August 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: rockchip: implement system suspend/resume for i2s
by Sugar Zhang 01 Sep '16

01 Sep '16

restore hw registers after power loss during a suspend/resume cycle. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/rockchip/rockchip_i2s.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 574c6af..53970ac 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -614,9 +614,35 @@ static const struct of_device_id rockchip_i2s_match[] = { {}, }; +#ifdef CONFIG_PM_SLEEP +static int rockchip_i2s_suspend(struct device *dev) +{ + struct rk_i2s_dev *i2s = dev_get_drvdata(dev); + + regcache_mark_dirty(i2s->regmap); + + return 0; +} + +static int rockchip_i2s_resume(struct device *dev) +{ + struct rk_i2s_dev *i2s = dev_get_drvdata(dev); + int ret; + + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; + ret = regcache_sync(i2s->regmap); + pm_runtime_put(dev); + + return ret; +} +#endif + static const struct dev_pm_ops rockchip_i2s_pm_ops = { SET_RUNTIME_PM_OPS(i2s_runtime_suspend, i2s_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(rockchip_i2s_suspend, rockchip_i2s_resume) }; static struct platform_driver rockchip_i2s_driver = { -- 1.9.1

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[alsa-devel] [PATCH] ASoC: tlv320dac33: Remove deprecated create_singlethread_workqueue
by Bhaktipriya Shridhar 01 Sep '16

01 Sep '16

The workqueue "dac33_wq" queues a single work item &dac33->work and hence doesn't require ordering. Also, it is not being used on a memory reclaim path. Hence, it has been converted to use system_wq. The work item has been flushed in dac33_soc_remove to ensure that there are no pending tasks while disconnecting the driver. The workqueue "post_msg_wq" queues a single work item &drv->ipc_post_msg_wq and hence doesn't require ordering. Also, it is not being used on a memory reclaim path. Hence, it has been converted to use system_wq. The work item has been flushed in sst_context_cleanup to ensure that there are no pending tasks while disconnecting the driver. System workqueues have been able to handle high level of concurrency for a long time now and hence it's not required to have a singlethreaded workqueue just to gain concurrency. Unlike a dedicated per-cpu workqueue created with create_singlethread_workqueue(), system_wq allows multiple work items to overlap executions even on the same CPU; however, a per-cpu workqueue doesn't have any CPU locality or global ordering guarantee unless the target CPU is explicitly specified and thus the increase of local concurrency shouldn't make any difference. Signed-off-by: Bhaktipriya Shridhar <bhaktipriya96(a)gmail.com> --- sound/soc/codecs/tlv320dac33.c | 17 ++++------------- sound/soc/intel/atom/sst/sst.c | 14 +++++--------- 2 files changed, 9 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f7a6ce7..6822ac1 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -90,7 +90,6 @@ static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { struct tlv320dac33_priv { struct mutex mutex; - struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec *codec; struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; @@ -771,7 +770,7 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev) /* Do not schedule the workqueue in Mode7 */ if (dac33->fifo_mode != DAC33_FIFO_MODE7) - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); return IRQ_HANDLED; } @@ -1127,7 +1126,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (dac33->fifo_mode) { dac33->state = DAC33_PREFILL; - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); } break; case SNDRV_PCM_TRIGGER_STOP: @@ -1135,7 +1134,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (dac33->fifo_mode) { dac33->state = DAC33_FLUSH; - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); } break; default: @@ -1410,14 +1409,6 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) dac33->irq = -1; } if (dac33->irq != -1) { - /* Setup work queue */ - dac33->dac33_wq = - create_singlethread_workqueue("tlv320dac33"); - if (dac33->dac33_wq == NULL) { - free_irq(dac33->irq, codec); - return -ENOMEM; - } - INIT_WORK(&dac33->work, dac33_work); } } @@ -1437,7 +1428,7 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) if (dac33->irq >= 0) { free_irq(dac33->irq, dac33->codec); - destroy_workqueue(dac33->dac33_wq); + flush_work(&dac33->work); } return 0; } diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index a4b458e..8cfd0cf 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -76,7 +76,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) spin_unlock(&drv->ipc_spin_lock); /* we can send more messages to DSP so trigger work */ - queue_work(drv->post_msg_wq, &drv->ipc_post_msg_wq); + schedule_work(&drv->ipc_post_msg_wq); retval = IRQ_HANDLED; } @@ -212,10 +212,6 @@ static int sst_workqueue_init(struct intel_sst_drv *ctx) INIT_WORK(&ctx->ipc_post_msg_wq, sst_process_pending_msg); init_waitqueue_head(&ctx->wait_queue); - ctx->post_msg_wq = - create_singlethread_workqueue("sst_post_msg_wq"); - if (!ctx->post_msg_wq) - return -EBUSY; return 0; } @@ -318,7 +314,6 @@ int sst_context_init(struct intel_sst_drv *ctx) return 0; do_free_mem: - destroy_workqueue(ctx->post_msg_wq); return ret; } EXPORT_SYMBOL_GPL(sst_context_init); @@ -330,7 +325,7 @@ void sst_context_cleanup(struct intel_sst_drv *ctx) sst_unregister(ctx->dev); sst_set_fw_state_locked(ctx, SST_SHUTDOWN); flush_scheduled_work(); - destroy_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); pm_qos_remove_request(ctx->qos); kfree(ctx->fw_sg_list.src); kfree(ctx->fw_sg_list.dst); @@ -414,7 +409,7 @@ static int intel_sst_runtime_suspend(struct device *dev) sst_set_fw_state_locked(ctx, SST_RESET); synchronize_irq(ctx->irq_num); - flush_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); ctx->ops->reset(ctx); /* save the shim registers because PMC doesn't save state */ @@ -445,8 +440,9 @@ static int intel_sst_suspend(struct device *dev) return -EBUSY; } } + synchronize_irq(ctx->irq_num); - flush_workqueue(ctx->post_msg_wq); + flush_work(ctx->ipc_post_msg_wq); /* Move the SST state to Reset */ sst_set_fw_state_locked(ctx, SST_RESET); -- 2.1.4

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[alsa-devel] Setting some clocks back to DUMMY fixes spdif output on imx6q wandboard rev B1
by Xavi Drudis Ferran 31 Aug '16

31 Aug '16

Hello. Sorry for my unfamiliarity with the kernel community. I'm afraid I'll be getting the procedure wrong... I was using linux-libre-3.19 (implies no working sdma) with a wandboard quad (Freescale imx6q). Spidf output worked fine. When I upgraded to linux-libre-4.7 spdif output was supressed without any error (precisely, with the same errors about sdma that 3.19 gave). I saw someone else reporting the same elsewhere with linux-4.4 https://forum.digikey.com/thread/34240 (but I don't have a login there) This patch fixes it for me and sound works again on spdif. But I don't know if it can break (or fix?) something for some other boards or kernels or cases... I hardly know what I'm doing. The commits that might have caused the problem for me might be commit 833f2cbf7091099baee28136dc68678e974c0ac5 Author: Shengjiu Wang <shengjiu.wang(a)freescale.com> Date: Sat Oct 10 18:15:07 2015 +0800 ARM: dts: imx6: change the core clock of spdif The correct core clock of spdif is SPDIF_GCLK, which is added to clock tree. So the dts also need to be updated. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> Signed-off-by: Shawn Guo <shawnguo(a)kernel.org> (the commit changed more clocks than SPDIF_GCLK) It might have to do with vendor commit http://git.freescale.com/git/cgit.cgi/imx/linux-2.6-imx.git/commit/arch/arm… The issue is fixed for me with this patch but I'm not sure what's the best way to help fix any issue someone else may have or what other info or test you might need. Any guidance welcome. Thank you for linux.

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Re: [alsa-devel] [PATCH] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by Tony Lindgren 31 Aug '16

31 Aug '16

* kbuild test robot <lkp(a)intel.com> [160829 21:31]: > vim +/dev_pm_qos_request_active +1108 sound/soc/omap/mcbsp.c > > 1102 clk_put(mcbsp->fclk); > 1103 return ret; > 1104 } > 1105 > 1106 void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) > 1107 { > > 1108 if (dev_pm_qos_request_active(&mcbsp->pm_qos_req)) > 1109 pm_qos_remove_request(&mcbsp->pm_qos_req); > 1110 > 1111 if (mcbsp->pdata->buffer_size) Oops sorry about that. Looks like I completely missed that warning after a copy-paste adding of the check to omap_mcbsp_cleanup. Looks like implementing dev_pm_qos needs more work as it needs the set_latency_tolerance implemented somewhere. Updated patch below, Mark let me know if you need an incremental patch to fix the warning. Regards, Tony 8< ----------------- From: Tony Lindgren <tony(a)atomide.com> Date: Mon, 29 Aug 2016 11:27:46 -0700 Subject: [PATCHv2] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches We can get audio errors if hitting deeper idle states on omaps: [alsa.c:230] error: Fatal problem with alsa output, error -5. [audio.c:614] error: Error in writing audio (Input/output error?)! This seems to happen with off mode idle enabled as power for the whole SoC may get cut off between filling the McBSP fifo using DMA. While active DMA blocks deeper idle states in hardware, McBSP activity does not seem to do so. Let's fix the issue by adding PM QoS latency requirement for McBSP. Based on my experiments a working latency value is somewhere around 35 ms with 40 ms breaking. So let's use a safer value of 30 ms. Note that this is different from snd_pcm_hw_constraint_step() as that can configure things based on the buffer and period size. However, that does not help to block idle between the fifo fills. Note that in theory some omaps are capable of playing audio while hitting deeper idle states. If somebody needs that and gets it working, we can set the latency requirements accordingly. Signed-off-by: Tony Lindgren <tony(a)atomide.com> diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -25,6 +25,7 @@ #include <linux/io.h> #include <linux/slab.h> #include <linux/pm_runtime.h> +#include <linux/pm_qos.h> #include <linux/platform_data/asoc-ti-mcbsp.h> @@ -643,6 +644,10 @@ void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) int enable_srg = 0; u16 w; + /* Prevent omap hardware from hitting off between fifo fills */ + pm_qos_add_request(&mcbsp->pm_qos_req, PM_QOS_CPU_DMA_LATENCY, + 30 * 1000); + if (mcbsp->st_data) omap_st_start(mcbsp); @@ -731,6 +736,8 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) if (mcbsp->st_data) omap_st_stop(mcbsp); + + pm_qos_remove_request(&mcbsp->pm_qos_req); } int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) @@ -1098,6 +1105,9 @@ err_thres: void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) { + if (pm_qos_request_active(&mcbsp->pm_qos_req)) + pm_qos_remove_request(&mcbsp->pm_qos_req); + if (mcbsp->pdata->buffer_size) sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -325,6 +325,8 @@ struct omap_mcbsp { unsigned int in_freq; int clk_div; int wlen; + + struct pm_qos_request pm_qos_req; }; void omap_mcbsp_config(struct omap_mcbsp *mcbsp,

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[alsa-devel] [PATCH v3] ASoC: Remove deprecated create_singlethread_workqueue
by Bhaktipriya Shridhar 31 Aug '16

31 Aug '16

The workqueue "dac33_wq" queues a single work item &dac33->work and hence doesn't require ordering. Also, it is not being used on a memory reclaim path. Hence, it has been converted to use system_wq. The work item has been flushed in dac33_soc_remove to ensure that there are no pending tasks while disconnecting the driver. The workqueue "post_msg_wq" queues a single work item &drv->ipc_post_msg_wq and hence doesn't require ordering. Also, it is not being used on a memory reclaim path. Hence, it has been converted to use system_wq. The work item has been flushed in sst_context_cleanup to ensure that there are no pending tasks while disconnecting the driver. System workqueues have been able to handle high level of concurrency for a long time now and hence it's not required to have a singlethreaded workqueue just to gain concurrency. Unlike a dedicated per-cpu workqueue created with create_singlethread_workqueue(), system_wq allows multiple work items to overlap executions even on the same CPU; however, a per-cpu workqueue doesn't have any CPU locality or global ordering guarantee unless the target CPU is explicitly specified and thus the increase of local concurrency shouldn't make any difference. Signed-off-by: Bhaktipriya Shridhar <bhaktipriya96(a)gmail.com> --- Changes in v3: -Added missing '&' sound/soc/codecs/tlv320dac33.c | 17 ++++------------- sound/soc/intel/atom/sst/sst.c | 14 +++++--------- 2 files changed, 9 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f7a6ce7..6822ac1 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -90,7 +90,6 @@ static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { struct tlv320dac33_priv { struct mutex mutex; - struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec *codec; struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; @@ -771,7 +770,7 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev) /* Do not schedule the workqueue in Mode7 */ if (dac33->fifo_mode != DAC33_FIFO_MODE7) - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); return IRQ_HANDLED; } @@ -1127,7 +1126,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (dac33->fifo_mode) { dac33->state = DAC33_PREFILL; - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); } break; case SNDRV_PCM_TRIGGER_STOP: @@ -1135,7 +1134,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (dac33->fifo_mode) { dac33->state = DAC33_FLUSH; - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); } break; default: @@ -1410,14 +1409,6 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) dac33->irq = -1; } if (dac33->irq != -1) { - /* Setup work queue */ - dac33->dac33_wq = - create_singlethread_workqueue("tlv320dac33"); - if (dac33->dac33_wq == NULL) { - free_irq(dac33->irq, codec); - return -ENOMEM; - } - INIT_WORK(&dac33->work, dac33_work); } } @@ -1437,7 +1428,7 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) if (dac33->irq >= 0) { free_irq(dac33->irq, dac33->codec); - destroy_workqueue(dac33->dac33_wq); + flush_work(&dac33->work); } return 0; } diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index a4b458e..1454603 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -76,7 +76,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) spin_unlock(&drv->ipc_spin_lock); /* we can send more messages to DSP so trigger work */ - queue_work(drv->post_msg_wq, &drv->ipc_post_msg_wq); + schedule_work(&drv->ipc_post_msg_wq); retval = IRQ_HANDLED; } @@ -212,10 +212,6 @@ static int sst_workqueue_init(struct intel_sst_drv *ctx) INIT_WORK(&ctx->ipc_post_msg_wq, sst_process_pending_msg); init_waitqueue_head(&ctx->wait_queue); - ctx->post_msg_wq = - create_singlethread_workqueue("sst_post_msg_wq"); - if (!ctx->post_msg_wq) - return -EBUSY; return 0; } @@ -318,7 +314,6 @@ int sst_context_init(struct intel_sst_drv *ctx) return 0; do_free_mem: - destroy_workqueue(ctx->post_msg_wq); return ret; } EXPORT_SYMBOL_GPL(sst_context_init); @@ -330,7 +325,7 @@ void sst_context_cleanup(struct intel_sst_drv *ctx) sst_unregister(ctx->dev); sst_set_fw_state_locked(ctx, SST_SHUTDOWN); flush_scheduled_work(); - destroy_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); pm_qos_remove_request(ctx->qos); kfree(ctx->fw_sg_list.src); kfree(ctx->fw_sg_list.dst); @@ -414,7 +409,7 @@ static int intel_sst_runtime_suspend(struct device *dev) sst_set_fw_state_locked(ctx, SST_RESET); synchronize_irq(ctx->irq_num); - flush_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); ctx->ops->reset(ctx); /* save the shim registers because PMC doesn't save state */ @@ -445,8 +440,9 @@ static int intel_sst_suspend(struct device *dev) return -EBUSY; } } + synchronize_irq(ctx->irq_num); - flush_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); /* Move the SST state to Reset */ sst_set_fw_state_locked(ctx, SST_RESET); -- 2.1.4

