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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

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[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

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[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

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[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

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[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

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[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(-)

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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

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[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

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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.

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[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

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[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

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[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

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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

2 26

[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

3 4

[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 29 Aug '16

29 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.

2 10

[alsa-devel] [PATCH 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 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

3 8

[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(-)

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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

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[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

3 3

[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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[alsa-devel] [PATCH 0/8] ASoC: simple-scu-card: tidyup and synchronize patches
by Kuninori Morimoto 25 Aug '16

25 Aug '16

Hi Mark These are cleanup/tidyup and synchronize patches for simple-scu-card. We have now simple-card and simple-scu-card. These are almost same, difference is very small. But, because of historical reason, these have "code" difference. This patch cleanup and synchronize these. These are based on mark/topic/rcar Kuninori Morimoto (8): ASoC: simple-scu-card: remove unused definitions ASoC: simple-scu-card: is GPL v2 ASoC: simple-scu-card: tidyup codec daifmt handling method ASoC: simple-scu-card: tidyup asoc_simple_card_parse_links() method ASoC: simple-scu-card: depends on CONFIG_OF ASoC: simple-scu-card: add 2 CPU 1 Codec example ASoC: simple-scu-card: code sync: move asoc_simple_card_of_match ASoC: simple-scu-card: code sync: tidyup TDM setting position .../devicetree/bindings/sound/simple-scu-card.txt | 32 +++++++++- sound/soc/generic/Kconfig | 1 + sound/soc/generic/simple-scu-card.c | 72 +++++++++------------- 3 files changed, 60 insertions(+), 45 deletions(-) -- 1.9.1

1 8

[alsa-devel] [PATCH 0/9] ASoC: simple-card: tidyup and synchronize patches
by Kuninori Morimoto 25 Aug '16

25 Aug '16

Hi Mark These are cleanup/tidyup and synchronize patches for simple-card. We have now simple-card and simple-scu-card. These are almost same, difference is very small. But, because of historical reason, these have "code" difference. This patch cleanup and synchronize these. Kuninori Morimoto (9): ASoC: simple-card: call of_node_put() for dai_link ASoC: simple-card: tidyup mclk-fs of_property_read_u32() usage ASoC: simple-card: code sync: tidyup simple_priv_to_xxx() macro ASoC: simple-card: code sync: use simple_priv_to_props() macro ASoC: simple-card: is GPL v2 ASoC: simple-card: code sync: tidyup white line ASoC: simple-card: code sync: use tag ASoC: simple-card: code sync: rename num_link to num ASoC: simple-card: use kzalloc() for dai_props / dai_link sound/soc/generic/simple-card.c | 87 ++++++++++++++++++++--------------------- 1 file changed, 43 insertions(+), 44 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: nau8810: Fix memory leak in nau8810_eq_put error path
by Axel Lin 25 Aug '16

25 Aug '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/nau8810.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index f9bcdc3..e455186 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -221,6 +221,7 @@ static int nau8810_eq_put(struct snd_kcontrol *kcontrol, if (ret) { dev_err(codec->dev, "EQ configuration fail, register: %x ret: %d\n", reg + i, ret); + kfree(data); return ret; } } -- 2.7.4

1 0

[alsa-devel] [PATCH v2 1/2] ASoC: simple-card-utils: add __printf attribute
by Nicolas Iooss 25 Aug '16

25 Aug '16

asoc_simple_card_set_dailink_name() uses devm_kvasprintf() to format some of its arguments. Adding a __printf attribute to this function makes it possible to detect at compile-time errors related to format strings. Signed-off-by: Nicolas Iooss <nicolas.iooss_linux(a)m4x.org> --- include/sound/simple_card_utils.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 86088aed9002..3207b1a70d38 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -27,6 +27,7 @@ int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *codec, char *prefix, unsigned int *retfmt); +__printf(3, 4) int asoc_simple_card_set_dailink_name(struct device *dev, struct snd_soc_dai_link *dai_link, const char *fmt, ...); -- 2.9.3

2 2

[alsa-devel] [PATCH] ASoC: tegra_rt5640: Correct a copy and paste typo in the comments
by Nicolin Chen 25 Aug '16

25 Aug '16

This patch corrects a copy and paste typo. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/tegra/tegra_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 773daec..e5ef4e9 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -1,5 +1,5 @@ /* -* tegra_rt5640.c - Tegra machine ASoC driver for boards using WM8903 codec. +* tegra_rt5640.c - Tegra machine ASoC driver for boards using RT5640 codec. * * Copyright (c) 2013, NVIDIA CORPORATION. All rights reserved. * -- 2.1.4

1 0

[alsa-devel] Spam in the wiki
by Tanu Kaskinen 24 Aug '16

24 Aug '16

I got a notification, because a wiki page that I had created was edited by a spammer. Looking at the recent changes, it seems that there's quite a lot of spam activity: http://www.alsa-project.org/main/index.php/Special:RecentChanges Can the wiki user registrations be restricted? There's already a captcha, but that doesn't seem very effective. -- Tanu

1 0

[alsa-devel] [PATCH] ASoC: rt5659: Enable IRQ output for GPIO1 pin
by Nicolin Chen 24 Aug '16

24 Aug '16

Since it's possible to have an IRQ without enabling JD3, the GPIO1 pin then would remain its default GPIO function which is set as an input direction (seeing from rt5659). Meanwhile, CPU would also listen this connection via its own GPIO input: [input] [input] CPU GPIO <--------> RT5659 GPIO1 The result for the IRQ on the CPU side will be unexpectable. So this patch enables the IRQ output for GPIO1 pin any way as long as there's an IRQ assigned from platform data or DT: [input] [IRQ output] CPU GPIO <--------< RT5659 GPIO1 Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/codecs/rt5659.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 4c0d2b8..db54550 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -4189,6 +4189,9 @@ static int rt5659_i2c_probe(struct i2c_client *i2c, if (ret) dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + /* Enable IRQ output for GPIO1 pin any way */ + regmap_update_bits(rt5659->regmap, RT5659_GPIO_CTRL_1, + RT5659_GP1_PIN_MASK, RT5659_GP1_PIN_IRQ); } return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5659, -- 2.1.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: simple-card-utils: add __printf attribute
by Nicolas Iooss 24 Aug '16

24 Aug '16

asoc_simple_card_set_dailink_name() uses devm_kvasprintf() to format some of its arguments. Adding a __printf attribute to this function makes it possible to detect at compile-time errors related to format strings. Signed-off-by: Nicolas Iooss <nicolas.iooss_linux(a)m4x.org> --- include/sound/simple_card_utils.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 86088aed9002..3207b1a70d38 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -27,6 +27,7 @@ int asoc_simple_card_parse_daifmt(struct device *dev, struct device_node *codec, char *prefix, unsigned int *retfmt); +__printf(3, 4) int asoc_simple_card_set_dailink_name(struct device *dev, struct snd_soc_dai_link *dai_link, const char *fmt, ...); -- 2.9.3

3 4

[alsa-devel] [PATCH -next] sound: rockchip: fix build when SPI_MASTER is not enabled
by Randy Dunlap 24 Aug '16

24 Aug '16

From: Randy Dunlap <rdunlap(a)infradead.org> Fix multiple build errors in sound/soc/rockchip/ drivers by adding "depends on SPI_MASTER". Fixes these build errors: sound/built-in.o: In function `rt5514_spi_burst_read': (.text+0x6c26c): undefined reference to `spi_sync' sound/built-in.o: In function `rt5514_spi_burst_write': (.text+0x6c96e): undefined reference to `spi_sync' sound/built-in.o: In function `rockchip_sound_probe': rk3399_gru_sound.c:(.text+0x9f579): undefined reference to `spi_bus_type' sound/built-in.o: In function `rt5514_spi_driver_init': rt5514-spi.c:(.init.text+0x983): undefined reference to `__spi_register_driver' Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Heiko Stuebner <heiko(a)sntech.de> Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-rockchip(a)lists.infradead.org --- sound/soc/rockchip/Kconfig | 1 + 1 file changed, 1 insertion(+) --- linux-next-20160824.orig/sound/soc/rockchip/Kconfig +++ linux-next-20160824/sound/soc/rockchip/Kconfig @@ -45,6 +45,7 @@ config SND_SOC_ROCKCHIP_RT5645 config SND_SOC_RK3399_GRU_SOUND tristate "ASoC support multiple codecs for Rockchip RK3399 GRU boards" depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP + depends on SPI_MASTER select SND_SOC_ROCKCHIP_I2S select SND_SOC_MAX98357A select SND_SOC_RT5514

1 0

[alsa-devel] [PATCH -next] sound: rockchip: fix build when SPI is not enabled
by Randy Dunlap 24 Aug '16

24 Aug '16

From: Randy Dunlap <rdunlap(a)infradead.org> Fix build errors found in rt5514 drivers when SPI is not enabled. Fixes these build errors: ERROR: "__spi_register_driver" [sound/soc/codecs/snd-soc-rt5514-spi.ko] undefined! ERROR: "spi_sync" [sound/soc/codecs/snd-soc-rt5514-spi.ko] undefined! Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/rockchip/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) This builds without errors, but if you always want to have the SPI interface available in this driver, change the first "depends" to include "&& SPI_MASTER". --- linux-next-20160822.orig/sound/soc/rockchip/Kconfig +++ linux-next-20160822/sound/soc/rockchip/Kconfig @@ -49,7 +49,7 @@ config SND_SOC_RK3399_GRU_SOUND select SND_SOC_MAX98357A select SND_SOC_RT5514 select SND_SOC_DA7219 - select SND_SOC_RT5514_SPI + select SND_SOC_RT5514_SPI if SPI_MASTER help Say Y or M here if you want to add support multiple codecs for SoC audio on Rockchip RK3399 GRU boards.

1 1

[alsa-devel] Questions about DAPM Routing for STA309A DSP Driver
by Mauricio Cirelli 24 Aug '16

24 Aug '16

Dear ALSA Community, I am new to this community and have a basic knowledge on ASoC architecture. I have worked a few times with Freescale's boards and the SGTL5000 Codec driver. Now, time has come for me to develop a ASoC Codec Driver for an Audio DSP (ST Microelectronics STA309A). Obviously, this driver should work only on the Playback stream. This chip supports up to 8 playback channels (I2S format) and has up to 8 output channels. There are also 8 internal channels, whose inputs can be one of the 8 playback channels and outputs are connected to one of each 8 output channels like this: Playback Channel X --> Input Channel Y --> Output Channel Z This DSP has several controls to apply effects and modify volume on each channel internal channel individually. I have been able to implement snd_kcontrol_new for them successfully and I am able to control them from the userspace correctly. The part I am still in doubt on how to proceed is related to this channel routing. From what I have researched, this could be designed on top of DAPM Widgets and DAPM Routes in a way that the machine-level driver could configure this routing according to the specific board and the userspace would not be required to deal with this stuff. I want to develop a device driver following the ASoC architecture so I can use it in different boards and in a way it could be useful for other developers. Here is what I was sketching up for this snd_soc_codec_driver: static const char *input_names[8] = { "DIN1", "DIN2", "DIN3", "DIN4", "DIN5", "DIN6", "DIN7", "DIN8" }; static const char *internal_names[8] = { "INT1", "INT2", "INT3", "INT4", "INT5", "INT6", "INT7", "INT8" }; /* Playback Channel to Internal DSP Channel Mapping */ static SOC_ENUM_SINGLE_DECL(int1mux, STA309A_C12IM, 0, input_names); static const struct snd_kcontrol_new int1mux_ctrl = SOC_DAPM_ENUM("INT1MUX", int1mux); .... replicated to all 8 internal channels /* Internal DSP Channel to Output Channel Mapping */ static SOC_ENUM_SINGLE_DECL(out1mux, STA309A_C12OM, 0, internal_names); static const struct snd_kcontrol_new out1mux_ctrl = SOC_DAPM_ENUM("OUT1MUX", out1mux); .... replicated to all 8 output channels static const struct snd_soc_dapm_widget sta309a_dsp_dapm_widgets[] = { SND_SOC_DAPM_MUX("INT1MUX", SND_SOC_NOPM, 0, 0, &int1mux_ctrl), .... added 8 internal channel muxes SND_SOC_DAPM_MUX("OUT1MUX", SND_SOC_NOPM, 0, 0, &out1mux_ctrl), .... added 8 output channel muxes }; Now I am lost on how to declare the possible routes. It should be possible to route each playback channel to any of the 8 possible internal channels and the internal channels to any of the 8 possible output channels. Note that this DSP supports Cross-Overs, so one can route a playback channel to one or more internal channels. Analogously, one can route an internal channel to one or more output channels. I have not started to develop the machine-level driver, but any thoughts on how to design it would be very helpful. In the machine-level driver, I would like to define the DSP routing for an specific board. How can I do it? Thank you very much. Mauricio Cirelli

1 0

[alsa-devel] [PATCH 0/2] ALSA: bebob: merge vendor-dependent codes for Yamaha/Terratec
by Takashi Sakamoto 24 Aug '16

24 Aug '16

Hi, This patchset merges vendor-dependent codes for Yamaha/Terratec, because each of these two pairs has the same configuration: - Yamaha GO44 / Terratec Phase 24 FW - Yamaha GO46 / Terratec Phase X24 FW The two companies might once have cooperated to develop some audio and music units on IEEE 1394 bus, based on DM1000E ASIC and BeBoB firmware. This conclusion comes from a reporter of a forum of Ubuntu Japanese Community. Thanks for his help to identify subunit formation and supported stream formats: Jack & Firewireオーディオインターフェイスの設定に関する質問です。 https://forums.ubuntulinux.jp/viewtopic.php?id=18657 Terratec has also produced the other models with DM ASIC series and BeBoB firmware. For the models, vendor-dependent codes are still left. Takashi Sakamoto (2): ALSA: bebob: rename file with vendor-dependent code so that it's for Yamaha/Terratec ALSA: bebob: unify configurations for some models produced by Yamaha/Terratec sound/firewire/bebob/Makefile | 5 ++-- sound/firewire/bebob/bebob.c | 8 +++--- sound/firewire/bebob/bebob.h | 3 +-- sound/firewire/bebob/bebob_terratec.c | 31 ---------------------- .../{bebob_yamaha.c => bebob_yamaha_terratec.c} | 9 ++++--- 5 files changed, 14 insertions(+), 42 deletions(-) rename sound/firewire/bebob/{bebob_yamaha.c => bebob_yamaha_terratec.c} (85%) -- 2.7.4

2 3

[alsa-devel] [PATCH -next] ASoC: samsung: i2s: Add missing clk_disable_unprepare() on error in samsung_i2s_probe()
by Wei Yongjun 24 Aug '16

24 Aug '16

Add the missing clk_disable_unprepare() before return from samsung_i2s_probe() in the error handling case. Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/samsung/i2s.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index fa3ff03..7e32cf4 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1318,7 +1318,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) sec_dai = i2s_alloc_dai(pdev, true); if (!sec_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_sec\n"); - return -ENOMEM; + ret = -ENOMEM; + goto err_disable_clk; } sec_dai->lock = &pri_dai->spinlock; @@ -1342,7 +1343,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (i2s_pdata && i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { dev_err(&pdev->dev, "Unable to configure gpio\n"); - return -EINVAL; + ret = -EINVAL; + goto err_disable_clk; } ret = devm_snd_soc_register_component(&pri_dai->pdev->dev, @@ -1366,6 +1368,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) err_free_dai: if (sec_dai) i2s_free_sec_dai(sec_dai); +err_disable_clk: + clk_disable_unprepare(pri_dai->clk); return ret; }

3 2

[alsa-devel] [PATCH] ASoC: tas5086: fix typo: "Inavlid" -> "Invalid"
by Colin King 24 Aug '16

24 Aug '16

From: Colin Ian King <colin.king(a)canonical.com> trivial typo fix in dev_err message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/tas5086.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index c297b9f..b7de857 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -387,7 +387,7 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream, val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios), priv->mclk / priv->rate); if (val < 0) { - dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n"); + dev_err(codec->dev, "Invalid MCLK / Fs ratio\n"); return -EINVAL; } -- 2.9.3

2 1

[alsa-devel] [PATCH -next] ASoC: omap-abe-twl6040: Use devm_snd_soc_register_card()
by Wei Yongjun 24 Aug '16

24 Aug '16

From: Wei Yongjun <weiyongjun1(a)huawei.com> Using devm_snd_soc_register_card() can make the code shorter and cleaner. Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/omap/omap-abe-twl6040.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index f61b3b5..89fe95e 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -305,23 +305,14 @@ static int omap_abe_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n", ret); return ret; } -static int omap_abe_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static const struct of_device_id omap_abe_of_match[] = { {.compatible = "ti,abe-twl6040", }, { }, @@ -335,7 +326,6 @@ static struct platform_driver omap_abe_driver = { .of_match_table = omap_abe_of_match, }, .probe = omap_abe_probe, - .remove = omap_abe_remove, }; static int __init omap_abe_init(void)

3 2

[alsa-devel] [PATCH] ASoC: sunxi: depend on ARCH_SUNXI
by Peter Robinson 24 Aug '16

24 Aug '16

Depend on ARCH_SUNXI or COMPILE_TEST as it doesn't make sense to enable these if the appropriate SoC they're embedded in isn't enabled. Signed-off-by: Peter Robinson <pbrobinson(a)gmail.com> --- sound/soc/sunxi/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 2a954bd..a2b62dc5 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -2,6 +2,7 @@ menu "Allwinner SoC Audio support" config SND_SUN4I_CODEC tristate "Allwinner A10 Codec Support" + depends on ARCH_SUNXI || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help @@ -10,6 +11,7 @@ config SND_SUN4I_CODEC config SND_SUN4I_I2S tristate "Allwinner A10 I2S Support" + depends on ARCH_SUNXI || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help @@ -19,7 +21,7 @@ config SND_SUN4I_I2S config SND_SUN4I_SPDIF tristate "Allwinner A10 SPDIF Support" - depends on OF + depends on OF && (ARCH_SUNXI || COMPILE_TEST) select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help -- 2.9.3

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[alsa-devel] [PATCH v2 0/4] ASoC: rsrc-card to simple-scu-card
by Kuninori Morimoto 24 Aug '16

24 Aug '16

Hi Mark Thesee are v2 of rsrc-card to simple-scu-card. rsrc-card was created for DPCM sound card for Renesas, but we can reuse this driver as more generic purpose. This driver is supporting Multi CPU and Single Codec DPCM system. Kuninori Morimoto (4): ASoC: rsrc-card: remove board specific compatible ASoC: rsrc-card: rename rsrc-card to simple-scu-card phase1 ASoC: rsrc-card: rename rsrc-card to simple-scu-card phase2 ASoC: rsrc-card: rename rsrc-card to simple-scu-card phase3 .../bindings/sound/renesas,rsrc-card.txt | 75 --------- .../devicetree/bindings/sound/simple-scu-card.txt | 80 ++++++++++ sound/soc/generic/Kconfig | 7 + sound/soc/generic/Makefile | 2 + .../rcar/rsrc-card.c => generic/simple-scu-card.c} | 172 +++++++++------------ sound/soc/sh/Kconfig | 6 - sound/soc/sh/rcar/Makefile | 3 - 7 files changed, 162 insertions(+), 183 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-scu-card.txt rename sound/soc/{sh/rcar/rsrc-card.c => generic/simple-scu-card.c} (57%) -- 1.9.1

2 7

Re: [alsa-devel] good ideas
by lemeunier gilles 24 Aug '16

24 Aug '16

Hi! I have some brilliant ideas regarding the issues we've discussed last time, please read 'em here <http://ndicrugafru.whonue.net/e6wutsxl> Pardon my monkey thumbs, lemeunier gilles

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - fix headset mic detection for MSI MS-B120
by Anisse Astier 24 Aug '16

24 Aug '16

MSI Cubi MS-B120 needs the same fixup as the Gigabyte BXBT-2807 for its mic to work. They both use a single 3-way jack for both mic and headset with an ALC283 codec, with the same pins used. Cc: Daniel Drake <drake(a)endlessm.com> Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- sound/pci/hda/patch_realtek.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 574b1b4..7100f05 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4828,7 +4828,7 @@ enum { ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, ALC292_FIXUP_TPT440, - ALC283_FIXUP_BXBT2807_MIC, + ALC283_FIXUP_HEADSET_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, ALC280_FIXUP_HP_GPIO4, @@ -5321,7 +5321,7 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC292_FIXUP_TPT440_DOCK, }, - [ALC283_FIXUP_BXBT2807_MIC] = { + [ALC283_FIXUP_HEADSET_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x19, 0x04a110f0 }, @@ -5651,7 +5651,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), - SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_BXBT2807_MIC), + SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), -- 2.5.5

2 1

[alsa-devel] [PATCH] ASoC: sunxi: depend on ARCH_SUNXI
by Peter Robinson 23 Aug '16

23 Aug '16

Depend on ARCH_SUNXI or COMPILE_TEST as it doesn't make sense to enable these if the appropriate SoC they're embedded in isn't enabled. Signed-off-by: Peter Robinson <pbrobinson(a)gmail.com> --- sound/soc/sunxi/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 2a954bd..dd23682 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -1,4 +1,5 @@ menu "Allwinner SoC Audio support" + depends on ARCH_SUNXI || COMPILE_TEST config SND_SUN4I_CODEC tristate "Allwinner A10 Codec Support" -- 2.9.3

1 0

[alsa-devel] CVE-2014-9892 fix is not upstream
by Luis Henriques 23 Aug '16

23 Aug '16

Hi! Digging through some old CVEs I came across this one that doesn't seem be in mainline. Was there a good reason for not being sent upstream? Maybe it was rejected for some reason and I failed to find the discussion. References: https://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2014-9892 https://source.android.com/security/bulletin/2016-08-01.html https://source.codeaurora.org/quic/la/kernel/msm-3.10/commit/?id=591b1f455c… Cheers, -- Luís

3 3

[alsa-devel] [PATCH v5 0/2] amidi delay option
by Felipe F. Tonello 23 Aug '16

23 Aug '16

Small improvement to amidi, adding useful functionality. 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 | 61 +++++++++++++++++++++++++++++++++++++++++++---------------- 2 files changed, 59 insertions(+), 16 deletions(-) -- 2.9.3

5 13

[alsa-devel] [PATCH] ASoC: nau8810: fix compile warning in loopback switch control
by John Hsu 23 Aug '16

23 Aug '16

Thank Stephen Rothwell for the message. The patch is to fix the following compile error. sound/soc/codecs/nau8810.c:441:3: error: initialization from incompatible pointer type [-Werror=incompatible-pointer-types] SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, &nau8810_loopback), ^ sound/soc/codecs/nau8810.c:441:3: note: (near initialization for 'nau8810_dapm_widgets[11].kcontrol_news') Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 7617dda..f9bcdc3 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -391,10 +391,9 @@ static const struct snd_kcontrol_new nau8810_mic_boost_controls = NAU8810_PMICBSTGAIN_SFT, 0x7, 0); /* Loopback Switch */ -static const struct snd_kcontrol_new nau8810_loopback[] = { +static const struct snd_kcontrol_new nau8810_loopback = SOC_DAPM_SINGLE("Switch", NAU8810_REG_COMP, - NAU8810_ADDAP_SFT, 1, 0), -}; + NAU8810_ADDAP_SFT, 1, 0); static int check_mclk_select_pll(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) -- 2.6.4

2 1

[alsa-devel] [PATCH for 4.8] ASoC: omap-mcpdm: Fix irq resource handling
by Peter Ujfalusi 23 Aug '16

23 Aug '16

Fixes: ddd17531ad908 ("ASoC: omap-mcpdm: Clean up with devm_* function") Managed irq request will not doing any good in ASoC probe level as it is not going to free up the irq when the driver is unbound from the sound card. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reported-by: Russell King <linux(a)armlinux.org.uk> --- sound/soc/omap/omap-mcpdm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 74d6e6fdcfd0..64609c77a79d 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -394,8 +394,8 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) pm_runtime_get_sync(mcpdm->dev); omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00); - ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler, - 0, "McPDM", (void *)mcpdm); + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, 0, "McPDM", + (void *)mcpdm); pm_runtime_put_sync(mcpdm->dev); @@ -420,6 +420,7 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); return 0; -- 2.9.2

2 1

[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: Parse manifest data
by Vinod Koul 23 Aug '16

