_drvdata(component); - struct snd_soc_card *crd = component->card; - int ret = 0; + struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); cs42l42->component = component; - ret = snd_soc_card_jack_new(crd, "CS42L42 Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, - &cs42l42->jack, NULL, 0); - if (ret < 0) - dev_err(component->dev, "Cannot create CS42L42 Headset: %d\n", ret); - - return ret; + return 0; } static const struct snd_soc_component_driver soc_component_dev_cs42l42 = { .probe = cs42l42_component_probe, + .set_jack = cs42l42_set_jack, .dapm_widgets = cs42l42_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cs42l42_dapm_widgets), .dapm_routes = cs42l42_audio_map, @@ -1410,11 +1417,11 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) switch(cs42l42->hs_type){ case CS42L42_PLUG_CTIA: case CS42L42_PLUG_OMTP: - snd_soc_jack_report(&cs42l42->jack, SND_JACK_HEADSET, + snd_soc_jack_report(cs42l42->jack, SND_JACK_HEADSET, SND_JACK_HEADSET); break; case CS42L42_PLUG_HEADPHONE: - snd_soc_jack_report(&cs42l42->jack, SND_JACK_HEADPHONE, + snd_soc_jack_report(cs42l42->jack, SND_JACK_HEADPHONE, SND_JACK_HEADPHONE); break; default: @@ -1442,10 +1449,10 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) switch(cs42l42->hs_type){ case CS42L42_PLUG_CTIA: case CS42L42_PLUG_OMTP: - snd_soc_jack_report(&cs42l42->jack, 0, SND_JACK_HEADSET); + snd_soc_jack_report(cs42l42->jack, 0, SND_JACK_HEADSET); break; case CS42L42_PLUG_HEADPHONE: - snd_soc_jack_report(&cs42l42->jack, 0, SND_JACK_HEADPHONE); + snd_soc_jack_report(cs42l42->jack, 0, SND_JACK_HEADPHONE); break; default: break; @@ -1472,7 +1479,7 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) report = cs42l42_handle_button_press(cs42l42); } - snd_soc_jack_report(&cs42l42->jack, report, SND_JACK_BTN_0 | SND_JACK_BTN_1 | + snd_soc_jack_report(cs42l42->jack, report, SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); } } @@ -1579,8 +1586,8 @@ static void cs42l42_set_interrupt_masks(struct cs42l42_private *cs42l42) CS42L42_TS_UNPLUG_MASK, (1 << CS42L42_RS_PLUG_SHIFT) | (1 << CS42L42_RS_UNPLUG_SHIFT) | - (0 << CS42L42_TS_PLUG_SHIFT) | - (0 << CS42L42_TS_UNPLUG_SHIFT)); + (1 << CS42L42_TS_PLUG_SHIFT) | + (1 << CS42L42_TS_UNPLUG_SHIFT)); } static void cs42l42_setup_hs_type_detect(struct cs42l42_private *cs42l42) diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 36b763f0d1a0..2e0d3836bd7e 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -773,7 +773,7 @@ struct cs42l42_private { struct regulator_bulk_data supplies[CS42L42_NUM_SUPPLIES]; struct gpio_desc *reset_gpio; struct completion pdn_done; - struct snd_soc_jack jack; + struct snd_soc_jack *jack; int bclk; u32 sclk; u32 srate; -- 2.31.1

2 3

[PATCH] ASoC: rt286: Remove redundant assignment to d_len_code
by Jiapeng Chong 11 May '21

11 May '21

Variable d_len_code is set to zero, but this value is never read as it is overwritten or not used later on, hence it is a redundant assignment and can be removed. Clean up the following clang-analyzer warning: sound/soc/codecs/rt286.c:728:2: warning: Value stored to 'd_len_code' is never read [clang-analyzer-deadcode.DeadStores]. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/rt286.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 802f485..caa7208 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -725,7 +725,6 @@ static int rt286_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - d_len_code = 0; switch (params_width(params)) { /* bit 6:4 Bits per Sample */ case 16: -- 1.8.3.1