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[alsa-devel] [PATCH v4] ALSA: fireworks: accessing to user space outside spinlock
by Takashi Sakamoto 31 Aug '16

31 Aug '16

Hi, This patch is a revised version of a part of my previous patchset: [alsa-devel] [PATCH v3 0/2] ALSA: fireworks/firewire-tascam: accessing to user space outside spinlock http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/112366.html In ALSA fireworks driver, current implementation for hwdep interface have issues for page fault handling. For example, when using single core processor, after executing lock_page() in page fault handler and call task scheduler, then the context never regain CPU again, because any interrupts are disabled by these drivers and the context can't catch waiting event. This patchset fixes the issues. The issues were reported by Vaishali Thakkar, with a help of coccinelle. Thanks a lot for her care to these minor drivers. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111887.html Changes in v4: * Give up counting the number of queued response messages to simplify ring-buffer management. Instead, comparing pull/push pointers is used consistently in any parts. * Assign current state to automatic variables in critical section of blocking processing part, then utilize it for later processing. Changes in v3: * Add a workaround against a case that several tasks are going to read from one response buffer. Changes in v2: * improve designated initialization for automatic variable of union type Takashi Sakamoto (1): ALSA: fireworks: accessing to user space outside spinlock sound/firewire/fireworks/fireworks.h | 1 - sound/firewire/fireworks/fireworks_hwdep.c | 71 +++++++++++++++++------- sound/firewire/fireworks/fireworks_proc.c | 4 +- sound/firewire/fireworks/fireworks_transaction.c | 5 +- 4 files changed, 56 insertions(+), 25 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v3 0/2] ALSA: fireworks/firewire-tascam: accessing to user space outside spinlock
by Takashi Sakamoto 31 Aug '16

31 Aug '16

Hi, This patchset updates my previous one: [alsa-devel] [PATCH v2 0/2] ALSA: fireworks/firewire-tascam: accessing to user space outside spinlock http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/112236.html In ALSA fireworks driver and firewire-tascam driver, current implementation for hwdep interface have issues for page fault handling. For example, when using single core processor, after executing lock_page() in page fault handler and call task scheduler, then the context never regain CPU again, because any interrupts are disabled by these drivers and the context can't catch waiting event. This patchset fixes the issues. The issues were reported by Vaishali Thakkar, with a help of coccinelle. Thanks a lot for her care to these minor drivers. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111887.html Changes in v3: * Add a workaround against a case that several tasks are going to read from one response buffer. Changes in v2: * improve designated initialization for automatic variable of union type Takashi Sakamoto (2): ALSA: firewire-tascam: accessing to user space outside spinlock ALSA: fireworks: accessing to user space outside spinlock sound/firewire/fireworks/fireworks_hwdep.c | 52 ++++++++++++++++++------ sound/firewire/fireworks/fireworks_transaction.c | 4 +- sound/firewire/tascam/tascam-hwdep.c | 33 +++++---------- 3 files changed, 53 insertions(+), 36 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2 0/3] drm/i2c: tda998x ASoC hdmi-codec support + BeagleBoneBlack audio support
by Jyri Sarha 31 Aug '16

31 Aug '16

Changes since first version [1] of this series: - "drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata" - Change audio in tda998x pdata to audio_params to match the naming in private data struct - Skip IEC958_AES2 byte when writing status bytes to registers - Turn AFMT-macros to enum - include linux/hdmi.h in drm/i2c/tda998x.h - "drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding" - Fix macronaming naming in dt-bindings/display/tda998x.h - Turn AFMT-macros to enum - Add locking to protect access to audio registers from ASoC thread - "dts/am335x-boneblack: SQUASH" - Use corrected macros from dt-bindings/display/tda998x.h [1] https://lists.freedesktop.org/archives/dri-devel/2016-August/114491.html Jyri Sarha (3): drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 18 ++ arch/arm/boot/dts/am335x-boneblack.dts | 71 ++++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 297 ++++++++++++++++++--- include/drm/i2c/tda998x.h | 29 +- include/dt-bindings/display/tda998x.h | 7 + 6 files changed, 368 insertions(+), 55 deletions(-) create mode 100644 include/dt-bindings/display/tda998x.h -- 1.9.1

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[alsa-devel] [PATCH v2] ALSA: seq: initialize whole fields of automatic variable with union type
by Takashi Sakamoto 31 Aug '16

31 Aug '16

Currently, automatic variable of 'union ioctl_arg' type is initialized by designated initialization. Although, the actual effect is interpretation of early element of int type and initialization of 'int pversion'. Therefore the first field corresponding to int type is initialized to zero. This is against my expectation to initialize whole fields. This commit uses memset() to initialize the variable, instead of designated initialization. Fixes: 04a56dd8ed0d ('ALSA: seq: change ioctl command operation to get data in kernel space') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/seq/seq_clientmgr.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 286394b..811b95b 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2084,7 +2084,7 @@ static long snd_seq_ioctl(struct file *file, unsigned int cmd, { struct snd_seq_client *client = file->private_data; /* To use kernel stack for ioctl data. */ - union ioctl_arg { + union { int pversion; int client_id; struct snd_seq_system_info system_info; @@ -2100,7 +2100,7 @@ static long snd_seq_ioctl(struct file *file, unsigned int cmd, struct snd_seq_client_pool client_pool; struct snd_seq_remove_events remove_events; struct snd_seq_query_subs query_subs; - } buf = {0}; + } buf; const struct ioctl_handler *handler; unsigned long size; int err; @@ -2114,6 +2114,9 @@ static long snd_seq_ioctl(struct file *file, unsigned int cmd, } if (handler->cmd == 0) return -ENOTTY; + + memset(&buf, 0, sizeof(buf)); + /* * All of ioctl commands for ALSA sequencer get an argument of size * within 13 bits. We can safely pick up the size from the command. -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: arizona: Don't change the FLLn_GAIN before entering FREERUN
by Charles Keepax 31 Aug '16

31 Aug '16

From: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> When reclocking an active FLL, to ensure a clean transition, do not change the gain setting until we have entered free run. Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ecfdbfc..0fc3b4b 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2218,11 +2218,11 @@ static int arizona_enable_fll(struct arizona_fll *fll) if (already_enabled) { /* Facilitate smooth refclk across the transition */ - regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9, - ARIZONA_FLL1_GAIN_MASK, 0); regmap_update_bits(fll->arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); udelay(32); + regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9, + ARIZONA_FLL1_GAIN_MASK, 0); } /* -- 2.1.4

2 1

[alsa-devel] [PATCH v6 1/2] mfd: Add DT bindings documentation for Samsung Exynos LPASS
by Sylwester Nawrocki 31 Aug '16

31 Aug '16

This patch adds documentation of the DT bindings for the Samsung Exynos SoC Low Power Audio Subsystem. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes since v5: - none. Changes since v4: - text reformatting, added VIC to the list of sub-devices, relative paths used for the reference DT binding documentation files, added example dts. Changes since v3: - moved to Documentation/devicetree/bindings/mfd, - added #address-cells, #size-cells, ranges as required properties, - added description of the IP block subnodes and links to their binding's documentation. Changes since v2: - none. --- .../bindings/mfd/samsung,exynos5433-lpass.txt | 70 ++++++++++++++++++++++ 1 file changed, 70 insertions(+) create mode 100644 Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt diff --git a/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt b/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt new file mode 100644 index 0000000..c110e11 --- /dev/null +++ b/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt @@ -0,0 +1,70 @@ +Samsung Exynos SoC Low Power Audio Subsystem (LPASS) + +Required properties: + + - compatible : "samsung,exynos5433-lpass" + - reg : should contain the LPASS top SFR region location + and size + - samsung,pmu-syscon : the phandle to the Power Management Unit node + - #address-cells : should be 1 + - #size-cells : should be 1 + - ranges : must be present + +Each IP block of the Low Power Audio Subsystem should be specified as +an optional sub-node. For "samsung,exynos5433-lpass" compatible this includes: +UART, SLIMBUS, PCM, I2S, DMAC, Timers 0...4, VIC, WDT 0...1 devices. + +Bindings of the sub-nodes are described in: + ../serial/samsung_uart.txt + ../sound/samsung-i2s.txt + ../dma/arm-pl330.txt + + +Example: + +audio-subsystem { + compatible = "samsung,exynos5433-lpass"; + reg = <0x11400000 0x100>, <0x11500000 0x08>; + samsung,pmu-syscon = <&pmu_system_controller>; + #address-cells = <1>; + #size-cells = <1>; + ranges; + + adma: adma@11420000 { + compatible = "arm,pl330", "arm,primecell"; + reg = <0x11420000 0x1000>; + interrupts = <0 73 0>; + clocks = <&cmu_aud CLK_ACLK_DMAC>; + clock-names = "apb_pclk"; + #dma-cells = <1>; + #dma-channels = <8>; + #dma-requests = <32>; + }; + + i2s0: i2s0@11440000 { + compatible = "samsung,exynos7-i2s"; + reg = <0x11440000 0x100>; + dmas = <&adma 0 &adma 2>; + dma-names = "tx", "rx"; + interrupts = <0 70 0>; + clocks = <&cmu_aud CLK_PCLK_AUD_I2S>, + <&cmu_aud CLK_SCLK_AUD_I2S>, + <&cmu_aud CLK_SCLK_I2S_BCLK>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + status = "disabled"; + }; + + serial_3: serial@11460000 { + compatible = "samsung,exynos5433-uart"; + reg = <0x11460000 0x100>; + interrupts = <0 67 0>; + clocks = <&cmu_aud CLK_PCLK_AUD_UART>, + <&cmu_aud CLK_SCLK_AUD_UART>; + clock-names = "uart", "clk_uart_baud0"; + pinctrl-names = "default"; + pinctrl-0 = <&uart_aud_bus>; + status = "disabled"; + }; + }; -- 1.9.1

3 4

[alsa-devel] [PATCH] ASoC: Intel: Atom: Fix message handling during drop stream
by Harsha Priya 31 Aug '16

31 Aug '16

If a stream is being dropped, the period_elapsed received then after need not be processed. Processing of the period_elapsed message causes a time out in drop stream response processing. This patch adds a condition to skip period_elapsed message processing if the stream is in INIT state. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Naveen M <naveen.m(a)intel.com> --- sound/soc/intel/atom/sst/sst_ipc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 8afa6fe..bfc8899 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -267,6 +267,9 @@ static void process_fw_async_msg(struct intel_sst_drv *sst_drv_ctx, "Period elapsed rcvd for pipe id 0x%x\n", pipe_id); stream = &sst_drv_ctx->streams[str_id]; + /* If stream is dropped, skip processing this message*/ + if (stream->status == STREAM_INIT) + break; if (stream->period_elapsed) stream->period_elapsed(stream->pcm_substream); if (stream->compr_cb) -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: Remove deprecated create_singlethread_workqueue
by Bhaktipriya Shridhar 30 Aug '16

30 Aug '16

The workqueue "dac33_wq" queues a single work item &dac33->work and hence doesn't require ordering. Also, it is not being used on a memory reclaim path. Hence, it has been converted to use system_wq. The work item has been flushed in dac33_soc_remove to ensure that there are no pending tasks while disconnecting the driver. The workqueue "post_msg_wq" queues a single work item &drv->ipc_post_msg_wq and hence doesn't require ordering. Also, it is not being used on a memory reclaim path. Hence, it has been converted to use system_wq. The work item has been flushed in sst_context_cleanup to ensure that there are no pending tasks while disconnecting the driver. System workqueues have been able to handle high level of concurrency for a long time now and hence it's not required to have a singlethreaded workqueue just to gain concurrency. Unlike a dedicated per-cpu workqueue created with create_singlethread_workqueue(), system_wq allows multiple work items to overlap executions even on the same CPU; however, a per-cpu workqueue doesn't have any CPU locality or global ordering guarantee unless the target CPU is explicitly specified and thus the increase of local concurrency shouldn't make any difference. Signed-off-by: Bhaktipriya Shridhar <bhaktipriya96(a)gmail.com> --- Changes in v2: - Fixed heading to indicate that changes were made in sound/soc sound/soc/codecs/tlv320dac33.c | 17 ++++------------- sound/soc/intel/atom/sst/sst.c | 14 +++++--------- 2 files changed, 9 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f7a6ce7..6822ac1 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -90,7 +90,6 @@ static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = { struct tlv320dac33_priv { struct mutex mutex; - struct workqueue_struct *dac33_wq; struct work_struct work; struct snd_soc_codec *codec; struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES]; @@ -771,7 +770,7 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev) /* Do not schedule the workqueue in Mode7 */ if (dac33->fifo_mode != DAC33_FIFO_MODE7) - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); return IRQ_HANDLED; } @@ -1127,7 +1126,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (dac33->fifo_mode) { dac33->state = DAC33_PREFILL; - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); } break; case SNDRV_PCM_TRIGGER_STOP: @@ -1135,7 +1134,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (dac33->fifo_mode) { dac33->state = DAC33_FLUSH; - queue_work(dac33->dac33_wq, &dac33->work); + schedule_work(&dac33->work); } break; default: @@ -1410,14 +1409,6 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) dac33->irq = -1; } if (dac33->irq != -1) { - /* Setup work queue */ - dac33->dac33_wq = - create_singlethread_workqueue("tlv320dac33"); - if (dac33->dac33_wq == NULL) { - free_irq(dac33->irq, codec); - return -ENOMEM; - } - INIT_WORK(&dac33->work, dac33_work); } } @@ -1437,7 +1428,7 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) if (dac33->irq >= 0) { free_irq(dac33->irq, dac33->codec); - destroy_workqueue(dac33->dac33_wq); + flush_work(&dac33->work); } return 0; } diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index a4b458e..8cfd0cf 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -76,7 +76,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) spin_unlock(&drv->ipc_spin_lock); /* we can send more messages to DSP so trigger work */ - queue_work(drv->post_msg_wq, &drv->ipc_post_msg_wq); + schedule_work(&drv->ipc_post_msg_wq); retval = IRQ_HANDLED; } @@ -212,10 +212,6 @@ static int sst_workqueue_init(struct intel_sst_drv *ctx) INIT_WORK(&ctx->ipc_post_msg_wq, sst_process_pending_msg); init_waitqueue_head(&ctx->wait_queue); - ctx->post_msg_wq = - create_singlethread_workqueue("sst_post_msg_wq"); - if (!ctx->post_msg_wq) - return -EBUSY; return 0; } @@ -318,7 +314,6 @@ int sst_context_init(struct intel_sst_drv *ctx) return 0; do_free_mem: - destroy_workqueue(ctx->post_msg_wq); return ret; } EXPORT_SYMBOL_GPL(sst_context_init); @@ -330,7 +325,7 @@ void sst_context_cleanup(struct intel_sst_drv *ctx) sst_unregister(ctx->dev); sst_set_fw_state_locked(ctx, SST_SHUTDOWN); flush_scheduled_work(); - destroy_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); pm_qos_remove_request(ctx->qos); kfree(ctx->fw_sg_list.src); kfree(ctx->fw_sg_list.dst); @@ -414,7 +409,7 @@ static int intel_sst_runtime_suspend(struct device *dev) sst_set_fw_state_locked(ctx, SST_RESET); synchronize_irq(ctx->irq_num); - flush_workqueue(ctx->post_msg_wq); + flush_work(&ctx->ipc_post_msg_wq); ctx->ops->reset(ctx); /* save the shim registers because PMC doesn't save state */ @@ -445,8 +440,9 @@ static int intel_sst_suspend(struct device *dev) return -EBUSY; } } + synchronize_irq(ctx->irq_num); - flush_workqueue(ctx->post_msg_wq); + flush_work(ctx->ipc_post_msg_wq); /* Move the SST state to Reset */ sst_set_fw_state_locked(ctx, SST_RESET); -- 2.1.4