23 Aug '16

From: Shreyas NC <shreyas.nc(a)intel.com> Topology manifest has lib names and lib count info. So, define tokens to represent module private data and parse these tokens to fill up the manifest structure in the driver accordingly. Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/uapi/sound/snd_sst_tokens.h | 8 +- sound/soc/intel/skylake/skl-topology.c | 194 ++++++++++++++++++++++++++++++++- 2 files changed, 200 insertions(+), 2 deletions(-) diff --git a/include/uapi/sound/snd_sst_tokens.h b/include/uapi/sound/snd_sst_tokens.h index f56a932736ca..1ee2e943d66a 100644 --- a/include/uapi/sound/snd_sst_tokens.h +++ b/include/uapi/sound/snd_sst_tokens.h @@ -153,6 +153,10 @@ * * %SKL_TKN_U32_PROC_DOMAIN: Specify processing domain * + * %SKL_TKN_U32_LIB_COUNT: Specifies the number of libraries + * + * %SKL_TKN_STR_LIB_NAME: Specifies the library name + * * module_id and loadable flags dont have tokens as these values will be * read from the DSP FW manifest */ @@ -202,7 +206,9 @@ enum SKL_TKNS { SKL_TKN_U32_CAPS_PARAMS_ID, SKL_TKN_U32_CAPS_SIZE, SKL_TKN_U32_PROC_DOMAIN, - SKL_TKN_MAX = SKL_TKN_U32_PROC_DOMAIN, + SKL_TKN_U32_LIB_COUNT, + SKL_TKN_STR_LIB_NAME, + SKL_TKN_MAX = SKL_TKN_STR_LIB_NAME, }; #endif diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 000482e6e2f7..108ebb9ab329 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2201,6 +2201,197 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, return 0; } +static int skl_tplg_fill_str_mfest_tkn(struct device *dev, + struct snd_soc_tplg_vendor_string_elem *str_elem, + struct skl_dfw_manifest *minfo) +{ + int tkn_count = 0; + static int ref_count; + + switch (str_elem->token) { + case SKL_TKN_STR_LIB_NAME: + if (ref_count > minfo->lib_count - 1) { + ref_count = 0; + return -EINVAL; + } + + strncpy(minfo->lib[ref_count].name, str_elem->string, + ARRAY_SIZE(minfo->lib[ref_count].name)); + ref_count++; + tkn_count++; + break; + + default: + dev_err(dev, "Not a string token %d", str_elem->token); + break; + } + + return tkn_count; +} + +static int skl_tplg_get_str_tkn(struct device *dev, + struct snd_soc_tplg_vendor_array *array, + struct skl_dfw_manifest *minfo) +{ + int tkn_count = 0, ret; + struct snd_soc_tplg_vendor_string_elem *str_elem; + + str_elem = (struct snd_soc_tplg_vendor_string_elem *)array->value; + while (tkn_count < array->num_elems) { + ret = skl_tplg_fill_str_mfest_tkn(dev, str_elem, minfo); + str_elem++; + + if (ret < 0) + return ret; + + tkn_count = tkn_count + ret; + } + + return tkn_count; +} + +static int skl_tplg_get_int_tkn(struct device *dev, + struct snd_soc_tplg_vendor_value_elem *tkn_elem, + struct skl_dfw_manifest *minfo) +{ + int tkn_count = 0; + + switch (tkn_elem->token) { + case SKL_TKN_U32_LIB_COUNT: + minfo->lib_count = tkn_elem->value; + tkn_count++; + break; + + default: + dev_err(dev, "Not a manifest token %d", tkn_elem->token); + return -EINVAL; + } + + return tkn_count; +} + +/* + * Fill the manifest structure by parsing the tokens based on the + * type. + */ +static int skl_tplg_get_manifest_tkn(struct device *dev, + char *pvt_data, struct skl_dfw_manifest *minfo, + int block_size) +{ + int tkn_count = 0, ret; + int off = 0, tuple_size = 0; + struct snd_soc_tplg_vendor_array *array; + struct snd_soc_tplg_vendor_value_elem *tkn_elem; + + if (block_size <= 0) + return -EINVAL; + + while (tuple_size < block_size) { + array = (struct snd_soc_tplg_vendor_array *)(pvt_data + off); + off += array->size; + switch (array->type) { + case SND_SOC_TPLG_TUPLE_TYPE_STRING: + ret = skl_tplg_get_str_tkn(dev, array, minfo); + + if (ret < 0) + return ret; + tkn_count += ret; + + tuple_size += tkn_count * + sizeof(struct snd_soc_tplg_vendor_string_elem); + continue; + + case SND_SOC_TPLG_TUPLE_TYPE_UUID: + dev_warn(dev, "no uuid tokens for skl tplf manifest"); + continue; + + default: + tkn_elem = array->value; + tkn_count = 0; + break; + } + + while (tkn_count <= array->num_elems - 1) { + ret = skl_tplg_get_int_tkn(dev, + tkn_elem, minfo); + if (ret < 0) + return ret; + + tkn_count = tkn_count + ret; + tkn_elem++; + tuple_size += tkn_count * + sizeof(struct snd_soc_tplg_vendor_value_elem); + break; + } + tkn_count = 0; + } + + return 0; +} + +/* + * Parse manifest private data for tokens. The private data block is + * preceded by descriptors for type and size of data block. + */ +static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, + struct device *dev, struct skl_dfw_manifest *minfo) +{ + struct snd_soc_tplg_vendor_array *array; + int num_blocks, block_size = 0, block_type, off = 0; + char *data; + int ret; + + /* Read the NUM_DATA_BLOCKS descriptor */ + array = (struct snd_soc_tplg_vendor_array *)manifest->priv.data; + ret = skl_tplg_get_desc_blocks(dev, array); + if (ret < 0) + return ret; + num_blocks = ret; + + off += array->size; + array = (struct snd_soc_tplg_vendor_array *) + (manifest->priv.data + off); + + /* Read the BLOCK_TYPE and BLOCK_SIZE descriptor */ + while (num_blocks > 0) { + ret = skl_tplg_get_desc_blocks(dev, array); + + if (ret < 0) + return ret; + block_type = ret; + off += array->size; + + array = (struct snd_soc_tplg_vendor_array *) + (manifest->priv.data + off); + + ret = skl_tplg_get_desc_blocks(dev, array); + + if (ret < 0) + return ret; + block_size = ret; + off += array->size; + + array = (struct snd_soc_tplg_vendor_array *) + (manifest->priv.data + off); + + data = (manifest->priv.data + off); + + if (block_type == SKL_TYPE_TUPLE) { + ret = skl_tplg_get_manifest_tkn(dev, data, minfo, + block_size); + + if (ret < 0) + return ret; + + --num_blocks; + } else { + return -EINVAL; + } + } + + return 0; +} + static int skl_manifest_load(struct snd_soc_component *cmpnt, struct snd_soc_tplg_manifest *manifest) { @@ -2211,7 +2402,8 @@ static int skl_manifest_load(struct snd_soc_component *cmpnt, int ret = 0; minfo = &skl->skl_sst->manifest; - memcpy(minfo, manifest->priv.data, sizeof(struct skl_dfw_manifest)); + + skl_tplg_get_manifest_data(manifest, bus->dev, minfo); if (minfo->lib_count > HDA_MAX_LIB) { dev_err(bus->dev, "Exceeding max Library count. Got:%d\n", -- 1.9.1

2 1

[alsa-devel] [PATCH 0/3] ASoC: rename rsrc-card to simple-dpcm-card
by Kuninori Morimoto 23 Aug '16

23 Aug '16

Hi Mark These patches moves current Renesas specific rsrc-card to more generic simple-dpcm-card, but by 1 patch was difficult. So, it uses 3 phases Kuninori Morimoto (3): ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase1 ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase2 ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase3 .../bindings/sound/renesas,rsrc-card.txt | 75 --------- .../devicetree/bindings/sound/simple-dpcm-card.txt | 79 ++++++++++ sound/soc/generic/Kconfig | 6 + sound/soc/generic/Makefile | 2 + .../rsrc-card.c => generic/simple-dpcm-card.c} | 173 +++++++++------------ sound/soc/sh/Kconfig | 6 - sound/soc/sh/rcar/Makefile | 3 - 7 files changed, 160 insertions(+), 184 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-dpcm-card.txt rename sound/soc/{sh/rcar/rsrc-card.c => generic/simple-dpcm-card.c} (57%) -- 1.9.1

2 10

[alsa-devel] [PATCH RESEND 0/3] conf: Add vendor tuples in conf
by Vinod Koul 23 Aug '16

23 Aug '16

Topology changes in alsa-lib allow driver data for modules to be be passed in topology conf file using tuples. These changes allow private data to represented by tuples of type UUID, byte, short, word and strings. This patch series adds changes to : - include tuple and token list in the conf file to define module private data. - Private data blobs for modules are no longer required. So, remove the binary blobs and the tool used to generate the blobs. - Then add bxt conf file which is based on tuples Shreyas NC (3): conf: sklrt286: define module private data through tuples in skylake conf: sklrt286: Remove tool to generate private data blobs conf: bxtrt298: Add topology conf file for bxt configure.ac | 2 +- src/conf/topology/Makefile.am | 2 +- src/conf/topology/bxtrt298/Makefile.am | 4 + src/conf/topology/bxtrt298/bxt_i2s.conf | 3323 ++++++++++++++++++++ src/conf/topology/sklrt286/Makefile.am | 3 +- src/conf/topology/sklrt286/codec0_in-cpr-1.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec0_in-mi.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec0_out-cpr-4.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec0_out-mo.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec1_out-cpr-5.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec1_out-mo.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/data/Makefile.am | 6 - src/conf/topology/sklrt286/data/pvt.c | 1815 ----------- src/conf/topology/sklrt286/data/pvt_data.c | 90 - src/conf/topology/sklrt286/data/pvt_local.h | 9 - .../topology/sklrt286/data/skl-tplg-interface.h | 232 -- .../topology/sklrt286/dmic01_hifi_in-cpr-3.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_in-cpr-0.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_in-mi.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_out-cpr-6.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_out-mo.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/skl_i2s.conf | 2694 +++++++++++++++- 29 files changed, 5988 insertions(+), 2192 deletions(-) create mode 100644 src/conf/topology/bxtrt298/Makefile.am create mode 100644 src/conf/topology/bxtrt298/bxt_i2s.conf delete mode 100644 src/conf/topology/sklrt286/codec0_in-cpr-1.bin delete mode 100644 src/conf/topology/sklrt286/codec0_in-mi.bin delete mode 100644 src/conf/topology/sklrt286/codec0_out-cpr-4.bin delete mode 100644 src/conf/topology/sklrt286/codec0_out-mo.bin delete mode 100644 src/conf/topology/sklrt286/codec1_out-cpr-5.bin delete mode 100644 src/conf/topology/sklrt286/codec1_out-mo.bin delete mode 100644 src/conf/topology/sklrt286/data/Makefile.am delete mode 100644 src/conf/topology/sklrt286/data/pvt.c delete mode 100644 src/conf/topology/sklrt286/data/pvt_data.c delete mode 100644 src/conf/topology/sklrt286/data/pvt_local.h delete mode 100644 src/conf/topology/sklrt286/data/skl-tplg-interface.h delete mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-cpr-3.bin delete mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin delete mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin delete mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin delete mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin delete mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin delete mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin delete mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin delete mode 100644 src/conf/topology/sklrt286/media0_in-cpr-0.bin delete mode 100644 src/conf/topology/sklrt286/media0_in-mi.bin delete mode 100644 src/conf/topology/sklrt286/media0_out-cpr-6.bin delete mode 100644 src/conf/topology/sklrt286/media0_out-mo.bin -- 1.9.1

1 3

[alsa-devel] [PATCH 0/2] ALSA: compress: Fine-tuning for snd_compr_set_params()
by SF Markus Elfring 23 Aug '16

23 Aug '16

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 21 Aug 2016 21:35:43 +0200 A few update suggestions were taken into account from static source code analysis. Markus Elfring (2): Use memdup_user() Reduce the scope for two variables sound/core/compress_offload.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) -- 2.9.3

6 13

[alsa-devel] [PATCH 0/4] ASoC: Intel: Skylake: Add token support for topology data
by Vinod Koul 23 Aug '16

23 Aug '16

With topology changes in alsa-lib, driver data for modules can now be passed in topology conf file using tuples. These changes allow private data to represented by tuples of type UUID, byte, short, word and strings. This patch series adds changes to: - Define vendor tokens to best describe module private data - Parse these token value to populate module private data - Remove the skl_dfw_config structure as it would no longer be required. - Define tokens for manifest private data and parse them to populate library names for library loading. Shreyas NC (4): ASoC: uapi: Intel: Skylake: Define vendor specific tokens ASoC: Intel: Skylake: Parse vendor tokens to build module data ASoC: Intel: Skylake: Remove dfw config and associated structures ASoC: Intel: Skylake: Parse manifest data include/uapi/sound/Kbuild | 1 + include/uapi/sound/snd_sst_tokens.h | 214 +++++++ sound/soc/intel/skylake/skl-sst-utils.c | 5 - sound/soc/intel/skylake/skl-topology.c | 839 +++++++++++++++++++++++---- sound/soc/intel/skylake/skl-tplg-interface.h | 83 +-- 5 files changed, 953 insertions(+), 189 deletions(-) create mode 100644 include/uapi/sound/snd_sst_tokens.h -- 1.9.1

4 14

[alsa-devel] [PATCH 0/3] conf: Add vendor tuples in conf
by Vinod Koul 23 Aug '16

23 Aug '16

Topology changes in alsa-lib allow driver data for modules to be be passed in topology conf file using tuples. These changes allow private data to represented by tuples of type UUID, byte, short, word and strings. This patch series adds changes to : - include tuple and token list in the conf file to define module private data. - Private data blobs for modules are no longer required. So, remove the binary blobs and the tool used to generate the blobs. - Then add bxt conf file which is based on tuples Takashi, This should be merged after the kernel changes are merged. Shreyas NC (3): conf: sklrt286: define module private data through tuples in skylake conf: sklrt286: Remove tool to generate private data blobs conf: bxtrt298: Add topology conf file for bxt configure.ac | 2 +- src/conf/topology/Makefile.am | 2 +- src/conf/topology/bxtrt298/Makefile.am | 4 + src/conf/topology/bxtrt298/bxt_i2s.conf | 3323 ++++++++++++++++++++ src/conf/topology/sklrt286/Makefile.am | 3 +- src/conf/topology/sklrt286/codec0_in-cpr-1.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec0_in-mi.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec0_out-cpr-4.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec0_out-mo.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec1_out-cpr-5.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/codec1_out-mo.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/data/Makefile.am | 6 - src/conf/topology/sklrt286/data/pvt.c | 1815 ----------- src/conf/topology/sklrt286/data/pvt_data.c | 90 - src/conf/topology/sklrt286/data/pvt_local.h | 9 - .../topology/sklrt286/data/skl-tplg-interface.h | 232 -- .../topology/sklrt286/dmic01_hifi_in-cpr-3.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_in-cpr-0.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_in-mi.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_out-cpr-6.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/media0_out-mo.bin | Bin 4244 -> 0 bytes src/conf/topology/sklrt286/skl_i2s.conf | 2694 +++++++++++++++- 29 files changed, 5988 insertions(+), 2192 deletions(-) create mode 100644 src/conf/topology/bxtrt298/Makefile.am create mode 100644 src/conf/topology/bxtrt298/bxt_i2s.conf delete mode 100644 src/conf/topology/sklrt286/codec0_in-cpr-1.bin delete mode 100644 src/conf/topology/sklrt286/codec0_in-mi.bin delete mode 100644 src/conf/topology/sklrt286/codec0_out-cpr-4.bin delete mode 100644 src/conf/topology/sklrt286/codec0_out-mo.bin delete mode 100644 src/conf/topology/sklrt286/codec1_out-cpr-5.bin delete mode 100644 src/conf/topology/sklrt286/codec1_out-mo.bin delete mode 100644 src/conf/topology/sklrt286/data/Makefile.am delete mode 100644 src/conf/topology/sklrt286/data/pvt.c delete mode 100644 src/conf/topology/sklrt286/data/pvt_data.c delete mode 100644 src/conf/topology/sklrt286/data/pvt_local.h delete mode 100644 src/conf/topology/sklrt286/data/skl-tplg-interface.h delete mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-cpr-3.bin delete mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin delete mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin delete mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin delete mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin delete mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin delete mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin delete mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin delete mode 100644 src/conf/topology/sklrt286/media0_in-cpr-0.bin delete mode 100644 src/conf/topology/sklrt286/media0_in-mi.bin delete mode 100644 src/conf/topology/sklrt286/media0_out-cpr-6.bin delete mode 100644 src/conf/topology/sklrt286/media0_out-mo.bin -- 1.9.1

2 3

[alsa-devel] [PATCH v05 63/72] include/uapi/sound/Kbuild: remove sound/asoc.h from userspace
by Mikko Rapeli 22 Aug '16

22 Aug '16

Based on git commit messages and the #error macro messages it should not be used in userspace. Fixes userspace compile errors: sound/asoc.h:21:2: error: #error This API is an early revision and not enabled in the current #error This API is an early revision and not enabled in the current ^~~~~ sound/asoc.h:22:2: error: #error kernel release, it will be enabled in a futur e kernel version #error kernel release, it will be enabled in a future kernel version ^~~~~ sound/asoc.h:23:2: error: #error with incompatible changes to what is here. #error with incompatible changes to what is here. ^~~~~ Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/Kbuild | 1 - 1 file changed, 1 deletion(-) diff --git a/include/uapi/sound/Kbuild b/include/uapi/sound/Kbuild index 691984c..6d8b203 100644 --- a/include/uapi/sound/Kbuild +++ b/include/uapi/sound/Kbuild @@ -1,6 +1,5 @@ # UAPI Header export list header-y += asequencer.h -header-y += asoc.h header-y += asound.h header-y += asound_fm.h header-y += compress_offload.h -- 2.8.1

1 0

[alsa-devel] [PATCH -next] ASoC: max98371: Add terminate entry for i2c_device_id tables
by Wei Yongjun 22 Aug '16

22 Aug '16

From: Wei Yongjun <weiyongjun1(a)huawei.com> Make sure i2c_device_id tables are NULL terminated. Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/max98371.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index 3053879..781be9ba 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -412,6 +412,7 @@ static int max98371_i2c_remove(struct i2c_client *client) static const struct i2c_device_id max98371_i2c_id[] = { { "max98371", 0 }, + { } }; MODULE_DEVICE_TABLE(i2c, max98371_i2c_id);

2 1

[alsa-devel] [PATCH] ASoC: nau8810: Add driver for Nuvoton codec chip NAU88C10
by John Hsu 22 Aug '16