2 1

[RESEND PATCH] soundwire: add slave device to linked list after device_register()
by Bard Liao 11 May '21

11 May '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> We currently add the slave device to a linked list before device_register(), and then remove it if device_register() fails. It's not clear why this sequence was necessary, this patch moves the linked list management to after the device_register(). Suggested-by: Keyon Jie <yang.jie(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/slave.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 0eed38a79c6d..ff01b96c6e56 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -65,9 +65,6 @@ int sdw_slave_add(struct sdw_bus *bus, for (i = 0; i < SDW_MAX_PORTS; i++) init_completion(&slave->port_ready[i]); - mutex_lock(&bus->bus_lock); - list_add_tail(&slave->node, &bus->slaves); - mutex_unlock(&bus->bus_lock); ret = device_register(&slave->dev); if (ret) { @@ -77,13 +74,15 @@ int sdw_slave_add(struct sdw_bus *bus, * On err, don't free but drop ref as this will be freed * when release method is invoked. */ - mutex_lock(&bus->bus_lock); - list_del(&slave->node); - mutex_unlock(&bus->bus_lock); put_device(&slave->dev); return ret; } + + mutex_lock(&bus->bus_lock); + list_add_tail(&slave->node, &bus->slaves); + mutex_unlock(&bus->bus_lock); + sdw_slave_debugfs_init(slave); return ret; -- 2.17.1

1 0

Re: broonie-sound/for-next bisection: baseline.bootrr.asoc-simple-card-probed on kontron-sl28-var3-ads2
by Guillaume Tucker 11 May '21

