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

[alsa-devel] [PATCH 1/2] ASoC: AMD: Fix simultaneous playback and capture on different channel
by Akshu Agrawal 10 Sep '18

10 Sep '18

If capture and playback are started on different channel (I2S/BT) there is a possibilty that channel information passed from machine driver is overwritten before the configuration is done in dma driver. Example: 113.597588: cz_max_startup: ---playback sets BT channel 113.597694: cz_dmic1_startup: ---capture sets I2S channel 113.597979: acp_dma_hw_params: ---configures capture for I2S channel 113.598114: acp_dma_hw_params: ---configures playback for I2S channel This is fixed by having lock between startup and prepare. This ensures no other codec startup gets called between a codec's startup(where channel info is set) and hw_params(where channel info is read). Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 3879ccc..b98ffbc 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -47,6 +47,7 @@ static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_clk; +static struct mutex instance_lock; extern int bt_uart_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) @@ -150,6 +151,7 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + mutex_lock(&instance_lock); machine->i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); @@ -160,6 +162,12 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_da7219_prepare(struct snd_pcm_substream *substream) +{ + mutex_unlock(&instance_lock); + return 0; +} + static int cz_max_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -177,6 +185,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + mutex_lock(&instance_lock); machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -186,6 +195,12 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_max_prepare(struct snd_pcm_substream *substream) +{ + mutex_unlock(&instance_lock); + return 0; +} + static int cz_dmic0_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -203,6 +218,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + mutex_lock(&instance_lock); machine->i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -224,6 +240,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + mutex_lock(&instance_lock); machine->i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); @@ -234,24 +251,34 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static int cz_dmic_prepare(struct snd_pcm_substream *substream) +{ + mutex_unlock(&instance_lock); + return 0; +} + static const struct snd_soc_ops cz_da7219_cap_ops = { .startup = cz_da7219_startup, .shutdown = cz_da7219_shutdown, + .prepare = cz_da7219_prepare, }; static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, .shutdown = cz_max_shutdown, + .prepare = cz_max_prepare, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, .shutdown = cz_dmic_shutdown, + .prepare = cz_dmic_prepare, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, .shutdown = cz_dmic_shutdown, + .prepare = cz_dmic_prepare, }; static struct snd_soc_dai_link cz_dai_7219_98357[] = { @@ -409,6 +436,7 @@ static int cz_probe(struct platform_device *pdev) card = &cz_card; cz_card.dev = &pdev->dev; platform_set_drvdata(pdev, card); + mutex_init(&instance_lock); snd_soc_card_set_drvdata(card, machine); ret = devm_snd_soc_register_card(&pdev->dev, &cz_card); if (ret) { -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: AMD: Fix simultaneous playback and capture on different channel
by Akshu Agrawal 10 Sep '18

10 Sep '18

If capture and playback are started on different channel (I2S/BT) there is a possibilty that channel information passed from machine driver is overwritten before the configuration is done in dma driver. Example: 113.597588: cz_max_startup: ---playback sets BT channel 113.597694: cz_dmic1_startup: ---capture sets I2S channel 113.597979: acp_dma_hw_params: ---configures capture for I2S channel 113.598114: acp_dma_hw_params: ---configures playback for I2S channel This is fixed by having 2 separate instance for playback and capture. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 40 +++++++++++++++++++++++++++++------- sound/soc/amd/acp-pcm-dma.c | 8 ++++++-- sound/soc/amd/acp.h | 3 ++- 3 files changed, 41 insertions(+), 10 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 55d7f61..0d97d00 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -133,7 +133,7 @@ static void da7219_clk_disable(void) .mask = 0, }; -static int cz_da7219_startup(struct snd_pcm_substream *substream) +static int cz_da7219_play_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -150,7 +150,28 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->play_i2s_instance = I2S_SP_INSTANCE; + return da7219_clk_enable(substream); +} + +static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + /* + * On this platform for PCM device we support stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); } @@ -177,7 +198,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->play_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -203,7 +224,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_BT_INSTANCE; + machine->cap_i2s_instance = I2S_BT_INSTANCE; return da7219_clk_enable(substream); } @@ -224,7 +245,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - machine->i2s_instance = I2S_SP_INSTANCE; + machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; return da7219_clk_enable(substream); } @@ -234,8 +255,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static const struct snd_soc_ops cz_da7219_play_ops = { + .startup = cz_da7219_play_startup, + .shutdown = cz_da7219_shutdown, +}; + static const struct snd_soc_ops cz_da7219_cap_ops = { - .startup = cz_da7219_startup, + .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, }; @@ -266,7 +292,7 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, - .ops = &cz_da7219_cap_ops, + .ops = &cz_da7219_play_ops, }, { .name = "amd-da7219-cap", diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index b43f0a1..b0e245c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -862,8 +862,12 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, return -EINVAL; if (pinfo) { - rtd->i2s_instance = pinfo->i2s_instance; - rtd->capture_channel = pinfo->capture_channel; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + rtd->i2s_instance = pinfo->play_i2s_instance; + } else { + rtd->i2s_instance = pinfo->cap_i2s_instance; + rtd->capture_channel = pinfo->capture_channel; + } } if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index be3963e..dbbb1a8 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -158,7 +158,8 @@ struct audio_drv_data { * and dma driver */ struct acp_platform_info { - u16 i2s_instance; + u16 play_i2s_instance; + u16 cap_i2s_instance; u16 capture_channel; }; -- 1.9.1