1 0

[alsa-devel] [PATCH v9 0/2] sun4i-codec: Add Line-In, FM-In, Mic 2, Capture Source, Differential Line-In
by Danny Milosavljevic 30 Aug '16

30 Aug '16

This patch adds support for some mixer controls: * Line-In * FM-In * Mic 2 * Capture Source * Differential Line-In v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (2): ASoC: sun4i-codec: Distinguish sun4i from sun7i Add mixer controls: Line-In, FM-In, Mic 2, Capture Source, Differential Line-In. sound/soc/sunxi/sun4i-codec.c | 278 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 259 insertions(+), 19 deletions(-)

3 6

Re: [alsa-devel] Questions about thinkpad_helper
by Takashi Iwai 30 Aug '16

30 Aug '16

[Changed Cc to proper alsa-devel ML] On Tue, 30 Aug 2016 12:02:56 +0200, Dennis Wassenberg wrote: > > Hi all, > > I wrote a driver which handles the function / special keys for > ThinKeyboard of Lenovo ThinkPad X1 Tablet 2016 including LED control. > This Keyboard is connected via USB to the Tablet. I added the driver to > hid-lenovo.c file because there are HID drivers included for different > Lenovo devices. > > For example this driver is able to handle LED control for mute and mic > mute keys. Now I want to make the other direction working. If someone > sets mute or mic mute at amixer, I want to control the mute and mic mute > LEDs. At other Thinkpad devices this is done by thinkpad_helper.c. The > thinkpad_helper let the thinkpad_apci driver to control the LEDs. > > Now I think about how to make this working for Lenovo Thinkpad X1 Tablet > 2016. There it is not possible to use the thinkpad_acpi driver. Instead > we have to use the hid-lenovo driver to make this working. > > Should I add this to thinkpad_helper because this is a thinkpad device > too or should I make a new helper? Can I use the same API than > thinkpad_acpi (static int (*led_set_func)(int, bool)) or should I make a > new one (e.g. in combination with LED triggers)? > > For the first shot I made something at > https://github.com/dwassenberg/linux/tree/development/hid-rmi > (dc05de0add955775434d18d4fe096ea78d1e63bd..5a88c66f2ae8db04147bb75ecf84a4f52e0da1df) > this shot is based on using the thinkpad_helper and the same API. Please > tell me your opinion. > > Maybe one additional remark. In case of using the USB connected > ThinKeyboard there is one additional effort to make this working at the > initial state if the Tablet is finished booting and after that the > ThinKeyboard is connected. So either there is an additional trigger > needed which tells the helper to publish the current mute and mic mute > state when the keyboard is connected. Or an other approach is to update > the hid-lenovo driver even when this keyboard is not connected such that > the hid-lenovo driver always knows the current state and is able to set > this if the keyboard is connected. Or is there there an other solution? I think your changes are OK. A slight concern I found while looking quickly through the git commits is the cosmetic build issues, either only one of CONFIG_THINKPAD_ACPI and CONFIG_HID_LENOVO is enabled. But other than that, it looks straightforward. Of course, another option would be leave everything to user-space. But I don't think it would work fluently on the current desktop, even in near future. So considering practically, we'll likely go to the kernel hook way, I suppose. thanks, Takashi

1 0

[alsa-devel] [PATCH] ASoC: Intel: Atom: Fix message handling during drop stream
by Harsha Priya 30 Aug '16

30 Aug '16

If a stream is being dropped, the period_elapsed received then after need not be processed. Processing of the period_elapsed message causes a time out in drop stream response processing. This patch adds a condition to skip period_elapsed message processing if the stream is in INIT state. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Naveen M <naveen.m(a)intel.com> --- sound/soc/intel/atom/sst/sst_ipc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 8afa6fe..bfc8899 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -267,6 +267,9 @@ static void process_fw_async_msg(struct intel_sst_drv *sst_drv_ctx, "Period elapsed rcvd for pipe id 0x%x\n", pipe_id); stream = &sst_drv_ctx->streams[str_id]; + /* If stream is dropped, skip processing this message*/ + if (stream->status == STREAM_INIT) + break; if (stream->period_elapsed) stream->period_elapsed(stream->pcm_substream); if (stream->compr_cb) -- 1.9.1

2 1

Re: [alsa-devel] [PATCH] ALSA: usb-audio: Add sample rate inquiry quirk for B850V3 CP2114
by Akshay Bhat 30 Aug '16

30 Aug '16

Hi, On Fri, Aug 12, 2016 at 2:08 PM, Akshay Bhat <akshay.bhat(a)timesys.com> wrote: > From: Ken Lin <ken.lin(a)advantech.com.tw> > > Avoid getting sample rate on B850V3 CP2114 as it is unsupported and > causes noisy "current rate is different from the runtime rate" messages > when playback starts. > > Signed-off-by: Ken Lin <ken.lin(a)advantech.com.tw> > Signed-off-by: Akshay Bhat <akshay.bhat(a)timesys.com> > --- Any feedback on the patch? If not can it be applied? Thanks.

2 5

[alsa-devel] [PATCH] ASoC: hdac_hdmi: fix spelling mistake "montior" -> "monitor"
by Colin King 30 Aug '16

30 Aug '16

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_warn message. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/hdac_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4e181b2..c602c49 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -614,7 +614,7 @@ static int hdac_hdmi_pcm_open(struct snd_pcm_substream *substream, (!pin->eld.eld_valid)) { dev_warn(&hdac->hdac.dev, - "Failed: montior present? %d ELD valid?: %d for pin: %d\n", + "Failed: monitor present? %d ELD valid?: %d for pin: %d\n", pin->eld.monitor_present, pin->eld.eld_valid, pin->nid); return 0; -- 2.9.3

2 1

[alsa-devel] sound: deadlock snd_rawmidi_kernel_open/check_and_subscribe_port
by Dmitry Vyukov 30 Aug '16

30 Aug '16

Hello, While running syzkaller fuzzer on f31494bd05b06b0cdb4da6aebe92eaafab970df6 (Aug 12), I've got the following deadlock report: ====================================================== [ INFO: possible circular locking dependency detected ] 4.8.0-rc1+ #11 Not tainted ------------------------------------------------------- syz-executor/7154 is trying to acquire lock: (register_mutex#5){+.+.+.}, at: [<ffffffff84fd6d4b>] snd_rawmidi_kernel_open+0x4b/0x260 sound/core/rawmidi.c:341 but task is already holding lock: (&grp->list_mutex){++++.+}, at: [<ffffffff850138bb>] check_and_subscribe_port+0x5b/0x5c0 sound/core/seq/seq_ports.c:495 which lock already depends on the new lock. the existing dependency chain (in reverse order) is: -> #1 (&grp->list_mutex){++++.+}: [<ffffffff8147a3a8>] lock_acquire+0x208/0x430 kernel/locking/lockdep.c:3746 [<ffffffff863f6199>] down_read+0x49/0xc0 kernel/locking/rwsem.c:22 [< inline >] deliver_to_subscribers sound/core/seq/seq_clientmgr.c:681 [<ffffffff85005c5e>] snd_seq_deliver_event+0x35e/0x890 sound/core/seq/seq_clientmgr.c:822 [<ffffffff85006e96>] snd_seq_kernel_client_dispatch+0x126/0x170 sound/core/seq/seq_clientmgr.c:2418 [<ffffffff85012c52>] snd_seq_system_broadcast+0xb2/0xf0 sound/core/seq/seq_system.c:101 [<ffffffff84fff70a>] snd_seq_create_kernel_client+0x24a/0x330 sound/core/seq/seq_clientmgr.c:2297 [< inline >] snd_virmidi_dev_attach_seq sound/core/seq/seq_virmidi.c:383 [<ffffffff8502d29f>] snd_virmidi_dev_register+0x29f/0x750 sound/core/seq/seq_virmidi.c:450 [<ffffffff84fd208c>] snd_rawmidi_dev_register+0x30c/0xd40 sound/core/rawmidi.c:1645 [<ffffffff84f816d3>] __snd_device_register.part.0+0x63/0xc0 sound/core/device.c:164 [< inline >] __snd_device_register sound/core/device.c:162 [<ffffffff84f8235d>] snd_device_register_all+0xad/0x110 sound/core/device.c:212 [<ffffffff84f7546f>] snd_card_register+0xef/0x6c0 sound/core/init.c:749 [<ffffffff85040b7f>] snd_virmidi_probe+0x3ef/0x590 sound/drivers/virmidi.c:123 [<ffffffff833ebf7b>] platform_drv_probe+0x8b/0x170 drivers/base/platform.c:564 [< inline >] really_probe drivers/base/dd.c:377 [<ffffffff833e5993>] driver_probe_device+0x563/0xcc0 drivers/base/dd.c:499 [<ffffffff833e653d>] __device_attach_driver+0x21d/0x2e0 drivers/base/dd.c:594 [<ffffffff833df38f>] bus_for_each_drv+0x13f/0x1d0 drivers/base/bus.c:463 [<ffffffff833e51ef>] __device_attach+0x1ef/0x300 drivers/base/dd.c:651 [<ffffffff833e669a>] device_initial_probe+0x1a/0x20 drivers/base/dd.c:698 [<ffffffff833e2999>] bus_probe_device+0x1e9/0x290 drivers/base/bus.c:557 [<ffffffff833dc459>] device_add+0xc49/0x14c0 drivers/base/core.c:1120 [<ffffffff833eb629>] platform_device_add+0x2f9/0x7b0 drivers/base/platform.c:403 [<ffffffff833ed3b2>] platform_device_register_full+0x392/0x4b0 drivers/base/platform.c:536 [< inline >] platform_device_register_resndata ./include/linux/platform_device.h:111 [< inline >] platform_device_register_simple ./include/linux/platform_device.h:140 [<ffffffff8870fabd>] alsa_card_virmidi_init+0x104/0x1da sound/drivers/virmidi.c:172 [<ffffffff81002330>] do_one_initcall+0xa0/0x2b0 init/main.c:778 [< inline >] do_initcall_level init/main.c:843 [< inline >] do_initcalls init/main.c:851 [< inline >] do_basic_setup init/main.c:869 [<ffffffff88604d11>] kernel_init_freeable+0x47b/0x534 init/main.c:1016 [<ffffffff863e2f53>] kernel_init+0x13/0x180 init/main.c:942 [<ffffffff863fbdcf>] ret_from_fork+0x1f/0x40 arch/x86/entry/entry_64.S:393 -> #0 (register_mutex#5){+.+.+.}: [< inline >] check_prev_add kernel/locking/lockdep.c:1829 [< inline >] check_prevs_add kernel/locking/lockdep.c:1939 [< inline >] validate_chain kernel/locking/lockdep.c:2266 [<ffffffff814791f4>] __lock_acquire+0x4d44/0x4d80 kernel/locking/lockdep.c:3335 [<ffffffff8147a3a8>] lock_acquire+0x208/0x430 kernel/locking/lockdep.c:3746 [< inline >] __mutex_lock_common kernel/locking/mutex.c:521 [<ffffffff863f0ef1>] mutex_lock_nested+0xb1/0xa20 kernel/locking/mutex.c:621 [<ffffffff84fd6d4b>] snd_rawmidi_kernel_open+0x4b/0x260 sound/core/rawmidi.c:341 [<ffffffff8502e7c7>] midisynth_subscribe+0xf7/0x350 sound/core/seq/seq_midi.c:188 [< inline >] subscribe_port sound/core/seq/seq_ports.c:427 [<ffffffff85013cc7>] check_and_subscribe_port+0x467/0x5c0 sound/core/seq/seq_ports.c:510 [<ffffffff85015da9>] snd_seq_port_connect+0x2c9/0x500 sound/core/seq/seq_ports.c:579 [<ffffffff850079b8>] snd_seq_ioctl_subscribe_port+0x1d8/0x2b0 sound/core/seq/seq_clientmgr.c:1480 [<ffffffff84ffe9e4>] snd_seq_do_ioctl+0x184/0x1e0 sound/core/seq/seq_clientmgr.c:2225 [<ffffffff84ffeae8>] snd_seq_kernel_client_ctl+0xa8/0x110 sound/core/seq/seq_clientmgr.c:2440 [<ffffffff85027664>] snd_seq_oss_midi_open+0x3b4/0x610 sound/core/seq/oss/seq_oss_midi.c:375 [<ffffffff85023d67>] snd_seq_oss_synth_setup_midi+0x107/0x4c0 sound/core/seq/oss/seq_oss_synth.c:281 [<ffffffff8501b0a8>] snd_seq_oss_open+0x748/0x8d0 sound/core/seq/oss/seq_oss_init.c:274 [<ffffffff85019d8a>] odev_open+0x6a/0x90 sound/core/seq/oss/seq_oss.c:138 [<ffffffff84f7040f>] soundcore_open+0x30f/0x640 sound/sound_core.c:639 [<ffffffff818069ea>] chrdev_open+0x22a/0x4c0 fs/char_dev.c:392 [<ffffffff817f146f>] do_dentry_open+0x69f/0xca0 fs/open.c:736 [<ffffffff817f4aa5>] vfs_open+0x105/0x220 fs/open.c:849 [< inline >] do_last fs/namei.c:3374 [<ffffffff8182c35d>] path_openat+0x1efd/0x2f60 fs/namei.c:3497 [<ffffffff81830b6e>] do_filp_open+0x18e/0x250 fs/namei.c:3532 [<ffffffff817f53c1>] do_sys_open+0x201/0x420 fs/open.c:1036 [< inline >] SYSC_open fs/open.c:1054 [<ffffffff817f560d>] SyS_open+0x2d/0x40 fs/open.c:1049 [<ffffffff863fbb80>] entry_SYSCALL_64_fastpath+0x23/0xc1 arch/x86/entry/entry_64.S:207 other info that might help us debug this: Possible unsafe locking scenario: CPU0 CPU1 ---- ---- lock(&grp->list_mutex); lock(register_mutex#5); lock(&grp->list_mutex); lock(register_mutex#5); *** DEADLOCK *** 2 locks held by syz-executor/7154: #0: (register_mutex#4){+.+.+.}, at: [<ffffffff85019d7f>] odev_open+0x5f/0x90 sound/core/seq/oss/seq_oss.c:137 #1: (&grp->list_mutex){++++.+}, at: [<ffffffff850138bb>] check_and_subscribe_port+0x5b/0x5c0 sound/core/seq/seq_ports.c:495 stack backtrace: CPU: 0 PID: 7154 Comm: syz-executor Not tainted 4.8.0-rc1+ #11 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff878b7680 ffff880038b670f0 ffffffff82a5f719 ffffffff00000000 fffffbfff0f16ed0 ffffffff8901a0d0 ffffffff8901a0d0 ffffffff890191a0 ffff88003da2cf08 ffff88003da2c6c0 ffff880038b67140 ffffffff814708a8 Call Trace: [< inline >] __dump_stack lib/dump_stack.c:15 [<ffffffff82a5f719>] dump_stack+0x12e/0x185 lib/dump_stack.c:51 [<ffffffff814708a8>] print_circular_bug+0x288/0x340 kernel/locking/lockdep.c:1202 [< inline >] check_prev_add kernel/locking/lockdep.c:1829 [< inline >] check_prevs_add kernel/locking/lockdep.c:1939 [< inline >] validate_chain kernel/locking/lockdep.c:2266 [<ffffffff814791f4>] __lock_acquire+0x4d44/0x4d80 kernel/locking/lockdep.c:3335 [<ffffffff8147a3a8>] lock_acquire+0x208/0x430 kernel/locking/lockdep.c:3746 [< inline >] __mutex_lock_common kernel/locking/mutex.c:521 [<ffffffff863f0ef1>] mutex_lock_nested+0xb1/0xa20 kernel/locking/mutex.c:621 [<ffffffff84fd6d4b>] snd_rawmidi_kernel_open+0x4b/0x260 sound/core/rawmidi.c:341 [<ffffffff8502e7c7>] midisynth_subscribe+0xf7/0x350 sound/core/seq/seq_midi.c:188 [< inline >] subscribe_port sound/core/seq/seq_ports.c:427 [<ffffffff85013cc7>] check_and_subscribe_port+0x467/0x5c0 sound/core/seq/seq_ports.c:510 [<ffffffff85015da9>] snd_seq_port_connect+0x2c9/0x500 sound/core/seq/seq_ports.c:579 [<ffffffff850079b8>] snd_seq_ioctl_subscribe_port+0x1d8/0x2b0 sound/core/seq/seq_clientmgr.c:1480 [<ffffffff84ffe9e4>] snd_seq_do_ioctl+0x184/0x1e0 sound/core/seq/seq_clientmgr.c:2225 [<ffffffff84ffeae8>] snd_seq_kernel_client_ctl+0xa8/0x110 sound/core/seq/seq_clientmgr.c:2440 [<ffffffff85027664>] snd_seq_oss_midi_open+0x3b4/0x610 sound/core/seq/oss/seq_oss_midi.c:375 [<ffffffff85023d67>] snd_seq_oss_synth_setup_midi+0x107/0x4c0 sound/core/seq/oss/seq_oss_synth.c:281 [<ffffffff8501b0a8>] snd_seq_oss_open+0x748/0x8d0 sound/core/seq/oss/seq_oss_init.c:274 [<ffffffff85019d8a>] odev_open+0x6a/0x90 sound/core/seq/oss/seq_oss.c:138 [<ffffffff84f7040f>] soundcore_open+0x30f/0x640 sound/sound_core.c:639 [<ffffffff818069ea>] chrdev_open+0x22a/0x4c0 fs/char_dev.c:392 [<ffffffff817f146f>] do_dentry_open+0x69f/0xca0 fs/open.c:736 [<ffffffff817f4aa5>] vfs_open+0x105/0x220 fs/open.c:849 [< inline >] do_last fs/namei.c:3374 [<ffffffff8182c35d>] path_openat+0x1efd/0x2f60 fs/namei.c:3497 [<ffffffff81830b6e>] do_filp_open+0x18e/0x250 fs/namei.c:3532 [<ffffffff817f53c1>] do_sys_open+0x201/0x420 fs/open.c:1036 [< inline >] SYSC_open fs/open.c:1054 [<ffffffff817f560d>] SyS_open+0x2d/0x40 fs/open.c:1049 [<ffffffff863fbb80>] entry_SYSCALL_64_fastpath+0x23/0xc1 arch/x86/entry/entry_64.S:207