22 Aug '16

The driver is for codec NAU88C10 of Nuvoton Technology Corporation. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8810.txt | 16 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8810.c | 884 +++++++++++++++++++++ sound/soc/codecs/nau8810.h | 281 +++++++ 5 files changed, 1188 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8810.txt create mode 100644 sound/soc/codecs/nau8810.c create mode 100644 sound/soc/codecs/nau8810.h diff --git a/Documentation/devicetree/bindings/sound/nau8810.txt b/Documentation/devicetree/bindings/sound/nau8810.txt new file mode 100644 index 0000000..05830e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8810.txt @@ -0,0 +1,16 @@ +NAU8810 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "nuvoton,nau8810" + + - reg : the I2C address of the device. + +Example: + +codec: nau8810@1a { + compatible = "nuvoton,nau8810"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1cd6ab3..b3fd110 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -92,6 +92,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C @@ -1064,6 +1065,10 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8810 + tristate "Nuvoton Technology Corporation NAU88C10 CODEC" + depends on I2C + config SND_SOC_NAU8825 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 58036af..ffaa523 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -86,6 +86,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8810-objs := nau8810.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o snd-soc-pcm1681-objs := pcm1681.o @@ -307,6 +308,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c new file mode 100644 index 0000000..7617dda --- /dev/null +++ b/sound/soc/codecs/nau8810.c @@ -0,0 +1,884 @@ +/* + * nau8810.c -- NAU8810 ALSA Soc Audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * + * Based on WM8974.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "nau8810.h" + +#define NAU_PLL_FREQ_MAX 100000000 +#define NAU_PLL_FREQ_MIN 90000000 +#define NAU_PLL_REF_MAX 33000000 +#define NAU_PLL_REF_MIN 8000000 +#define NAU_PLL_OPTOP_MIN 6 + + +static const int nau8810_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 }; + +static const struct reg_default nau8810_reg_defaults[] = { + { NAU8810_REG_POWER1, 0x0000 }, + { NAU8810_REG_POWER2, 0x0000 }, + { NAU8810_REG_POWER3, 0x0000 }, + { NAU8810_REG_IFACE, 0x0050 }, + { NAU8810_REG_COMP, 0x0000 }, + { NAU8810_REG_CLOCK, 0x0140 }, + { NAU8810_REG_SMPLR, 0x0000 }, + { NAU8810_REG_DAC, 0x0000 }, + { NAU8810_REG_DACGAIN, 0x00FF }, + { NAU8810_REG_ADC, 0x0100 }, + { NAU8810_REG_ADCGAIN, 0x00FF }, + { NAU8810_REG_EQ1, 0x012C }, + { NAU8810_REG_EQ2, 0x002C }, + { NAU8810_REG_EQ3, 0x002C }, + { NAU8810_REG_EQ4, 0x002C }, + { NAU8810_REG_EQ5, 0x002C }, + { NAU8810_REG_DACLIM1, 0x0032 }, + { NAU8810_REG_DACLIM2, 0x0000 }, + { NAU8810_REG_NOTCH1, 0x0000 }, + { NAU8810_REG_NOTCH2, 0x0000 }, + { NAU8810_REG_NOTCH3, 0x0000 }, + { NAU8810_REG_NOTCH4, 0x0000 }, + { NAU8810_REG_ALC1, 0x0038 }, + { NAU8810_REG_ALC2, 0x000B }, + { NAU8810_REG_ALC3, 0x0032 }, + { NAU8810_REG_NOISEGATE, 0x0000 }, + { NAU8810_REG_PLLN, 0x0008 }, + { NAU8810_REG_PLLK1, 0x000C }, + { NAU8810_REG_PLLK2, 0x0093 }, + { NAU8810_REG_PLLK3, 0x00E9 }, + { NAU8810_REG_ATTEN, 0x0000 }, + { NAU8810_REG_INPUT_SIGNAL, 0x0003 }, + { NAU8810_REG_PGAGAIN, 0x0010 }, + { NAU8810_REG_ADCBOOST, 0x0100 }, + { NAU8810_REG_OUTPUT, 0x0002 }, + { NAU8810_REG_SPKMIX, 0x0001 }, + { NAU8810_REG_SPKGAIN, 0x0039 }, + { NAU8810_REG_MONOMIX, 0x0001 }, + { NAU8810_REG_POWER4, 0x0000 }, + { NAU8810_REG_TSLOTCTL1, 0x0000 }, + { NAU8810_REG_TSLOTCTL2, 0x0020 }, + { NAU8810_REG_DEVICE_REVID, 0x0000 }, + { NAU8810_REG_I2C_DEVICEID, 0x001A }, + { NAU8810_REG_ADDITIONID, 0x00CA }, + { NAU8810_REG_RESERVE, 0x0124 }, + { NAU8810_REG_OUTCTL, 0x0001 }, + { NAU8810_REG_ALC1ENHAN1, 0x0010 }, + { NAU8810_REG_ALC1ENHAN2, 0x0000 }, + { NAU8810_REG_MISCCTL, 0x0000 }, + { NAU8810_REG_OUTTIEOFF, 0x0000 }, + { NAU8810_REG_AGCP2POUT, 0x0000 }, + { NAU8810_REG_AGCPOUT, 0x0000 }, + { NAU8810_REG_AMTCTL, 0x0000 }, + { NAU8810_REG_OUTTIEOFFMAN, 0x0000 }, +}; + +static bool nau8810_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET ... NAU8810_REG_SMPLR: + case NAU8810_REG_DAC ... NAU8810_REG_DACGAIN: + case NAU8810_REG_ADC ... NAU8810_REG_ADCGAIN: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_DACLIM1 ... NAU8810_REG_DACLIM2: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + case NAU8810_REG_ALC1 ... NAU8810_REG_ATTEN: + case NAU8810_REG_INPUT_SIGNAL ... NAU8810_REG_PGAGAIN: + case NAU8810_REG_ADCBOOST: + case NAU8810_REG_OUTPUT ... NAU8810_REG_SPKMIX: + case NAU8810_REG_SPKGAIN: + case NAU8810_REG_MONOMIX: + case NAU8810_REG_POWER4 ... NAU8810_REG_TSLOTCTL2: + case NAU8810_REG_DEVICE_REVID ... NAU8810_REG_RESERVE: + case NAU8810_REG_OUTCTL ... NAU8810_REG_ALC1ENHAN2: + case NAU8810_REG_MISCCTL: + case NAU8810_REG_OUTTIEOFF ... NAU8810_REG_OUTTIEOFFMAN: + return true; + default: + return false; + } +} + +static bool nau8810_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET ... NAU8810_REG_SMPLR: + case NAU8810_REG_DAC ... NAU8810_REG_DACGAIN: + case NAU8810_REG_ADC ... NAU8810_REG_ADCGAIN: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_DACLIM1 ... NAU8810_REG_DACLIM2: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + case NAU8810_REG_ALC1 ... NAU8810_REG_ATTEN: + case NAU8810_REG_INPUT_SIGNAL ... NAU8810_REG_PGAGAIN: + case NAU8810_REG_ADCBOOST: + case NAU8810_REG_OUTPUT ... NAU8810_REG_SPKMIX: + case NAU8810_REG_SPKGAIN: + case NAU8810_REG_MONOMIX: + case NAU8810_REG_POWER4 ... NAU8810_REG_TSLOTCTL2: + case NAU8810_REG_OUTCTL ... NAU8810_REG_ALC1ENHAN2: + case NAU8810_REG_MISCCTL: + case NAU8810_REG_OUTTIEOFF ... NAU8810_REG_OUTTIEOFFMAN: + return true; + default: + return false; + } +} + +static bool nau8810_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET: + case NAU8810_REG_DEVICE_REVID ... NAU8810_REG_RESERVE: + return true; + default: + return false; + } +} + +/* The EQ parameters get function is to get the 5 band equalizer control. + * The regmap raw read can't work here because regmap doesn't provide + * value format for value width of 9 bits. Therefore, the driver reads data + * from cache and makes value format according to the endianness of + * bytes type control element. + */ +static int nau8810_eq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + int i, reg, reg_val; + u16 *val; + + val = (u16 *)ucontrol->value.bytes.data; + reg = NAU8810_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + regmap_read(nau8810->regmap, reg + i, &reg_val); + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + reg_val = cpu_to_be16(reg_val); + memcpy(val + i, &reg_val, sizeof(reg_val)); + } + + return 0; +} + +/* The EQ parameters put function is to make configuration of 5 band equalizer + * control. These configuration includes central frequency, equalizer gain, + * cut-off frequency, bandwidth control, and equalizer path. + * The regmap raw write can't work here because regmap doesn't provide + * register and value format for register with address 7 bits and value 9 bits. + * Therefore, the driver makes value format according to the endianness of + * bytes type control element and writes data to codec. + */ +static int nau8810_eq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + u16 *val, value; + int i, reg, ret; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + val = (u16 *)data; + reg = NAU8810_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + /* conversion of 16-bit integers between native CPU format + * and big endian format + */ + value = be16_to_cpu(*(val + i)); + ret = regmap_write(nau8810->regmap, reg + i, value); + if (ret) { + dev_err(codec->dev, "EQ configuration fail, register: %x ret: %d\n", + reg + i, ret); + return ret; + } + } + kfree(data); + + return 0; +} + +static const char * const nau8810_companding[] = { + "Off", "NC", "u-law", "A-law" }; + +static const struct soc_enum nau8810_companding_adc_enum = + SOC_ENUM_SINGLE(NAU8810_REG_COMP, NAU8810_ADCCM_SFT, + ARRAY_SIZE(nau8810_companding), nau8810_companding); + +static const struct soc_enum nau8810_companding_dac_enum = + SOC_ENUM_SINGLE(NAU8810_REG_COMP, NAU8810_DACCM_SFT, + ARRAY_SIZE(nau8810_companding), nau8810_companding); + +static const char * const nau8810_deemp[] = { + "None", "32kHz", "44.1kHz", "48kHz" }; + +static const struct soc_enum nau8810_deemp_enum = + SOC_ENUM_SINGLE(NAU8810_REG_DAC, NAU8810_DEEMP_SFT, + ARRAY_SIZE(nau8810_deemp), nau8810_deemp); + +static const char * const nau8810_eqmode[] = {"Capture", "Playback" }; + +static const struct soc_enum nau8810_eqmode_enum = + SOC_ENUM_SINGLE(NAU8810_REG_EQ1, NAU8810_EQM_SFT, + ARRAY_SIZE(nau8810_eqmode), nau8810_eqmode); + +static const char * const nau8810_alc[] = {"Normal", "Limiter" }; + +static const struct soc_enum nau8810_alc_enum = + SOC_ENUM_SINGLE(NAU8810_REG_ALC3, NAU8810_ALCM_SFT, + ARRAY_SIZE(nau8810_alc), nau8810_alc); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); + +static const struct snd_kcontrol_new nau8810_snd_controls[] = { + SOC_ENUM("ADC Companding", nau8810_companding_adc_enum), + SOC_ENUM("DAC Companding", nau8810_companding_dac_enum), + SOC_ENUM("DAC De-emphasis", nau8810_deemp_enum), + + SOC_ENUM("EQ Function", nau8810_eqmode_enum), + SND_SOC_BYTES_EXT("EQ Parameters", 10, + nau8810_eq_get, nau8810_eq_put), + + SOC_SINGLE("DAC Inversion Switch", NAU8810_REG_DAC, + NAU8810_DACPL_SFT, 1, 0), + SOC_SINGLE_TLV("Playback Volume", NAU8810_REG_DACGAIN, + NAU8810_DACGAIN_SFT, 0xff, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", NAU8810_REG_ADC, + NAU8810_HPFEN_SFT, 1, 0), + SOC_SINGLE("High Pass Cut Off", NAU8810_REG_ADC, + NAU8810_HPF_SFT, 0x7, 0), + + SOC_SINGLE("ADC Inversion Switch", NAU8810_REG_ADC, + NAU8810_ADCPL_SFT, 1, 0), + SOC_SINGLE_TLV("Capture Volume", NAU8810_REG_ADCGAIN, + NAU8810_ADCGAIN_SFT, 0xff, 0, digital_tlv), + + SOC_SINGLE_TLV("EQ1 Volume", NAU8810_REG_EQ1, + NAU8810_EQ1GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ2 Volume", NAU8810_REG_EQ2, + NAU8810_EQ2GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ3 Volume", NAU8810_REG_EQ3, + NAU8810_EQ3GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ4 Volume", NAU8810_REG_EQ4, + NAU8810_EQ4GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ5 Volume", NAU8810_REG_EQ5, + NAU8810_EQ5GC_SFT, 0x18, 1, eq_tlv), + + SOC_SINGLE("DAC Limiter Switch", NAU8810_REG_DACLIM1, + NAU8810_DACLIMEN_SFT, 1, 0), + SOC_SINGLE("DAC Limiter Decay", NAU8810_REG_DACLIM1, + NAU8810_DACLIMDCY_SFT, 0xf, 0), + SOC_SINGLE("DAC Limiter Attack", NAU8810_REG_DACLIM1, + NAU8810_DACLIMATK_SFT, 0xf, 0), + SOC_SINGLE("DAC Limiter Threshold", NAU8810_REG_DACLIM2, + NAU8810_DACLIMTHL_SFT, 0x7, 0), + SOC_SINGLE("DAC Limiter Boost", NAU8810_REG_DACLIM2, + NAU8810_DACLIMBST_SFT, 0xf, 0), + + SOC_ENUM("ALC Mode", nau8810_alc_enum), + SOC_SINGLE("ALC Enable Switch", NAU8810_REG_ALC1, + NAU8810_ALCEN_SFT, 1, 0), + SOC_SINGLE("ALC Max Volume", NAU8810_REG_ALC1, + NAU8810_ALCMXGAIN_SFT, 0x7, 0), + SOC_SINGLE("ALC Min Volume", NAU8810_REG_ALC1, + NAU8810_ALCMINGAIN_SFT, 0x7, 0), + SOC_SINGLE("ALC ZC Switch", NAU8810_REG_ALC2, + NAU8810_ALCZC_SFT, 1, 0), + SOC_SINGLE("ALC Hold", NAU8810_REG_ALC2, + NAU8810_ALCHT_SFT, 0xf, 0), + SOC_SINGLE("ALC Target", NAU8810_REG_ALC2, + NAU8810_ALCSL_SFT, 0xf, 0), + SOC_SINGLE("ALC Decay", NAU8810_REG_ALC3, + NAU8810_ALCDCY_SFT, 0xf, 0), + SOC_SINGLE("ALC Attack", NAU8810_REG_ALC3, + NAU8810_ALCATK_SFT, 0xf, 0), + SOC_SINGLE("ALC Noise Gate Switch", NAU8810_REG_NOISEGATE, + NAU8810_ALCNEN_SFT, 1, 0), + SOC_SINGLE("ALC Noise Gate Threshold", NAU8810_REG_NOISEGATE, + NAU8810_ALCNTH_SFT, 0x7, 0), + + SOC_SINGLE("PGA ZC Switch", NAU8810_REG_PGAGAIN, + NAU8810_PGAZC_SFT, 1, 0), + SOC_SINGLE_TLV("PGA Volume", NAU8810_REG_PGAGAIN, + NAU8810_PGAGAIN_SFT, 0x3f, 0, inpga_tlv), + + SOC_SINGLE("Speaker ZC Switch", NAU8810_REG_SPKGAIN, + NAU8810_SPKZC_SFT, 1, 0), + SOC_SINGLE("Speaker Mute Switch", NAU8810_REG_SPKGAIN, + NAU8810_SPKMT_SFT, 1, 0), + SOC_SINGLE_TLV("Speaker Volume", NAU8810_REG_SPKGAIN, + NAU8810_SPKGAIN_SFT, 0x3f, 0, spk_tlv), + + SOC_SINGLE("Capture Boost(+20dB)", NAU8810_REG_ADCBOOST, + NAU8810_PGABST_SFT, 1, 0), + SOC_SINGLE("Mono Mute Switch", NAU8810_REG_MONOMIX, + NAU8810_MOUTMXMT_SFT, 1, 0), + + SOC_SINGLE("DAC Oversampling Rate(128x) Switch", NAU8810_REG_DAC, + NAU8810_DACOS_SFT, 1, 0), + SOC_SINGLE("ADC Oversampling Rate(128x) Switch", NAU8810_REG_ADC, + NAU8810_ADCOS_SFT, 1, 0), +}; + +/* Speaker Output Mixer */ +static const struct snd_kcontrol_new nau8810_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_SPKMIX, + NAU8810_BYPSPK_SFT, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_SPKMIX, + NAU8810_DACSPK_SFT, 1, 0), +}; + +/* Mono Output Mixer */ +static const struct snd_kcontrol_new nau8810_mono_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_MONOMIX, + NAU8810_BYPMOUT_SFT, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_MONOMIX, + NAU8810_DACMOUT_SFT, 1, 0), +}; + +/* PGA Mute */ +static const struct snd_kcontrol_new nau8810_inpga_mute[] = { + SOC_DAPM_SINGLE("PGA Mute Switch", NAU8810_REG_PGAGAIN, + NAU8810_PGAMT_SFT, 1, 0), +}; + +/* Input PGA */ +static const struct snd_kcontrol_new nau8810_inpga[] = { + SOC_DAPM_SINGLE("MicN Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_NMICPGA_SFT, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_PMICPGA_SFT, 1, 0), +}; + +/* Mic Input boost vol */ +static const struct snd_kcontrol_new nau8810_mic_boost_controls = + SOC_DAPM_SINGLE("Mic Volume", NAU8810_REG_ADCBOOST, + NAU8810_PMICBSTGAIN_SFT, 0x7, 0); + +/* Loopback Switch */ +static const struct snd_kcontrol_new nau8810_loopback[] = { + SOC_DAPM_SINGLE("Switch", NAU8810_REG_COMP, + NAU8810_ADDAP_SFT, 1, 0), +}; + +static int check_mclk_select_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + regmap_read(nau8810->regmap, NAU8810_REG_CLOCK, &value); + return (value & NAU8810_CLKM_MASK); +} + +static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Speaker Mixer", NAU8810_REG_POWER3, + NAU8810_SPKMX_EN_SFT, 0, &nau8810_speaker_mixer_controls[0], + ARRAY_SIZE(nau8810_speaker_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", NAU8810_REG_POWER3, + NAU8810_MOUTMX_EN_SFT, 0, &nau8810_mono_mixer_controls[0], + ARRAY_SIZE(nau8810_mono_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", NAU8810_REG_POWER3, + NAU8810_DAC_EN_SFT, 0), + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", NAU8810_REG_POWER2, + NAU8810_ADC_EN_SFT, 0), + SND_SOC_DAPM_PGA("SpkN Out", NAU8810_REG_POWER3, + NAU8810_NSPK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SpkP Out", NAU8810_REG_POWER3, + NAU8810_PSPK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", NAU8810_REG_POWER3, + NAU8810_MOUT_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Input PGA", NAU8810_REG_POWER2, + NAU8810_PGA_EN_SFT, 0, nau8810_inpga, + ARRAY_SIZE(nau8810_inpga)), + SND_SOC_DAPM_MIXER("Input Boost Stage", NAU8810_REG_POWER2, + NAU8810_BST_EN_SFT, 0, nau8810_inpga_mute, + ARRAY_SIZE(nau8810_inpga_mute)), + + SND_SOC_DAPM_SUPPLY("Mic Bias", NAU8810_REG_POWER1, + NAU8810_MICBIAS_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", NAU8810_REG_POWER1, + NAU8810_PLL_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, + &nau8810_loopback), + + SND_SOC_DAPM_INPUT("MICN"), + SND_SOC_DAPM_INPUT("MICP"), + SND_SOC_DAPM_OUTPUT("MONOOUT"), + SND_SOC_DAPM_OUTPUT("SPKOUTP"), + SND_SOC_DAPM_OUTPUT("SPKOUTN"), +}; + +static const struct snd_soc_dapm_route nau8810_dapm_routes[] = { + {"DAC", NULL, "PLL", check_mclk_select_pll}, + + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Line Bypass Switch", "Input Boost Stage"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Line Bypass Switch", "Input Boost Stage"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Input Boost Stage */ + {"ADC", NULL, "Input Boost Stage"}, + {"ADC", NULL, "PLL", check_mclk_select_pll}, + {"Input Boost Stage", NULL, "Input PGA"}, + {"Input Boost Stage", NULL, "MICP"}, + + /* Input PGA */ + {"Input PGA", NULL, "Mic Bias"}, + {"Input PGA", "MicN Switch", "MICN"}, + {"Input PGA", "MicP Switch", "MICP"}, + + /* Digital Looptack */ + {"Digital Loopback", "Switch", "ADC"}, + {"DAC", NULL, "Digital Loopback"}, +}; + +static int nau8810_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + + nau8810->clk_id = clk_id; + nau8810->sysclk = freq; + dev_dbg(nau8810->dev, "master sysclk %dHz, source %s\n", + freq, clk_id == NAU8810_SCLK_PLL ? "PLL" : "MCLK"); + + return 0; +} + +static int nau88l0_calc_pll(unsigned int pll_in, + unsigned int fs, struct nau8810_pll *pll_param) +{ + u64 f2, f2_max, pll_ratio; + int i, scal_sel; + + if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN) + return -EINVAL; + + f2_max = 0; + scal_sel = ARRAY_SIZE(nau8810_mclk_scaler); + for (i = 0; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { + f2 = 256 * fs * 4 * nau8810_mclk_scaler[i] / 10; + if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && + f2_max < f2) { + f2_max = f2; + scal_sel = i; + } + } + if (ARRAY_SIZE(nau8810_mclk_scaler) == scal_sel) + return -EINVAL; + pll_param->mclk_scaler = scal_sel; + f2 = f2_max; + + /* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional + * input; round up the 24+4bit. + */ + pll_ratio = div_u64(f2 << 28, pll_in); + pll_param->pre_factor = 0; + if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) { + pll_ratio <<= 1; + pll_param->pre_factor = 1; + } + pll_param->pll_int = (pll_ratio >> 28) & 0xF; + pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4); + + return 0; +} + +static int nau8810_set_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct regmap *map = nau8810->regmap; + struct nau8810_pll *pll_param = &nau8810->pll; + int ret, fs; + + fs = freq_out / 256; + ret = nau88l0_calc_pll(freq_in, fs, pll_param); + if (ret < 0) { + dev_err(nau8810->dev, "Unsupported input clock %d\n", freq_in); + return ret; + } + dev_info(nau8810->dev, "pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n", + pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, + pll_param->pre_factor); + + regmap_update_bits(map, NAU8810_REG_PLLN, + NAU8810_PLLMCLK_DIV2 | NAU8810_PLLN_MASK, + (pll_param->pre_factor ? NAU8810_PLLMCLK_DIV2 : 0) | + pll_param->pll_int); + regmap_write(map, NAU8810_REG_PLLK1, + (pll_param->pll_frac >> NAU8810_PLLK1_SFT) & + NAU8810_PLLK1_MASK); + regmap_write(map, NAU8810_REG_PLLK2, + (pll_param->pll_frac >> NAU8810_PLLK2_SFT) & + NAU8810_PLLK2_MASK); + regmap_write(map, NAU8810_REG_PLLK3, + pll_param->pll_frac & NAU8810_PLLK3_MASK); + regmap_update_bits(map, NAU8810_REG_CLOCK, NAU8810_MCLKSEL_MASK, + pll_param->mclk_scaler << NAU8810_MCLKSEL_SFT); + regmap_update_bits(map, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_PLL); + + return 0; +} + +static int nau8810_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + u16 ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8810_CLKIO_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8810_AIFMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8810_AIFMT_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8810_AIFMT_PCM_A; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1_val |= NAU8810_BCLKP_IB | NAU8810_FSP_IF; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8810_BCLKP_IB; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1_val |= NAU8810_FSP_IF; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8810->regmap, NAU8810_REG_IFACE, + NAU8810_AIFMT_MASK | NAU8810_FSP_IF | + NAU8810_BCLKP_IB, ctrl1_val); + regmap_update_bits(nau8810->regmap, NAU8810_REG_CLOCK, + NAU8810_CLKIO_MASK, ctrl2_val); + + return 0; +} + +static int nau8810_mclk_clkdiv(struct nau8810 *nau8810, int rate) +{ + int i, sclk, imclk = rate * 256, div = 0; + + if (!nau8810->sysclk) { + dev_err(nau8810->dev, "Make mclk div configuration fail because of invalid system clock\n"); + return -EINVAL; + } + + /* Configure the master clock prescaler div to make system + * clock to approximate the internal master clock (IMCLK); + * and large or equal to IMCLK. + */ + for (i = 1; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { + sclk = (nau8810->sysclk * 10) / + nau8810_mclk_scaler[i]; + if (sclk < imclk) + break; + div = i; + } + dev_dbg(nau8810->dev, + "master clock prescaler %x for fs %d\n", div, rate); + + /* master clock from MCLK and disable PLL */ + regmap_update_bits(nau8810->regmap, NAU8810_REG_CLOCK, + NAU8810_MCLKSEL_MASK, (div << NAU8810_MCLKSEL_SFT)); + regmap_update_bits(nau8810->regmap, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_MCLK); + + return 0; +} + +static int nau8810_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + int val_len = 0, val_rate = 0, ret = 0; + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= NAU8810_WLEN_20; + break; + case 24: + val_len |= NAU8810_WLEN_24; + break; + case 32: + val_len |= NAU8810_WLEN_32; + break; + } + + switch (params_rate(params)) { + case 8000: + val_rate |= NAU8810_SMPLR_8K; + break; + case 11025: + val_rate |= NAU8810_SMPLR_12K; + break; + case 16000: + val_rate |= NAU8810_SMPLR_16K; + break; + case 22050: + val_rate |= NAU8810_SMPLR_24K; + break; + case 32000: + val_rate |= NAU8810_SMPLR_32K; + break; + case 44100: + case 48000: + break; + } + + regmap_update_bits(nau8810->regmap, NAU8810_REG_IFACE, + NAU8810_WLEN_MASK, val_len); + regmap_update_bits(nau8810->regmap, NAU8810_REG_SMPLR, + NAU8810_SMPLR_MASK, val_rate); + + /* If the master clock is from MCLK, provide the runtime FS for driver + * to get the master clock prescaler configuration. + */ + if (nau8810->clk_id == NAU8810_SCLK_MCLK) { + ret = nau8810_mclk_clkdiv(nau8810, params_rate(params)); + if (ret < 0) + dev_err(nau8810->dev, "MCLK div configuration fail\n"); + } + + return ret; +} + +static int nau8810_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct regmap *map = nau8810->regmap; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_80K); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_IOBUF_EN | NAU8810_ABIAS_EN, + NAU8810_IOBUF_EN | NAU8810_ABIAS_EN); + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + regcache_sync(map); + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_3K); + mdelay(100); + } + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_300K); + break; + + case SND_SOC_BIAS_OFF: + regmap_write(map, NAU8810_REG_POWER1, 0); + regmap_write(map, NAU8810_REG_POWER2, 0); + regmap_write(map, NAU8810_REG_POWER3, 0); + break; + } + + return 0; +} + + +#define NAU8810_RATES (SNDRV_PCM_RATE_8000_48000) + +#define NAU8810_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8810_ops = { + .hw_params = nau8810_pcm_hw_params, + .set_fmt = nau8810_set_dai_fmt, + .set_sysclk = nau8810_set_sysclk, + .set_pll = nau8810_set_pll, +}; + +static struct snd_soc_dai_driver nau8810_dai = { + .name = "nau8810-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = NAU8810_RATES, + .formats = NAU8810_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = NAU8810_RATES, + .formats = NAU8810_FORMATS, + }, + .ops = &nau8810_ops, + .symmetric_rates = 1, +}; + +static const struct regmap_config nau8810_regmap_config = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = NAU8810_REG_MAX, + .readable_reg = nau8810_readable_reg, + .writeable_reg = nau8810_writeable_reg, + .volatile_reg = nau8810_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8810_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8810_reg_defaults), +}; + +static struct snd_soc_codec_driver nau8810_codec_driver = { + .set_bias_level = nau8810_set_bias_level, + .suspend_bias_off = true, + + .component_driver = { + .controls = nau8810_snd_controls, + .num_controls = ARRAY_SIZE(nau8810_snd_controls), + .dapm_widgets = nau8810_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8810_dapm_widgets), + .dapm_routes = nau8810_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8810_dapm_routes), + }, +}; + +static int nau8810_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8810 *nau8810 = dev_get_platdata(dev); + + if (!nau8810) { + nau8810 = devm_kzalloc(dev, sizeof(*nau8810), GFP_KERNEL); + if (!nau8810) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8810); + + nau8810->regmap = devm_regmap_init_i2c(i2c, &nau8810_regmap_config); + if (IS_ERR(nau8810->regmap)) + return PTR_ERR(nau8810->regmap); + nau8810->dev = dev; + + regmap_write(nau8810->regmap, NAU8810_REG_RESET, 0x00); + + return snd_soc_register_codec(dev, + &nau8810_codec_driver, &nau8810_dai, 1); +} + +static int nau8810_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id nau8810_i2c_id[] = { + { "nau8810", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8810_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id nau8810_of_match[] = { + { .compatible = "nuvoton,nau8810", }, + { } +}; +MODULE_DEVICE_TABLE(of, nau8810_of_match); +#endif + +static struct i2c_driver nau8810_i2c_driver = { + .driver = { + .name = "nau8810", + .of_match_table = of_match_ptr(nau8810_of_match), + }, + .probe = nau8810_i2c_probe, + .remove = nau8810_i2c_remove, + .id_table = nau8810_i2c_id, +}; + +module_i2c_driver(nau8810_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8810 driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8810.h b/sound/soc/codecs/nau8810.h new file mode 100644 index 0000000..df88265 --- /dev/null +++ b/sound/soc/codecs/nau8810.h @@ -0,0 +1,281 @@ +/* + * NAU8810 ALSA SoC audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * Author: David Lin <ctlin0(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8810_H__ +#define __NAU8810_H__ + +#define NAU8810_REG_RESET 0x00 +#define NAU8810_REG_POWER1 0x01 +#define NAU8810_REG_POWER2 0x02 +#define NAU8810_REG_POWER3 0x03 +#define NAU8810_REG_IFACE 0x04 +#define NAU8810_REG_COMP 0x05 +#define NAU8810_REG_CLOCK 0x06 +#define NAU8810_REG_SMPLR 0x07 +#define NAU8810_REG_DAC 0x0A +#define NAU8810_REG_DACGAIN 0x0B +#define NAU8810_REG_ADC 0x0E +#define NAU8810_REG_ADCGAIN 0x0F +#define NAU8810_REG_EQ1 0x12 +#define NAU8810_REG_EQ2 0x13 +#define NAU8810_REG_EQ3 0x14 +#define NAU8810_REG_EQ4 0x15 +#define NAU8810_REG_EQ5 0x16 +#define NAU8810_REG_DACLIM1 0x18 +#define NAU8810_REG_DACLIM2 0x19 +#define NAU8810_REG_NOTCH1 0x1B +#define NAU8810_REG_NOTCH2 0x1C +#define NAU8810_REG_NOTCH3 0x1D +#define NAU8810_REG_NOTCH4 0x1E +#define NAU8810_REG_ALC1 0x20 +#define NAU8810_REG_ALC2 0x21 +#define NAU8810_REG_ALC3 0x22 +#define NAU8810_REG_NOISEGATE 0x23 +#define NAU8810_REG_PLLN 0x24 +#define NAU8810_REG_PLLK1 0x25 +#define NAU8810_REG_PLLK2 0x26 +#define NAU8810_REG_PLLK3 0x27 +#define NAU8810_REG_ATTEN 0x28 +#define NAU8810_REG_INPUT_SIGNAL 0x2C +#define NAU8810_REG_PGAGAIN 0x2D +#define NAU8810_REG_ADCBOOST 0x2F +#define NAU8810_REG_OUTPUT 0x31 +#define NAU8810_REG_SPKMIX 0x32 +#define NAU8810_REG_SPKGAIN 0x36 +#define NAU8810_REG_MONOMIX 0x38 +#define NAU8810_REG_POWER4 0x3A +#define NAU8810_REG_TSLOTCTL1 0x3B +#define NAU8810_REG_TSLOTCTL2 0x3C +#define NAU8810_REG_DEVICE_REVID 0x3E +#define NAU8810_REG_I2C_DEVICEID 0x3F +#define NAU8810_REG_ADDITIONID 0x40 +#define NAU8810_REG_RESERVE 0x41 +#define NAU8810_REG_OUTCTL 0x45 +#define NAU8810_REG_ALC1ENHAN1 0x46 +#define NAU8810_REG_ALC1ENHAN2 0x47 +#define NAU8810_REG_MISCCTL 0x49 +#define NAU8810_REG_OUTTIEOFF 0x4B +#define NAU8810_REG_AGCP2POUT 0x4C +#define NAU8810_REG_AGCPOUT 0x4D +#define NAU8810_REG_AMTCTL 0x4E +#define NAU8810_REG_OUTTIEOFFMAN 0x4F +#define NAU8810_REG_MAX NAU8810_REG_OUTTIEOFFMAN + + +/* NAU8810_REG_POWER1 (0x1) */ +#define NAU8810_DCBUF_EN (0x1 << 8) +#define NAU8810_PLL_EN_SFT 5 +#define NAU8810_MICBIAS_EN_SFT 4 +#define NAU8810_ABIAS_EN (0x1 << 3) +#define NAU8810_IOBUF_EN (0x1 << 2) +#define NAU8810_REFIMP_MASK 0x3 +#define NAU8810_REFIMP_DIS 0x0 +#define NAU8810_REFIMP_80K 0x1 +#define NAU8810_REFIMP_300K 0x2 +#define NAU8810_REFIMP_3K 0x3 + +/* NAU8810_REG_POWER2 (0x2) */ +#define NAU8810_BST_EN_SFT 4 +#define NAU8810_PGA_EN_SFT 2 +#define NAU8810_ADC_EN_SFT 0 + +/* NAU8810_REG_POWER3 (0x3) */ +#define NAU8810_DAC_EN_SFT 0 +#define NAU8810_SPKMX_EN_SFT 2 +#define NAU8810_MOUTMX_EN_SFT 3 +#define NAU8810_PSPK_EN_SFT 5 +#define NAU8810_NSPK_EN_SFT 6 +#define NAU8810_MOUT_EN_SFT 7 + +/* NAU8810_REG_IFACE (0x4) */ +#define NAU8810_AIFMT_SFT 3 +#define NAU8810_AIFMT_MASK (0x3 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_RIGHT (0x0 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_LEFT (0x1 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_I2S (0x2 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_PCM_A (0x3 << NAU8810_AIFMT_SFT) +#define NAU8810_WLEN_SFT 5 +#define NAU8810_WLEN_MASK (0x3 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_16 (0x0 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_20 (0x1 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_24 (0x2 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_32 (0x3 << NAU8810_WLEN_SFT) +#define NAU8810_FSP_IF (0x1 << 7) +#define NAU8810_BCLKP_IB (0x1 << 8) + +/* NAU8810_REG_COMP (0x5) */ +#define NAU8810_ADDAP_SFT 0 +#define NAU8810_ADCCM_SFT 1 +#define NAU8810_DACCM_SFT 3 + +/* NAU8810_REG_CLOCK (0x6) */ +#define NAU8810_CLKIO_MASK 0x1 +#define NAU8810_CLKIO_SLAVE 0x0 +#define NAU8810_CLKIO_MASTER 0x1 +#define NAU8810_BCLKSEL_SFT 2 +#define NAU8810_BCLKSEL_MASK (0x7 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_1 (0x0 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_2 (0x1 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_4 (0x2 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_8 (0x3 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_16 (0x4 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_32 (0x5 << NAU8810_BCLKSEL_SFT) +#define NAU8810_MCLKSEL_SFT 5 +#define NAU8810_MCLKSEL_MASK (0x7 << NAU8810_MCLKSEL_SFT) +#define NAU8810_CLKM_SFT 8 +#define NAU8810_CLKM_MASK (0x1 << NAU8810_CLKM_SFT) +#define NAU8810_CLKM_MCLK (0x0 << NAU8810_CLKM_SFT) +#define NAU8810_CLKM_PLL (0x1 << NAU8810_CLKM_SFT) + +/* NAU8810_REG_SMPLR (0x7) */ +#define NAU8810_SMPLR_SFT 1 +#define NAU8810_SMPLR_MASK (0x7 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_48K (0x0 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_32K (0x1 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_24K (0x2 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_16K (0x3 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_12K (0x4 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_8K (0x5 << NAU8810_SMPLR_SFT) + +/* NAU8810_REG_DAC (0xA) */ +#define NAU8810_DACPL_SFT 0 +#define NAU8810_DACOS_SFT 3 +#define NAU8810_DEEMP_SFT 4 + +/* NAU8810_REG_DACGAIN (0xB) */ +#define NAU8810_DACGAIN_SFT 0 + +/* NAU8810_REG_ADC (0xE) */ +#define NAU8810_ADCPL_SFT 0 +#define NAU8810_ADCOS_SFT 3 +#define NAU8810_HPF_SFT 4 +#define NAU8810_HPFEN_SFT 8 + +/* NAU8810_REG_ADCGAIN (0xF) */ +#define NAU8810_ADCGAIN_SFT 0 + +/* NAU8810_REG_EQ1 (0x12) */ +#define NAU8810_EQ1GC_SFT 0 +#define NAU8810_EQ1CF_SFT 5 +#define NAU8810_EQM_SFT 8 + +/* NAU8810_REG_EQ2 (0x13) */ +#define NAU8810_EQ2GC_SFT 0 +#define NAU8810_EQ2CF_SFT 5 +#define NAU8810_EQ2BW_SFT 8 + +/* NAU8810_REG_EQ3 (0x14) */ +#define NAU8810_EQ3GC_SFT 0 +#define NAU8810_EQ3CF_SFT 5 +#define NAU8810_EQ3BW_SFT 8 + +/* NAU8810_REG_EQ4 (0x15) */ +#define NAU8810_EQ4GC_SFT 0 +#define NAU8810_EQ4CF_SFT 5 +#define NAU8810_EQ4BW_SFT 8 + +/* NAU8810_REG_EQ5 (0x16) */ +#define NAU8810_EQ5GC_SFT 0 +#define NAU8810_EQ5CF_SFT 5 + +/* NAU8810_REG_DACLIM1 (0x18) */ +#define NAU8810_DACLIMATK_SFT 0 +#define NAU8810_DACLIMDCY_SFT 4 +#define NAU8810_DACLIMEN_SFT 8 + +/* NAU8810_REG_DACLIM2 (0x19) */ +#define NAU8810_DACLIMBST_SFT 0 +#define NAU8810_DACLIMTHL_SFT 4 + +/* NAU8810_REG_ALC1 (0x20) */ +#define NAU8810_ALCMINGAIN_SFT 0 +#define NAU8810_ALCMXGAIN_SFT 3 +#define NAU8810_ALCEN_SFT 8 + +/* NAU8810_REG_ALC2 (0x21) */ +#define NAU8810_ALCSL_SFT 0 +#define NAU8810_ALCHT_SFT 4 +#define NAU8810_ALCZC_SFT 8 + +/* NAU8810_REG_ALC3 (0x22) */ +#define NAU8810_ALCATK_SFT 0 +#define NAU8810_ALCDCY_SFT 4 +#define NAU8810_ALCM_SFT 8 + +/* NAU8810_REG_NOISEGATE (0x23) */ +#define NAU8810_ALCNTH_SFT 0 +#define NAU8810_ALCNEN_SFT 3 + +/* NAU8810_REG_PLLN (0x24) */ +#define NAU8810_PLLN_MASK 0xF +#define NAU8810_PLLMCLK_DIV2 (0x1 << 4) + +/* NAU8810_REG_PLLK1 (0x25) */ +#define NAU8810_PLLK1_SFT 18 +#define NAU8810_PLLK1_MASK 0x3F + +/* NAU8810_REG_PLLK2 (0x26) */ +#define NAU8810_PLLK2_SFT 9 +#define NAU8810_PLLK2_MASK 0x1FF + +/* NAU8810_REG_PLLK3 (0x27) */ +#define NAU8810_PLLK3_MASK 0x1FF + +/* NAU8810_REG_INPUT_SIGNAL (0x2C) */ +#define NAU8810_PMICPGA_SFT 0 +#define NAU8810_NMICPGA_SFT 1 + +/* NAU8810_REG_PGAGAIN (0x2D) */ +#define NAU8810_PGAGAIN_SFT 0 +#define NAU8810_PGAMT_SFT 6 +#define NAU8810_PGAZC_SFT 7 + +/* NAU8810_REG_ADCBOOST (0x2F) */ +#define NAU8810_PMICBSTGAIN_SFT 4 +#define NAU8810_PGABST_SFT 8 + +/* NAU8810_REG_SPKMIX (0x32) */ +#define NAU8810_DACSPK_SFT 0 +#define NAU8810_BYPSPK_SFT 1 + +/* NAU8810_REG_SPKGAIN (0x36) */ +#define NAU8810_SPKGAIN_SFT 0 +#define NAU8810_SPKMT_SFT 6 +#define NAU8810_SPKZC_SFT 7 + +/* NAU8810_REG_MONOMIX (0x38) */ +#define NAU8810_DACMOUT_SFT 0 +#define NAU8810_BYPMOUT_SFT 1 +#define NAU8810_MOUTMXMT_SFT 6 + + +/* System Clock Source */ +enum { + NAU8810_SCLK_MCLK, + NAU8810_SCLK_PLL, +}; + +struct nau8810_pll { + int pre_factor; + int mclk_scaler; + int pll_frac; + int pll_int; +}; + +struct nau8810 { + struct device *dev; + struct regmap *regmap; + struct nau8810_pll pll; + int sysclk; + int clk_id; +}; + +#endif -- 2.6.4