11 May '21

Hi Morimoto-san, Please see the bisection report below about the asoc-simple-card driver failing to probe on kontron-sl28-var3-ads2 in today's broonie-sound tree. I believe this has not reached linux-next yet. Reports aren't automatically sent to the public while we're trialing new bisection features on kernelci.org but this one looks valid. Here's the full boot log: https://storage.kernelci.org/broonie-sound/for-next/v5.12-rc8-542-g80e0ab42… More details can be found here: https://linux.kernelci.org/test/case/id/608089974135ccea439b779c/ Please let us know if you need any help debugging the issue or to try a fix. Best wishes, Guillaume On 22/04/2021 04:06, KernelCI bot wrote: > * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * > * This automated bisection report was sent to you on the basis * > * that you may be involved with the breaking commit it has * > * found. No manual investigation has been done to verify it, * > * and the root cause of the problem may be somewhere else. * > * * > * If you do send a fix, please include this trailer: * > * Reported-by: "kernelci.org bot" <bot(a)kernelci.org> * > * * > * Hope this helps! * > * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * > > broonie-sound/for-next bisection: baseline.bootrr.asoc-simple-card-probed on kontron-sl28-var3-ads2 > > Summary: > Start: 80e0ab429149 Merge remote-tracking branch 'asoc/for-5.13' into asoc-next > Plain log: https://storage.kernelci.org/broonie-sound/for-next/v5.12-rc8-542-g80e0ab42… > HTML log: https://storage.kernelci.org/broonie-sound/for-next/v5.12-rc8-542-g80e0ab42… > Result: 59c35c44a9cf ASoC: simple-card: add simple_parse_node() > > Checks: > revert: PASS > verify: PASS > > Parameters: > Tree: broonie-sound > URL: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git > Branch: for-next > Target: kontron-sl28-var3-ads2 > CPU arch: arm64 > Lab: lab-kontron > Compiler: gcc-8 > Config: defconfig+CONFIG_RANDOMIZE_BASE=y > Test case: baseline.bootrr.asoc-simple-card-probed > > Breaking commit found: > > ------------------------------------------------------------------------------- > commit 59c35c44a9cf89a83a92a8d26749e59497d0317d > Author: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > Date: Wed Apr 21 14:45:48 2021 +0900 > > ASoC: simple-card: add simple_parse_node() > > Parse dai/tdm/clk are common for both CPU/Codec node. > This patch creates simple_parse_node() for it and share the code. > > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > Link: https://lore.kernel.org/r/87czuoi41f.wl-kuninori.morimoto.gx@renesas.com > Signed-off-by: Mark Brown <broonie(a)kernel.org> > > diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c > index a1373be4558f..91af4eca9c86 100644 > --- a/sound/soc/generic/simple-card.c > +++ b/sound/soc/generic/simple-card.c > @@ -93,12 +93,11 @@ static void simple_parse_convert(struct device *dev, > } > > static void simple_parse_mclk_fs(struct device_node *top, > - struct device_node *cpu, > - struct device_node *codec, > + struct device_node *np, > struct simple_dai_props *props, > char *prefix) > { > - struct device_node *node = of_get_parent(cpu); > + struct device_node *node = of_get_parent(np); > char prop[128]; > > snprintf(prop, sizeof(prop), "%smclk-fs", PREFIX); > @@ -106,12 +105,57 @@ static void simple_parse_mclk_fs(struct device_node *top, > > snprintf(prop, sizeof(prop), "%smclk-fs", prefix); > of_property_read_u32(node, prop, &props->mclk_fs); > - of_property_read_u32(cpu, prop, &props->mclk_fs); > - of_property_read_u32(codec, prop, &props->mclk_fs); > + of_property_read_u32(np, prop, &props->mclk_fs); > > of_node_put(node); > } > > +static int simple_parse_node(struct asoc_simple_priv *priv, > + struct device_node *np, > + struct link_info *li, > + char *prefix, > + int is_cpu) > +{ > + struct device *dev = simple_priv_to_dev(priv); > + struct device_node *top = dev->of_node; > + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); > + struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); > + struct snd_soc_dai_link_component *dlc; > + struct asoc_simple_dai *dai; > + int ret, single = 0; > + > + if (is_cpu) { > + dlc = asoc_link_to_cpu(dai_link, 0); > + dai = simple_props_to_dai_cpu(dai_props, 0); > + } else { > + dlc = asoc_link_to_codec(dai_link, 0); > + dai = simple_props_to_dai_codec(dai_props, 0); > + } > + > + simple_parse_mclk_fs(top, np, dai_props, prefix); > + > + ret = asoc_simple_parse_dai(np, dlc, &single); > + if (ret) > + return ret; > + > + ret = asoc_simple_parse_clk(dev, np, dai, dlc); > + if (ret) > + return