2 4

[alsa-devel] [PATCH 0/2] Support non-ssi cpu-dai for sgtl5000
by Winter Wang 30 Aug '16

30 Aug '16

These patches support sgtl5000 to be attached to other non-ssi cpu-dais like SAI. Winter Wang (2): ASoC: imx-sgtl5000: Support non-ssi cpu-dai ARM: dts: modify sgtl5000 property to support non-ssi cpu-dai .../bindings/sound/imx-audio-sgtl5000.txt | 4 +-- arch/arm/boot/dts/imx25-pdk.dts | 2 +- arch/arm/boot/dts/imx51-babbage.dts | 2 +- arch/arm/boot/dts/imx53-m53evk.dts | 2 +- arch/arm/boot/dts/imx53-mba53.dts | 2 +- arch/arm/boot/dts/imx53-qsb-common.dtsi | 2 +- arch/arm/boot/dts/imx53-tx53.dtsi | 2 +- arch/arm/boot/dts/imx53-voipac-bsb.dts | 2 +- arch/arm/boot/dts/imx6dl-riotboard.dts | 2 +- arch/arm/boot/dts/imx6q-bx50v3.dtsi | 2 +- arch/arm/boot/dts/imx6q-gw5400-a.dts | 2 +- arch/arm/boot/dts/imx6q-h100.dts | 2 +- arch/arm/boot/dts/imx6q-icore-rqs.dts | 2 +- arch/arm/boot/dts/imx6q-tbs2910.dts | 2 +- arch/arm/boot/dts/imx6qdl-apalis.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-apf6dev.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-gw52xx.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-gw53xx.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-gw54xx.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-hummingboard.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-nit6xlite.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-nitrogen6_max.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-nitrogen6x.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-rex.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-sabrelite.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-tx6.dtsi | 2 +- arch/arm/boot/dts/imx6qdl-wandboard.dtsi | 2 +- sound/soc/fsl/imx-sgtl5000.c | 41 ++++++++++++++-------- 28 files changed, 55 insertions(+), 42 deletions(-) -- 1.9.1

3 4

[alsa-devel] [PATCH v6 0/2] amidi delay option
by Felipe F. Tonello 30 Aug '16

30 Aug '16

Small improvement to amidi, adding useful functionality. changes from v5: * wait until MIDI message is competely sent to sleep changes from v4: * simplified code changes from v3: * fixed a bug if memchr() returns NULL changes from v2: * only check of end of sysex (0xf7) * use memchr() instead of direct loop approach * fixed commit message changes from v1: * remove delay in between any MIDI message, only SysEx * added patch to use getopt_long() arguments Felipe F. Tonello (2): amidi: add delay option amidi: use GNU getopt_long() proper argument names amidi/amidi.1 | 14 ++++++++++ amidi/amidi.c | 82 ++++++++++++++++++++++++++++++++++++++++++++++------------- 2 files changed, 79 insertions(+), 17 deletions(-) -- 2.9.3

3 6

Re: [alsa-devel] No sound for several minutes after reboot
by Takashi Iwai 30 Aug '16

30 Aug '16

On Sat, 27 Aug 2016 13:39:08 +0200, Markus Trippelsdorf wrote: > > On 2016.08.27 at 11:50 +0200, Markus Trippelsdorf wrote: > > Hi, > > > > since several kernel releases I get no sound for several minutes after > > reboot. After this delay it starts working normally. > > > > Is this some kind of speaker protection? Is there a parameter that I can > > set to get sound immediately? > > > > % aplay -l > > **** List of PLAYBACK Hardware Devices **** > > card 0: SB [HDA ATI SB], device 0: VT1708S Analog [VT1708S Analog] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 0: SB [HDA ATI SB], device 3: VT1708S Digital [VT1708S Digital] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 1: HDMI [HDA ATI HDMI], device 3: Generic Digital [Generic Digital] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > > > card 0, device 3 [VT1708S Digital] is the device that I use. It is a > > TOSLINK-S/PDIF link to my DAC. > > Here is the diff (bad (after reboot) vs. good (after a few minutes)): > > @@ -121,7 +121,7 @@ > Control: name="Surround Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x2a, nsteps=0x2a, stepsize=0x05, mute=0 > - Amp-Out vals: [0x1c 0x1c] > + Amp-Out vals: [0x26 0x26] > Converter: stream=0, channel=0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > @@ -174,12 +174,12 @@ > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power states: D0 D1 D2 D3 > - Power: setting=D0, actual=D0 > + Power: setting=D3, actual=D3 > Connection: 1 > 0x1e So this looks like the only significant change. Others are about the volume level, and should lead to the complete mute. Could you try to adjust the power state manually while it happens? hda-verb /dev/snd/hwC0D0 0xXX SET_POWER 0x00 where 0xXX is filled with the node number of this widget. BTW, at the next time, take a diff with -p option. This will show the node ID in the diff. Takashi

2 14

[alsa-devel] iMX6SX having SSI TXC from external and TXFS from internal, is it possible?
by Roberto Fichera 30 Aug '16

30 Aug '16

Hi, I'm playing with a custom iMX6SX board having the SSI connected to a SLIC in TDM mode. SSI has TXC (PCLK) coming from an external oscillator running at 2.048MHz while TXFS (FSYNC) is set to be generated from internal or at least this is what I'm trying to do. The SSI documentation is not really clear in case, so wondering to know if it is possible to generate FSYNC with this particular configuration: SSI set in I2S normal mode, synchronous transfers, network mode, TXC taken from external oscillator. The relevant part reported in iMX6SXRM doc (62.8.4 SSI Clocking) is: "In master and synchronous mode, the unused port SRCK is used as network clock (oversampling clock) enabled by the SCR register bit 15, SYS_CLK_EN. This network clock is an oversampling clock of the frame sync clock (STFS). In this mode, the word length (WL), Prescaler Range (PSR), Prescaler Modulus (PM) and Frame rate (DC) selects the ratio of network clock to sampling clock STFS. In case of I2S mode, the network clock (oversampling clock) can be made available on this port if the SYS_CLK_EN bit is set." Accordingly with Figure 62-22 in the iMX6SX reference manual network clock on SRCK can be activated only when both TXDIR and SYS_CLK_EN are set. But if TXDIR is set I cannot get STCK from external oscillator because SSI is now generating it from internal. I've also changed the way how audmux is routing FYSNC, internal port is SSI1 and external is SSI6 int audmux_config(unsigned int master, unsigned int slave) { unsigned int ptcr, pdcr; ptcr = IMX_AUDMUX_V2_PTCR_SYN /* Synchronous mode */ | IMX_AUDMUX_V2_PTCR_TCSEL(slave); /* transmit clock select from external port */ pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(slave); audmux_v2_configure_port(master, ptcr, pdcr); /* configure internal port */ ptcr = IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR /* Transmit Frame Sync from Internal */ | IMX_AUDMUX_V2_PTCR_TFSEL(master); /* from master port */ pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(master); audmux_v2_configure_port(slave, ptcr, pdcr); /* configure external port */ return 0; } and SSI1 is set as: scr = CCSR_SSI_SCR_I2S_MODE_NORMAL | CCSR_SSI_SCR_SYN | /* Synchronous mode */ CCSR_SSI_SCR_NET; /* Network mode to enable TDM */ stcr = CCSR_SSI_STCR_TXBIT0 | /* LSB Aligned */ CCSR_SSI_STCR_TFDIR | /* Transmit Frame Direction Internal - generated internally */ CCSR_SSI_STCR_TFEN0 | /* Enable TX FIFO0 */ CCSR_SSI_STCR_TSCKP; /* Transmit Clock Polarity - Data Clocked out on falling edge */ Doesn't anyone know if it's possible to have this particular TDM configuration? Thanks in advance, Roberto Fichera.