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[alsa-devel] [PATCH] ALSA: emu10k1: fix possible info leak to userspace on SNDRV_EMU10K1_IOCTL_INFO
by Takashi Iwai 10 Sep '18

10 Sep '18

From: Willy Tarreau <w(a)1wt.eu> snd_emu10k1_fx8010_ioctl(SNDRV_EMU10K1_IOCTL_INFO) allocates memory using kmalloc() and partially fills it by calling snd_emu10k1_fx8010_info() before returning the resulting structure to userspace, leaving uninitialized holes. Let's just use kzalloc() here. BugLink: http://blog.infosectcbr.com.au/2018/09/linux-kernel-infoleaks.html Signed-off-by: Willy Tarreau <w(a)1wt.eu> Cc: Jann Horn <jannh(a)google.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/emu10k1/emufx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 90713741c2dc..6ebe817801ea 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -2540,7 +2540,7 @@ static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, un emu->support_tlv = 1; return put_user(SNDRV_EMU10K1_VERSION, (int __user *)argp); case SNDRV_EMU10K1_IOCTL_INFO: - info = kmalloc(sizeof(*info), GFP_KERNEL); + info = kzalloc(sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; snd_emu10k1_fx8010_info(emu, info); -- 2.18.0

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[alsa-devel] RaspberryPi vc4-hdmi softvol for alsamixer
by Andreas Müller 10 Sep '18

10 Sep '18

Hi, This is my first post here and guess what I am looking for support :) Up to now I used audioinjector and hifiberry on RaspberryPi and that worked without issues (from alsa point of view). Now I would like to select vc4hdmi on alsamixer but that complains for missing controls. I think vc4hdmi does not support hard volume setting - like some other devices. So I read here and there and found softvol. To test I modified vc4-hdmi.conf shipped with alsal-iib from: # default with dmix vc4-hdmi.pcm.default { @args [ CARD ] @args.CARD { type string } type asym playback.pcm { type plug slave.pcm { @func concat strings [ "dmix:" $CARD ] } } } to (stolen from PC-Speaker.conf) # default with dmix vc4-hdmi.pcm.default { @args [ CARD ] @args.CARD { type string } type asym playback.pcm { type plug slave.pcm { type softvol control { name "Master Playback Volume" card $CARD } slave.pcm { @func concat strings [ "dmix:" $CARD ] } } } } after reboot nothing changes! I build my images with Openembedded/Yocto and from my experience it is possible that there is some misconfiguration / path issue causes problems. I know the card works with the help of pulseaudio. So now my questions: * Would you expect that my modification of vc4-hdmi.conf adds softvol control to vc4hdmi? * If yes: The only information I have currently: It does not work as expected. So where can I ask alsa for some feedback what went wrong (log-files/dmesg/journal...)? Thanks in advance Andreas

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[alsa-devel] [PATCH] ASoC: dapm: Fix a couple uninitialized ret variables
by Dan Carpenter 10 Sep '18