2 1

[alsa-devel] [PATCH v2 0/2] alsabat: test scripts and reference amixer config files
by focus.luo＠linux.intel.com 22 Aug '16

22 Aug '16

From: Focus Luo <focus.luo(a)linux.intel.com> The patch set supplies some automated test scripts and reference amixer config files for linux audio driver testing by using alsa-bat as test tool on Intel platforms. The first patch includes automated test scripts for linux audio driver based on alsa-lib interface by using alsabat as test tool. It supports analog audio and display audio(HDMI/DP) interface. The second patch supplies some amixer reference config file on Intel platforms. Focus Luo (2): alsabat: automation test scripts alsabat: add amixer config files bat/Makefile.am | 1 + bat/tests/Makefile.am | 11 + bat/tests/README | 32 ++ bat/tests/alsabat_main.sh | 178 ++++++++++ bat/tests/analog_audio_playback_and_capture.sh | 84 +++++ bat/tests/asound_state/Makefile.am | 6 + bat/tests/asound_state/asound.state.Broadwell | 439 +++++++++++++++++++++++++ bat/tests/asound_state/asound.state.Haswell | 342 +++++++++++++++++++ bat/tests/asound_state/asound.state.Skylake | 437 ++++++++++++++++++++++++ bat/tests/dp_audio_playback.sh | 83 +++++ bat/tests/dp_audio_subdevice_number.sh | 68 ++++ bat/tests/hdmi_audio_playback.sh | 81 +++++ bat/tests/hdmi_audio_subdevice_number.sh | 65 ++++ bat/tests/map_test_case | 8 + configure.ac | 3 +- 15 files changed, 1837 insertions(+), 1 deletion(-) create mode 100644 bat/tests/Makefile.am create mode 100644 bat/tests/README create mode 100755 bat/tests/alsabat_main.sh create mode 100755 bat/tests/analog_audio_playback_and_capture.sh create mode 100644 bat/tests/asound_state/Makefile.am create mode 100644 bat/tests/asound_state/asound.state.Broadwell create mode 100644 bat/tests/asound_state/asound.state.Haswell create mode 100644 bat/tests/asound_state/asound.state.Skylake create mode 100755 bat/tests/dp_audio_playback.sh create mode 100755 bat/tests/dp_audio_subdevice_number.sh create mode 100755 bat/tests/hdmi_audio_playback.sh create mode 100755 bat/tests/hdmi_audio_subdevice_number.sh create mode 100644 bat/tests/map_test_case -- 1.9.1

1 2

[alsa-devel] [PATCH] pcm: Fix shm initialization race-condition
by Ismael Luceno 22 Aug '16

22 Aug '16

Easily seen when two threads try at the same time, one of them will fail. The bug was identified by using apulse with Skype. Fixes: dec428c35221 ("pcm: fix 'unable to create IPC shm instance' caused by fork from a thread") Fixes: https://github.com/i-rinat/apulse/issues/38 Signed-off-by: Ismael Luceno <ismael(a)iodev.co.uk> --- src/pcm/pcm_direct.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index c3925cc20fd3..643498375b34 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -96,11 +96,12 @@ int snd_pcm_direct_shm_create_or_connect(snd_pcm_direct_t *dmix) retryget: dmix->shmid = shmget(dmix->ipc_key, sizeof(snd_pcm_direct_share_t), dmix->ipc_perm); - if (dmix->shmid < 0) { - if (errno == ENOENT) + if (dmix->shmid < 0 && errno == ENOENT) { if ((dmix->shmid = shmget(dmix->ipc_key, sizeof(snd_pcm_direct_share_t), IPC_CREAT | IPC_EXCL | dmix->ipc_perm)) != -1) first_instance = 1; + else if (errno == EEXIST) + goto retryget; } err = -errno; if (dmix->shmid < 0) { -- 2.9.2

2 2

[alsa-devel] [PATCH 0/2] ASoC: Extended SPDIF support to Allwinner A31 SoCs
by codekipper＠gmail.com 22 Aug '16

22 Aug '16

From: Marcus Cooper <codekipper(a)gmail.com> This patch set extends the current support of the Allwinner SPDIF transceiver to include the A31 SoC. For now just the SPDIF transmitter has been tested on the Mele I7. In order for this patch set to be functional then we require changes to the clocks and the device tree. For those that are interested I've pushed all the required patches here https://github.com/codekipper/linux-sunxi/commits/a31-spdif Thanks in advance, CK Marcus Cooper (2): ASoC: sunxi: Add bindings for sun6i to SPDIF ASoC: sunxi: compatibility for sun6i to SPDIF .../devicetree/bindings/sound/sunxi,sun4i-spdif.txt | 3 +++ sound/soc/sunxi/sun4i-spdif.c | 17 ++++++++++++++++- 2 files changed, 19 insertions(+), 1 deletion(-) -- 2.9.2

8 15

Re: [alsa-devel] [PATCH] pcm: Fix shm initialization race-condition
by Takashi Iwai 22 Aug '16

22 Aug '16

On Sun, 14 Aug 2016 02:28:52 +0200, Ismael Luceno wrote: > > Easily seen when two threads try at the same time, one of them will fail. > > The bug was identified by using apulse with Skype. > > Fixes: dec428c35221 ("pcm: fix 'unable to create IPC shm instance' caused by fork from a thread") > Fixes: https://github.com/i-rinat/apulse/issues/38 > Signed-off-by: Ismael Luceno <ismael(a)iodev.co.uk> > --- > src/pcm/pcm_direct.c | 2 ++ > 1 file changed, 2 insertions(+) > > diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c > index c3925cc20fd3..b5215ba35406 100644 > --- a/src/pcm/pcm_direct.c > +++ b/src/pcm/pcm_direct.c > @@ -101,6 +101,8 @@ retryget: > if ((dmix->shmid = shmget(dmix->ipc_key, sizeof(snd_pcm_direct_share_t), > IPC_CREAT | IPC_EXCL | dmix->ipc_perm)) != -1) > first_instance = 1; > + if (dmix->shmid < 0 && errno == EEXIST) > + goto retryget; Hrm, but this would result in an endless loop if the shm was already taken persistently. Also, which call does give a negative shmid, actually? It's from the first shmget() or the second shmget()? (The existing code is ugly and we need to fix the indentation there...) thanks, Takashi

2 3

[alsa-devel] [PATCH] ASoC: core: Clean up DAPM before the card debugfs
by Mark Brown 22 Aug '16

22 Aug '16

Both the card and DAPM cleanups recursively delete their debugfs directories. Since the DAPM debugfs subdirectory for the card is located within the card debugfs this means we end up trying to double free the DAPM subdirectory. Reorder the cleanup to free the card debugfs after we've cleaned up DAPM and it has deleted its own subdirectory. Reported-by: Russell King - ARM Linux <linux(a)armlinux.org.uk> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Not tested at all yet, just about done for today. sound/soc/soc-core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16369cad4803..66a33f1e4881 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2083,14 +2083,13 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) /* remove auxiliary devices */ soc_remove_aux_devices(card); + snd_soc_dapm_free(&card->dapm); soc_cleanup_card_debugfs(card); /* remove the card */ if (card->remove) card->remove(card); - snd_soc_dapm_free(&card->dapm); - snd_card_free(card->snd_card); return 0; -- 2.8.1

3 7

[alsa-devel] [PATCH] ALSA: hdspm: fix spelling mistake "Externel" -> "External"
by Colin King 22 Aug '16

22 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/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a4a999a..0e51a00 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1666,7 +1666,7 @@ static int hdspm_set_rate(struct hdspm * hdspm, int rate, int called_internally) HDSPM_AUTOSYNC_FROM_NONE) { dev_warn(hdspm->card->dev, - "Detected no Externel Sync\n"); + "Detected no External Sync\n"); not_set = 1; } else if (rate != external_freq) { -- 2.9.3

2 1

Re: [alsa-devel] ASoC: sun4i-codec: playback stall and I/O error with DAPM paths all disabled
by Maxime Ripard 22 Aug '16

22 Aug '16

On Tue, Aug 16, 2016 at 12:45:39PM +0800, wens Tsai wrote: > On Mon, Aug 15, 2016 at 7:42 PM, Mark Brown <broonie(a)kernel.org> wrote: > > On Mon, Aug 15, 2016 at 05:43:55PM +0800, wens Tsai wrote: > > > >> What is unexpected is any attempt to play anything under this state makes > >> the playback software (in my case mpg321) stall, and later report an I/O > >> error. My guess is that the DAC is still disabled by DAPM, so it doesn't > >> send any DRQs, and thus the DMA engine is not consuming any data from > >> userspace. > > > > This is normal for ASoC - like you say it'll be becasue the hardware > > isn't powered up. > > > >> I think we should just enable the digital bits of the DAC/ADC all the > >> time. Or maybe transfer and then discard data if the DAC is off. Not > >> sure if this is doable though. I expect playback software to work, and > >> not block, regardless of the hardware status. > > > > Powering things up all the time will have a major effect on battery life > > for systems that care about that. The expectation is that systems with > > this sort of hardware won't normally be offering end users direct > > control of the routing, it'll be something that's handled during system > > integration. > > Ok. So I guess one solution would be to move the mute controls out of > DAPM, and maybe change some other mux like paths into actual muxes, so > there's at least one usable path at all times. I'm not sure that's even a requirement. IIRC, but Mark might correct me, you just need to expose whatever the reset state is. Maxime -- Maxime Ripard, Free Electrons Embedded Linux and Kernel engineering http://free-electrons.com

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: rmove print for failure of kmalloc
by Shawn Lin 22 Aug '16

22 Aug '16

kmalloc already print similar error once failing to alloc enough memory, so let's remove this dump here. Signed-off-by: Shawn Lin <shawn.lin(a)rock-chips.com> --- sound/usb/format.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 789d19e..2c44386 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -176,10 +176,8 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof int r, idx; fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); - if (fp->rate_table == NULL) { - usb_audio_err(chip, "cannot malloc\n"); + if (fp->rate_table == NULL) return -ENOMEM; - } fp->nr_rates = 0; fp->rate_min = fp->rate_max = 0; -- 2.3.7

2 1

[alsa-devel] [PATCH 1/3] ALSA: usb: move udh01_fb_quirk setting to quirks.c
by Daniel Mack 22 Aug '16

22 Aug '16

That's a quirk, after all, so move it where to all the other quirks live. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/usb/endpoint.c | 4 ---- sound/usb/quirks.c | 5 +++++ 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index c07a7ed..4b4ffa0 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -502,10 +502,6 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, ep->syncinterval = 3; ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); - - if (chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ && - ep->syncmaxsize == 4) - ep->udh01_fb_quirk = 1; } list_add_tail(&ep->list, &chip->ep_list); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6cf1f35..299813f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1216,6 +1216,11 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) ep->chip->usb_id == USB_ID(0x0763, 0x2031)) && ep->type == SND_USB_ENDPOINT_TYPE_DATA) ep->skip_packets = 16; + + /* Work around devices that report unreasonable feedback data */ + if (ep->chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ && + ep->syncmaxsize == 4) + ep->udh01_fb_quirk = 1; } void snd_usb_set_interface_quirk(struct usb_device *dev) -- 2.5.5

2 3

[alsa-devel] [PATCH] mixer: Fix rounding mode documentation
by David Henningsson 22 Aug '16

22 Aug '16

The documentation specifies three different rounding modes, but this is wrong, because they all end up calling snd_tlv_convert_from_dB, which only has two rounding modes. Signed-off-by: David Henningsson <diwic(a)ubuntu.com> --- src/mixer/simple.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/src/mixer/simple.c b/src/mixer/simple.c index fd9ba93..82476d5 100644 --- a/src/mixer/simple.c +++ b/src/mixer/simple.c @@ -376,7 +376,7 @@ int snd_mixer_selem_ask_playback_vol_dB(snd_mixer_elem_t *elem, long value, long * \brief Return corresponding integer playback volume for given dB value for a mixer simple element * \param elem Mixer simple element handle * \param value value to be converted to dB range - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \param dBvalue pointer to returned dB value * \return 0 on success otherwise a negative error code */ @@ -454,7 +454,7 @@ int snd_mixer_selem_set_playback_volume(snd_mixer_elem_t *elem, snd_mixer_selem_ * \param elem Mixer simple element handle * \param channel mixer simple element channel identifier * \param value control value in dB * 100 - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_set_playback_dB(snd_mixer_elem_t *elem, snd_mixer_selem_channel_id_t channel, long value, int dir) @@ -491,7 +491,7 @@ int snd_mixer_selem_set_playback_volume_all(snd_mixer_elem_t *elem, long value) * \brief Set value in dB of playback volume control for all channels of a mixer simple element * \param elem Mixer simple element handle * \param value control value in dB * 100 - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_set_playback_dB_all(snd_mixer_elem_t *elem, long value, int dir) @@ -706,7 +706,7 @@ int snd_mixer_selem_ask_capture_vol_dB(snd_mixer_elem_t *elem, long value, long * \param elem Mixer simple element handle * \param dBvalue dB value to be converted to integer range * \param value pointer to returned integer value - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_ask_capture_dB_vol(snd_mixer_elem_t *elem, long dBvalue, int dir, long *value) @@ -777,7 +777,7 @@ int snd_mixer_selem_set_capture_volume(snd_mixer_elem_t *elem, snd_mixer_selem_c * \param elem Mixer simple element handle * \param channel mixer simple element channel identifier * \param value control value in dB * 100 - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_set_capture_dB(snd_mixer_elem_t *elem, snd_mixer_selem_channel_id_t channel, long value, int dir) @@ -814,7 +814,7 @@ int snd_mixer_selem_set_capture_volume_all(snd_mixer_elem_t *elem, long value) * \brief Set value in dB of capture volume control for all channels of a mixer simple element * \param elem Mixer simple element handle * \param value control value in dB * 100 - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_set_capture_dB_all(snd_mixer_elem_t *elem, long value, int dir) -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] amidi: fix handling of ignored messages
by Clemens Ladisch 22 Aug '16

22 Aug '16

Clemens Ladisch (2): amidi: ignore not only Active Sensing but also Clock bytes amidi: fix timeout handling amidi/amidi.1 | 19 ++++++++--- amidi/amidi.c | 82 +++++++++++++++++++++++++++++++++++++++----------- 2 files changed, 78 insertions(+), 23 deletions(-)

4 11

[alsa-devel] [PATCH v4 0/4] ALSA: seq: obsolete change of address limit
by Takashi Sakamoto 22 Aug '16

22 Aug '16

ALSA sequencer core has two types of clients; application and kernel. For kernel clients, the core should handle data in kernel space, meanwhile this is achieved with a hack of change of address limit. This should be purged. This patchset is a revised version of my previous one: [alsa-devel] obsolete change of address limit http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/110937.html Changes in v2: - Use '_IOC_SIZE' macro to calculate the size of argument of ioctl command. - Keep redundant longer name. - Improve comments. - Merge some patches relevant to the same features. Changes in v3: - Improve series of patch - Use 'dir' field of ioctl command to detect the length of argument Changes in v4: - Update kernel API documentation - Use logical AND to detect ioctl direction Takashi Sakamoto (4): ALSA: seq: add documentation for snd_seq_kernel_client_ctl ALSA: seq: add an alternative way to handle ioctl requests ALSA: seq: change ioctl command operation to get data in kernel space ALSA: seq: obsolete change of address limit sound/core/seq/seq_clientmgr.c | 721 ++++++++++++++++++----------------------- sound/core/seq/seq_compat.c | 7 +- 2 files changed, 317 insertions(+), 411 deletions(-) -- 2.7.4

2 6

[alsa-devel] [PATCH -next] ASoC: sun4i-codec: Fix error return code in sun4i_codec_probe()
by Wei Yongjun 22 Aug '16

22 Aug '16

Fix to return error code -ENOMEM instead of 0 when create card failed, as done elsewhere in this function. Fixes: 45fb6b6f2aa3 ("ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs") Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/sunxi/sun4i-codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 0e19c50..e741659 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -846,6 +846,7 @@ static int sun4i_codec_probe(struct platform_device *pdev) card = sun4i_codec_create_card(&pdev->dev); if (!card) { dev_err(&pdev->dev, "Failed to create our card\n"); + ret = -ENOMEM; goto err_unregister_codec; }

3 3

[alsa-devel] [PATCH 000/127] ASoC: codec cleanup - codec probe/remove
by Kuninori Morimoto 22 Aug '16

22 Aug '16

Hi Mark These are codec duplicate functions cleanup patches - v2 We would like to switch to "component" style base ASoC in the future instead of current "CPU/Codec/Platform/Card" style. Current "codec" has its .probe/.remove, and "component" has it too. codec side is just relayed it to component side. This is the time to cleanup. These are very big patch-set, but it doesn't add new features. [001/127] - [126/127] move callback functions from codec to component, and last [127/127] will remove codec side .probe/.remove. We need this kind of cleanup for Platform too. I build-tested these patches with make allyesconfig on x86, and tested on Lager board. Best regards --- Kuninori Morimoto

3 137

Re: [alsa-devel] [PATCH] Try to support analog audio codec at firefly reload
by John Keeping 20 Aug '16

20 Aug '16

On Tue, Aug 16, 2016 at 04:42:06AM +0800, Randy Li wrote: > This patch could enabled the es8328 codec at firefly reload board, > but it doesn't not really work. I disable the mute of channel > Left mix and Right mix in alsamixer, I could hear the noise from > my headphone, but it never be a music. > I just send this patch here to see whether somebody could figure > out what is wrong, I am really not familiar with sound. Have you tried adding an mclk-fs setting to the simple-audio-card? You probably want to set it to 256, which will allow 44.1kHz, 48kHz, 88.2kHz and 96kHz at the codec. This will give more functionality than pinning the clock to a specific rate. For the Rock2 Square, I write a machine driver for the sound because the headphone GPIO setup cannot be made to work with simple-audio-card; I found the Firefly Reload schematic and it looks like the configuration hardware is very similar. It's a bit of a mess but you can find it on Github [1] if you want to take a look. [1] https://github.com/johnkeeping/linux/tree/topic/audio

1 0

[alsa-devel] [PATCH 1/3] ASoC: fsl_asrc: Propagate the real error code
by Fabio Estevam 19 Aug '16

19 Aug '16

Instead of returning -EINVAL on error, return the real error code. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index c1a0e01..15f8ba5 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -892,7 +892,7 @@ static int fsl_asrc_probe(struct platform_device *pdev) ret = fsl_asrc_init(asrc_priv); if (ret) { dev_err(&pdev->dev, "failed to init asrc %d\n", ret); - return -EINVAL; + return ret; } asrc_priv->channel_avail = 10; @@ -901,14 +901,14 @@ static int fsl_asrc_probe(struct platform_device *pdev) &asrc_priv->asrc_rate); if (ret) { dev_err(&pdev->dev, "failed to get output rate\n"); - return -EINVAL; + return ret; } ret = of_property_read_u32(np, "fsl,asrc-width", &asrc_priv->asrc_width); if (ret) { dev_err(&pdev->dev, "failed to get output width\n"); - return -EINVAL; + return ret; } if (asrc_priv->asrc_width != 16 && asrc_priv->asrc_width != 24) { -- 1.9.1

2 4

[alsa-devel] [PATCH for 4.8] ASoC: omap-abe-twl6040: Correct dmic-codec device registration
by Peter Ujfalusi 19 Aug '16

19 Aug '16

The dmic-codec was registered within the platform_driver's probe function, which can cause deferred probe to run in loops as reported and analyzed by Russell King. Use module_init/exit in the driver and handle the dmic-codec device registration and removal at that level instead of the platform_driver probe/remove. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reported-by: Russell King <linux(a)arm.linux.org.uk> --- sound/soc/omap/omap-abe-twl6040.c | 61 ++++++++++++++++++++------------------- 1 file changed, 32 insertions(+), 29 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 0843a68f277c..f61b3b58083b 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -38,10 +38,10 @@ struct abe_twl6040 { int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ - - struct platform_device *dmic_codec_dev; }; +struct platform_device *dmic_codec_dev; + static int omap_abe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -258,8 +258,6 @@ static int omap_abe_probe(struct platform_device *pdev) if (priv == NULL) return -ENOMEM; - priv->dmic_codec_dev = ERR_PTR(-EINVAL); - if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; @@ -284,13 +282,6 @@ static int omap_abe_probe(struct platform_device *pdev) num_links = 2; abe_twl6040_dai_links[1].cpu_of_node = dai_node; abe_twl6040_dai_links[1].platform_of_node = dai_node; - - priv->dmic_codec_dev = platform_device_register_simple( - "dmic-codec", -1, NULL, 0); - if (IS_ERR(priv->dmic_codec_dev)) { - dev_err(&pdev->dev, "Can't instantiate dmic-codec\n"); - return PTR_ERR(priv->dmic_codec_dev); - } } else { num_links = 1; } @@ -299,16 +290,14 @@ static int omap_abe_probe(struct platform_device *pdev) of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency not provided\n"); - ret = -EINVAL; - goto err_unregister; + return -EINVAL; } card->fully_routed = 1; if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); - ret = -ENODEV; - goto err_unregister; + return -ENODEV; } card->dai_link = abe_twl6040_dai_links; @@ -317,17 +306,9 @@ static int omap_abe_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); ret = snd_soc_register_card(card); - if (ret) { + if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); - goto err_unregister; - } - - return 0; - -err_unregister: - if (!IS_ERR(priv->dmic_codec_dev)) - platform_device_unregister(priv->dmic_codec_dev); return ret; } @@ -335,13 +316,9 @@ err_unregister: static int omap_abe_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); snd_soc_unregister_card(card); - if (!IS_ERR(priv->dmic_codec_dev)) - platform_device_unregister(priv->dmic_codec_dev); - return 0; } @@ -361,7 +338,33 @@ static struct platform_driver omap_abe_driver = { .remove = omap_abe_remove, }; -module_platform_driver(omap_abe_driver); +static int __init omap_abe_init(void) +{ + int ret; + + dmic_codec_dev = platform_device_register_simple("dmic-codec", -1, NULL, + 0); + if (IS_ERR(dmic_codec_dev)) { + pr_err("%s: dmic-codec device registration failed\n", __func__); + return PTR_ERR(dmic_codec_dev); + } + + ret = platform_driver_register(&omap_abe_driver); + if (ret) { + pr_err("%s: platform driver registration failed\n", __func__); + platform_device_unregister(dmic_codec_dev); + } + + return ret; +} +module_init(omap_abe_init); + +static void __exit omap_abe_exit(void) +{ + platform_driver_unregister(&omap_abe_driver); + platform_device_unregister(dmic_codec_dev); +} +module_exit(omap_abe_exit); MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez(a)ti.com>"); MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec"); -- 2.9.2

3 3

Re: [alsa-devel] [PATCH] Try to support analog audio codec at firefly reload
by Heiko Stuebner 19 Aug '16

19 Aug '16

Am Dienstag, 16. August 2016, 04:42:06 CEST schrieb Randy Li: > Hello: > This patch could enabled the es8328 codec at firefly reload board, > but it doesn't not really work. I disable the mute of channel > Left mix and Right mix in alsamixer, I could hear the noise from > my headphone, but it never be a music. > I just send this patch here to see whether somebody could figure > out what is wrong, I am really not familiar with sound. I can't really help here as I have neither the board nor that much sound experience, but it's definitly customary to label patches as rfc when they're expected to need more work, so make that [PATCH RFC] ... for stuff that requires more work before it can actually be merged. Heiko

1 0

[alsa-devel] [PATCH] ASoC: rockchip: Add support rt5514 dsp summy dailink
by Xing Zheng 19 Aug '16