1 0

[alsa-devel] [PATCH 0/4] ASoC: dapm: Various fixes for dapm kcontrols
by Chen-Yu Tsai 30 Aug '16

30 Aug '16

Hi everyone, This series is a bunch of fixes for some problems I encountered while experimenting with various dapm kcontrols for Allwinner A31 support in sun4i-codec. Patch 1 fixes a possible uninitialized variable usage in snd_soc_dapm_get_volsw(). This doesn't show up as a compiler warning but I'm fairly certain it might happen. Patch 2 fixes setting the value for an enum kcontrol's second channel. Patch 3 fixes kcontrol creation for output driver widgets. An output driver widget is the same as a PGA widget, just with a different priority. Patch 4 adds an error message if anyone attempts to add hardware backed kcontrols on PGA widgets. Originally kcontrols on PGAs weren't supported at all. Support for them on virtual widgets were added later, but a guard for real widgets was missing. Regards ChenYu Chen-Yu Tsai (4): ASoC: dapm: Fix possible uninitialized variable in snd_soc_dapm_get_volsw() ASoC: dapm: Fix value setting for _ENUM_DOUBLE MUX's second channel ASoC: dapm: Fix kcontrol creation for output driver widget ASoC: dapm: Give error when adding hardware backed controls for PGA widgets sound/soc/soc-dapm.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) -- 2.9.3

2 5

[alsa-devel] [PATCH v2] ASoC: core: fix shift used for second item in snd_soc_get_enum_double
by Jaswinder Jassal 30 Aug '16

30 Aug '16

Incorrect shift value was being used to extract the second item. Signed-off-by: Jaswinder Jassal <jjassal(a)opensource.wolfsonmicro.com> --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index a513a34..9fc1a7b 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -77,7 +77,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[0] = item; if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_r) & e->mask; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = item; } -- 2.1.4

2 1

[alsa-devel] Range of pitchbend MIDI events
by Maurizio Berti 30 Aug '16

30 Aug '16

Hello, I was playing with pyalsa and I realized an unexpected behavior with pitchbend events. The pitchbend range should be, by MIDI standard, between -8192 and +8191, or 0-16384. Being a 3 byte event, it should ignore any subsequent byte received. Using pyalsa I'm able to create events with values greater than 127 and they are translated using as many bytes as possible; in fact, if I create a note event with a note value of 8193, I receive a value 1 (using the first byte of a 2 byte value with 7bit format). That's fine, I suppose it's by architecture (I'm not a programmer, I just play around with python, I'm just assuming this). The problem comes with pitchbend values, since it looks like alsa accepts values greater than 8191 and does not ignore the exceeding bytes as it should; that is, I'm able to send 8192, and that's what vkeybd does. The transformation here is the issue, since receiving programs seem to translate only the latest two bytes, that's why vkeybd (and other software too) shouldn't have a 8192 value. That can be confusing, for example, using a2jmidi: the end of the pitchbend scale is suddenly a 0 value; I know that vkeybd is not the ultimate software in music production, but, still, it shouldn't be able to do so, I think :) I don't know if, maybe, there's some less known aspect regarding pitchbend, and, as I mentioned before, I don't know a thing about programming, nor how actually works ALSA under the hood. Anyway, I think there should be some kind of range control, at least with pyalsa, if not by alsa at all. What do you think? Regards, Maurizio -- È difficile avere una convinzione precisa quando si parla delle ragioni del cuore. - "Sostiene Pereira", Antonio Tabucchi http://www.jidesk.net

3 5

[alsa-devel] [PATCH v2] ALSA: hda - Enable subwoofer on Dell Inspiron 7559
by Kai-Heng Feng 30 Aug '16

30 Aug '16

The subwoofer on Inspiron 7559 was disabled originally. Applying a pin fixup to node 0x1b can enable it and make it work. Old pin: 0x411111f0 New pin: 0x90170151 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7100f05..2a3dd18 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4855,6 +4855,7 @@ enum { ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, ALC298_FIXUP_SPK_VOLUME, + ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, }; static const struct hda_fixup alc269_fixups[] = { @@ -5516,6 +5517,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, }, + [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x90170151 }, + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5560,6 +5570,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), -- 2.9.3

2 1

[alsa-devel] [PATCH 00/12] ASoC: hdac: Add hdac generic driver
by Subhransu S. Prusty 30 Aug '16

30 Aug '16

HDA devices generically can be modelled with DAPM in ASoC. We would like to create a framework for the HDA devices in ASoC. This will follow HDA standard. This will be registered as generic "virtual" class device (Not implemented in this series). If there is no match found for vendor id and device id, then the class driver will be loaded. For vendor specific devices, vendor quirks will be loaded based on the match. Additional widgets, controls and route will be created in a vendor specific way through vendor ops. This series adds a framework in ASoC to model HDA codecs. HDA widgets are enumerated and one or multiple DAPM widget(s) are created. Connection list is queried for each widget to identify the connection between two endpoints and modelled using DAPM graph. Set of event handlers are defined for each widget type. Based on DAPM events required verbs are sent to program codec. Finally a function is exported to query for the device endpoint configuration to create machine controls. Features yet to be added: - Optional features of widgets will be added using additional dapm widgets or controls - Pin retasking will be done by adding all possible dapm route map - Class driver to load the generic driver if no match found - Jack detection Subhransu S. Prusty (12): ALSA: hdac: Add codec helper library ALSA: hda - Add macro to test pin widget's input capability ASoC: hdac: Add a generic hdac driver framework ASoC: hdac: Create DAPM model for HDA widgets ALSA: hdac: Move 'to_hda_ext_device' helper to core ASoC: hdac_hdmi: use get_edev helper from core ASoC: hdac: Build DAPM graph by querying through widget connection list ASoC: hdac: Register widget event handlers ALSA: hda - macro to get default config device of pin widgets ASoC: hdac: Export API to create machine controls ALSA: hdac: Implement a match function ASoC: hdac: Add coefficient init callback for RT286 include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 1 + sound/hda/ext/Makefile | 3 +- sound/hda/ext/hdac_codec.c | 182 ++++ sound/hda/ext/hdac_codec.h | 60 ++ sound/hda/ext/hdac_ext_bus.c | 12 + sound/hda/local.h | 13 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_generic.c | 1754 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_generic.h | 36 + sound/soc/codecs/hdac_hdmi.c | 8 +- 12 files changed, 2069 insertions(+), 8 deletions(-) create mode 100644 sound/hda/ext/hdac_codec.c create mode 100644 sound/hda/ext/hdac_codec.h create mode 100644 sound/soc/codecs/hdac_generic.c create mode 100644 sound/soc/codecs/hdac_generic.h -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Enable subwoofer on Dell Inspiron 7559
by Kai-Heng Feng 30 Aug '16

30 Aug '16

The subwoofer on Inspiron 7559 does not work originally. Applying a pin fixup can make it work. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7100f05..2a3dd18 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4855,6 +4855,7 @@ enum { ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, ALC298_FIXUP_SPK_VOLUME, + ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, }; static const struct hda_fixup alc269_fixups[] = { @@ -5516,6 +5517,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, }, + [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x90170151 }, + { } + }, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5560,6 +5570,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), -- 2.9.3

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[alsa-devel] [PATCH v2 0/2] ALSA: fireworks/firewire-tascam: accessing to user space outside spinlock
by Takashi Sakamoto 30 Aug '16

30 Aug '16

Hi, In ALSA fireworks driver and firewire-tascam driver, current implementation for hwdep interface have issues for page fault handling. For example, when using single core processor, after executing lock_page() in page fault handler and call task scheduler, then the context never regain CPU again, because any interrupts are disabled by these drivers and the context can't catch waiting event. This patchset fixes the issues. The issues were reported by Vaishali Thakkar, with a help of coccinelle. Thanks a lot for her care to these minor drivers. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111887.html Takashi Sakamoto (2): ALSA: fireworks: accessing to user space outside spinlock ALSA: firewire-tascam: accessing to user space outside spinlock sound/firewire/fireworks/fireworks_hwdep.c | 21 ++++++++++++++----- sound/firewire/tascam/tascam-hwdep.c | 33 ++++++++++-------------------- 2 files changed, 27 insertions(+), 27 deletions(-) -- 2.7.4

3 6

[alsa-devel] [PATCH] ALSA: seq: initialize whole fields of automatic variable with union type
by Takashi Sakamoto 30 Aug '16

30 Aug '16

Currently, automatic variable of 'union ioctl_arg' type is initialized by designated initialization. Although, the actual effect is interpretation of early element of int type and initialization of 'int pversion'. Therefore the first field corresponding to int type is initialized to zero. This is against my expectation to initialize whole fields. This commit uses memset() to initialize the variable, instead of designated initialization. Fixes: 04a56dd8ed0d ('ALSA: seq: change ioctl command operation to get data in kernel space') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/seq/seq_clientmgr.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 286394b..fd50c38 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2100,7 +2100,7 @@ static long snd_seq_ioctl(struct file *file, unsigned int cmd, struct snd_seq_client_pool client_pool; struct snd_seq_remove_events remove_events; struct snd_seq_query_subs query_subs; - } buf = {0}; + } buf; const struct ioctl_handler *handler; unsigned long size; int err; @@ -2114,6 +2114,9 @@ static long snd_seq_ioctl(struct file *file, unsigned int cmd, } if (handler->cmd == 0) return -ENOTTY; + + memset(&buf, 0, sizeof(union ioctl_arg)); + /* * All of ioctl commands for ALSA sequencer get an argument of size * within 13 bits. We can safely pick up the size from the command. -- 2.7.4

2 2

[alsa-devel] [PATCH 0/2] alsabat: add power management S3 test
by keqiao.zhang＠intel.com 30 Aug '16

30 Aug '16

From: "Keqiao, Zhang" <keqiao.zhang(a)intel.com> The patch set provide a method for QA to quick test audio during system S3 The first patch fix -86 error when system suspend/resume The second patch provide a test script Keqiao, Zhang (2): alsabat: fix alsabat -86 error alsabat: add system power management S3 test bat/alsa.c | 10 ++++++++++ bat/alsabat-test.sh | 34 ++++++++++++++++++++++++++++++++++ 2 files changed, 44 insertions(+) -- 2.5.0

2 3

[alsa-devel] [PATCH] ALSA: hda - Add headset mic quirk for Dell Inspiron 5468
by Shrirang Bagul 30 Aug '16

30 Aug '16

This patch enables headset microphone on some variants of Dell Inspiron 5468. (Dell SSID 0x07ad) BugLink: https://bugs.launchpad.net/bugs/1617900 Signed-off-by: Shrirang Bagul <shrirang.bagul(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7100f05..02fd252 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5895,6 +5895,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60170}, {0x14, 0x90170120}, {0x21, 0x02211030}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell Inspiron 5468", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60180}, + {0x14, 0x90170120}, + {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, -- 2.7.4

3 2

[alsa-devel] [PATCH v9 0/2] sun4i-codec: Add Line-In, FM-In, Mic 2, Capture Source, Differential Line-In
by Danny Milosavljevic 30 Aug '16

30 Aug '16

This patch adds support for some mixer controls: - Line-In - FM-In - Mic 2 - Capture Source - Differential Line-In v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (2): ASoC: sun4i-codec: Distinguish sun4i from sun7i Add mixer controls: Line-In, FM-In, Mic 2, Capture Source, Differential Line-In. sound/soc/sunxi/sun4i-codec.c | 278 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 259 insertions(+), 19 deletions(-)

3 5

[alsa-devel] [PATCH 0/2] ALSA: fireworks/firewire-tascam: accessing to user space outside spinlock
by Takashi Sakamoto 30 Aug '16

30 Aug '16

Hi, In ALSA fireworks driver and firewire-tascam driver, current implementation for hwdep interface have issues for page fault handling. For example, when using single core processor, after executing lock_page() in page fault handler and call task scheduler, then the context never run again, because any interrupts are disabled by these drivers and the context can't catch waiting event. This patchset fixes the issues. The issues were reported by Vaishali Thakkar, with a help of coccinelle. Thanks a lot for his care to these minor drivers. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111887.html Takashi Sakamoto (2): ALSA: fireworks: accessing to user space outside spinlock ALSA: firewire-tascam: accessing to user space outside spinlock sound/firewire/fireworks/fireworks_hwdep.c | 20 ++++++++++++++----- sound/firewire/tascam/tascam-hwdep.c | 32 ++++++++++-------------------- 2 files changed, 25 insertions(+), 27 deletions(-) -- 2.7.4

2 4

[alsa-devel] Translating USB device names to ALSA client numbers
by Dejay 30 Aug '16

30 Aug '16

Hello, I'm trying to write some udev wrappers to automatically connect USB MIDI devices to each other. My logic could be really robust if I had a way of mapping ALSA client numbers (e.g. from 'aconnect -l', such as '20:0') to physical USB devices (either by bus ID, /dev/ symlink name, /proc/asound/ files, etc.) My ultimate goal is so that when a udev rule executes upon hotplugging a MIDI device, I can reference that MIDI device by client number and then execute an aconnect command. Thus, I need some way to map USB device information to the ALSA client numbers that get reported by aconnect. Is there any way to accomplish this? I tried to review the ALSA source code, but my eyes got jumbly after awhile. Thanks, Dejay

1 1

[alsa-devel] [PATCH] ASoC: wm8962: fix spelling mistake "mesurement" -> "measurement"
by Colin King 29 Aug '16

29 Aug '16

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index f3109da..97b0228 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3713,7 +3713,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, ARRAY_SIZE(wm8962_dc_measure)); if (ret != 0) dev_err(&i2c->dev, - "Failed to configure for DC mesurement: %d\n", + "Failed to configure for DC measurement: %d\n", ret); } -- 2.9.3

2 1

[alsa-devel] [RFC PATCH v2 0/7] AC97 device/driver model revamp
by Robert Jarzmik 29 Aug '16