ret; > + > + ret = asoc_simple_parse_tdm(np, dai); > + if (ret) > + return ret; > + > + if (is_cpu) { > + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); > + > + asoc_simple_canonicalize_cpu(dlc, single); > + asoc_simple_canonicalize_platform(platforms, dlc); > + } > + > + return 0; > +} > + > static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > struct device_node *np, > struct device_node *codec, > @@ -121,10 +165,8 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > struct device *dev = simple_priv_to_dev(priv); > struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); > struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); > - struct asoc_simple_dai *dai; > struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); > struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); > - struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); > struct device_node *top = dev->of_node; > struct device_node *node = of_get_parent(np); > char *prefix = ""; > @@ -132,28 +174,18 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > > dev_dbg(dev, "link_of DPCM (%pOF)\n", np); > > - li->link++; > - > /* For single DAI link & old style of DT node */ > if (is_top) > prefix = PREFIX; > > if (li->cpu) { > - int is_single_links = 0; > - > /* Codec is dummy */ > > /* FE settings */ > dai_link->dynamic = 1; > dai_link->dpcm_merged_format = 1; > > - dai = simple_props_to_dai_cpu(dai_props, 0); > - > - ret = asoc_simple_parse_dai(np, cpus, &is_single_links); > - if (ret) > - goto out_put_node; > - > - ret = asoc_simple_parse_clk(dev, np, dai, cpus); > + ret = simple_parse_node(priv, np, li, prefix, 1); > if (ret < 0) > goto out_put_node; > > @@ -162,9 +194,6 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > cpus->dai_name); > if (ret < 0) > goto out_put_node; > - > - asoc_simple_canonicalize_cpu(cpus, is_single_links); > - asoc_simple_canonicalize_platform(platforms, cpus); > } else { > struct snd_soc_codec_conf *cconf; > > @@ -174,14 +203,9 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > dai_link->no_pcm = 1; > dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; > > - dai = simple_props_to_dai_codec(dai_props, 0); > cconf = simple_props_to_codec_conf(dai_props, 0); > > - ret = asoc_simple_parse_dai(np, codecs, NULL); > - if (ret < 0) > - goto out_put_node; > - > - ret = asoc_simple_parse_clk(dev, np, dai, codecs); > + ret = simple_parse_node(priv, np, li, prefix, 0); > if (ret < 0) > goto out_put_node; > > @@ -201,11 +225,6 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > } > > simple_parse_convert(dev, np, &dai_props->adata); > - simple_parse_mclk_fs(top, np, codec, dai_props, prefix); > - > - ret = asoc_simple_parse_tdm(np, dai); > - if (ret) > - goto out_put_node; > > ret = asoc_simple_parse_daifmt(dev, node, codec, > prefix, &dai_link->dai_fmt); > @@ -218,6 +237,8 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > dai_link->init = asoc_simple_dai_init; > > out_put_node: > + li->link++; > + > of_node_put(node); > return ret; > } > @@ -230,23 +251,18 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, > { > struct device *dev = simple_priv_to_dev(priv); > struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); > - struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); > - struct asoc_simple_dai *cpu_dai = simple_props_to_dai_cpu(dai_props, 0); > - struct asoc_simple_dai *codec_dai = simple_props_to_dai_codec(dai_props, 0); > struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); > struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); > struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); > - struct device_node *top = dev->of_node; > struct device_node *cpu = NULL; > struct device_node *node = NULL; > struct device_node *plat = NULL; > char prop[128]; > char *prefix = ""; > - int ret, single_cpu = 0; > + int ret; > > cpu = np; > node = of_get_parent(np); > - li->link++; > > dev_dbg(dev, "link_of (%pOF)\n", node); > > @@ -262,13 +278,11 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, > if (ret < 0) > goto dai_link_of_err; > > - simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); > - > - ret = asoc_simple_parse_dai(cpu, cpus, &single_cpu); > + ret = simple_parse_node(priv, cpu, li, prefix, 1); > if (ret < 0) > goto dai_link_of_err; > > - ret = asoc_simple_parse_dai(codec, codecs, NULL); > + ret = simple_parse_node(priv, codec, li, prefix, 0); > if (ret < 0) > goto dai_link_of_err; > > @@ -276,22 +290,6 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, > if (ret < 0) > goto dai_link_of_err; > > - ret = asoc_simple_parse_tdm(cpu, cpu_dai); > - if (ret < 0) > - goto dai_link_of_err; > - > - ret = asoc_simple_parse_tdm(codec, codec_dai); > - if (ret < 0) > - goto dai_link_of_err; > - > - ret = asoc_simple_parse_clk(dev, cpu, cpu_dai, cpus); > - if (ret < 0) > - goto dai_link_of_err; > - > - ret = asoc_simple_parse_clk(dev, codec, codec_dai, codecs); > - if (ret < 0) > - goto dai_link_of_err; > - > ret = asoc_simple_set_dailink_name(dev, dai_link, > "%s-%s", > cpus->dai_name, > @@ -302,13 +300,12 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, > dai_link->ops = &simple_ops; > dai_link->init = asoc_simple_dai_init; > > - asoc_simple_canonicalize_cpu(cpus, single_cpu); > - asoc_simple_canonicalize_platform(platforms, cpus); > - > dai_link_of_err: > of_node_put(plat); > of_node_put(node); > > + li->link++; > + > return ret; > } > ------------------------------------------------------------------------------- > > > Git bisection log: > > ------------------------------------------------------------------------------- > git bisect start > # good: [bf05bf16c76bb44ab5156223e1e58e26dfe30a88] Linux 5.12-rc8 > git bisect good bf05bf16c76bb44ab5156223e1e58e26dfe30a88 > # bad: [80e0ab4291498248157d2801c994dfaa09ef3082] Merge remote-tracking branch 'asoc/for-5.13' into asoc-next > git bisect bad 80e0ab4291498248157d2801c994dfaa09ef3082 > # good: [d479f00b795ac62b24ef90f4ec421e65c3178ca7] ASoC: soc-pcm: indicate error message at dpcm_path_get() > git bisect good d479f00b795ac62b24ef90f4ec421e65c3178ca7 > # good: [a96d2ba2d8248d5e8170f2f44f98d4a33329b08a] ASoC: codecs: tlv320aic3x: move I2C to separated file > git bisect good a96d2ba2d8248d5e8170f2f44f98d4a33329b08a > # good: [8577bf61a6359bf2387f85a2fda713a4f05185c3] Merge series "ASoC: rsnd: tidyup Renesas sound" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: > git bisect good 8577bf61a6359bf2387f85a2fda713a4f05185c3 > # good: [87143bfdb9f7ddc14b129fda610e114d29077596] Merge series "ASoC: remove more cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: > git bisect good 87143bfdb9f7ddc14b129fda610e114d29077596 > # good: [23b16df6c9c91f70df070be43c5b13ef4016c1e7] Merge series "ASoC: audio-graph: cleanups" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: > git bisect good 23b16df6c9c91f70df070be43c5b13ef4016c1e7 > # good: [1fa27f35ee23b52e0bd708d00c272c5df805afc8] Merge series "ASoC: rt286/rt298: Fixes for DMIC2 config and combo jack" from David Ward <david.ward(a)gatech.edu>: > git bisect good 1fa27f35ee23b52e0bd708d00c272c5df805afc8 > # bad: [434392271afcff350fe11730f12b831fffaf33eb] ASoC: simple-card: add simple_link_init() > git bisect bad 434392271afcff350fe11730f12b831fffaf33eb > # good: [e51237b8d3052251421770468903fa6e4446d158] ASoC: audio-graph: add graph_link_init() > git bisect good e51237b8d3052251421770468903fa6e4446d158 > # bad: [59c35c44a9cf89a83a92a8d26749e59497d0317d] ASoC: simple-card: add simple_parse_node() > git bisect bad 59c35c44a9cf89a83a92a8d26749e59497d0317d > # first bad commit: [59c35c44a9cf89a83a92a8d26749e59497d0317d] ASoC: simple-card: add simple_parse_node() > ------------------------------------------------------------------------------- > > > -=-=-=-=-=-=-=-=-=-=-=- > Groups.io Links: You receive all messages sent to this group. > View/Reply Online (#10236): https://groups.io/g/kernelci-results/message/10236 > Mute This Topic: https://groups.io/mt/82277947/924702 > Group Owner: kernelci-results+owner(a)groups.io > Unsubscribe: https://groups.io/g/kernelci-results/unsub [guillaume.tucker(a)collabora.com] > -=-=-=-=-=-=-=-=-=-=-=- > >

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[PATCH] ALSA: usb-audio: Validate MS endpoint descriptors
by Takashi Iwai 10 May '21

10 May '21

snd_usbmidi_get_ms_info() may access beyond the border when a malformed descriptor is passed. This patch adds the sanity checks of the given MS endpoint descriptors, and skips invalid ones. Reported-by: syzbot+6bb23a5d5548b93c94aa(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index a10ac75969a8..649eb8d1ab7d 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1956,8 +1956,12 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ms_ep = find_usb_ms_endpoint_descriptor(hostep); if (!ms_ep) continue; + if (ms_ep->bLength <= sizeof(*ms_ep)) + continue; if (ms_ep->bNumEmbMIDIJack > 0x10) continue; + if (ms_ep->bLength < sizeof(*ms_ep) + ms_ep->bNumEmbMIDIJack) + continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { if (++epidx >= MIDI_MAX_ENDPOINTS) { -- 2.26.2

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