19 Aug '16

This patch can attach automaticlly rt5514 spi DAI with driver name "rt5514" in the snd_soc_find_dai process. Turn this feature on, we can enable the voice wake up via rt5514 dsp for RK3399 Gru Boards. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- sound/soc/rockchip/Kconfig | 1 + sound/soc/rockchip/rk3399_gru_sound.c | 39 ++++++++++++++++++++++++++++++++- 2 files changed, 39 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index 6d39032..d82d763 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -49,6 +49,7 @@ config SND_SOC_RK3399_GRU_SOUND select SND_SOC_MAX98357A select SND_SOC_RT5514 select SND_SOC_DA7219 + select SND_SOC_RT5514_SPI help Say Y or M here if you want to add support multiple codecs for SoC audio on Rockchip RK3399 GRU boards. diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 9933703..164b6da 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -230,8 +230,11 @@ enum { DAILINK_MAX98357A, DAILINK_RT5514, DAILINK_DA7219, + DAILINK_RT5514_DSP, }; +#define DAILINK_ENTITIES (DAILINK_DA7219 + 1) + static struct snd_soc_dai_link rockchip_dailinks[] = { [DAILINK_MAX98357A] = { .name = "MAX98357A", @@ -261,6 +264,13 @@ static struct snd_soc_dai_link rockchip_dailinks[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, }, + /* RT5514 DSP for voice wakeup via spi bus */ + [DAILINK_RT5514_DSP] = { + .name = "RT5514 DSP", + .stream_name = "Wake on Voice", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + }, }; static struct snd_soc_card rockchip_sound_card = { @@ -276,10 +286,17 @@ static struct snd_soc_card rockchip_sound_card = { .num_controls = ARRAY_SIZE(rockchip_controls), }; +static int rockchip_sound_match_stub(struct device *dev, void *data) +{ + return 1; +} + static int rockchip_sound_probe(struct platform_device *pdev) { struct snd_soc_card *card = &rockchip_sound_card; struct device_node *cpu_node; + struct device *dev; + struct device_driver *drv; int i, ret; cpu_node = of_parse_phandle(pdev->dev.of_node, "rockchip,cpu", 0); @@ -288,7 +305,7 @@ static int rockchip_sound_probe(struct platform_device *pdev) return -EINVAL; } - for (i = 0; i < card->num_links; i++) { + for (i = 0; i < DAILINK_ENTITIES; i++) { rockchip_dailinks[i].platform_of_node = cpu_node; rockchip_dailinks[i].cpu_of_node = cpu_node; @@ -301,6 +318,26 @@ static int rockchip_sound_probe(struct platform_device *pdev) } } + /** + * To acquire the spi driver of the rt5514 and set the dai-links names + * for soc_bind_dai_link + */ + drv = driver_find("rt5514", &spi_bus_type); + if (!drv) { + dev_err(&pdev->dev, "Can not find the rt5514 driver at the spi bus\n"); + return -EINVAL; + } + + dev = driver_find_device(drv, NULL, NULL, rockchip_sound_match_stub); + if (!dev) { + dev_err(&pdev->dev, "Can not find the rt5514 device\n"); + return -ENODEV; + } + + rockchip_dailinks[DAILINK_RT5514_DSP].cpu_name = kstrdup_const(dev_name(dev), GFP_KERNEL); + rockchip_dailinks[DAILINK_RT5514_DSP].cpu_dai_name = kstrdup_const(dev_name(dev), GFP_KERNEL); + rockchip_dailinks[DAILINK_RT5514_DSP].platform_name = kstrdup_const(dev_name(dev), GFP_KERNEL); + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

1 0

[alsa-devel] Use of copy_to_user in fireworks_hwdep.c while holding a spin_lock
by Vaishali Thakkar 19 Aug '16

19 Aug '16

Hello, I was wondering about the call to copy_to_user in function hwdep_read_locked and hwdep_read_resp_buf for driver sound/firewire/fireworks/fireworks_hwdep.c. The function hwdep_read calls both of these functions while holding a spinlock[1], which is not normally allowed due to the possibility of a deadlock. This seems to be coming from the commit 555e8a8f7f149544eb7d4aa3a6420bc4c3055638 while adding a command/response functionality into hwdep interface. Is there some reason that I am overlooking, why it is OK in this case? Is there some code in the same file which ensures that page fault will not occur when we are calling these functions while holding a spin_lock_irq? The same issue is there with the driver sound/firewire/tascam/tascam-hwdep.c for obvious reasons. Coccinelle script is used to detect this issue. Thank you. [1] http://lxr.free-electrons.com/source/sound/firewire/fireworks/fireworks_hwd… -- Vaishali

2 1

[alsa-devel] [PATCH 0/3] line6 locking and segfault fixes
by Andrej Krutak 19 Aug '16

19 Aug '16

These are already seen patches fixing a few locking issues, now split out of the POD-X3 patchset. In addition, pod sysfs attributes touched something else than they thought, causing segfaults - fixed it (tested with the same code in podhd)... Andrej Krutak (3): ALSA: line6: Remove double line6_pcm_release() after failed acquire. ALSA: line6: Give up on the lock while URBs are released. ALSA: line6: Fix POD sysfs attributes segfault sound/usb/line6/pcm.c | 3 ++- sound/usb/line6/pod.c | 12 ++++++------ 2 files changed, 8 insertions(+), 7 deletions(-) -- 1.9.1

3 6

[alsa-devel] [PATCH] ASoC: nau8810: Add driver for Nuvoton codec chip NAU88C10
by John Hsu 19 Aug '16

19 Aug '16

The driver is for codec NAU88C10 of Nuvoton Technology Corporation. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8810.txt | 16 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8810.c | 908 +++++++++++++++++++++ sound/soc/codecs/nau8810.h | 288 +++++++ 5 files changed, 1219 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8810.txt create mode 100644 sound/soc/codecs/nau8810.c create mode 100644 sound/soc/codecs/nau8810.h diff --git a/Documentation/devicetree/bindings/sound/nau8810.txt b/Documentation/devicetree/bindings/sound/nau8810.txt new file mode 100644 index 0000000..05830e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8810.txt @@ -0,0 +1,16 @@ +NAU8810 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "nuvoton,nau8810" + + - reg : the I2C address of the device. + +Example: + +codec: nau8810@1a { + compatible = "nuvoton,nau8810"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb57f8f..bbd7324 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -91,6 +91,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C @@ -1046,6 +1047,10 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8810 + tristate "Nuvoton Technology Corporation NAU88C10 CODEC" + depends on I2C + config SND_SOC_NAU8825 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 550a174..3c15bc5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -84,6 +84,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8810-objs := nau8810.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o snd-soc-pcm1681-objs := pcm1681.o @@ -301,6 +302,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c new file mode 100644 index 0000000..aa25286 --- /dev/null +++ b/sound/soc/codecs/nau8810.c @@ -0,0 +1,908 @@ +/* + * nau8810.c -- NAU8810 ALSA Soc Audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * + * Based on WM8974.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "nau8810.h" + +#define NAU_PLL_FREQ_MAX 100000000 +#define NAU_PLL_FREQ_MIN 90000000 +#define NAU_PLL_REF_MAX 33000000 +#define NAU_PLL_REF_MIN 8000000 +#define NAU_PLL_OPTOP_MIN 6 + + +static const int nau8810_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 }; + +static const struct reg_default nau8810_reg_defaults[] = { + { NAU8810_REG_POWER1, 0x0000 }, + { NAU8810_REG_POWER2, 0x0000 }, + { NAU8810_REG_POWER3, 0x0000 }, + { NAU8810_REG_IFACE, 0x0050 }, + { NAU8810_REG_COMP, 0x0000 }, + { NAU8810_REG_CLOCK, 0x0140 }, + { NAU8810_REG_SMPLR, 0x0000 }, + { NAU8810_REG_DAC, 0x0000 }, + { NAU8810_REG_DACGAIN, 0x00FF }, + { NAU8810_REG_ADC, 0x0100 }, + { NAU8810_REG_ADCGAIN, 0x00FF }, + { NAU8810_REG_EQ1, 0x012C }, + { NAU8810_REG_EQ2, 0x002C }, + { NAU8810_REG_EQ3, 0x002C }, + { NAU8810_REG_EQ4, 0x002C }, + { NAU8810_REG_EQ5, 0x002C }, + { NAU8810_REG_DACLIM1, 0x0032 }, + { NAU8810_REG_DACLIM2, 0x0000 }, + { NAU8810_REG_NOTCH1, 0x0000 }, + { NAU8810_REG_NOTCH2, 0x0000 }, + { NAU8810_REG_NOTCH3, 0x0000 }, + { NAU8810_REG_NOTCH4, 0x0000 }, + { NAU8810_REG_ALC1, 0x0038 }, + { NAU8810_REG_ALC2, 0x000B }, + { NAU8810_REG_ALC3, 0x0032 }, + { NAU8810_REG_NOISEGATE, 0x0000 }, + { NAU8810_REG_PLLN, 0x0008 }, + { NAU8810_REG_PLLK1, 0x000C }, + { NAU8810_REG_PLLK2, 0x0093 }, + { NAU8810_REG_PLLK3, 0x00E9 }, + { NAU8810_REG_ATTEN, 0x0000 }, + { NAU8810_REG_INPUT_SIGNAL, 0x0003 }, + { NAU8810_REG_PGAGAIN, 0x0010 }, + { NAU8810_REG_ADCBOOST, 0x0100 }, + { NAU8810_REG_OUTPUT, 0x0002 }, + { NAU8810_REG_SPKMIX, 0x0001 }, + { NAU8810_REG_SPKGAIN, 0x0039 }, + { NAU8810_REG_MONOMIX, 0x0001 }, + { NAU8810_REG_POWER4, 0x0000 }, + { NAU8810_REG_TSLOTCTL1, 0x0000 }, + { NAU8810_REG_TSLOTCTL2, 0x0020 }, + { NAU8810_REG_DEVICE_REVID, 0x0000 }, + { NAU8810_REG_I2C_DEVICEID, 0x001A }, + { NAU8810_REG_ADDITIONID, 0x00CA }, + { NAU8810_REG_RESERVE, 0x0124 }, + { NAU8810_REG_OUTCTL, 0x0001 }, + { NAU8810_REG_ALC1ENHAN1, 0x0010 }, + { NAU8810_REG_ALC1ENHAN2, 0x0000 }, + { NAU8810_REG_MISCCTL, 0x0000 }, + { NAU8810_REG_OUTTIEOFF, 0x0000 }, + { NAU8810_REG_AGCP2POUT, 0x0000 }, + { NAU8810_REG_AGCPOUT, 0x0000 }, + { NAU8810_REG_AMTCTL, 0x0000 }, + { NAU8810_REG_OUTTIEOFFMAN, 0x0000 }, +}; + +static bool nau8810_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET ... NAU8810_REG_SMPLR: + case NAU8810_REG_DAC ... NAU8810_REG_DACGAIN: + case NAU8810_REG_ADC ... NAU8810_REG_ADCGAIN: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_DACLIM1 ... NAU8810_REG_DACLIM2: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + case NAU8810_REG_ALC1 ... NAU8810_REG_ATTEN: + case NAU8810_REG_INPUT_SIGNAL ... NAU8810_REG_PGAGAIN: + case NAU8810_REG_ADCBOOST: + case NAU8810_REG_OUTPUT ... NAU8810_REG_SPKMIX: + case NAU8810_REG_SPKGAIN: + case NAU8810_REG_MONOMIX: + case NAU8810_REG_POWER4 ... NAU8810_REG_TSLOTCTL2: + case NAU8810_REG_DEVICE_REVID ... NAU8810_REG_RESERVE: + case NAU8810_REG_OUTCTL ... NAU8810_REG_ALC1ENHAN2: + case NAU8810_REG_MISCCTL: + case NAU8810_REG_OUTTIEOFF ... NAU8810_REG_OUTTIEOFFMAN: + return true; + default: + return false; + } +} + +static bool nau8810_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET ... NAU8810_REG_SMPLR: + case NAU8810_REG_DAC ... NAU8810_REG_DACGAIN: + case NAU8810_REG_ADC ... NAU8810_REG_ADCGAIN: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_DACLIM1 ... NAU8810_REG_DACLIM2: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + case NAU8810_REG_ALC1 ... NAU8810_REG_ATTEN: + case NAU8810_REG_INPUT_SIGNAL ... NAU8810_REG_PGAGAIN: + case NAU8810_REG_ADCBOOST: + case NAU8810_REG_OUTPUT ... NAU8810_REG_SPKMIX: + case NAU8810_REG_SPKGAIN: + case NAU8810_REG_MONOMIX: + case NAU8810_REG_POWER4 ... NAU8810_REG_TSLOTCTL2: + case NAU8810_REG_OUTCTL ... NAU8810_REG_ALC1ENHAN2: + case NAU8810_REG_MISCCTL: + case NAU8810_REG_OUTTIEOFF ... NAU8810_REG_OUTTIEOFFMAN: + return true; + default: + return false; + } +} + +static bool nau8810_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET: + case NAU8810_REG_DEVICE_REVID ... NAU8810_REG_RESERVE: + return true; + default: + return false; + } +} + +static int nau8810_eq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + int i, reg, reg_val; + u16 *val; + + val = (u16 *)ucontrol->value.bytes.data; + reg = NAU8810_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + regmap_read(nau8810->regmap, reg + i, &reg_val); + reg_val = cpu_to_be16(reg_val); + memcpy(val + i, &reg_val, sizeof(reg_val)); + } + + return 0; +} + +static int nau8810_eq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + u16 *val, value; + int i, reg, ret; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + val = (u16 *)data; + reg = NAU8810_REG_EQ1; + for (i = 0; i < params->max / sizeof(u16); i++) { + value = be16_to_cpu(*(val + i)); + ret = regmap_write(nau8810->regmap, reg + i, value); + if (ret) { + dev_err(codec->dev, "EQ configuration fail, register: %x ret: %d\n", + reg + i, ret); + return ret; + } + } + kfree(data); + + return 0; +} + +static const char * const nau8810_companding[] = { + "Off", "NC", "u-law", "A-law" }; + +static const struct soc_enum nau8810_companding_adc_enum = + SOC_ENUM_SINGLE(NAU8810_REG_COMP, NAU8810_ADCCM_SFT, + ARRAY_SIZE(nau8810_companding), nau8810_companding); + +static const struct soc_enum nau8810_companding_dac_enum = + SOC_ENUM_SINGLE(NAU8810_REG_COMP, NAU8810_DACCM_SFT, + ARRAY_SIZE(nau8810_companding), nau8810_companding); + +static const char * const nau8810_deemp[] = { + "None", "32kHz", "44.1kHz", "48kHz" }; + +static const struct soc_enum nau8810_deemp_enum = + SOC_ENUM_SINGLE(NAU8810_REG_DAC, NAU8810_DEEMP_SFT, + ARRAY_SIZE(nau8810_deemp), nau8810_deemp); + +static const char * const nau8810_eqmode[] = {"Capture", "Playback" }; + +static const struct soc_enum nau8810_eqmode_enum = + SOC_ENUM_SINGLE(NAU8810_REG_EQ1, NAU8810_EQM_SFT, + ARRAY_SIZE(nau8810_eqmode), nau8810_eqmode); + +static const char * const nau8810_alc[] = {"Normal", "Limiter" }; + +static const struct soc_enum nau8810_alc_enum = + SOC_ENUM_SINGLE(NAU8810_REG_ALC3, NAU8810_ALCM_SFT, + ARRAY_SIZE(nau8810_alc), nau8810_alc); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); + +static const struct snd_kcontrol_new nau8810_snd_controls[] = { + SOC_ENUM("ADC Companding", nau8810_companding_adc_enum), + SOC_ENUM("DAC Companding", nau8810_companding_dac_enum), + SOC_ENUM("DAC De-emphasis", nau8810_deemp_enum), + + SOC_ENUM("EQ Function", nau8810_eqmode_enum), + SND_SOC_BYTES_EXT("EQ Parameters", 10, + nau8810_eq_get, nau8810_eq_put), + + SOC_SINGLE("DAC Inversion Switch", NAU8810_REG_DAC, + NAU8810_DACPL_SFT, 1, 0), + SOC_SINGLE_TLV("Playback Volume", NAU8810_REG_DACGAIN, + NAU8810_DACGAIN_SFT, 0xff, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", NAU8810_REG_ADC, + NAU8810_HPFEN_SFT, 1, 0), + SOC_SINGLE("High Pass Cut Off", NAU8810_REG_ADC, + NAU8810_HPF_SFT, 0x7, 0), + + SOC_SINGLE("ADC Inversion Switch", NAU8810_REG_ADC, + NAU8810_ADCPL_SFT, 1, 0), + SOC_SINGLE_TLV("Capture Volume", NAU8810_REG_ADCGAIN, + NAU8810_ADCGAIN_SFT, 0xff, 0, digital_tlv), + + SOC_SINGLE_TLV("EQ1 Volume", NAU8810_REG_EQ1, + NAU8810_EQ1GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ2 Volume", NAU8810_REG_EQ2, + NAU8810_EQ2GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ3 Volume", NAU8810_REG_EQ3, + NAU8810_EQ3GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ4 Volume", NAU8810_REG_EQ4, + NAU8810_EQ4GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ5 Volume", NAU8810_REG_EQ5, + NAU8810_EQ5GC_SFT, 0x18, 1, eq_tlv), + + SOC_SINGLE("DAC Limiter Switch", NAU8810_REG_DACLIM1, + NAU8810_DACLIMEN_SFT, 1, 0), + SOC_SINGLE("DAC Limiter Decay", NAU8810_REG_DACLIM1, + NAU8810_DACLIMDCY_SFT, 0xf, 0), + SOC_SINGLE("DAC Limiter Attack", NAU8810_REG_DACLIM1, + NAU8810_DACLIMATK_SFT, 0xf, 0), + SOC_SINGLE("DAC Limiter Threshold", NAU8810_REG_DACLIM2, + NAU8810_DACLIMTHL_SFT, 0x7, 0), + SOC_SINGLE("DAC Limiter Boost", NAU8810_REG_DACLIM2, + NAU8810_DACLIMBST_SFT, 0xf, 0), + + SOC_ENUM("ALC Mode", nau8810_alc_enum), + SOC_SINGLE("ALC Enable Switch", NAU8810_REG_ALC1, + NAU8810_ALCEN_SFT, 1, 0), + SOC_SINGLE("ALC Max Volume", NAU8810_REG_ALC1, + NAU8810_ALCMXGAIN_SFT, 0x7, 0), + SOC_SINGLE("ALC Min Volume", NAU8810_REG_ALC1, + NAU8810_ALCMINGAIN_SFT, 0x7, 0), + SOC_SINGLE("ALC ZC Switch", NAU8810_REG_ALC2, + NAU8810_ALCZC_SFT, 1, 0), + SOC_SINGLE("ALC Hold", NAU8810_REG_ALC2, + NAU8810_ALCHT_SFT, 0xf, 0), + SOC_SINGLE("ALC Target", NAU8810_REG_ALC2, + NAU8810_ALCSL_SFT, 0xf, 0), + SOC_SINGLE("ALC Decay", NAU8810_REG_ALC3, + NAU8810_ALCDCY_SFT, 0xf, 0), + SOC_SINGLE("ALC Attack", NAU8810_REG_ALC3, + NAU8810_ALCATK_SFT, 0xf, 0), + SOC_SINGLE("ALC Noise Gate Switch", NAU8810_REG_NOISEGATE, + NAU8810_ALCNEN_SFT, 1, 0), + SOC_SINGLE("ALC Noise Gate Threshold", NAU8810_REG_NOISEGATE, + NAU8810_ALCNTH_SFT, 0x7, 0), + + SOC_SINGLE("PGA ZC Switch", NAU8810_REG_PGAGAIN, + NAU8810_PGAZC_SFT, 1, 0), + SOC_SINGLE_TLV("PGA Volume", NAU8810_REG_PGAGAIN, + NAU8810_PGAGAIN_SFT, 0x3f, 0, inpga_tlv), + + SOC_SINGLE("Speaker ZC Switch", NAU8810_REG_SPKGAIN, + NAU8810_SPKZC_SFT, 1, 0), + SOC_SINGLE("Speaker Mute Switch", NAU8810_REG_SPKGAIN, + NAU8810_SPKMT_SFT, 1, 0), + SOC_SINGLE_TLV("Speaker Volume", NAU8810_REG_SPKGAIN, + NAU8810_SPKGAIN_SFT, 0x3f, 0, spk_tlv), + + SOC_SINGLE("Capture Boost(+20dB)", NAU8810_REG_ADCBOOST, + NAU8810_PGABST_SFT, 1, 0), + SOC_SINGLE("Mono Mute Switch", NAU8810_REG_MONOMIX, + NAU8810_MOUTMXMT_SFT, 1, 0), + + SOC_SINGLE("DAC Oversampling Rate(128x) Switch", NAU8810_REG_DAC, + NAU8810_DACOS_SFT, 1, 0), + SOC_SINGLE("ADC Oversampling Rate(128x) Switch", NAU8810_REG_ADC, + NAU8810_ADCOS_SFT, 1, 0), +}; + +/* Speaker Output Mixer */ +static const struct snd_kcontrol_new nau8810_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_SPKMIX, + NAU8810_BYPSPK_SFT, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_SPKMIX, + NAU8810_DACSPK_SFT, 1, 0), +}; + +/* Mono Output Mixer */ +static const struct snd_kcontrol_new nau8810_mono_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_MONOMIX, + NAU8810_BYPMOUT_SFT, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_MONOMIX, + NAU8810_DACMOUT_SFT, 1, 0), +}; + +/* PGA Mute */ +static const struct snd_kcontrol_new nau8810_inpga_mute[] = { + SOC_DAPM_SINGLE("PGA Mute Switch", NAU8810_REG_PGAGAIN, + NAU8810_PGAMT_SFT, 1, 0), +}; + +/* Input PGA */ +static const struct snd_kcontrol_new nau8810_inpga[] = { + SOC_DAPM_SINGLE("MicN Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_NMICPGA_SFT, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_PMICPGA_SFT, 1, 0), +}; + +/* Mic Input boost vol */ +static const struct snd_kcontrol_new nau8810_mic_boost_controls = + SOC_DAPM_SINGLE("Mic Volume", NAU8810_REG_ADCBOOST, + NAU8810_PMICBSTGAIN_SFT, 0x7, 0); + +/* Loopback Switch */ +static const struct snd_kcontrol_new nau8810_loopback[] = { + SOC_DAPM_SINGLE("Switch", NAU8810_REG_COMP, + NAU8810_ADDAP_SFT, 1, 0), +}; + +static int check_mclk_select_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + regmap_read(nau8810->regmap, NAU8810_REG_CLOCK, &value); + return (value & NAU8810_CLKM_MASK); +} + +static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Speaker Mixer", NAU8810_REG_POWER3, + NAU8810_SPKMX_EN_SFT, 0, &nau8810_speaker_mixer_controls[0], + ARRAY_SIZE(nau8810_speaker_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", NAU8810_REG_POWER3, + NAU8810_MOUTMX_EN_SFT, 0, &nau8810_mono_mixer_controls[0], + ARRAY_SIZE(nau8810_mono_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", NAU8810_REG_POWER3, + NAU8810_DAC_EN_SFT, 0), + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", NAU8810_REG_POWER2, + NAU8810_ADC_EN_SFT, 0), + SND_SOC_DAPM_PGA("SpkN Out", NAU8810_REG_POWER3, + NAU8810_NSPK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SpkP Out", NAU8810_REG_POWER3, + NAU8810_PSPK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", NAU8810_REG_POWER3, + NAU8810_MOUT_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Input PGA", NAU8810_REG_POWER2, + NAU8810_PGA_EN_SFT, 0, nau8810_inpga, + ARRAY_SIZE(nau8810_inpga)), + SND_SOC_DAPM_MIXER("Input Boost Stage", NAU8810_REG_POWER2, + NAU8810_BST_EN_SFT, 0, nau8810_inpga_mute, + ARRAY_SIZE(nau8810_inpga_mute)), + + SND_SOC_DAPM_SUPPLY("Mic Bias", NAU8810_REG_POWER1, + NAU8810_MICBIAS_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", NAU8810_REG_POWER1, + NAU8810_PLL_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Digital Loopback", SND_SOC_NOPM, 0, 0, + &nau8810_loopback), + + SND_SOC_DAPM_INPUT("MICN"), + SND_SOC_DAPM_INPUT("MICP"), + SND_SOC_DAPM_OUTPUT("MONOOUT"), + SND_SOC_DAPM_OUTPUT("SPKOUTP"), + SND_SOC_DAPM_OUTPUT("SPKOUTN"), +}; + +static const struct snd_soc_dapm_route nau8810_dapm_routes[] = { + {"DAC", NULL, "PLL", check_mclk_select_pll}, + + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Line Bypass Switch", "Input Boost Stage"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Line Bypass Switch", "Input Boost Stage"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Input Boost Stage */ + {"ADC", NULL, "Input Boost Stage"}, + {"ADC", NULL, "PLL", check_mclk_select_pll}, + {"Input Boost Stage", NULL, "Input PGA"}, + {"Input Boost Stage", NULL, "MICP"}, + + /* Input PGA */ + {"Input PGA", NULL, "Mic Bias"}, + {"Input PGA", "MicN Switch", "MICN"}, + {"Input PGA", "MicP Switch", "MICP"}, + + /* Digital Looptack */ + {"Digital Loopback", "Switch", "ADC"}, + {"DAC", NULL, "Digital Loopback"}, +}; + +static int nau8810_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + + nau8810->sysclk = freq; + dev_dbg(nau8810->dev, "master sysclk %dHz\n", nau8810->sysclk); + + return 0; +} + +static int nau8810_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = nau8810->regmap; + struct nau8810_pll *pll = &nau8810->pll; + + switch (nau8810->div_id) { + case NAU8810_MCLK_DIV_PLL: + /* master clock from PLL and enable PLL */ + if (pll->mclk_scaler != div) { + dev_err(nau8810->dev, "Master clock prescaler not meet PLL parameters\n"); + return -EINVAL; + } + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_MCLKSEL_MASK, (div << NAU8810_MCLKSEL_SFT)); + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_PLL); + break; + + case NAU8810_MCLK_DIV_MCLK: + /* Defer the master clock prescaler configuration to DAI + * hardware parameter if master clock from MCLK because + * it needs runtime fs information to get the proper div. + */ + break; + + case NAU8810_BCLK_DIV: + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_BCLKSEL_MASK, (div << NAU8810_BCLKSEL_SFT)); + break; + + default: + return -EINVAL; + } + nau8810->div_id = div_id; + + return 0; +} + +static int nau8810_mclk_clkdiv(struct nau8810 *nau8810, int rate) +{ + struct regmap *regmap = nau8810->regmap; + int i, sclk, imclk = rate * 256, div = 0; + + if (!nau8810->sysclk) { + dev_err(nau8810->dev, "Make mclk div configuration fail because of invalid system clock\n"); + return -EINVAL; + } + + /* Configure the master clock prescaler div to make system + * clock to approximate the internal master clock (IMCLK); + * and large or equal to IMCLK. + */ + for (i = 1; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { + sclk = (nau8810->sysclk * 10) / + nau8810_mclk_scaler[i]; + if (sclk < imclk) + break; + div = i; + } + dev_dbg(nau8810->dev, + "master clock prescaler %x for fs %d\n", div, rate); + + /* master clock from MCLK and disable PLL */ + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_MCLKSEL_MASK, (div << NAU8810_MCLKSEL_SFT)); + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_MCLK); + + return 0; +} + +static int nau88l0_calc_pll(unsigned int pll_in, + unsigned int fs, struct nau8810_pll *pll_param) +{ + u64 f2, f2_max, pll_ratio; + int i, scal_sel; + + if (pll_in > NAU_PLL_REF_MAX || pll_in < NAU_PLL_REF_MIN) + return -EINVAL; + + f2_max = 0; + scal_sel = ARRAY_SIZE(nau8810_mclk_scaler); + for (i = 0; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { + f2 = 256 * fs * 4 * nau8810_mclk_scaler[i] / 10; + if (f2 > NAU_PLL_FREQ_MIN && f2 < NAU_PLL_FREQ_MAX && + f2_max < f2) { + f2_max = f2; + scal_sel = i; + } + } + if (ARRAY_SIZE(nau8810_mclk_scaler) == scal_sel) + return -EINVAL; + pll_param->mclk_scaler = scal_sel; + f2 = f2_max; + + /* Calculate the PLL 4-bit integer input and the PLL 24-bit fractional + * input; round up the 24+4bit. + */ + pll_ratio = div_u64(f2 << 28, pll_in); + pll_param->pre_factor = 0; + if (((pll_ratio >> 28) & 0xF) < NAU_PLL_OPTOP_MIN) { + pll_ratio <<= 1; + pll_param->pre_factor = 1; + } + pll_param->pll_int = (pll_ratio >> 28) & 0xF; + pll_param->pll_frac = ((pll_ratio & 0xFFFFFFF) >> 4); + + return 0; +} + +static int nau8810_set_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct regmap *map = nau8810->regmap; + struct nau8810_pll *pll_param = &nau8810->pll; + int ret, fs; + + fs = freq_out / 256; + ret = nau88l0_calc_pll(freq_in, fs, pll_param); + if (ret < 0) { + dev_err(nau8810->dev, "Unsupported input clock %d\n", freq_in); + return ret; + } + dev_info(nau8810->dev, "pll_int=%x pll_frac=%x mclk_scaler=%x pre_factor=%x\n", + pll_param->pll_int, pll_param->pll_frac, pll_param->mclk_scaler, + pll_param->pre_factor); + + regmap_update_bits(map, NAU8810_REG_PLLN, + NAU8810_PLLMCLK_DIV2 | NAU8810_PLLN_MASK, + (pll_param->pre_factor ? NAU8810_PLLMCLK_DIV2 : 0) | + pll_param->pll_int); + regmap_write(map, NAU8810_REG_PLLK1, + (pll_param->pll_frac >> NAU8810_PLLK1_SFT) & + NAU8810_PLLK1_MASK); + regmap_write(map, NAU8810_REG_PLLK2, + (pll_param->pll_frac >> NAU8810_PLLK2_SFT) & + NAU8810_PLLK2_MASK); + regmap_write(map, NAU8810_REG_PLLK3, + pll_param->pll_frac & NAU8810_PLLK3_MASK); + regmap_update_bits(map, NAU8810_REG_CLOCK, NAU8810_MCLKSEL_MASK, + pll_param->mclk_scaler << NAU8810_MCLKSEL_SFT); + regmap_update_bits(map, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_PLL); + + return 0; +} + +static int nau8810_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + u16 ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8810_CLKIO_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8810_AIFMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8810_AIFMT_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8810_AIFMT_PCM_A; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1_val |= NAU8810_BCLKP_IB | NAU8810_FSP_IF; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8810_BCLKP_IB; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1_val |= NAU8810_FSP_IF; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8810->regmap, NAU8810_REG_IFACE, + NAU8810_AIFMT_MASK | NAU8810_FSP_IF | + NAU8810_BCLKP_IB, ctrl1_val); + regmap_update_bits(nau8810->regmap, NAU8810_REG_CLOCK, + NAU8810_CLKIO_MASK, ctrl2_val); + + return 0; +} + +static int nau8810_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + int val_len = 0, val_rate = 0, ret = 0; + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= NAU8810_WLEN_20; + break; + case 24: + val_len |= NAU8810_WLEN_24; + break; + case 32: + val_len |= NAU8810_WLEN_32; + break; + } + + switch (params_rate(params)) { + case 8000: + val_rate |= NAU8810_SMPLR_8K; + break; + case 11025: + val_rate |= NAU8810_SMPLR_12K; + break; + case 16000: + val_rate |= NAU8810_SMPLR_16K; + break; + case 22050: + val_rate |= NAU8810_SMPLR_24K; + break; + case 32000: + val_rate |= NAU8810_SMPLR_32K; + break; + case 44100: + case 48000: + break; + } + + regmap_update_bits(nau8810->regmap, NAU8810_REG_IFACE, + NAU8810_WLEN_MASK, val_len); + regmap_update_bits(nau8810->regmap, NAU8810_REG_SMPLR, + NAU8810_SMPLR_MASK, val_rate); + + /* If the master clock is from MCLK, provide the runtime FS for driver + * to get the master clock prescaler configuration. + */ + if (nau8810->div_id == NAU8810_MCLK_DIV_MCLK) { + ret = nau8810_mclk_clkdiv(nau8810, params_rate(params)); + if (ret < 0) + dev_err(nau8810->dev, "MCLK div configuration fail\n"); + } + + return ret; +} + +static int nau8810_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct regmap *map = nau8810->regmap; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_80K); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_IOBUF_EN | NAU8810_ABIAS_EN, + NAU8810_IOBUF_EN | NAU8810_ABIAS_EN); + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + regcache_sync(map); + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_3K); + mdelay(100); + } + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_300K); + break; + + case SND_SOC_BIAS_OFF: + regmap_write(map, NAU8810_REG_POWER1, 0); + regmap_write(map, NAU8810_REG_POWER2, 0); + regmap_write(map, NAU8810_REG_POWER3, 0); + break; + } + + return 0; +} + + +#define NAU8810_RATES (SNDRV_PCM_RATE_8000_48000) + +#define NAU8810_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8810_ops = { + .hw_params = nau8810_pcm_hw_params, + .set_fmt = nau8810_set_dai_fmt, + .set_sysclk = nau8810_set_sysclk, + .set_clkdiv = nau8810_set_clkdiv, + .set_pll = nau8810_set_pll, +}; + +static struct snd_soc_dai_driver nau8810_dai = { + .name = "nau8810-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = NAU8810_RATES, + .formats = NAU8810_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = NAU8810_RATES, + .formats = NAU8810_FORMATS, + }, + .ops = &nau8810_ops, + .symmetric_rates = 1, +}; + +static const struct regmap_config nau8810_regmap_config = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = NAU8810_REG_MAX, + .readable_reg = nau8810_readable_reg, + .writeable_reg = nau8810_writeable_reg, + .volatile_reg = nau8810_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8810_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8810_reg_defaults), +}; + +static int nau8810_probe(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_nau8810 = { + .probe = nau8810_probe, + .set_bias_level = nau8810_set_bias_level, + .suspend_bias_off = true, + + .controls = nau8810_snd_controls, + .num_controls = ARRAY_SIZE(nau8810_snd_controls), + .dapm_widgets = nau8810_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8810_dapm_widgets), + .dapm_routes = nau8810_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8810_dapm_routes), +}; + +static int nau8810_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8810 *nau8810 = dev_get_platdata(dev); + + if (!nau8810) { + nau8810 = devm_kzalloc(dev, sizeof(*nau8810), GFP_KERNEL); + if (!nau8810) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8810); + + nau8810->regmap = devm_regmap_init_i2c(i2c, &nau8810_regmap_config); + if (IS_ERR(nau8810->regmap)) + return PTR_ERR(nau8810->regmap); + nau8810->dev = dev; + + regmap_write(nau8810->regmap, NAU8810_REG_RESET, 0x00); + + return snd_soc_register_codec(dev, + &soc_codec_dev_nau8810, &nau8810_dai, 1); +} + +static int nau8810_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id nau8810_i2c_id[] = { + { "nau8810", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8810_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id nau8810_of_match[] = { + { .compatible = "nuvoton,nau8810", }, + { } +}; +MODULE_DEVICE_TABLE(of, nau8810_of_match); +#endif + +static struct i2c_driver nau8810_i2c_driver = { + .driver = { + .name = "nau8810", + .of_match_table = of_match_ptr(nau8810_of_match), + }, + .probe = nau8810_i2c_probe, + .remove = nau8810_i2c_remove, + .id_table = nau8810_i2c_id, +}; + +module_i2c_driver(nau8810_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8810 driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8810.h b/sound/soc/codecs/nau8810.h new file mode 100644 index 0000000..14b8ae3 --- /dev/null +++ b/sound/soc/codecs/nau8810.h @@ -0,0 +1,288 @@ +/* + * NAU8810 ALSA SoC audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * Author: David Lin <ctlin0(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8810_H__ +#define __NAU8810_H__ + +#define NAU8810_REG_RESET 0x00 +#define NAU8810_REG_POWER1 0x01 +#define NAU8810_REG_POWER2 0x02 +#define NAU8810_REG_POWER3 0x03 +#define NAU8810_REG_IFACE 0x04 +#define NAU8810_REG_COMP 0x05 +#define NAU8810_REG_CLOCK 0x06 +#define NAU8810_REG_SMPLR 0x07 +#define NAU8810_REG_DAC 0x0A +#define NAU8810_REG_DACGAIN 0x0B +#define NAU8810_REG_ADC 0x0E +#define NAU8810_REG_ADCGAIN 0x0F +#define NAU8810_REG_EQ1 0x12 +#define NAU8810_REG_EQ2 0x13 +#define NAU8810_REG_EQ3 0x14 +#define NAU8810_REG_EQ4 0x15 +#define NAU8810_REG_EQ5 0x16 +#define NAU8810_REG_DACLIM1 0x18 +#define NAU8810_REG_DACLIM2 0x19 +#define NAU8810_REG_NOTCH1 0x1B +#define NAU8810_REG_NOTCH2 0x1C +#define NAU8810_REG_NOTCH3 0x1D +#define NAU8810_REG_NOTCH4 0x1E +#define NAU8810_REG_ALC1 0x20 +#define NAU8810_REG_ALC2 0x21 +#define NAU8810_REG_ALC3 0x22 +#define NAU8810_REG_NOISEGATE 0x23 +#define NAU8810_REG_PLLN 0x24 +#define NAU8810_REG_PLLK1 0x25 +#define NAU8810_REG_PLLK2 0x26 +#define NAU8810_REG_PLLK3 0x27 +#define NAU8810_REG_ATTEN 0x28 +#define NAU8810_REG_INPUT_SIGNAL 0x2C +#define NAU8810_REG_PGAGAIN 0x2D +#define NAU8810_REG_ADCBOOST 0x2F +#define NAU8810_REG_OUTPUT 0x31 +#define NAU8810_REG_SPKMIX 0x32 +#define NAU8810_REG_SPKGAIN 0x36 +#define NAU8810_REG_MONOMIX 0x38 +#define NAU8810_REG_POWER4 0x3A +#define NAU8810_REG_TSLOTCTL1 0x3B +#define NAU8810_REG_TSLOTCTL2 0x3C +#define NAU8810_REG_DEVICE_REVID 0x3E +#define NAU8810_REG_I2C_DEVICEID 0x3F +#define NAU8810_REG_ADDITIONID 0x40 +#define NAU8810_REG_RESERVE 0x41 +#define NAU8810_REG_OUTCTL 0x45 +#define NAU8810_REG_ALC1ENHAN1 0x46 +#define NAU8810_REG_ALC1ENHAN2 0x47 +#define NAU8810_REG_MISCCTL 0x49 +#define NAU8810_REG_OUTTIEOFF 0x4B +#define NAU8810_REG_AGCP2POUT 0x4C +#define NAU8810_REG_AGCPOUT 0x4D +#define NAU8810_REG_AMTCTL 0x4E +#define NAU8810_REG_OUTTIEOFFMAN 0x4F +#define NAU8810_REG_MAX NAU8810_REG_OUTTIEOFFMAN + + +/* NAU8810_REG_POWER1 (0x1) */ +#define NAU8810_DCBUF_EN (0x1 << 8) +#define NAU8810_PLL_EN_SFT 5 +#define NAU8810_MICBIAS_EN_SFT 4 +#define NAU8810_ABIAS_EN (0x1 << 3) +#define NAU8810_IOBUF_EN (0x1 << 2) +#define NAU8810_REFIMP_MASK 0x3 +#define NAU8810_REFIMP_DIS 0x0 +#define NAU8810_REFIMP_80K 0x1 +#define NAU8810_REFIMP_300K 0x2 +#define NAU8810_REFIMP_3K 0x3 + +/* NAU8810_REG_POWER2 (0x2) */ +#define NAU8810_BST_EN_SFT 4 +#define NAU8810_PGA_EN_SFT 2 +#define NAU8810_ADC_EN_SFT 0 + +/* NAU8810_REG_POWER3 (0x3) */ +#define NAU8810_DAC_EN_SFT 0 +#define NAU8810_SPKMX_EN_SFT 2 +#define NAU8810_MOUTMX_EN_SFT 3 +#define NAU8810_PSPK_EN_SFT 5 +#define NAU8810_NSPK_EN_SFT 6 +#define NAU8810_MOUT_EN_SFT 7 + +/* NAU8810_REG_IFACE (0x4) */ +#define NAU8810_AIFMT_SFT 3 +#define NAU8810_AIFMT_MASK (0x3 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_RIGHT (0x0 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_LEFT (0x1 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_I2S (0x2 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_PCM_A (0x3 << NAU8810_AIFMT_SFT) +#define NAU8810_WLEN_SFT 5 +#define NAU8810_WLEN_MASK (0x3 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_16 (0x0 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_20 (0x1 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_24 (0x2 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_32 (0x3 << NAU8810_WLEN_SFT) +#define NAU8810_FSP_IF (0x1 << 7) +#define NAU8810_BCLKP_IB (0x1 << 8) + +/* NAU8810_REG_COMP (0x5) */ +#define NAU8810_ADDAP_SFT 0 +#define NAU8810_ADCCM_SFT 1 +#define NAU8810_DACCM_SFT 3 + +/* NAU8810_REG_CLOCK (0x6) */ +#define NAU8810_CLKIO_MASK 0x1 +#define NAU8810_CLKIO_SLAVE 0x0 +#define NAU8810_CLKIO_MASTER 0x1 +#define NAU8810_BCLKSEL_SFT 2 +#define NAU8810_BCLKSEL_MASK (0x7 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_1 (0x0 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_2 (0x1 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_4 (0x2 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_8 (0x3 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_16 (0x4 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_32 (0x5 << NAU8810_BCLKSEL_SFT) +#define NAU8810_MCLKSEL_SFT 5 +#define NAU8810_MCLKSEL_MASK (0x7 << NAU8810_MCLKSEL_SFT) +#define NAU8810_CLKM_SFT 8 +#define NAU8810_CLKM_MASK (0x1 << NAU8810_CLKM_SFT) +#define NAU8810_CLKM_MCLK (0x0 << NAU8810_CLKM_SFT) +#define NAU8810_CLKM_PLL (0x1 << NAU8810_CLKM_SFT) + +/* NAU8810_REG_SMPLR (0x7) */ +#define NAU8810_SMPLR_SFT 1 +#define NAU8810_SMPLR_MASK (0x7 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_48K (0x0 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_32K (0x1 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_24K (0x2 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_16K (0x3 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_12K (0x4 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_8K (0x5 << NAU8810_SMPLR_SFT) + +/* NAU8810_REG_DAC (0xA) */ +#define NAU8810_DACPL_SFT 0 +#define NAU8810_DACOS_SFT 3 +#define NAU8810_DEEMP_SFT 4 + +/* NAU8810_REG_DACGAIN (0xB) */ +#define NAU8810_DACGAIN_SFT 0 + +/* NAU8810_REG_ADC (0xE) */ +#define NAU8810_ADCPL_SFT 0 +#define NAU8810_ADCOS_SFT 3 +#define NAU8810_HPF_SFT 4 +#define NAU8810_HPFEN_SFT 8 + +/* NAU8810_REG_ADCGAIN (0xF) */ +#define NAU8810_ADCGAIN_SFT 0 + +/* NAU8810_REG_EQ1 (0x12) */ +#define NAU8810_EQ1GC_SFT 0 +#define NAU8810_EQ1CF_SFT 5 +#define NAU8810_EQM_SFT 8 + +/* NAU8810_REG_EQ2 (0x13) */ +#define NAU8810_EQ2GC_SFT 0 +#define NAU8810_EQ2CF_SFT 5 +#define NAU8810_EQ2BW_SFT 8 + +/* NAU8810_REG_EQ3 (0x14) */ +#define NAU8810_EQ3GC_SFT 0 +#define NAU8810_EQ3CF_SFT 5 +#define NAU8810_EQ3BW_SFT 8 + +/* NAU8810_REG_EQ4 (0x15) */ +#define NAU8810_EQ4GC_SFT 0 +#define NAU8810_EQ4CF_SFT 5 +#define NAU8810_EQ4BW_SFT 8 + +/* NAU8810_REG_EQ5 (0x16) */ +#define NAU8810_EQ5GC_SFT 0 +#define NAU8810_EQ5CF_SFT 5 + +/* NAU8810_REG_DACLIM1 (0x18) */ +#define NAU8810_DACLIMATK_SFT 0 +#define NAU8810_DACLIMDCY_SFT 4 +#define NAU8810_DACLIMEN_SFT 8 + +/* NAU8810_REG_DACLIM2 (0x19) */ +#define NAU8810_DACLIMBST_SFT 0 +#define NAU8810_DACLIMTHL_SFT 4 + +/* NAU8810_REG_ALC1 (0x20) */ +#define NAU8810_ALCMINGAIN_SFT 0 +#define NAU8810_ALCMXGAIN_SFT 3 +#define NAU8810_ALCEN_SFT 8 + +/* NAU8810_REG_ALC2 (0x21) */ +#define NAU8810_ALCSL_SFT 0 +#define NAU8810_ALCHT_SFT 4 +#define NAU8810_ALCZC_SFT 8 + +/* NAU8810_REG_ALC3 (0x22) */ +#define NAU8810_ALCATK_SFT 0 +#define NAU8810_ALCDCY_SFT 4 +#define NAU8810_ALCM_SFT 8 + +/* NAU8810_REG_NOISEGATE (0x23) */ +#define NAU8810_ALCNTH_SFT 0 +#define NAU8810_ALCNEN_SFT 3 + +/* NAU8810_REG_PLLN (0x24) */ +#define NAU8810_PLLN_MASK 0xF +#define NAU8810_PLLMCLK_DIV2 (0x1 << 4) + +/* NAU8810_REG_PLLK1 (0x25) */ +#define NAU8810_PLLK1_SFT 18 +#define NAU8810_PLLK1_MASK 0x3F + +/* NAU8810_REG_PLLK2 (0x26) */ +#define NAU8810_PLLK2_SFT 9 +#define NAU8810_PLLK2_MASK 0x1FF + +/* NAU8810_REG_PLLK3 (0x27) */ +#define NAU8810_PLLK3_MASK 0x1FF + +/* NAU8810_REG_INPUT_SIGNAL (0x2C) */ +#define NAU8810_PMICPGA_SFT 0 +#define NAU8810_NMICPGA_SFT 1 + +/* NAU8810_REG_PGAGAIN (0x2D) */ +#define NAU8810_PGAGAIN_SFT 0 +#define NAU8810_PGAMT_SFT 6 +#define NAU8810_PGAZC_SFT 7 + +/* NAU8810_REG_ADCBOOST (0x2F) */ +#define NAU8810_PMICBSTGAIN_SFT 4 +#define NAU8810_PGABST_SFT 8 + +/* NAU8810_REG_SPKMIX (0x32) */ +#define NAU8810_DACSPK_SFT 0 +#define NAU8810_BYPSPK_SFT 1 + +/* NAU8810_REG_SPKGAIN (0x36) */ +#define NAU8810_SPKGAIN_SFT 0 +#define NAU8810_SPKMT_SFT 6 +#define NAU8810_SPKZC_SFT 7 + +/* NAU8810_REG_MONOMIX (0x38) */ +#define NAU8810_DACMOUT_SFT 0 +#define NAU8810_BYPMOUT_SFT 1 +#define NAU8810_MOUTMXMT_SFT 6 + + +/* System Clock Source */ +enum { + NAU8810_CLK_MCLK, + NAU8810_CLK_PLL, +}; + +/* Clock divider Id's */ +enum { + NAU8810_MCLK_DIV_PLL, + NAU8810_MCLK_DIV_MCLK, + NAU8810_BCLK_DIV, +}; + +struct nau8810_pll { + int pre_factor; + int mclk_scaler; + int pll_frac; + int pll_int; +}; + +struct nau8810 { + struct device *dev; + struct regmap *regmap; + struct nau8810_pll pll; + int sysclk; + int div_id; +}; + +#endif -- 2.6.4