29 Aug '16

It all started in the pxa device-tree submission here : https://lkml.org/lkml/2016/2/25/965 It will be maintained in : git fetch https://github.com/rjarzmik/linux.git work/ac97 And now it transformed into this RFC, which would bring a ground for AC'97 devices closer to the linux device/driver model. The driving ideas are still the same, and I put them in [1] for memory. This is the second opus of the RFC. I'm intending to stop the RFC cycle here if possible and have a true PATCH v1 submission after this RFC v2 unless there is a deep design flaw uncovered. I took into account Mark's and Takashi's remarks, I hope I forgot none. All the changes should be documented in the first 2 patches mainly. I also added : - the AC97 link clock For now, bus code doesn't disable it in suspend and re-enable in resume, leaving the controller decide. I'm still pondering if for S2RAM (as opposed to runtime suspend), bus code should disable the clock. - the .h guards I'm not particularly happy with my naming even for v2, feel free to propose anything better, the codec.h is particularly ugly. - statics and namespace pollution I made more functions static, limiting further the namespace pollution. - Kconfig/Makefile layout change I get the feeling that the KConfig "select" flavor would have better been a "depends on" one. And yet I didn't find a good way to enforce it, because of the way sound/soc/codecs/Kconfig is designed, and I'm afraid to break the structure by doing a "depends on AC97_BUS_NEW" in WM9713 configuration. Let's have another review cycle, that will let me test this serie more thoroughly, especially the suspend/resume and the reset in the resume, which doesn't work, and requires me to test more deeply. I'll also test by removing the wm9713 change to see how robust the implementation is. Happy review. -- Robert Robert Jarzmik (7): ALSA: ac97: split out the generic ac97 registers ALSA: ac97: add an ac97 bus ASoC: add new ac97 bus support ASoC: wm9713: add ac97 new bus support ASoC: pxa: switch to new ac97 bus support ARM: pxa: mioa701 convert to the new AC97 bus ASoC: mioa701_wm9713: convert to new ac97 bus arch/arm/mach-pxa/mioa701.c | 12 +- include/sound/ac97/codec.h | 111 ++++++++++++ include/sound/ac97/compat.h | 21 +++ include/sound/ac97/controller.h | 80 +++++++++ include/sound/ac97/regs.h | 262 +++++++++++++++++++++++++++ include/sound/ac97_codec.h | 239 +------------------------ include/sound/pxa2xx-lib.h | 15 +- sound/Kconfig | 2 + sound/Makefile | 1 + sound/ac97/Kconfig | 18 ++ sound/ac97/Makefile | 8 + sound/ac97/ac97_core.h | 10 ++ sound/ac97/bus.c | 381 ++++++++++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 ++ sound/ac97/snd_ac97_compat.c | 104 +++++++++++ sound/arm/Kconfig | 1 - sound/arm/pxa2xx-ac97-lib.c | 39 ++-- sound/soc/Kconfig | 4 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/wm9713.c | 62 ++++++- sound/soc/pxa/Kconfig | 4 +- sound/soc/pxa/mioa701_wm9713.c | 4 +- sound/soc/pxa/pxa2xx-ac97.c | 20 ++- 23 files changed, 1132 insertions(+), 282 deletions(-) create mode 100644 include/sound/ac97/codec.h create mode 100644 include/sound/ac97/compat.h create mode 100644 include/sound/ac97/controller.h create mode 100644 include/sound/ac97/regs.h create mode 100644 sound/ac97/Kconfig create mode 100644 sound/ac97/Makefile create mode 100644 sound/ac97/ac97_core.h create mode 100644 sound/ac97/bus.c create mode 100644 sound/ac97/codec.c create mode 100644 sound/ac97/snd_ac97_compat.c -- 2.1.4 [1] RFC PATCH reminder: Amongst the driving ideas behind this RFC : - use device/driver model for AC'97 device/driver binding is based on vendor_id{1,2} - use auto-probe of the AC'97 bus to enumerate codecs and create devices - be compatible with previous platform_data model for codecs => this enables a smooth transition, where a codec (here wm9713) can be used either through platform_device or automatic discovery - struct snd_ac97 is not used This structure is really heavy, and doesn't represent an AC97 device, but rather an AC'97 codec internals IIUC. I think it's right place would be in include/sound/ac97/codec.h. This is also a problem for regmap adherence, hence the include/sound/ac97/compat.h. - a new AC'97 bus Kconfig is created This was done for a smooth transition ... let's see if it's a good idea. - split the AC'97 into : - the codec - the digital controller - the bus Amongst the things that are not all touched yet : - ac97_codec.c The generic AC'97 codec in sound/pci/ac97/ac97_codec.c has really a lot features, which are not at all PCI related. They could be transfered to sound/ac97, without the pci specific field. => this deserves a deep ahead thinking, which I'll do if the current approach is accepted by the communauty. The most important patch is 2/7 "ALSA: ac97: add an ac97 bus". This is the main evaluation point for the serie, the others are here so that I can test it all. Well, this is a long term effort, which might need a complete rewrite according to the comments it'll get. Let's expose it for comments and see how I can progress with it.

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[alsa-devel] [PATCH 1/1] ASoC: Intel: Atom: add a missing star in a memcpy call
by Nicolas Iooss 29 Aug '16

29 Aug '16

In sst_prepare_and_post_msg(), when a response is received in "block", the following code gets executed: *data = kzalloc(block->size, GFP_KERNEL); memcpy(data, (void *) block->data, block->size); The memcpy() call overwrites the content of the *data pointer instead of filling the newly-allocated memory (which pointer is hold by *data). Fix this by using *data in the memcpy() call. Fixes: 60dc8dbacb00 ("ASoC: Intel: sst: Add some helper functions") Cc: stable(a)vger.kernel.org # 3.19.x Signed-off-by: Nicolas Iooss <nicolas.iooss_linux(a)m4x.org> --- sound/soc/intel/atom/sst/sst_pvt.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index adb32fefd693..7c398b7c9d4b 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -289,7 +289,7 @@ int sst_prepare_and_post_msg(struct intel_sst_drv *sst, ret = -ENOMEM; goto out; } else - memcpy(data, (void *) block->data, block->size); + memcpy(*data, (void *) block->data, block->size); } } out: -- 2.9.3

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[alsa-devel] sound: division by 0 in snd_hrtimer_callback
by Dmitry Vyukov 28 Aug '16

28 Aug '16

Hello, The following program triggers divide error in snd_hrtimer_callback: divide error: 0000 [#1] SMP DEBUG_PAGEALLOC KASAN Dumping ftrace buffer: (ftrace buffer empty) Modules linked in: CPU: 3 PID: 17469 Comm: syz-executor Not tainted 4.8.0-rc3+ #33 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 task: ffff88003dc4a640 task.stack: ffff88003c690000 RIP: 0010:[<ffffffff8575b49c>] [< inline >] ktime_divns ./include/linux/ktime.h:195 [<ffffffff8575b49c>] snd_hrtimer_callback+0x1bc/0x3c0 sound/core/hrtimer.c:62 RSP: 0018:ffff88006d507d98 EFLAGS: 00010006 RAX: 0000000000005f28 RBX: ffff88006c4acb48 RCX: 0000000000000000 RDX: 0000000000000000 RSI: ffff88006c4acb60 RDI: ffffffff884b7a68 RBP: ffff88006d507df0 R08: 000000687d030efd R09: ffff88007fff70c8 R10: ffff88007fff70d8 R11: ffff88007fff70d0 R12: ffff88003cda70c0 R13: ffff88003cda7174 R14: ffff88003cda71f0 R15: ffff88006c4acb40 FS: 00007f131f945700(0000) GS:ffff88006d500000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000451ed0 CR3: 000000005bb88000 CR4: 00000000000006e0 Stack: 0000000000000000 ffff88006c4acb60 ffff88003cda7178 0000000000005f28 ffff88006c4acba0 0000000000000000 ffff88006c4acb48 0000000000000003 ffff88006d51b400 dffffc0000000000 dffffc0000000000 ffff88006d507ee8 Call Trace: <IRQ> [< inline >] __run_hrtimer kernel/time/hrtimer.c:1238 [<ffffffff814f48b5>] __hrtimer_run_queues+0x325/0xe70 kernel/time/hrtimer.c:1302 [<ffffffff814f726b>] hrtimer_interrupt+0x18b/0x420 kernel/time/hrtimer.c:1336 [<ffffffff81268f8f>] local_apic_timer_interrupt+0x6f/0xe0 arch/x86/kernel/apic/apic.c:933 [<ffffffff86c27d96>] smp_apic_timer_interrupt+0x76/0xa0 arch/x86/kernel/apic/apic.c:957 [<ffffffff86c26e4c>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:455 <EOI> [< inline >] spin_unlock_irqrestore ./include/linux/spinlock.h:362 [<ffffffff85750f7d>] snd_timer_start1+0xdd/0x590 sound/core/timer.c:464 [<ffffffff857532b5>] snd_timer_continue+0x45/0x80 sound/core/timer.c:593 [< inline >] snd_timer_user_continue sound/core/timer.c:1821 [< inline >] __snd_timer_user_ioctl sound/core/timer.c:1888 [<ffffffff85758ce9>] snd_timer_user_ioctl+0xe99/0x2470 sound/core/timer.c:1903 [< inline >] vfs_ioctl fs/ioctl.c:43 [<ffffffff8186f1fc>] do_vfs_ioctl+0x18c/0x1080 fs/ioctl.c:675 [< inline >] SYSC_ioctl fs/ioctl.c:690 [<ffffffff8187017f>] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:681 [<ffffffff86c25480>] entry_SYSCALL_64_fastpath+0x23/0xc1 arch/x86/entry/entry_64.S:207 Code: 6b 45 e5 fb 8b 0d e5 ca 64 05 48 0f af 4d d0 48 85 c9 0f 88 6a 01 00 00 48 89 4d a8 e8 4e 45 e5 fb 48 8b 45 c0 48 8b 4d a8 48 99 <48> f7 f9 48 01 45 d0 e8 38 45 e5 fb 4c 89 fa 48 b8 00 00 00 00 RIP [< inline >] ktime_divns ./include/linux/ktime.h:195 RIP [<ffffffff8575b49c>] snd_hrtimer_callback+0x1bc/0x3c0 sound/core/hrtimer.c:62 RSP <ffff88006d507d98> ---[ end trace b12f88d370b1ccee ]--- Kernel panic - not syncing: Fatal exception in interrupt On commit 61c04572de404e52a655a36752e696bbcb483cf5 (Aug 25). // autogenerated by syzkaller (http://github.com/google/syzkaller) #ifndef __NR_mmap #define __NR_mmap 9 #endif #ifndef __NR_syz_open_dev #define __NR_syz_open_dev 1000001 #endif #ifndef __NR_ioctl #define __NR_ioctl 16 #endif #ifndef __NR_syz_fuse_mount #define __NR_syz_fuse_mount 1000003 #endif #ifndef __NR_syz_fuseblk_mount #define __NR_syz_fuseblk_mount 1000004 #endif #ifndef __NR_syz_open_pts #define __NR_syz_open_pts 1000002 #endif #include <fcntl.h> #include <pthread.h> #include <setjmp.h> #include <signal.h> #include <stddef.h> #include <stdint.h> #include <stdio.h> #include <stdlib.h> #include <string.h> #include <sys/ioctl.h> #include <sys/stat.h> #include <sys/syscall.h> #include <sys/types.h> #include <unistd.h> __thread int skip_segv; __thread jmp_buf segv_env; static void segv_handler(int sig, siginfo_t* info, void* uctx) { if (__atomic_load_n(&skip_segv, __ATOMIC_RELAXED)) _longjmp(segv_env, 1); exit(sig); } static void install_segv_handler() { struct sigaction sa; memset(&sa, 0, sizeof(sa)); sa.sa_sigaction = segv_handler; sa.sa_flags = SA_NODEFER | SA_SIGINFO; sigaction(SIGSEGV, &sa, NULL); sigaction(SIGBUS, &sa, NULL); } #define NONFAILING(...) \ { \ __atomic_fetch_add(&skip_segv, 1, __ATOMIC_SEQ_CST); \ if (_setjmp(segv_env) == 0) { \ __VA_ARGS__; \ } \ __atomic_fetch_sub(&skip_segv, 1, __ATOMIC_SEQ_CST); \ } static uintptr_t syz_open_dev(uintptr_t a0, uintptr_t a1, uintptr_t a2) { if (a0 == 0xc || a0 == 0xb) { char buf[128]; sprintf(buf, "/dev/%s/%d:%d", a0 == 0xc ? "char" : "block", (uint8_t)a1, (uint8_t)a2); return open(buf, O_RDWR, 0); } else { char buf[1024]; char* hash; strncpy(buf, (char*)a0, sizeof(buf)); buf[sizeof(buf) - 1] = 0; while ((hash = strchr(buf, '#'))) { *hash = '0' + (char)(a1 % 10); a1 /= 10; } return open(buf, a2, 0); } } static uintptr_t syz_open_pts(uintptr_t a0, uintptr_t a1) { int ptyno = 0; if (ioctl(a0, TIOCGPTN, &ptyno)) return -1; char buf[128]; sprintf(buf, "/dev/pts/%d", ptyno); return open(buf, a1, 0); } static uintptr_t syz_fuse_mount(uintptr_t a0, uintptr_t a1, uintptr_t a2, uintptr_t a3, uintptr_t a4, uintptr_t a5) { uint64_t target = a0; uint64_t mode = a1; uint64_t uid = a2; uint64_t gid = a3; uint64_t maxread = a4; uint64_t flags = a5; int fd = open("/dev/fuse", O_RDWR); if (fd == -1) return fd; char buf[1024]; sprintf(buf, "fd=%d,user_id=%ld,group_id=%ld,rootmode=0%o", fd, (long)uid, (long)gid, (unsigned)mode & ~3u); if (maxread != 0) sprintf(buf + strlen(buf), ",max_read=%ld", (long)maxread); if (mode & 1) strcat(buf, ",default_permissions"); if (mode & 2) strcat(buf, ",allow_other"); syscall(SYS_mount, "", target, "fuse", flags, buf); return fd; } static uintptr_t syz_fuseblk_mount(uintptr_t a0, uintptr_t a1, uintptr_t a2, uintptr_t a3, uintptr_t a4, uintptr_t a5, uintptr_t a6, uintptr_t a7) { uint64_t target = a0; uint64_t blkdev = a1; uint64_t mode = a2; uint64_t uid = a3; uint64_t gid = a4; uint64_t maxread = a5; uint64_t blksize = a6; uint64_t flags = a7; int fd = open("/dev/fuse", O_RDWR); if (fd == -1) return fd; if (syscall(SYS_mknodat, AT_FDCWD, blkdev, S_IFBLK, makedev(7, 199))) return fd; char buf[256]; sprintf(buf, "fd=%d,user_id=%ld,group_id=%ld,rootmode=0%o", fd, (long)uid, (long)gid, (unsigned)mode & ~3u); if (maxread != 0) sprintf(buf + strlen(buf), ",max_read=%ld", (long)maxread); if (blksize != 0) sprintf(buf + strlen(buf), ",blksize=%ld", (long)blksize); if (mode & 1) strcat(buf, ",default_permissions"); if (mode & 2) strcat(buf, ",allow_other"); syscall(SYS_mount, blkdev, target, "fuseblk", flags, buf); return fd; } static uintptr_t execute_syscall(int nr, uintptr_t a0, uintptr_t a1, uintptr_t a2, uintptr_t a3, uintptr_t a4, uintptr_t a5, uintptr_t a6, uintptr_t a7, uintptr_t a8) { switch (nr) { default: return syscall(nr, a0, a1, a2, a3, a4, a5); case __NR_syz_open_dev: return syz_open_dev(a0, a1, a2); case __NR_syz_open_pts: return syz_open_pts(a0, a1); case __NR_syz_fuse_mount: return syz_fuse_mount(a0, a1, a2, a3, a4, a5); case __NR_syz_fuseblk_mount: return syz_fuseblk_mount(a0, a1, a2, a3, a4, a5, a6, a7); } } long r[42]; int main() { install_segv_handler(); memset(r, -1, sizeof(r)); r[0] = execute_syscall(__NR_mmap, 0x20000000ul, 0xd000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul, 0, 0, 0); NONFAILING(memcpy( (void*)0x20003000, "\x2f\x64\x65\x76\x2f\x73\x6e\x64\x2f\x74\x69\x6d\x65\x72\x00", 15)); r[2] = execute_syscall(__NR_syz_open_dev, 0x20003000ul, 0x0ul, 0x1ul, 0, 0, 0, 0, 0, 0); NONFAILING(*(uint32_t*)0x2000bfcc = (uint32_t)0x1); NONFAILING(*(uint32_t*)0x2000bfd0 = (uint32_t)0x4); NONFAILING(*(uint32_t*)0x2000bfd4 = (uint32_t)0x2); NONFAILING(*(uint32_t*)0x2000bfd8 = (uint32_t)0x3); NONFAILING(*(uint32_t*)0x2000bfdc = (uint32_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe0 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe1 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe2 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe3 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe4 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe5 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe6 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe7 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe8 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfe9 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfea = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfeb = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfec = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfed = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfee = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfef = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff0 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff1 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff2 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff3 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff4 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff5 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff6 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff7 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff8 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bff9 = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bffa = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bffb = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bffc = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bffd = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bffe = (uint8_t)0x0); NONFAILING(*(uint8_t*)0x2000bfff = (uint8_t)0x0); r[40] = execute_syscall(__NR_ioctl, r[2], 0x40345410ul, 0x2000bfccul, 0, 0, 0, 0, 0, 0); r[41] = execute_syscall(__NR_ioctl, r[2], 0x54a2ul, 0, 0, 0, 0, 0, 0, 0); return 0; }