10 Sep '18

Smatch complains that these variables could be uninitialized. The first one in snd_soc_dai_link_event() is probably a false positive, because probably we know the lists are not empty. I would normally ignore the warning, but GCC complains here as well so I just silenced the warning. The "ret" in snd_soc_dapm_new_dai() does need to be initialized or it leads to a bogus dereference in the caller. Fixes: 3bbf5d34fd4a ("ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9f4edcd19e02..0dcdcc23dcfd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3623,7 +3623,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; unsigned int fmt; - int ret; + int ret = 0; config = rtd->dai_link->params + rtd->params_select; @@ -3938,8 +3938,10 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); - if (IS_ERR(w)) + if (IS_ERR(w)) { + ret = PTR_ERR(w); goto outfree_kcontrol_news; + } w->priv = rtd;

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[alsa-devel] Applied "ASoC: q6core: Use kmemdup to replace kzalloc + memcpy" to the asoc tree
by Mark Brown 10 Sep '18

10 Sep '18

The patch ASoC: q6core: Use kmemdup to replace kzalloc + memcpy 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 18fbe800e6066050ab6ae7751708da04975cdc22 Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Sat, 8 Sep 2018 16:36:19 +0800 Subject: [PATCH] ASoC: q6core: Use kmemdup to replace kzalloc + memcpy kmemdup has implemented the function that kzalloc() + memcpy() will do. and we prefer to use the kmemdup rather than the open coded implementation. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6core.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c index 06f03a5fe9bd..ca1be7305524 100644 --- a/sound/soc/qcom/qdsp6/q6core.c +++ b/sound/soc/qcom/qdsp6/q6core.c @@ -105,12 +105,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) bytes = sizeof(*fwk) + fwk->num_services * sizeof(fwk->svc_api_info[0]); - core->fwk_version = kzalloc(bytes, GFP_ATOMIC); + core->fwk_version = kmemdup(data->payload, bytes, GFP_ATOMIC); if (!core->fwk_version) return -ENOMEM; - memcpy(core->fwk_version, data->payload, bytes); - core->fwk_version_supported = true; core->resp_received = true; @@ -124,12 +122,10 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]); - core->svc_version = kzalloc(len, GFP_ATOMIC); + core->svc_version = kmemdup(data->payload, len, GFP_ATOMIC); if (!core->svc_version) return -ENOMEM; - memcpy(core->svc_version, data->payload, len); - core->get_version_supported = true; core->resp_received = true; -- 2.19.0.rc1

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[alsa-devel] Applied "ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy" to the asoc tree
by Mark Brown 10 Sep '18

10 Sep '18

The patch ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy 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 ca92cc4636fdedf0d7ee88a5e50cd2b85c246a3b Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Sat, 8 Sep 2018 16:36:20 +0800 Subject: [PATCH] ASoC: skl-topology: Use kmemdup to replace kzalloc + memcpy kmemdup has implemented the function that kzalloc() + memcpy() will do. and we prefer to kmemdup rather than the open coded implementation. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 2620d77729c5..52a9915da0f5 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -898,11 +898,10 @@ static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, bc = (struct skl_algo_data *)sb->dobj.private; if (bc->set_params == SKL_PARAM_BIND) { - params = kzalloc(bc->max, GFP_KERNEL); + params = kmemdup(bc->params, bc->max, GFP_KERNEL); if (!params) return -ENOMEM; - memcpy(params, bc->params, bc->max); skl_fill_sink_instance_id(ctx, params, bc->max, mconfig); -- 2.19.0.rc1

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[alsa-devel] [PATCH] ASoC: qdsp6: q6asm-dai: clean up a return
by Dan Carpenter 10 Sep '18

10 Sep '18

Smatch complains that if both "psubstream" and "csubstream" are NULL then "ret" is uninitialized. That probably can't happen, but it's cleaner to just return zero anyway so let's do that. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 9db9a2944ef2..c75fab38905d 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -493,7 +493,7 @@ static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd) } } - return ret; + return 0; } static void q6asm_dai_pcm_free(struct snd_pcm *pcm)

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[alsa-devel] [PATCH] ALSA: fireface: fix memory leak in ff400_switch_fetching_mode()
by Takashi Sakamoto 10 Sep '18