2 8

[alsa-devel] [PATCH 00/17] Baytrail audio fixes
by Pierre-Louis Bossart 18 Aug '16

18 Aug '16

This second set of patches solves problems with Baytrail-CR, specifically the auto-detection of IRQ index and SSP0 routing. These patches enable sound on the Asus T100TAF with the support of the MCLK handled by the clock framework (Credits to Irina Tirdea) If you prefer git access, the patches are available at https://github.com/plbossart/sound/commits/t100taf-6 The corresponding UCM file is at https://github.com/plbossart/UCM/commits/t100-test Thanks to Luka Karinja and Antonio Ospite for early tests and suggestions, as well as Liam Girdwood and Vinod Koul for internal code reviews. Additional fixes for UCM files will be coming soon to support mono speakers, IN3 input and other fixes. Irina Tirdea (3): clk: x86: Add Atom PMC platform clocks ASoC: Intel: bytcr_rt5640: add MCLK support AsoC: Intel: Add quirks for MinnowBoard MAX Pierre-Louis Bossart (14): ASoC: Intel: bytcr-rt5640: add Asus T100TAF quirks ASoC: Intel: bytcr_rt5640: quirk for mono speaker ASoC: Intel: bytcr_rt5640: enable differential mic quirk ASoC: Intel: Atom: auto-detection of Baytrail-CR ASoC: Intel: Atom: add definitions for modem/SSP0 interface ASoC: Intel: atom: enable configuration of SSP0 ASoC: Intel: bytcr_rt5640: add SSP2_AIF2 routing ASoC: Intel: bytcr_rt56040: additional routing quirks ASoC: Intel: bytcr_rt5640: fix dai/clock setup for SSP0 routing ASoC: Intel: bytcr_rt5640: default routing and quirks on Baytrail-CR ASoC: Intel: bytcr_rt5640: add IN3 map ASoC: rt5640: add internal clock source support ASoC: bytcr_rt5640: Add quirk for Teclast X98 Air 3G tablet ASoC: Intel: bytcr_rt5640: log quirks arch/x86/Kconfig | 1 + arch/x86/include/asm/pmc_atom.h | 6 + arch/x86/platform/atom/pmc_atom.c | 55 +++- drivers/clk/x86/Makefile | 1 + drivers/clk/x86/clk-byt-plt.c | 413 +++++++++++++++++++++++++ sound/soc/codecs/rt5640.c | 3 + sound/soc/codecs/rt5640.h | 1 + sound/soc/intel/atom/sst-atom-controls.c | 32 +- sound/soc/intel/atom/sst-atom-controls.h | 6 + sound/soc/intel/atom/sst/sst_acpi.c | 82 ++++- sound/soc/intel/boards/bytcr_rt5640.c | 496 ++++++++++++++++++++++++++++--- 11 files changed, 1036 insertions(+), 60 deletions(-) create mode 100644 drivers/clk/x86/clk-byt-plt.c -- 2.5.0

2 39

[alsa-devel] [PATCH for 4.8] Revert "ASoC: omap-mcpdm: Add support for pdmclk clock handling"
by Peter Ujfalusi 18 Aug '16

18 Aug '16

This reverts commit 65aca64d05b5eaa5ce15e18b458a8d338ddbd478. The patches for twl6040 MFD and clk missed the merge window and causing the McPDM driver to never probe since it is put back to the deferred list because the missing drivers. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Mark, OMAP4/5 audio regressed in 4.8 due to the hack that the twl6040 MFD and clk driver changes were not take in time. The only clean solution I can think is to revert the pdmclk clock handling from the McPDM driver for now. Regards, Peter Documentation/devicetree/bindings/sound/omap-mcpdm.txt | 10 ---------- sound/soc/omap/omap-mcpdm.c | 17 ----------------- 2 files changed, 27 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt index 6f6c2f8e908d..0741dff048dd 100644 --- a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt +++ b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt @@ -8,8 +8,6 @@ Required properties: - interrupts: Interrupt number for McPDM - interrupt-parent: The parent interrupt controller - ti,hwmods: Name of the hwmod associated to the McPDM -- clocks: phandle for the pdmclk provider, likely <&twl6040> -- clock-names: Must be "pdmclk" Example: @@ -21,11 +19,3 @@ mcpdm: mcpdm@40132000 { interrupt-parent = <&gic>; ti,hwmods = "mcpdm"; }; - -In board DTS file the pdmclk needs to be added: - -&mcpdm { - clocks = <&twl6040>; - clock-names = "pdmclk"; - status = "okay"; -}; diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index e7cdc51fd806..74d6e6fdcfd0 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -31,7 +31,6 @@ #include <linux/err.h> #include <linux/io.h> #include <linux/irq.h> -#include <linux/clk.h> #include <linux/slab.h> #include <linux/pm_runtime.h> #include <linux/of_device.h> @@ -55,7 +54,6 @@ struct omap_mcpdm { unsigned long phys_base; void __iomem *io_base; int irq; - struct clk *pdmclk; struct mutex mutex; @@ -390,7 +388,6 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); int ret; - clk_prepare_enable(mcpdm->pdmclk); pm_runtime_enable(mcpdm->dev); /* Disable lines while request is ongoing */ @@ -425,7 +422,6 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) pm_runtime_disable(mcpdm->dev); - clk_disable_unprepare(mcpdm->pdmclk); return 0; } @@ -445,8 +441,6 @@ static int omap_mcpdm_suspend(struct snd_soc_dai *dai) mcpdm->pm_active_count++; } - clk_disable_unprepare(mcpdm->pdmclk); - return 0; } @@ -454,8 +448,6 @@ static int omap_mcpdm_resume(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); - clk_prepare_enable(mcpdm->pdmclk); - if (mcpdm->pm_active_count) { while (mcpdm->pm_active_count--) pm_runtime_get_sync(mcpdm->dev); @@ -549,15 +541,6 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dev = &pdev->dev; - mcpdm->pdmclk = devm_clk_get(&pdev->dev, "pdmclk"); - if (IS_ERR(mcpdm->pdmclk)) { - if (PTR_ERR(mcpdm->pdmclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; - dev_warn(&pdev->dev, "Error getting pdmclk (%ld)!\n", - PTR_ERR(mcpdm->pdmclk)); - mcpdm->pdmclk = NULL; - } - ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcpdm_component, &omap_mcpdm_dai, 1); -- 2.9.2

2 1

[alsa-devel] [PATCH] ASoC: Intel: sst: fix ix spelling mistake: "capablities" -> "capabilites"
by Colin King 18 Aug '16

18 Aug '16

From: Colin Ian King <colin.king(a)canonical.com> trivial fix to spelling mistake in dev_err message and reformat code to avoid being over 80 chars wide per line Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/atom/sst/sst.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index a4b458e..498b5f7 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -190,7 +190,8 @@ int sst_driver_ops(struct intel_sst_drv *sst) default: dev_err(sst->dev, - "SST Driver capablities missing for dev_id: %x", sst->dev_id); + "SST Driver capabilities missing for dev_id: %x", + sst->dev_id); return -EINVAL; }; } -- 2.8.1

2 1

[alsa-devel] [PATCH 0/3] Baytrail followup patches
by Pierre-Louis Bossart 17 Aug '16

17 Aug '16

fixes for warnings detected by 0-Day New quirk for Acer Aspire platforms Resending clock framework support (for-next audio not working on all platforms enabling MCLK if this patch is not applied) Irina Tirdea (1): clk: x86: Add Atom PMC platform clocks Pierre-Louis Bossart (2): ASoC: Intel: atom: fix 0-day warnings ASoC: Intel: bytcr_rt5640: quirk for Acer Aspire SWS-012 arch/x86/Kconfig | 1 + arch/x86/include/asm/pmc_atom.h | 6 + arch/x86/platform/atom/pmc_atom.c | 55 ++++- drivers/clk/x86/Makefile | 1 + drivers/clk/x86/clk-byt-plt.c | 413 ++++++++++++++++++++++++++++++++++ sound/soc/intel/atom/sst/sst_acpi.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 15 +- 7 files changed, 487 insertions(+), 6 deletions(-) create mode 100644 drivers/clk/x86/clk-byt-plt.c -- 2.7.4