2 1

[alsa-devel] [PATCH v2 1/1] ASoC: Intel: Atom: add a missing star in a memcpy call
by Nicolas Iooss 28 Aug '16

28 Aug '16

In sst_prepare_and_post_msg(), when a response is received in "block", the following code gets executed: *data = kzalloc(block->size, GFP_KERNEL); memcpy(data, (void *) block->data, block->size); The memcpy() call overwrites the content of the *data pointer instead of filling the newly-allocated memory (which pointer is hold by *data). Fix this by merging kzalloc+memcpy into a single kmemdup() call. Thanks Joe Perches for suggesting using kmemdup() Fixes: 60dc8dbacb00 ("ASoC: Intel: sst: Add some helper functions") Cc: stable(a)vger.kernel.org # 3.19.x Signed-off-by: Nicolas Iooss <nicolas.iooss_linux(a)m4x.org> --- sound/soc/intel/atom/sst/sst_pvt.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index adb32fefd693..b1e6b8f34a6a 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -279,17 +279,15 @@ int sst_prepare_and_post_msg(struct intel_sst_drv *sst, if (response) { ret = sst_wait_timeout(sst, block); - if (ret < 0) { + if (ret < 0) goto out; - } else if(block->data) { - if (!data) - goto out; - *data = kzalloc(block->size, GFP_KERNEL); - if (!(*data)) { + + if (data && block->data) { + *data = kmemdup(block->data, block->size, GFP_KERNEL); + if (!*data) { ret = -ENOMEM; goto out; - } else - memcpy(data, (void *) block->data, block->size); + } } } out: -- 2.9.3

1 0

[alsa-devel] [PATCH] ARM: dts: am335x-wega: Add audio support
by Stefan Müller-Klieser 26 Aug '16

26 Aug '16

The wega board has a TLV320AIC3007 connected via McASP0. In the default configuration, no external crystal is mounted. We run a system clock of 25 MHz, so we use the audio codec PLL for audio clock generation. Signed-off-by: Stefan Müller-Klieser <s.mueller-klieser(a)phytec.de> --- arch/arm/boot/dts/am335x-wega.dtsi | 62 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 62 insertions(+) diff --git a/arch/arm/boot/dts/am335x-wega.dtsi b/arch/arm/boot/dts/am335x-wega.dtsi index 282f6d4..7f5f8ed 100644 --- a/arch/arm/boot/dts/am335x-wega.dtsi +++ b/arch/arm/boot/dts/am335x-wega.dtsi @@ -11,6 +11,10 @@ model = "Phytec AM335x phyBOARD-WEGA"; compatible = "phytec,am335x-wega", "phytec,am335x-phycore-som", "ti,am33xx"; + sound: sound_iface { + compatible = "ti,da830-evm-audio"; + }; + regulators { compatible = "simple-bus"; @@ -24,6 +28,58 @@ }; }; +/* Audio */ +&am33xx_pinmux { + mcasp0_pins: pinmux_mcasp0 { + pinctrl-single,pins = < + AM33XX_IOPAD(0x9AC, PIN_OUTPUT_PULLDOWN | MUX_MODE0) /* mcasp0_ahclkx.mcasp0_ahclkx */ + AM33XX_IOPAD(0x990, PIN_INPUT_PULLDOWN | MUX_MODE0) /* mcasp0_aclkx.mcasp0_aclkx */ + AM33XX_IOPAD(0x994, PIN_INPUT_PULLDOWN | MUX_MODE0) /* mcasp0_fsx.mcasp0_fsx */ + AM33XX_IOPAD(0x998, PIN_INPUT_PULLDOWN | MUX_MODE0) /* mcasp0_axr0.mcasp0_axr0 */ + AM33XX_IOPAD(0x9A8, PIN_OUTPUT_PULLDOWN | MUX_MODE0) /* mcasp0_axr1.mcasp0_axr1 */ + >; + }; +}; + +&i2c0 { + tlv320aic3007: tlv320aic3007@18 { + compatible = "ti,tlv320aic3007"; + reg = <0x18>; + AVDD-supply = <&vcc3v3>; + IOVDD-supply = <&vcc3v3>; + DRVDD-supply = <&vcc3v3>; + DVDD-supply = <&vdig1_reg>; + status = "okay"; + }; +}; + +&mcasp0 { + pinctrl-names = "default"; + pinctrl-0 = <&mcasp0_pins>; + op-mode = <0>; /* DAVINCI_MCASP_IIS_MODE */ + tdm-slots = <2>; + serial-dir = < + 2 1 0 0 /* # 0: INACTIVE, 1: TX, 2: RX */ + >; + tx-num-evt = <16>; + rt-num-evt = <16>; + status = "okay"; +}; + +&sound { + ti,model = "AM335x-Wega"; + ti,audio-codec = <&tlv320aic3007>; + ti,mcasp-controller = <&mcasp0>; + ti,audio-routing = + "Line Out", "LLOUT", + "Line Out", "RLOUT", + "LINE1L", "Line In", + "LINE1R", "Line In"; + clocks = <&mcasp0_fck>; + clock-names = "mclk"; + status = "okay"; +}; + /* CAN Busses */ &am33xx_pinmux { dcan1_pins: pinmux_dcan1 { @@ -99,6 +155,12 @@ status = "okay"; }; +/* Power */ +&vdig1_reg { + regulator-boot-on; + regulator-always-on; +}; + /* UARTs */ &am33xx_pinmux { uart0_pins: pinmux_uart0 { -- 1.9.1

3 2

[alsa-devel] Toggle GPIO if ALSA playbacks begins or ends
by Jörg Krause 26 Aug '16

26 Aug '16

Hi, I am looking for a way to toggle a GPIO if ALSA is about to play or has stopped to turn an external amplifier on/off. Can it be done using libasound or can I use a plugin or write one? Best regards Jörg Krause

1 1

[alsa-devel] [PATCH v9 0/2] sun4i-codec: Add Line-In, FM-In, Mic 2, Capture Source, Differential Line-In
by Danny Milosavljevic 26 Aug '16

26 Aug '16

This patch adds support for some mixer controls: - Line-In - FM-In - Mic 2 - Capture Source - Differential Line-In v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (2): ASoC: sun4i-codec: Distinguish sun4i from sun7i Add mixer controls: Line-In, FM-In, Mic 2, Capture Source, Differential Line-In. sound/soc/sunxi/sun4i-codec.c | 278 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 259 insertions(+), 19 deletions(-)

2 3

Re: [alsa-devel] [PATCH 8/9] ASoC: simple-card: code sync: rename num_link to num
by Kuninori Morimoto 26 Aug '16

26 Aug '16

Hi Mark This is my fault. I will post v2 patch. > [auto build test ERROR on asoc/for-next] > [also build test ERROR on next-20160824] > [cannot apply to v4.8-rc3] > [if your patch is applied to the wrong git tree, please drop us a note to help improve the system] > [Suggest to use git(>=2.9.0) format-patch --base=<commit> (or --base=auto for convenience) to record what (public, well-known) commit your patch series was built on] > [Check https://git-scm.com/docs/git-format-patch for more information] > > url: https://github.com/0day-ci/linux/commits/Kuninori-Morimoto/ASoC-simple-card… > base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next > config: i386-randconfig-x011-201634 (attached as .config) > compiler: gcc-6 (Debian 6.1.1-9) 6.1.1 20160705 > reproduce: > # save the attached .config to linux build tree > make ARCH=i386 > > Note: the linux-review/Kuninori-Morimoto/ASoC-simple-card-tidyup-and-synchronize-patches/20160825-100058 HEAD 36fb10f61ffb92eb287328e22cad00a486495747 builds fine. > It only hurts bisectibility. > > All errors (new ones prefixed by >>): > > sound/soc/generic/simple-card.c: In function 'asoc_simple_card_probe': > >> sound/soc/generic/simple-card.c:398:40: error: 'num_links' undeclared (first use in this function) > sizeof(*priv) + sizeof(*dai_link) * num_links, > ^~~~~~~~~ > sound/soc/generic/simple-card.c:398:40: note: each undeclared identifier is reported only once for each function it appears in > > vim +/num_links +398 sound/soc/generic/simple-card.c > > 6805a759b Kuninori Morimoto 2016-08-25 392 num = of_get_child_count(np); > 2080437d3 Xiubo Li 2014-09-03 393 else > 6805a759b Kuninori Morimoto 2016-08-25 394 num = 1; > f2390880e Kuninori Morimoto 2012-04-08 395 > 0dd4fc3c2 Xiubo Li 2014-09-10 396 /* Allocate the private data and the DAI link array */ > cf7dc23cb Jean-Francois Moine 2014-03-20 397 priv = devm_kzalloc(dev, > 6a91a17bd Jean-Francois Moine 2014-03-20 @398 sizeof(*priv) + sizeof(*dai_link) * num_links, > cf7dc23cb Jean-Francois Moine 2014-03-20 399 GFP_KERNEL); > ca65b492c Jean-Francois Moine 2014-01-15 400 if (!priv) > ca919fe4b Xiubo Li 2014-01-14 401 return -ENOMEM; > > :::::: The code at line 398 was first introduced by commit > :::::: 6a91a17bd7b92b2d2aa9ece85457f52a62fd7708 ASoC: simple-card: Handle many DAI links > > :::::: TO: Jean-Francois Moine <moinejf(a)free.fr> > :::::: CC: Mark Brown <broonie(a)linaro.org> > > --- > 0-DAY kernel test infrastructure Open Source Technology Center > https://lists.01.org/pipermail/kbuild-all Intel Corporation > [2 .config.gz <application/octet-stream (base64)>] > Best regards --- Kuninori Morimoto

1 0

[alsa-devel] ALSA hooks: dlopen failed: snd_dlsym_start: symbol not found
by Jörg Krause 26 Aug '16

26 Aug '16

Hi, I am trying to build a custom hook library. The minimal hook is: #include <alsa/asoundlib.h> int _snd_pcm_hook_audio_status_install(snd_pcm_t *pcm, snd_config_t *conf) { return 0; } SND_DLSYM_BUILD_VERSION(_snd_pcm_hook_audio_status_install, SND_CONFIG_DLSYM_VERSION_HOOK); I compile this as a shared library: $(CC) -fPIC -lasound -shared -Wl,-soname,lib$(NAME).so $^ -o $@ If I try to load this library using aplay I get the following error: dlopen failed: Error relocating /usr/lib/alsa-lib/libaudiostatus.so: snd_dlsym_start: symbol not found This is the output of the preprocessor: int _snd_pcm_hook_audio_status_install(snd_pcm_t *pcm, snd_config_t *conf) { return 0; } static struct snd_dlsym_link _snd_dlsym__snd_pcm_hook_audio_status_install_dlsym_config_hook_001; void _snd_dlsym_constructor__snd_pcm_hook_audio_status_install_dlsym_conf ig_hook_001 (void) __attribute__ ((constructor)); void _snd_dlsym_constructor__snd_pcm_hook_audio_status_install_dlsym_conf ig_hook_001 (void) { _snd_dlsym__snd_pcm_hook_audio_status_install_dlsym_config_hook _001.next = snd_dlsym_start; _snd_dlsym__snd_pcm_hook_audio_status_install_dlsym_config_hook _001.dlsym_name = "_snd_pcm_hook_audio_status_install"; _snd_dlsym__snd_pcm_hook_audio_status_install_dlsym_config_hook _001.dlsym_ptr = (void *)&_snd_pcm_hook_audio_status_install; snd_dlsym_start = &_snd_dlsym__snd_pcm_hook_audio_status_install_dlsym_config_hook_001 ; }; What do I miss? Best regards Jörg Krause

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[alsa-devel] Alsa-lib 'cancelability'
by Alan Horstmann 25 Aug '16

25 Aug '16

Hi, I am wondering if there is a policy on the thread 'cancelability' / 'cancellability' of the functions in Alsa-lib? We are dissecting an issue on the Portaudio list which has appeared with the 'Locking API' version 1.1.2, when using the Portaudio callback API. The callback runs in it's own thread; when a user requests 'AbortStream' a thread cancel is requested. Sometimes it seems this can take effect during snd_pcm_mmap_commit(), so the function never completes and a lock is not released. Then, during the cleanup, snd_pcm_drop() attempts to aquire the lock, which is already held, and a lock-up occurs. Is it up to the application to ensure each Alsa function completes, eg by disabling cancelability on the thread for those periods, or is it reasonable to expect the functions to ensure they complete adequately? Any comment would be appreciated. Regards Alan

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[alsa-devel] [GIT PULL] ASoC fixes for v4.8
by Mark Brown 25 Aug '16