10 Sep '18

An allocated memory forgets to be released. Fixes: 76fdb3a9e13 ('ALSA: fireface: add support for Fireface 400') Cc: <stable(a)vger.kernel.org> # 4.12+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireface/ff-protocol-ff400.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/firewire/fireface/ff-protocol-ff400.c b/sound/firewire/fireface/ff-protocol-ff400.c index ad7a0a32557d..64c3cb0fb926 100644 --- a/sound/firewire/fireface/ff-protocol-ff400.c +++ b/sound/firewire/fireface/ff-protocol-ff400.c @@ -146,6 +146,7 @@ static int ff400_switch_fetching_mode(struct snd_ff *ff, bool enable) { __le32 *reg; int i; + int err; reg = kcalloc(18, sizeof(__le32), GFP_KERNEL); if (reg == NULL) @@ -163,9 +164,11 @@ static int ff400_switch_fetching_mode(struct snd_ff *ff, bool enable) reg[i] = cpu_to_le32(0x00000001); } - return snd_fw_transaction(ff->unit, TCODE_WRITE_BLOCK_REQUEST, - FF400_FETCH_PCM_FRAMES, reg, - sizeof(__le32) * 18, 0); + err = snd_fw_transaction(ff->unit, TCODE_WRITE_BLOCK_REQUEST, + FF400_FETCH_PCM_FRAMES, reg, + sizeof(__le32) * 18, 0); + kfree(reg); + return err; } static void ff400_dump_sync_status(struct snd_ff *ff, -- 2.17.1

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[alsa-devel] [PATCH] ALSA: bebob: use address returned by kmalloc() instead of kernel stack for streaming DMA mapping
by Takashi Sakamoto 10 Sep '18

10 Sep '18

When executing 'fw_run_transaction()' with 'TCODE_WRITE_BLOCK_REQUEST', an address of 'payload' argument is used for streaming DMA mapping by 'firewire_ohci' module if 'size' argument is larger than 8 byte. Although in this case the address should not be on kernel stack, current implementation of ALSA bebob driver uses data in kernel stack for a cue to boot M-Audio devices. This often brings unexpected result, especially for a case of CONFIG_VMAP_STACK=y. This commit fixes the bug. Reference: https://bugzilla.kernel.org/show_bug.cgi?id=201021 Reference: https://forum.manjaro.org/t/firewire-m-audio-410-driver-wont-load-firmware/… Fixes: a2b2a7798fb6('ALSA: bebob: Send a cue to load firmware for M-Audio Firewire series') Cc: <stable(a)vger.kernel.org> # v3.16+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_maudio.c | 24 ++++++++++++++---------- 1 file changed, 14 insertions(+), 10 deletions(-) diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index bd55620c6a47..0c5a4cbb99ba 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -96,17 +96,13 @@ int snd_bebob_maudio_load_firmware(struct fw_unit *unit) struct fw_device *device = fw_parent_device(unit); int err, rcode; u64 date; - __le32 cues[3] = { - cpu_to_le32(MAUDIO_BOOTLOADER_CUE1), - cpu_to_le32(MAUDIO_BOOTLOADER_CUE2), - cpu_to_le32(MAUDIO_BOOTLOADER_CUE3) - }; + __le32 *cues; /* check date of software used to build */ err = snd_bebob_read_block(unit, INFO_OFFSET_SW_DATE, &date, sizeof(u64)); if (err < 0) - goto end; + return err; /* * firmware version 5058 or later has date later than "20070401", but * 'date' is not null-terminated. @@ -114,20 +110,28 @@ int snd_bebob_maudio_load_firmware(struct fw_unit *unit) if (date < 0x3230303730343031LL) { dev_err(&unit->device, "Use firmware version 5058 or later\n"); - err = -ENOSYS; - goto end; + return -ENXIO; } + cues = kmalloc_array(3, sizeof(*cues), GFP_KERNEL); + if (!cues) + return -ENOMEM; + + cues[0] = cpu_to_le32(MAUDIO_BOOTLOADER_CUE1); + cues[1] = cpu_to_le32(MAUDIO_BOOTLOADER_CUE2); + cues[2] = cpu_to_le32(MAUDIO_BOOTLOADER_CUE3); + rcode = fw_run_transaction(device->card, TCODE_WRITE_BLOCK_REQUEST, device->node_id, device->generation, device->max_speed, BEBOB_ADDR_REG_REQ, - cues, sizeof(cues)); + cues, 3 * sizeof(*cues)); + kfree(cues); if (rcode != RCODE_COMPLETE) { dev_err(&unit->device, "Failed to send a cue to load firmware\n"); err = -EIO; } -end: + return err; } -- 2.17.1

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