2 6

[alsa-devel] [PATCH] ASoC: fsl_sai: Use 'np' variable
by Fabio Estevam 17 Aug '16

17 Aug '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> The 'np' variable is already assigned to 'pdev->dev.of_node', so use it to improve readability. No functional change. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 2147994..9fadf7e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -801,8 +801,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") || - of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) + if (of_device_is_compatible(np, "fsl,imx6sx-sai") || + of_device_is_compatible(np, "fsl,imx6ul-sai")) sai->sai_on_imx = true; sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -883,7 +883,7 @@ static int fsl_sai_probe(struct platform_device *pdev) } if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) && - of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) { + of_device_is_compatible(np, "fsl,imx6ul-sai")) { gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr"); if (IS_ERR(gpr)) { dev_err(&pdev->dev, "cannot find iomuxc registers\n"); -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: dapm: Fix typo in comment
by Peter Meerwald-Stadler 16 Aug '16

16 Aug '16

Signed-off-by: Peter Meerwald-Stadler <pmeerw(a)pmeerw.net> --- sound/soc/soc-dapm.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8698c26..c46b1ff 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1169,7 +1169,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, * @custom_stop_condition: (optional) a function meant to stop the widget graph * walk based on custom logic. * - * Queries DAPM graph as to whether an valid audio stream path exists for + * Queries DAPM graph as to whether a valid audio stream path exists for * the initial stream specified by name. This takes into account * current mixer and mux kcontrol settings. Creates list of valid widgets. * @@ -1294,8 +1294,7 @@ static int dapm_widget_power_check(struct snd_soc_dapm_widget *w) return w->new_power; } -/* Generic check to see if a widget should be powered. - */ +/* Generic check to see if a widget should be powered. */ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) { int in, out; @@ -1646,7 +1645,7 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) struct snd_soc_dapm_context *d = data; int ret; - /* If we're off and we're not supposed to be go into STANDBY */ + /* If we're off and we're not supposed to go into STANDBY */ if (d->bias_level == SND_SOC_BIAS_OFF && d->target_bias_level != SND_SOC_BIAS_OFF) { if (d->dev) @@ -1798,7 +1797,7 @@ static bool dapm_idle_bias_off(struct snd_soc_dapm_context *dapm) * A complete path is a route that has valid endpoints i.e.:- * * o DAC to output pin. - * o Input Pin to ADC. + * o Input pin to ADC. * o Input pin to Output pin (bypass, sidetone) * o DAC to ADC (loopback). */ @@ -2114,7 +2113,7 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) * soc_dapm_connect_path() - Connects or disconnects a path * @path: The path to update * @connect: The new connect state of the path. True if the path is connected, - * false if it is disconneted. + * false if it is disconnected. * @reason: The reason why the path changed (for debugging only) */ static void soc_dapm_connect_path(struct snd_soc_dapm_path *path, @@ -2233,7 +2232,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, if (w->dapm != dapm) continue; - /* only display widgets that burnm power */ + /* only display widgets that burn power */ switch (w->id) { case snd_soc_dapm_hp: case snd_soc_dapm_mic: @@ -2461,7 +2460,7 @@ static void dapm_update_widget_flags(struct snd_soc_dapm_widget *w) switch (w->id) { case snd_soc_dapm_input: - /* On a fully routed card a input is never a source */ + /* On a fully routed card an input is never a source */ if (w->dapm->card->fully_routed) return; ep = SND_SOC_DAPM_EP_SOURCE; @@ -3445,7 +3444,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->endpoints[dir] = -1; } - /* machine layer set ups unconnected pins and insertions */ + /* machine layer sets up unconnected pins and insertions */ w->connected = 1; return w; } -- 1.7.10.4

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Re: [alsa-devel] linux-next: Tree for Aug 16 (mellanox/mlx5 & sound/oss/)
by Randy Dunlap 16 Aug '16

16 Aug '16

On 08/15/16 21:01, Stephen Rothwell wrote: > Hi all, > > Changes since 20160815: > on i386: sound/built-in.o:(.data+0x5c): multiple definition of `dev_list' drivers/built-in.o:(.data+0x6aee8): first defined here ld: Warning: size of symbol `dev_list' changed from 8 in drivers/built-in.o to 16 in sound/built-in.o sound/ is an old oss driver (soundcard.c): struct oss_minor_dev { unsigned short minor; unsigned int enabled; } dev_list[] = { { SND_DEV_DSP16 }, { SND_DEV_AUDIO }, }; ===== drivers/ is net/ethernet/mellanox/mlx5/core/mlx5_core.h: extern struct list_head dev_list; -- ~Randy

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[alsa-devel] [PATCH] ASoC: compress: Fix leak of a widget list in soc_compr_open_fe
by Charles Keepax 16 Aug '16

16 Aug '16

After we have called dpcm_path_get we should make sure to call dpcm_path_put on all error paths. This was not happening causing the allocated widget list to be leaked, this patch corrects this by adding a dpcm_path_put to the error path. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index d2df46c..bf7b52f 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -121,7 +121,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) dpcm_be_disconnect(fe, stream); fe->dpcm[stream].runtime = NULL; - goto fe_err; + goto path_err; } dpcm_clear_pending_state(fe, stream); @@ -136,6 +136,8 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) return 0; +path_err: + dpcm_path_put(&list); fe_err: if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown) fe->dai_link->compr_ops->shutdown(cstream); -- 2.1.4

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[alsa-devel] [PATCH v2] ASoC: atmel_ssc_dai: Don't unconditionally reset SSC on stream startup
by Peter Meerwald-Stadler 16 Aug '16

16 Aug '16

From: Christoph Huber <c.huber(a)bct-electronic.com> commit cbaadf0f90d6 ("ASoC: atmel_ssc_dai: refactor the startup and shutdown") refactored code such that the SSC is reset on every startup; this breaks duplex audio (e.g. first start audio playback, then start record, causing the playback to stop/hang) Signed-off-by: Christoph Huber <c.huber(a)bct-electronic.com> Signed-off-by: Peter Meerwald-Stadler <p.meerwald(a)bct-electronic.com> Cc: Songjun Wu <songjun.wu(a)microchip.com> Cc: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- v2 fixes formatting issues and rephrases the comment sound/soc/atmel/atmel_ssc_dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 54c09ac..16e459a 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -299,8 +299,9 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, clk_enable(ssc_p->ssc->clk); ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); - /* Reset the SSC to keep it at a clean status */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Reset the SSC unless initialized to keep it in a clean state */ + if (!ssc_p->initialized) + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dir = 0; -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: max98371: Delete unnecessary assignment for the field "owner"
by SF Markus Elfring 16 Aug '16

16 Aug '16

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 15 Aug 2016 16:16:00 +0200 The field "owner" is set by the core. Thus delete an unneeded initialisation. Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/codecs/max98371.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index cf0a39b..3053879 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -425,7 +425,6 @@ MODULE_DEVICE_TABLE(of, max98371_of_match); static struct i2c_driver max98371_i2c_driver = { .driver = { .name = "max98371", - .owner = THIS_MODULE, .pm = NULL, .of_match_table = of_match_ptr(max98371_of_match), }, -- 2.9.2

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[alsa-devel] [PATCH 1149/1285] Replace numeric parameter like 0444 with macro
by Baole Ni 16 Aug '16

16 Aug '16

I find that the developers often just specified the numeric value when calling a macro which is defined with a parameter for access permission. As we know, these numeric value for access permission have had the corresponding macro, and that using macro can improve the robustness and readability of the code, thus, I suggest replacing the numeric parameter with the macro. Signed-off-by: Chuansheng Liu <chuansheng.liu(a)intel.com> Signed-off-by: Baole Ni <baolex.ni(a)intel.com> --- sound/core/oss/pcm_oss.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index ebc9fdf..0ff0dd8 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -53,11 +53,11 @@ static bool nonblock_open = 1; MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>, Abramo Bagnara <abramo(a)alsa-project.org>"); MODULE_DESCRIPTION("PCM OSS emulation for ALSA."); MODULE_LICENSE("GPL"); -module_param_array(dsp_map, int, NULL, 0444); +module_param_array(dsp_map, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(dsp_map, "PCM device number assigned to 1st OSS device."); -module_param_array(adsp_map, int, NULL, 0444); +module_param_array(adsp_map, int, NULL, S_IRUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(adsp_map, "PCM device number assigned to 2nd OSS device."); -module_param(nonblock_open, bool, 0644); +module_param(nonblock_open, bool, S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH); MODULE_PARM_DESC(nonblock_open, "Don't block opening busy PCM devices."); MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_PCM); MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_PCM1); -- 2.9.2

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[alsa-devel] [PATCH v1 00/14] clk/mfd/ASoC/ARM: OMAP4/5: McPDM/twl6040 pdmclk support
by Peter Ujfalusi 15 Aug '16

15 Aug '16

Hi, Changes since v1: - Added Rob's Acked-by to patch 5 and 14 - Check the return value of clk_prepare_enable() calls in the last patch this is something I wanted to do for a long time... First of all: sorry for the cross domain series. I did tested alone all the domain patches and they are not causing any regression. When they come together we will have ability to control the pdmclk needed by McPDM. CLK patches: The driver was w/o DT support and needed some cleanup so we can use the driver. I have also renamed it to twl6040-pdmclk from twl6040-clk to be precise. MFD patches: The regmap_sync() was not working since the twl6040 does not support bulk access and now we are going to create the needed child for the pdmclk. ARM patches: Updatding the board DTS files and add the needed lines for the pdmclk. ASoC patches: The machine driver needs to select the twl6040-clk driver as it will be used by the omap-mcpdm. I needed to do some cleanup and add also suspend/resume support before adding the code for pdmclk handling since I don't want to keep the twl6040 powered on when the board is suspended. At the moment it is not possible to do true dynamic twl6040 power up/down due to pop noises, but I will be looking at that later. The driver in the future will fail if it is not able to get the pdmclk to avoid kernel crash (McPDM registers are not accessible when pdmclk is not available). I have tested the patches by domain and also in all (I hope) permutation they could be. I have not experienced any regression. Regards, Peter --- Peter Ujfalusi (14): clk: twl6040: Correct clk_ops clk: twl6040: Register the clock as of_clk_provider clk: twl6040: Rename the driver and use consistent names in the code mfd: twl6040: The chip does not support bulk access mfd: twl6040: Register child device for twl6040-pdmclk ARM: dts: omap5-board-common: Add pdmclk binding for audio ARM: dts: omap4-panda-common: Add pdmclk binding for audio ARM: dts: omap4-sdp: Add pdmclk binding for audio ARM: dts: omap4-var-som-om44: Add pdmclk binding for audio ARM: dts: omap4-duovero: Add pdmclk binding for audio ASoC: omap: Kconfig: SND_OMAP_SOC_OMAP_ABE_TWL6040 to select CLK_TWL6040 ASoC: omap-mcpdm: Move the WD enable write inside omap_mcpdm_open_streams() ASoC: omap-mcpdm: Support for suspend resume ASoC: omap-mcpdm: Add support for pdmclk clock handling Documentation/devicetree/bindings/mfd/twl6040.txt | 1 + .../devicetree/bindings/sound/omap-mcpdm.txt | 10 +++ arch/arm/boot/dts/omap4-duovero.dtsi | 5 ++ arch/arm/boot/dts/omap4-panda-common.dtsi | 5 ++ arch/arm/boot/dts/omap4-sdp.dts | 5 ++ arch/arm/boot/dts/omap4-var-som-om44.dtsi | 5 ++ arch/arm/boot/dts/omap5-board-common.dtsi | 5 ++ drivers/clk/clk-twl6040.c | 80 ++++++++++++--------- drivers/mfd/twl6040.c | 6 ++ include/linux/mfd/twl6040.h | 2 +- sound/soc/omap/Kconfig | 1 + sound/soc/omap/omap-mcpdm.c | 84 ++++++++++++++++++++-- 12 files changed, 170 insertions(+), 39 deletions(-) -- 2.8.3

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[alsa-devel] [PATCH v4 0/2] amidi delay option
by Felipe F. Tonello 15 Aug '16

15 Aug '16

Small improvement to amidi, adding useful functionality. 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 | 63 ++++++++++++++++++++++++++++++++++++++++++++--------------- 2 files changed, 61 insertions(+), 16 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH v3 0/2] amidi delay option
by Felipe F. Tonello 15 Aug '16

15 Aug '16

Small improvement to amidi, adding useful functionality. change 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 | 60 +++++++++++++++++++++++++++++++++++++++++++---------------- 2 files changed, 58 insertions(+), 16 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH v2 0/2] amidi delay option
by Felipe F. Tonello 15 Aug '16

15 Aug '16

Small improvement to amidi, adding useful functionality. 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 | 66 ++++++++++++++++++++++++++++++++++++++++++++--------------- 2 files changed, 64 insertions(+), 16 deletions(-) -- 2.9.2

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[alsa-devel] [PATCH v8 0/5] ASoC: MT8173 HDMI jack detection
by Philipp Zabel 15 Aug '16

15 Aug '16

Hi, now that kernel v4.8-rc1 is released and includes the new CEC framework in drivers/staging/media/cec, I wonder how to proceed with this. To allow the ASoC drivers to present HDMI cable connection state as an ALSA jack, and to allow them to update sink capabilities when it signals an EDID change, we need a notification mechanism between CEC/HDMI, and ALSA drivers. I had used Russell's hdmi notification prototype to let the CEC/HDMI drivers notify the hdmi-codec and ASoC machine drivers of changes. The mtk_cec driver currently is just a placeholder that only handles the HPD interrupts. Should the notifications be integrated with the CEC framework for those devices that do have CEC support? Changes since v7 ("ASoC: MT8173 HDMI codec support"): - Dropped the already applied ELD control and machine driver patches - Dropped the N/CTS helper patch for now, as the helpers don't exist yet. - Lock ELD mutex while copying ELD in hdmi_eld_ctl_get regards Philipp Philipp Zabel (5): video: rmk's HDMI notification prototype ASoC: hdmi-codec: Use HDMI notifications to add jack support ASoC: mediatek: Add jack detection support to mt8173-rt5650-rt5676 machine driver ASoC: mediatek: Add jack detection support to the mt8173-rt5650 machine driver drm/mediatek: hdmi: issue notifications drivers/gpu/drm/mediatek/mtk_cec.c | 11 +++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 3 + drivers/video/Kconfig | 3 + drivers/video/Makefile | 1 + drivers/video/hdmi-notifier.c | 61 ++++++++++++++++ include/linux/hdmi-notifier.h | 44 ++++++++++++ include/sound/hdmi-codec.h | 6 ++ sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdmi-codec.c | 88 +++++++++++++++++++++--- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 21 ++++++ sound/soc/mediatek/mt8173/mt8173-rt5650.c | 21 ++++++ 11 files changed, 252 insertions(+), 8 deletions(-) create mode 100644 drivers/video/hdmi-notifier.c create mode 100644 include/linux/hdmi-notifier.h -- 2.8.1

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[alsa-devel] [GIT PULL]: intel audio firmware updates
by Vinod Koul 15 Aug '16

15 Aug '16

Hi, Please PULL to receive the latest Broxton and Kabylake audio firmware. The following changes since commit c170c8d95794d6aedbaeea44674daaa96baf04f7: linux-firmware: intel: Update Skylake audio firmware (2016-08-04 16:09:21 +0530) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git apl for you to fetch changes up to 92b8db544104bf8f4fa8b5b2bea2f3681ed2e713: linux-firmware: intel: Update Broxton audio firmware (2016-08-12 16:07:01 +0530) ---------------------------------------------------------------- Vinod Koul (2): linux-firmware: intel: Update Kabylake audio firmware linux-firmware: intel: Update Broxton audio firmware WHENCE | 8 ++++---- intel/dsp_fw_bxtn.bin | 2 +- intel/dsp_fw_bxtn_v1118.bin | Bin 0 -> 505608 bytes intel/dsp_fw_kbl.bin | 2 +- intel/dsp_fw_kbl_v1037.bin | Bin 0 -> 238920 bytes 5 files changed, 6 insertions(+), 6 deletions(-) create mode 100644 intel/dsp_fw_bxtn_v1118.bin create mode 100644 intel/dsp_fw_kbl_v1037.bin Thanks -- ~Vinod

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Re: [alsa-devel] ASoC: sun4i-codec: playback stall and I/O error with DAPM paths all disabled
by Mark Brown 15 Aug '16

15 Aug '16

On Mon, Aug 15, 2016 at 05:43:55PM +0800, wens Tsai wrote: > What is unexpected is any attempt to play anything under this state makes > the playback software (in my case mpg321) stall, and later report an I/O > error. My guess is that the DAC is still disabled by DAPM, so it doesn't > send any DRQs, and thus the DMA engine is not consuming any data from > userspace. This is normal for ASoC - like you say it'll be becasue the hardware isn't powered up. > I think we should just enable the digital bits of the DAC/ADC all the > time. Or maybe transfer and then discard data if the DAC is off. Not > sure if this is doable though. I expect playback software to work, and > not block, regardless of the hardware status. Powering things up all the time will have a major effect on battery life for systems that care about that. The expectation is that systems with this sort of hardware won't normally be offering end users direct control of the routing, it'll be something that's handled during system integration.

1 0

[alsa-devel] Bug in 3.10 alsa routing?
by noman pouigt 15 Aug '16

15 Aug '16

Hello folks, I am working on enabling a new alsa codec driver and I am stuck with this below routing problem. SND_SOC_DAPM_AIF_OUT("AIF1TX1",) { "AIF3RX1", NULL, "AIF3 Playback" }, { "AIF1TX1 Input 1", NULL, "AIF3RX1" }, { "AIF1TX1", AIF1TX1, "AIF1TX1 Input 1" }, { "AIF1TX1", AIF1TX2, "AIF1TX1 Input 1" }, { "AIF1 Capture", NULL, "AIF1TX1" }, AIF3 Playback and AIF1 Capture is stream name of AIF3 and AIF1 dai respectively. After setting the paths using mixer command and triggering AIF3 playback dai, I see that it is not powering AIF3 Playback widget. After debugging I found the problem to be that ALSA core searches for physically connected or active widget path before it can power on the widget. So I think the codec developer wanted me start the AIF1 capturing to turn on the whole path. However in our system we just want to connect AIF3RX1 to AIF1TX1. So i changed to below and now it is powering on the AIF3 Playback widget. SND_SOC_DAPM_OUTPUT("DUMMY_NODE"), // { "AIF1 Capture", NULL, "AIF1TX1" }, { "DUMMY_NODE", NULL, "AIF1TX1" }, Is ALSA core working as designed as I am using 3.11? Thanks, Variksla

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[alsa-devel] [PATCH v2 0/2] alsabat: test scripts and reference amixer config files
by focus.luo＠linux.intel.com 15 Aug '16

15 Aug '16

From: Focus Luo <focus.luo(a)linux.intel.com> The patch set supplies some automated test scripts and reference amixer config files for linux audio driver testing by using alsa-bat as test tool on Intel platforms. The first patch includes automated test scripts for linux audio driver based on alsa-lib interface by using alsabat as test tool. It supports analog audio and display audio(HDMI/DP) interface. The second patch supplies some amixer reference config file on Intel platforms. Focus Luo (2): alsabat: automation test scripts alsabat: add amixer config files bat/Makefile.am | 1 + bat/tests/Makefile.am | 11 + bat/tests/README | 32 ++ bat/tests/alsabat_main.sh | 178 ++++++++++ bat/tests/analog_audio_playback_and_capture.sh | 84 +++++ bat/tests/asound_state/Makefile.am | 6 + bat/tests/asound_state/asound.state.Broadwell | 439 +++++++++++++++++++++++++ bat/tests/asound_state/asound.state.Haswell | 342 +++++++++++++++++++ bat/tests/asound_state/asound.state.Skylake | 437 ++++++++++++++++++++++++ bat/tests/dp_audio_playback.sh | 83 +++++ bat/tests/dp_audio_subdevice_number.sh | 68 ++++ bat/tests/hdmi_audio_playback.sh | 81 +++++ bat/tests/hdmi_audio_subdevice_number.sh | 65 ++++ bat/tests/map_test_case | 8 + configure.ac | 3 +- 15 files changed, 1837 insertions(+), 1 deletion(-) create mode 100644 bat/tests/Makefile.am create mode 100644 bat/tests/README create mode 100755 bat/tests/alsabat_main.sh create mode 100755 bat/tests/analog_audio_playback_and_capture.sh create mode 100644 bat/tests/asound_state/Makefile.am create mode 100644 bat/tests/asound_state/asound.state.Broadwell create mode 100644 bat/tests/asound_state/asound.state.Haswell create mode 100644 bat/tests/asound_state/asound.state.Skylake create mode 100755 bat/tests/dp_audio_playback.sh create mode 100755 bat/tests/dp_audio_subdevice_number.sh create mode 100755 bat/tests/hdmi_audio_playback.sh create mode 100755 bat/tests/hdmi_audio_subdevice_number.sh create mode 100644 bat/tests/map_test_case -- 1.9.1

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Re: [alsa-devel] twl4030: runtime audio capture channel swapping (kernel 3.18+)
by Peter Ujfalusi 15 Aug '16

15 Aug '16

Hi Akram, On 08/11/16 04:07, Akram Hameed wrote: > Hei Peter, > > Sorry to bother you directly, but I got no response to my questions on the > alsa-devel list and since you seem to be a maintainer of the omap mcbsp and > pcm code, I thought you would: Sorry, it looks like I have missed your mail in alsa-devel. Adding it to CC and also looping Jarkko. > a) Want to be aware of my issues > > b) Might help point me in a direction to fix things. > > In short: I am capturing a stereo audio stream, 16-bit, 48kHz using a gumstix > overo that is quite similar to the older beagleboards (we are using two > variants - one is OMAP3530 based and the other, DM3730). Periodically, left > channel audio will move to right channel and vice versa over time. > > Previously my company was using a branch from kernel 3.0 (made by Steve > Sakoman) and our audio capture was fine - no swapping, ever. Due to > instabilities with the old mmc interface (unpredictable failures using newer > UHS-3 cards), we have been forced to migrate to a newer kernel, and here I > encounter the channel swap issue. > > To me, it seems the problem is like what is described > here: http://processors.wiki.ti.com/index.php/McBSP_Channel_Swapping > Which I presume draws on a discussion you had years ago with one > Ying: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-February/036895.h… > > After spending quite some time reading and investigating the various > registers, I come to the following conclusions: > > 1) Audio codec (TWL) appears to be configured in the same between working > (3.0) and non working (3.18) kernels, except that voice mode is active also > and some gain settings are different (different defaults, I suppose). I have a > comparison of these register maps if you want to see. > > 2) McBSP2 is also configured the same between old and new kernels, though I > note for some reason the ID pulled by dev_dbg looks spurious, though I guess > unrelated to my issue. > Configuring McBSP255 phys_base: 0x49022000 > **** McBSP255 regs **** > DRR2: 0xf88a > DRR1: 0x0000 > DXR2: 0x0000 > DXR1: 0x0000 > SPCR2: 0x0230 > SPCR1: 0x0031 > RCR2: 0x8041 > RCR1: 0x0040 > XCR2: 0x8041 > XCR1: 0x0040 > SRGR2: 0x001f > SRGR1: 0x0f00 > PCR0: 0x000f > *********************** We must drop the DRR1/2 (and probably DXR1/2) dump... By reading the DRR register with the debug code we introduce underflow. The data must be only moved by the DMA from the DATA registers. If we have low threshold it is more likely to cause channel swap and it is for sure going to introduce missing data - the data is going to the kernel log instead of the receive buffer. > 3) My channel swapping occurs when using dma_op_mode element OR threshold mode > in kernel 3.18+ (I have only tested on 3.18 and 4.4, for the record). Setting > a threshold of 960 with period size 10ms reduces the frequency of the swap. > > I did notice that DMA CCR is different between kernel 3 and 3.18 when > operating in ELEMENT mode. > > Version 3 kernel: > > DMA4:Chan5 @ 48056200:CCR=/dev/mem opened. > Memory mapped at address 0x4013b000. > Read at address 0x48056200 (0x4013b200): 0x01084482 > > > Version 3.18 kernel: > > DMA4:Chan5 @ 48056200:CCR=/dev/mem opened. > Memory mapped at address 0xb6fa1000. > Read at address 0x48056200 (0xb6fa1200): 0x010C44A2 <- McBSP2 RX request. > > > Interestingly, it appears the new kernel (3.18+) is using packet mode even > when 'element' is selected. If I choose dma_op_mode THRESHOLD in the older > kernel 3.0, I see the same 0x010C44A2 in the CCR as I do in 3.18+. Yes, I have made a change that if we are in element mode we will still use packet mode to transfer one sample per DMA request. Real element mode will be used only in case of mono audio. If the audio is 2+ channels we are using packet mode to transfer the sample. This does helped to avoid swaps as we send/receive one sample at a time from/to McBSP. > 4) I have not observed playback channel swapping. I am unsure why. > > 5) I enabled underflow and overflow reporting in the McBSP IRQ handler (kernel > 3.18+ since older 3 kernel did not have such a handler and I have not patched > one in yet), and found several interesting things. I can not recall the history, but between 3.0 and 3.18 we might moved to dmaengine based PCM for OMAPs. But I still have no recollection to experience capture side channel swap. > a) Swap definitely is happening due to Overflow, but so far I only observe an > overflow when using dma_op_mode THRESHOLD. When using dma_op_mode THRESHOLD, > the swap does not seem to coincide with an underflow, so I am at a loss to > explain further. How does you application works? What are the parameters ALSA is configured for capture/playback (cat /proc/asound/card0/pcm0p/sub0/hw_params; cat /proc/asound/card0/pcm0c/sub0/hw_params) - number of periods, etc? > b) In dma_op_mode ELEMENT, I get Underflow reported almost every second in > this newer kernel (again, using 10ms period size). Channel swap only seems to > occur ever few hours, however, and I do not believe I have observed the > overflow at all in ELEMENT mode. So, the swap occurs due to underflow, maybe? Underflow in McBSP or Underflow by ALSA? Underflow in McBSP RX tells that DMA is trying to read data when the FIFO is empty. This can only happen if something else is reading data also from the McBSP since we configure the McBSP and sDMA in sync. Overflow in McBSP happens when DAM is failing to read the data out from McBSP FIFO in time - FIFO is full and McBSP discards the incoming data) ALSA underflow is different, it happens when the application fails to process the period in time and the DMA starts to rewrite the buffer. This happens when you have one thread to capture/process/play. If the process/play takes more time than we go round in the boffer (buffer time) we have ALSA underflow. You can try to increase the number of periods or separate the capture and process/play jobs into separate threads. > My plan from now is to get my JTAG up and running and try and make the change > "If you are working with 16-bit stereo data a nice solution is to configure > the McBSP for a single 32-bit element instead of 2 16-bit elements. " The McBSP driver does not have support for this ATM. There is a side effect of this AFAIK: channels will be swapped, but I have not tested it. Using 32bit element instead of 2x16bit helps in case when you do not have FIFO for sure to give more time for the DMA. Also in case of underrun/overrun you will be loosing both channel's data so the swap would not happen. > If you have any helpful suggestions, I would really appreciate them. I am > operating under the assumption I can leave the TWL codec settings as they are > and just 'lie' to McBSP about the data framing. Namely, specify single phase > frame with 1 word of 32 bits, and set DMA packet size to a 32bit word also. > Since 16 bit mode operates by default in dual phase, I am not sure if the > signalling will work correctly, but I guess I will find out. Yeah, this is also going to be a problem. As twl4030 is using I2S when in stereo mode, with fake mono 32bit you can not use I2S signals. But since McBSP is looking for the start condition, configuring it for DSP_A mode will probably going to start the reception/playback at the correct time. > Again, my apologies for contacting you directly. Hopefully the information I > have provided can help diagnosing the real problem in the kernel. No problem. Sorry for that I have missed your mail in alsa-devel. > For me, I > must find a solution in the near-term so I will try to figure out some hacks > as described in the Wiki article. The wiki page was written for daVinci devices AFAIK where they don't have FIFO for their McBSP. -- Péter