25 Aug '16

The following changes since commit fa8410b355251fd30341662a40ac6b22d3e38468: Linux 4.8-rc3 (2016-08-21 16:14:10 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.8-rc4 for you to fetch changes up to cfb89f2e7505c6823020a18bbdc5410284305234: Merge remote-tracking branches 'asoc/fix/max98371', 'asoc/fix/nau8825', 'asoc/fix/omap', 'asoc/fix/samsung', 'asoc/fix/simple' and 'asoc/fix/wm2000' into asoc-linus (2016-08-24 19:05:25 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.8 A clutch of fixes for v4.8. These are mainly driver specific, the most notable ones being those for OMAP which fix a series of issues that broke boot on some platforms there when deferred probe kicked in. There's also one core fix for an issue when unbinding a card which for some reason had managed to not manifest until recently. ---------------------------------------------------------------- Adam Thomson (1): ASoC: da7213: Default to 64 BCLKs per WCLK to support all formats Charles Keepax (3): ASoC: Fix leak of rtd in soc_bind_dai_link ASoC: wm2000: Fix return of uninitialised varible ASoC: compress: Fix leak of a widget list in soc_compr_open_fe Christoph Huber (1): ASoC: atmel_ssc_dai: Don't unconditionally reset SSC on stream startup Hiroyuki Yokoyama (1): ASoC: rsnd: Fixup SRCm_IFSVR calculate method John Hsu (2): ASoC: nau8825: fix bug in playback when suspend ASoC: nau8825: fix static check error about semaphone control Kuninori Morimoto (1): ASoC: simple-card-utils: add missing MODULE_xxx() Mark Brown (7): ASoC: core: Clean up DAPM before the card debugfs Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/compress', 'asoc/fix/da7213' and 'asoc/fix/debugfs' into asoc-linus Merge remote-tracking branches 'asoc/fix/max98371', 'asoc/fix/nau8825', 'asoc/fix/omap', 'asoc/fix/samsung', 'asoc/fix/simple' and 'asoc/fix/wm2000' into asoc-linus Nicolin Chen (1): ASoC: dapm: Add a dummy snd_pcm_runtime to avoid NULL pointer access Peter Ujfalusi (3): ASoC: omap-mcpdm: Drop pdmclk clock handling ASoC: omap-abe-twl6040: Correct dmic-codec device registration ASoC: omap-mcpdm: Fix irq resource handling Sylwester Nawrocki (1): ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card Vinod Koul (1): ASoC: Intel: Skylake: Check list empty while getting module info Wei Yongjun (2): ASoC: Intel: Skylake: Fix error return code in skl_probe() ASoC: max98371: Add terminate entry for i2c_device_id tables .../devicetree/bindings/sound/omap-mcpdm.txt | 10 --- sound/soc/atmel/atmel_ssc_dai.c | 5 +- sound/soc/codecs/da7213.c | 4 +- sound/soc/codecs/max98371.c | 1 + sound/soc/codecs/nau8825.c | 78 +++++++++------------- sound/soc/codecs/wm2000.c | 2 +- sound/soc/generic/Makefile | 6 +- sound/soc/generic/simple-card-utils.c | 6 ++ sound/soc/intel/skylake/skl-sst-utils.c | 5 ++ sound/soc/intel/skylake/skl.c | 4 +- sound/soc/omap/omap-abe-twl6040.c | 61 +++++++++-------- sound/soc/omap/omap-mcpdm.c | 22 +----- sound/soc/samsung/s3c24xx_uda134x.c | 7 +- sound/soc/sh/rcar/src.c | 6 +- sound/soc/soc-compress.c | 4 +- sound/soc/soc-core.c | 5 +- sound/soc/soc-dapm.c | 10 +++ 17 files changed, 115 insertions(+), 121 deletions(-)

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[alsa-devel] [PATCH] ALSA: snd-aoa: enable sound on PowerBook G4 12"
by Aaro Koskinen 25 Aug '16

25 Aug '16

Enable sound on PowerBook G4 12". Signed-off-by: Aaro Koskinen <aaro.koskinen(a)iki.fi> --- sound/aoa/fabrics/layout.c | 8 ++++++++ sound/aoa/soundbus/i2sbus/core.c | 2 +- 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 8f71f7e..edc8681 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -112,6 +112,7 @@ MODULE_ALIAS("sound-layout-100"); MODULE_ALIAS("aoa-device-id-14"); MODULE_ALIAS("aoa-device-id-22"); +MODULE_ALIAS("aoa-device-id-31"); MODULE_ALIAS("aoa-device-id-35"); MODULE_ALIAS("aoa-device-id-44"); @@ -362,6 +363,13 @@ static struct layout layouts[] = { .connections = tas_connections_nolineout, }, }, + /* PowerBook6,1 */ + { .device_id = 31, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nolineout, + }, + }, /* PowerBook6,5 */ { .device_id = 44, .codecs[0] = { diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 1cbf210..000b585 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -197,7 +197,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, * so restrict to those we do handle for now. */ if (id && (*id == 22 || *id == 14 || *id == 35 || - *id == 44)) { + *id == 31 || *id == 44)) { snprintf(dev->sound.modalias, 32, "aoa-device-id-%d", *id); ok = 1; -- 2.9.2

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[alsa-devel] [RFC PATCH] mfd: arizona: Add gating of external MCLKn clocks
by Sylwester Nawrocki 25 Aug '16

25 Aug '16

This patch adds handling of the CODEC's external MCLK{1,2} clocks needed for board configurations where these clocks are not always on oscillators. The 32k source MCLKn clock is basically kept permanently enabled while the other MCLKn clock is being gated in the MFD's runtime suspend/resume callbacks. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- I'm not sure it's a correct approach hence only an RFC patch. --- drivers/mfd/arizona-core.c | 105 ++++++++++++++++++++++++++++++++------- include/linux/mfd/arizona/core.h | 9 ++++ 2 files changed, 96 insertions(+), 18 deletions(-) diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index e4f97b3..a733b61 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -10,6 +10,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/delay.h> #include <linux/err.h> #include <linux/gpio.h> @@ -49,7 +50,15 @@ int arizona_clk32k_enable(struct arizona *arizona) case ARIZONA_32KZ_MCLK1: ret = pm_runtime_get_sync(arizona->dev); if (ret != 0) - goto out; + goto err_ref; + ret = clk_prepare_enable(arizona->mclk[ARIZONA_MCLK1]); + if (ret != 0) + goto err_pm; + break; + case ARIZONA_32KZ_MCLK2: + ret = clk_prepare_enable(arizona->mclk[ARIZONA_MCLK2]); + if (ret != 0) + goto err_ref; break; } @@ -58,7 +67,9 @@ int arizona_clk32k_enable(struct arizona *arizona) ARIZONA_CLK_32K_ENA); } -out: +err_pm: + pm_runtime_put_sync(arizona->dev); +err_ref: if (ret != 0) arizona->clk32k_ref--; @@ -83,6 +94,10 @@ int arizona_clk32k_disable(struct arizona *arizona) switch (arizona->pdata.clk32k_src) { case ARIZONA_32KZ_MCLK1: pm_runtime_put_sync(arizona->dev); + clk_disable_unprepare(arizona->mclk[ARIZONA_MCLK1]); + break; + case ARIZONA_32KZ_MCLK2: + clk_disable_unprepare(arizona->mclk[ARIZONA_MCLK2]); break; } } @@ -93,6 +108,34 @@ int arizona_clk32k_disable(struct arizona *arizona) } EXPORT_SYMBOL_GPL(arizona_clk32k_disable); +/* Enable all external MCLKn clocks except the 32k source clock */ +int arizona_mclk_enable(struct arizona *arizona) +{ + int ret; + + if (arizona->pdata.clk32k_src != ARIZONA_32KZ_MCLK1) { + ret = clk_prepare_enable(arizona->mclk[ARIZONA_MCLK1]); + if (ret != 0) + return ret; + } + if (arizona->pdata.clk32k_src != ARIZONA_32KZ_MCLK2) { + ret = clk_prepare_enable(arizona->mclk[ARIZONA_MCLK2]); + if (ret != 0) + return ret; + } + return 0; +} + +/* Disable all external MCLKn clocks except the 32k source clock */ +void arizona_mclk_disable(struct arizona *arizona) +{ + if (arizona->pdata.clk32k_src != ARIZONA_32KZ_MCLK1) + clk_disable_unprepare(arizona->mclk[ARIZONA_MCLK1]); + + if (arizona->pdata.clk32k_src != ARIZONA_32KZ_MCLK2) + clk_disable_unprepare(arizona->mclk[ARIZONA_MCLK2]); +} + static irqreturn_t arizona_clkgen_err(int irq, void *data) { struct arizona *arizona = data; @@ -533,6 +576,12 @@ static int arizona_runtime_resume(struct device *dev) return ret; } + ret = arizona_mclk_enable(arizona); + if (ret != 0) { + dev_err(dev, "Failed to enable mclk: %d\n", ret); + goto err_reg; + } + if (arizona->has_fully_powered_off) { arizona_disable_reset(arizona); enable_irq(arizona->irq); @@ -546,14 +595,14 @@ static int arizona_runtime_resume(struct device *dev) if (arizona->external_dcvdd) { ret = arizona_connect_dcvdd(arizona); if (ret != 0) - goto err; + goto err_clk; } ret = wm5102_patch(arizona); if (ret != 0) { dev_err(arizona->dev, "Failed to apply patch: %d\n", ret); - goto err; + goto err_clk; } ret = wm5102_apply_hardware_patch(arizona); @@ -561,19 +610,19 @@ static int arizona_runtime_resume(struct device *dev) dev_err(arizona->dev, "Failed to apply hardware patch: %d\n", ret); - goto err; + goto err_clk; } break; case WM5110: case WM8280: ret = arizona_wait_for_boot(arizona); if (ret) - goto err; + goto err_clk; if (arizona->external_dcvdd) { ret = arizona_connect_dcvdd(arizona); if (ret != 0) - goto err; + goto err_clk; } else { /* * As this is only called for the internal regulator @@ -586,7 +635,7 @@ static int arizona_runtime_resume(struct device *dev) dev_err(arizona->dev, "Failed to set resume voltage: %d\n", ret); - goto err; + goto err_clk; } } @@ -595,24 +644,24 @@ static int arizona_runtime_resume(struct device *dev) dev_err(arizona->dev, "Failed to re-apply sleep patch: %d\n", ret); - goto err; + goto err_clk; } break; case WM1831: case CS47L24: ret = arizona_wait_for_boot(arizona); if (ret != 0) - goto err; + goto err_clk; break; default: ret = arizona_wait_for_boot(arizona); if (ret != 0) - goto err; + goto err_clk; if (arizona->external_dcvdd) { ret = arizona_connect_dcvdd(arizona); if (ret != 0) - goto err; + goto err_clk; } break; } @@ -620,12 +669,14 @@ static int arizona_runtime_resume(struct device *dev) ret = regcache_sync(arizona->regmap); if (ret != 0) { dev_err(arizona->dev, "Failed to restore register cache\n"); - goto err; + goto err_clk; } return 0; -err: +err_clk: + arizona_mclk_disable(arizona); +err_reg: regcache_cache_only(arizona->regmap, true); regulator_disable(arizona->dcvdd); return ret; @@ -707,6 +758,7 @@ static int arizona_runtime_suspend(struct device *dev) regcache_cache_only(arizona->regmap, true); regcache_mark_dirty(arizona->regmap); regulator_disable(arizona->dcvdd); + arizona_mclk_disable(arizona); /* Allow us to completely power down if no jack detection */ if (!jd_active) { @@ -1000,6 +1052,7 @@ static const struct mfd_cell wm8998_devs[] = { int arizona_dev_init(struct arizona *arizona) { + const char * const mclk_name[] = { "mclk1", "mclk2" }; struct device *dev = arizona->dev; const char *type_name = NULL; unsigned int reg, val, mask; @@ -1016,6 +1069,15 @@ int arizona_dev_init(struct arizona *arizona) else arizona_of_get_core_pdata(arizona); + for (i = 0; i < ARRAY_SIZE(arizona->mclk); i++) { + arizona->mclk[i] = devm_clk_get(arizona->dev, mclk_name[i]); + if (IS_ERR(arizona->mclk[i])) { + dev_info(arizona->dev, "Failed to get mclk%d: %ld\n", + i, PTR_ERR(arizona->mclk[i])); + arizona->mclk[i] = NULL; + } + } + regcache_cache_only(arizona->regmap, true); switch (arizona->type) { @@ -1101,6 +1163,16 @@ int arizona_dev_init(struct arizona *arizona) goto err_enable; } + /* Chip default */ + if (!arizona->pdata.clk32k_src) + arizona->pdata.clk32k_src = ARIZONA_32KZ_MCLK2; + + ret = arizona_mclk_enable(arizona); + if (ret != 0) { + dev_err(dev, "Failed to enable mclk: %d\n", ret); + goto err_enable; + } + arizona_disable_reset(arizona); regcache_cache_only(arizona->regmap, false); @@ -1332,10 +1404,6 @@ int arizona_dev_init(struct arizona *arizona) arizona->pdata.gpio_defaults[i]); } - /* Chip default */ - if (!arizona->pdata.clk32k_src) - arizona->pdata.clk32k_src = ARIZONA_32KZ_MCLK2; - switch (arizona->pdata.clk32k_src) { case ARIZONA_32KZ_MCLK1: case ARIZONA_32KZ_MCLK2: @@ -1490,6 +1558,7 @@ err_pm: pm_runtime_disable(arizona->dev); err_reset: arizona_enable_reset(arizona); + arizona_mclk_disable(arizona); regulator_disable(arizona->dcvdd); err_enable: regulator_bulk_disable(arizona->num_core_supplies, diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 58ab4c0..b9909bb 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -13,6 +13,7 @@ #ifndef _WM_ARIZONA_CORE_H #define _WM_ARIZONA_CORE_H +#include <linux/clk.h> #include <linux/interrupt.h> #include <linux/notifier.h> #include <linux/regmap.h> @@ -21,6 +22,12 @@ #define ARIZONA_MAX_CORE_SUPPLIES 2 +enum { + ARIZONA_MCLK1, + ARIZONA_MCLK2, + ARIZONA_NUM_MCLK +}; + enum arizona_type { WM5102 = 1, WM5110 = 2, @@ -139,6 +146,8 @@ struct arizona { struct mutex clk_lock; int clk32k_ref; + struct clk *mclk[ARIZONA_NUM_MCLK]; + bool ctrlif_error; struct snd_soc_dapm_context *dapm; -- 1.9.1

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