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[alsa-devel] [PATCH 0/2] ALSA: rawmidi: add runtime error state for disorder of data transmission
by Takashi Sakamoto 14 Aug '16

14 Aug '16

Hi, This patchset is based on an idea addressed in this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-February/104589.h… For packet-oriented drivers such as USB Audio Device class, message transmission can fail due to data bus issue. In this case, corresponding rawmidi runtime should be closed. Although, current ALSA rawmidi core has no mechanism to help applications/drivers to handle this situation. This commit adds a helper functions for this aim. Takashi Sakamoto (2): ALSA: rawmidi: handle error state for broken logical channel ALSA: rawmidi: add a helper to set runtime error include/sound/rawmidi.h | 3 ++ sound/core/rawmidi.c | 86 ++++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 84 insertions(+), 5 deletions(-) -- 2.7.4

1 3

[alsa-devel] [PATCH v2] drm/i915/dp: DP audio API changes for MST
by Dhinakaran Pandiyan 13 Aug '16

13 Aug '16

DP MST provides the capability to send multiple video and audio streams through a single port. This requires the API's between i915 and audio drivers to distinguish between multiple audio capable displays that can be connected to a port. Currently only the port identity is shared in the APIs. This patch adds support for MST with an additional parameter 'int pipe'. The existing parameter 'port' does not change it's meaning. pipe = MST : display pipe that the stream originates from Non-MST : -1 Affected APIs: struct i915_audio_component_ops - int (*sync_audio_rate)(struct device *, int port, int rate); + int (*sync_audio_rate)(struct device *, int port, int pipe, + int rate); - int (*get_eld)(struct device *, int port, bool *enabled, - unsigned char *buf, int max_bytes); + int (*get_eld)(struct device *, int port, int pipe, + bool *enabled, unsigned char *buf, int max_bytes); struct i915_audio_component_audio_ops - void (*pin_eld_notify)(void *audio_ptr, int port); + void (*pin_eld_notify)(void *audio_ptr, int port, int pipe); This patch makes dummy changes in the audio drivers (Libin) for build to succeed. The audio side drivers will send the right 'pipe' values in patches that will follow. v2: Renamed the new API parameter from 'dev_id' to 'pipe'. (Jim, Ville) Included Asoc driver API compatibility changes from Jeeja. Added WARN_ON() for invalid pipe in get_saved_encoder(). (Takashi) Added comment for av_enc_map[] definition. (Takashi) Signed-off-by: Dhinakaran Pandiyan <dhinakaran.pandiyan(a)intel.com> --- drivers/gpu/drm/i915/i915_drv.h | 3 +- drivers/gpu/drm/i915/intel_audio.c | 92 ++++++++++++++++++++++++++------------ include/drm/i915_component.h | 6 +-- include/sound/hda_i915.h | 11 ++--- sound/hda/hdac_i915.c | 9 ++-- sound/pci/hda/patch_hdmi.c | 7 +-- sound/soc/codecs/hdac_hdmi.c | 2 +- 7 files changed, 85 insertions(+), 45 deletions(-) diff --git a/drivers/gpu/drm/i915/i915_drv.h b/drivers/gpu/drm/i915/i915_drv.h index c36d176..8e4a88f 100644 --- a/drivers/gpu/drm/i915/i915_drv.h +++ b/drivers/gpu/drm/i915/i915_drv.h @@ -2036,7 +2036,8 @@ struct drm_i915_private { /* perform PHY state sanity checks? */ bool chv_phy_assert[2]; - struct intel_encoder *dig_port_map[I915_MAX_PORTS]; + /* Used to save the pipe-to-encoder mapping for audio */ + struct intel_encoder *av_enc_map[I915_MAX_PIPES]; /* * NOTE: This is the dri1/ums dungeon, don't add stuff here. Your patch diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index ef20875..240dad2 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -500,6 +500,7 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) struct i915_audio_component *acomp = dev_priv->audio_component; struct intel_digital_port *intel_dig_port = enc_to_dig_port(encoder); enum port port = intel_dig_port->port; + enum pipe pipe = -1; connector = drm_select_eld(encoder); if (!connector) @@ -524,12 +525,18 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) mutex_lock(&dev_priv->av_mutex); intel_encoder->audio_connector = connector; + /* referred in audio callbacks */ - dev_priv->dig_port_map[port] = intel_encoder; + dev_priv->av_enc_map[pipe] = intel_encoder; mutex_unlock(&dev_priv->av_mutex); + + if (intel_encoder->type == INTEL_OUTPUT_DP_MST) + pipe = crtc->pipe; + if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port); + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, + (int) port, (int) pipe); } /** @@ -542,22 +549,28 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) void intel_audio_codec_disable(struct intel_encoder *intel_encoder) { struct drm_encoder *encoder = &intel_encoder->base; + struct intel_crtc *crtc = to_intel_crtc(encoder->crtc); struct drm_device *dev = encoder->dev; struct drm_i915_private *dev_priv = to_i915(dev); struct i915_audio_component *acomp = dev_priv->audio_component; struct intel_digital_port *intel_dig_port = enc_to_dig_port(encoder); enum port port = intel_dig_port->port; + enum pipe pipe = -1; if (dev_priv->display.audio_codec_disable) dev_priv->display.audio_codec_disable(intel_encoder); mutex_lock(&dev_priv->av_mutex); intel_encoder->audio_connector = NULL; - dev_priv->dig_port_map[port] = NULL; + dev_priv->av_enc_map[pipe] = NULL; mutex_unlock(&dev_priv->av_mutex); + if (intel_encoder->type == INTEL_OUTPUT_DP_MST) + pipe = crtc->pipe; + if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port); + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, + (int) port, (int) pipe); } /** @@ -632,15 +645,39 @@ static int i915_audio_component_get_cdclk_freq(struct device *dev) return dev_priv->cdclk_freq; } -static int i915_audio_component_sync_audio_rate(struct device *dev, - int port, int rate) +static struct intel_encoder *get_saved_encoder(struct intel_encoder *av_enc_map[], + int port, int pipe) +{ + struct drm_encoder *encoder; + + if (WARN_ON(pipe >= I915_MAX_PIPES)) + return NULL; + + /* MST */ + if (pipe != -1) + return av_enc_map[pipe]; + + /* Non-MST */ + for (pipe = PIPE_A; pipe < I915_MAX_PIPES; pipe++) { + if (!av_enc_map[pipe]) + continue; + + encoder = &av_enc_map[pipe]->base; + if (port == enc_to_dig_port(encoder)->port) + return av_enc_map[pipe]; + } + + return NULL; +} + +static int i915_audio_component_sync_audio_rate(struct device *dev, int port, + int pipe, int rate) { struct drm_i915_private *dev_priv = dev_to_i915(dev); struct intel_encoder *intel_encoder; struct intel_crtc *crtc; struct drm_display_mode *mode; struct i915_audio_component *acomp = dev_priv->audio_component; - enum pipe pipe = INVALID_PIPE; u32 tmp; int n; int err = 0; @@ -654,25 +691,20 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, i915_audio_component_get_power(dev); mutex_lock(&dev_priv->av_mutex); + /* 1. get the pipe */ - intel_encoder = dev_priv->dig_port_map[port]; - /* intel_encoder might be NULL for DP MST */ + intel_encoder = get_saved_encoder(dev_priv->av_enc_map, port, pipe); if (!intel_encoder || !intel_encoder->base.crtc || intel_encoder->type != INTEL_OUTPUT_HDMI) { - DRM_DEBUG_KMS("no valid port %c\n", port_name(port)); + DRM_DEBUG_KMS("Not valid for port %c\n", port_name(port)); err = -ENODEV; goto unlock; } + + /* pipe passed from the audio driver will be -1 for non-mst case */ crtc = to_intel_crtc(intel_encoder->base.crtc); pipe = crtc->pipe; - if (pipe == INVALID_PIPE) { - DRM_DEBUG_KMS("no pipe for the port %c\n", port_name(port)); - err = -ENODEV; - goto unlock; - } - DRM_DEBUG_KMS("pipe %c connects port %c\n", - pipe_name(pipe), port_name(port)); mode = &crtc->config->base.adjusted_mode; /* port must be valid now, otherwise the pipe will be invalid */ @@ -708,7 +740,7 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, } static int i915_audio_component_get_eld(struct device *dev, int port, - bool *enabled, + int pipe, bool *enabled, unsigned char *buf, int max_bytes) { struct drm_i915_private *dev_priv = dev_to_i915(dev); @@ -717,16 +749,20 @@ static int i915_audio_component_get_eld(struct device *dev, int port, int ret = -EINVAL; mutex_lock(&dev_priv->av_mutex); - intel_encoder = dev_priv->dig_port_map[port]; - /* intel_encoder might be NULL for DP MST */ - if (intel_encoder) { - ret = 0; - *enabled = intel_encoder->audio_connector != NULL; - if (*enabled) { - eld = intel_encoder->audio_connector->eld; - ret = drm_eld_size(eld); - memcpy(buf, eld, min(max_bytes, ret)); - } + + intel_encoder = get_saved_encoder(dev_priv->av_enc_map, port, pipe); + if (!intel_encoder) { + DRM_DEBUG_KMS("Not valid for port %c\n", port_name(port)); + mutex_unlock(&dev_priv->av_mutex); + return ret; + } + + ret = 0; + *enabled = intel_encoder->audio_connector != NULL; + if (*enabled) { + eld = intel_encoder->audio_connector->eld; + ret = drm_eld_size(eld); + memcpy(buf, eld, min(max_bytes, ret)); } mutex_unlock(&dev_priv->av_mutex); diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index b46fa0e..545c6e0 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -64,7 +64,7 @@ struct i915_audio_component_ops { * Called from audio driver. After audio driver sets the * sample rate, it will call this function to set n/cts */ - int (*sync_audio_rate)(struct device *, int port, int rate); + int (*sync_audio_rate)(struct device *, int port, int pipe, int rate); /** * @get_eld: fill the audio state and ELD bytes for the given port * @@ -77,7 +77,7 @@ struct i915_audio_component_ops { * Note that the returned size may be over @max_bytes. Then it * implies that only a part of ELD has been copied to the buffer. */ - int (*get_eld)(struct device *, int port, bool *enabled, + int (*get_eld)(struct device *, int port, int pipe, bool *enabled, unsigned char *buf, int max_bytes); }; @@ -97,7 +97,7 @@ struct i915_audio_component_audio_ops { * status accordingly (even when the HDA controller is in power save * mode). */ - void (*pin_eld_notify)(void *audio_ptr, int port); + void (*pin_eld_notify)(void *audio_ptr, int port, int pipe); }; /** diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index 796cabf..07fd64e 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -10,8 +10,9 @@ int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); int snd_hdac_display_power(struct hdac_bus *bus, bool enable); void snd_hdac_i915_set_bclk(struct hdac_bus *bus); -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate); -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, + int pipe, int rate); +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int pipe, bool *audio_enabled, char *buffer, int max_bytes); int snd_hdac_i915_init(struct hdac_bus *bus); int snd_hdac_i915_exit(struct hdac_bus *bus); @@ -29,13 +30,13 @@ static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { } static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec, - hda_nid_t nid, int rate) + hda_nid_t nid, int pipe, int rate) { return 0; } static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, - bool *audio_enabled, char *buffer, - int max_bytes) + int pipe, bool *audio_enabled, + char *buffer, int max_bytes) { return -ENODEV; } diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index c9af022..b99994b 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -201,7 +201,8 @@ static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) * This function sets N/CTS value based on the given sample rate. * Returns zero for success, or a negative error code. */ -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate) +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, + int pipe, int rate) { struct hdac_bus *bus = codec->bus; struct i915_audio_component *acomp = bus->audio_component; @@ -212,7 +213,7 @@ int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate) port = pin2port(codec, nid); if (port < 0) return -EINVAL; - return acomp->ops->sync_audio_rate(acomp->dev, port, rate); + return acomp->ops->sync_audio_rate(acomp->dev, port, pipe, rate); } EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); @@ -236,7 +237,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); * thus it may be over @max_bytes. If it's over @max_bytes, it implies * that only a part of ELD bytes have been fetched. */ -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int pipe, bool *audio_enabled, char *buffer, int max_bytes) { struct hdac_bus *bus = codec->bus; @@ -249,7 +250,7 @@ int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, port = pin2port(codec, nid); if (port < 0) return -EINVAL; - return acomp->ops->get_eld(acomp->dev, port, audio_enabled, + return acomp->ops->get_eld(acomp->dev, port, pipe, audio_enabled, buffer, max_bytes); } EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d0d5ad8..67890df 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1485,7 +1485,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, mutex_lock(&per_pin->lock); eld->monitor_present = false; - size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, + size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, -1, &eld->monitor_present, eld->eld_buffer, ELD_MAX_SIZE); if (size > 0) { @@ -1739,7 +1739,8 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, /* Call sync_audio_rate to set the N/CTS/M manually if necessary */ /* Todo: add DP1.2 MST audio support later */ if (codec_has_acomp(codec)) - snd_hdac_sync_audio_rate(&codec->core, pin_nid, runtime->rate); + snd_hdac_sync_audio_rate(&codec->core, pin_nid, -1, + runtime->rate); non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); mutex_lock(&per_pin->lock); @@ -2285,7 +2286,7 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, snd_hda_codec_set_power_to_all(codec, fg, power_state); } -static void intel_pin_eld_notify(void *audio_ptr, int port) +static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) { struct hda_codec *codec = audio_ptr; int pin_nid; diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 2abb742..cf57ab3 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1366,7 +1366,7 @@ static int hdac_hdmi_parse_and_map_nid(struct hdac_ext_device *edev, return hdac_hdmi_init_dai_map(edev); } -static void hdac_hdmi_eld_notify_cb(void *aptr, int port) +static void hdac_hdmi_eld_notify_cb(void *aptr, int port, int pipe) { struct hdac_ext_device *edev = aptr; struct hdac_hdmi_priv *hdmi = edev->private_data; -- 2.5.0

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Re: [alsa-devel] Correct modules for Bay Trail MAX98090 soc?
by Pierre-Louis Bossart 12 Aug '16

12 Aug '16

On 8/11/16 3:42 PM, Laura Abbott wrote: > Hi, > > Fedora got a bug report[1] that sound was failing on Intel Bay Trail Socs > that use the MAX98090 module. I found > > Author: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > Date: Mon Feb 8 10:45:38 2016 +0530 > > ASoC: Intel: Load the atom DPCM driver only > DPCM driver is recommended for BYT, CHT based platforms, so if > CONFIG_SND_SST_IPC_ACPI is selected then don't compile the BYT > Device IDs in common ACPI driver to avoid probe conflicts. > Signed-off-by: Pierre-Louis Bossart > <pierre-louis.bossart(a)linux.intel.com> > Acked-by: Jie Yang <yang.jie(a)intel.com> > Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> > Signed-off-by: Mark Brown <broonie(a)kernel.org> > > > which changed the dependencies for CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH. > The set of options Fedora selects means that > CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH > can't be selected. Is there another driver that's supposed to replace > CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH on Bay Trail or do the dependencies > need to be updated? The bugzilla has alsa-info for working and non- > working cases and the Fedora config is attached. If you remove support for all other baytrail options this driver should still be there and selectable. We just can't support both this driver for Chromebooks and the rest for other machines with the same distribution at the moment.

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[alsa-devel] [PATCH] amidi: add delay option
by Felipe F. Tonello 12 Aug '16

12 Aug '16

This patch adds a new options to amidi tool to add a delay (in milliseconds) to each MIDI message. This is very useful when sending firmware updates to a remote device via SysEx or any other use that requires this delay in between messages. Signed-off-by: Felipe F. Tonello <eu(a)felipetonello.com> --- amidi/amidi.1 | 14 ++++++++++++ amidi/amidi.c | 72 ++++++++++++++++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 81 insertions(+), 5 deletions(-) diff --git a/amidi/amidi.1 b/amidi/amidi.1 index 1b4cfb1ea956..40402c17e374 100644 --- a/amidi/amidi.1 +++ b/amidi/amidi.1 @@ -111,6 +111,12 @@ to stop receiving data. Does not ignore Active Sensing bytes (FEh) when saving or printing received MIDI commands. +.TP +.I \-D, \-\-delay=mseconds +Adds a delay in between each MIDI message sent to a device. It is +useful when sending firmware updates via SysEx messages to a remote +device. + .SH EXAMPLES .TP @@ -121,6 +127,14 @@ to port .I hw:0. .TP +.B amidi \-p hw:1,0,0 -s firmware.syx \-D 100 +will send the MIDI commands in +.I firmware.syx +to port +.I hw:1,0,0 +with 100 milliseconds delay in between MIDI messages. + +.TP .B amidi \-S 'F0 43 10 4C 00 00 7E 00 F7' sends an XG Reset to the default port. diff --git a/amidi/amidi.c b/amidi/amidi.c index cedf18c5fd39..f45abfa7eac0 100644 --- a/amidi/amidi.c +++ b/amidi/amidi.c @@ -48,6 +48,7 @@ static int receive_file; static int dump; static int timeout; static int stop; +static int delay; static snd_rawmidi_t *input, **inputp; static snd_rawmidi_t *output, **outputp; @@ -77,7 +78,8 @@ static void usage(void) "-d, --dump print received data as hexadecimal bytes\n" "-t, --timeout=seconds exits when no data has been received\n" " for the specified duration\n" - "-a, --active-sensing don't ignore active sensing bytes\n"); + "-a, --active-sensing don't ignore active sensing bytes\n" + "-D, --delay=mseconds delay in between each sent MIDI message\n"); } static void version(void) @@ -305,6 +307,35 @@ static void parse_data(void) send_data_length = i; } +static int message_length(unsigned char byte) +{ + static int rstate_len = 0; + + switch (byte) { + case 0xf8 ... 0xff: + return 1; + case 0xf0: + return -1; /* variable size */ + case 0xf1: + case 0xf3: + return 2; + case 0xf2: + return 3; + case 0xf4: + case 0xf5: + return 0; /* ignore */ + case 0xf6: + return 1; + case 0xf7: + return 0; /* ignore */ + case 0x80 ... 0xef: + rstate_len = (byte >= 0xc0 && byte <= 0xdf) ? 2 : 3; + return rstate_len; + case 0x00 ... 0x7f: + return rstate_len ?: rstate_len - 1; + } +} + /* * prints MIDI commands, formatting them nicely */ @@ -406,7 +437,7 @@ static void add_send_hex_data(const char *str) int main(int argc, char *argv[]) { - static const char short_options[] = "hVlLp:s:r:S::dt:a"; + static const char short_options[] = "hVlLp:s:r:S::dt:aD:"; static const struct option long_options[] = { {"help", 0, NULL, 'h'}, {"version", 0, NULL, 'V'}, @@ -419,6 +450,7 @@ int main(int argc, char *argv[]) {"dump", 0, NULL, 'd'}, {"timeout", 1, NULL, 't'}, {"active-sensing", 0, NULL, 'a'}, + {"delay", 1, NULL, 'D'}, { } }; int c, err, ok = 0; @@ -463,6 +495,9 @@ int main(int argc, char *argv[]) case 'a': ignore_active_sensing = 0; break; + case 'D': + delay = atoi(optarg); + break; default: error("Try `amidi --help' for more information."); return 1; @@ -538,9 +573,36 @@ int main(int argc, char *argv[]) error("cannot set blocking mode: %s", snd_strerror(err)); goto _exit; } - if ((err = snd_rawmidi_write(output, send_data, send_data_length)) < 0) { - error("cannot send data: %s", snd_strerror(err)); - goto _exit; + if (!delay) { + if ((err = snd_rawmidi_write(output, send_data, send_data_length)) < 0) { + error("cannot send data: %s", snd_strerror(err)); + return err; + } + } else { + unsigned char *data = (unsigned char *)send_data; + + while (data < ((unsigned char *)send_data + send_data_length)) { + + if (data > (unsigned char *)send_data) + usleep(delay * 1000); + + int len = message_length(*data); + + /* check for variable length */ + if (len == -1) { + int i = 0; + while (data[i++] != 0xf7); + len = i; + } + + if (len > 0 && + (err = snd_rawmidi_write(output, data, len)) < 0) { + error("cannot send data: %s", snd_strerror(err)); + return err; + } + + data += len ?: 1; + } } } -- 2.9.2

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[alsa-devel] [PATCH] ALSA: hda - repoll jack presence state on resume
by Jaroslav Kysela 12 Aug '16

12 Aug '16

The jack detection on resume might fail on some hardware (Lenovo T430s, T440s and X1 Carbon 2016). The codec registers reports that the jack is not used and the unsolicited event is not sent on change (or perhaps, it's lost?). This patch will repoll the jack presence state after 100ms again. Because the jack status is updated from two workqueues now, create a new codec->jack_mutex to protect the concurrent access. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/hda_bind.c | 2 ++ sound/pci/hda/hda_codec.c | 9 ++++++++- sound/pci/hda/hda_codec.h | 1 + 3 files changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 6efadbf..81a9284 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -42,8 +42,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) { struct hda_codec *codec = container_of(dev, struct hda_codec, core); + mutex_lock(&codec->jack_mutex); if (codec->patch_ops.unsol_event) codec->patch_ops.unsol_event(codec, ev); + mutex_unlock(&codec->jack_mutex); } /** diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9913be8..4339e98 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -592,8 +592,10 @@ static void hda_jackpoll_work(struct work_struct *work) struct hda_codec *codec = container_of(work, struct hda_codec, jackpoll_work.work); + mutex_lock(&codec->jack_mutex); snd_hda_jack_set_dirty_all(codec); snd_hda_jack_poll_all(codec); + mutex_unlock(&codec->jack_mutex); if (!codec->jackpoll_interval) return; @@ -837,6 +839,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); mutex_init(&codec->control_mutex); + mutex_init(&codec->jack_mutex); snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); @@ -3002,8 +3005,12 @@ static void hda_call_codec_resume(struct hda_codec *codec) if (codec->jackpoll_interval) hda_jackpoll_work(&codec->jackpoll_work.work); - else + else { snd_hda_jack_report_sync(codec); + /* Some hw requires a little time to settle things */ + schedule_delayed_work(&codec->jackpoll_work, + msecs_to_jiffies(100)); + } atomic_dec(&codec->core.in_pm); } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 373fcad..6a4a631 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -213,6 +213,7 @@ struct hda_codec { struct mutex spdif_mutex; struct mutex control_mutex; + struct mutex jack_mutex; struct snd_array spdif_out; unsigned int spdif_in_enable; /* SPDIF input enable? */ const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ -- 2.5.5

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[alsa-devel] [PATCH -next] ASoC: atmel-pdmic: add missing clk_disable_unprepare() on error in atmel_pdmic_cpu_dai_startup()
by Wei Yongjun 12 Aug '16

12 Aug '16

Add the missing clk_disable_unprepare() before return from atmel_pdmic_cpu_dai_startup() in the error handling case. Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/atmel/atmel-pdmic.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index 2a3a41f..a67e31b 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -115,8 +115,10 @@ static int atmel_pdmic_cpu_dai_startup(struct snd_pcm_substream *substream, return ret; ret = clk_prepare_enable(dd->pclk); - if (ret) + if (ret) { + clk_disable_unprepare(dd->gclk); return ret; + } /* Clear all bits in the Control Register(PDMIC_CR) */ regmap_write(dd->regmap, PDMIC_CR, 0);

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[alsa-devel] [PATCH -next] ASoC: topology: Fix error return code in soc_tplg_dapm_widget_create()
by Wei Yongjun 12 Aug '16

12 Aug '16

Fix to return error code -ENOMEM instead of 0 when failed to create widget, as done elsewhere in this function. Fixes: 8a9782346dcc ("ASoC: topology: Add topology core") Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/soc-topology.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 05a18f6..a9e83a2 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1476,6 +1476,7 @@ widget: if (widget == NULL) { dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n", w->name); + ret = -ENOMEM; goto hdr_err; }

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[alsa-devel] [PATCH] ASoC: omap-mcbsp: Enable TX/RX under and overflow interrupts
by Peter Ujfalusi 12 Aug '16

12 Aug '16

FIFO under or overflow can cause channel swaps and data loss. Reporting them can help to identify such events. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/mcbsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 76ce33199bf9..06fec5699cc8 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -221,7 +221,8 @@ void omap_mcbsp_config(struct omap_mcbsp *mcbsp, /* Enable TX/RX sync error interrupts by default */ if (mcbsp->irq) - MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN); + MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN | + RUNDFLEN | ROVFLEN | XUNDFLEN | XOVFLEN); } /** -- 2.9.2

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[alsa-devel] [PATCH v2 0/2] ASoC: Extended SPDIF support to Allwinner A31 SoCs
by codekipper＠gmail.com 12 Aug '16

12 Aug '16

From: Marcus Cooper <codekipper(a)gmail.com> This patch set extends the current support of the Allwinner SPDIF transceiver to include the A31 SoC. For now just the SPDIF transmitter has been tested on the Mele I7. In order for this patch set to be functional then we require changes to the clocks and the device tree. For those that are interested I've pushed all the required patches here https://github.com/codekipper/linux-sunxi/commits/a31-spdif Thanks in advance, CK --- Changes since v1: - Corrected mistake in bindings documentation Marcus Cooper (2): ASoC: sunxi: Add bindings for sun6i to SPDIF ASoC: sunxi: compatibility for sun6i to SPDIF .../devicetree/bindings/sound/sunxi,sun4i-spdif.txt | 3 +++ sound/soc/sunxi/sun4i-spdif.c | 17 ++++++++++++++++- 2 files changed, 19 insertions(+), 1 deletion(-) -- 2.9.2

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[alsa-devel] [PATCH -next] ASoC: Intel: Skylake: Fix error return code in skl_probe()
by Wei Yongjun 12 Aug '16

12 Aug '16

Fix to return error code -ENODEV from the error handling case instead of 0, as done elsewhere in this function. Fixes: 87b2bdf02278 ("ASoC: Intel: Skylake: Initialize NHLT table") Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/intel/skylake/skl.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index a893ca1..2989c16 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -672,8 +672,10 @@ static int skl_probe(struct pci_dev *pci, skl->nhlt = skl_nhlt_init(bus->dev); - if (skl->nhlt == NULL) + if (skl->nhlt == NULL) { + err = -ENODEV; goto out_free; + } skl_nhlt_update_topology_bin(skl);

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Re: [alsa-devel] Underrun with 2 periods on skylake HDA driver
by Dharageswari R 12 Aug '16

12 Aug '16

On Tue, Aug 09, 2016 at 11:54:22PM -0700, Clemens Ladisch wrote: > Dharageswari R wrote: > > While doing the validation of Skylake HDA driver using aplay and > > following parameters, we noticed underrun issues. > > > > buffer_size : 9600 > > period_size : 4800 > > > > After doing further debugging, what I found is > > > > 1.DMA complete interrupt occurs for the first period. > > 2.Driver reads DMA position in buffer pointer and gets value of 4784. > > The interrupt says that the buffer descriptor list entry has been > completed, but the position in buffer pointer says that the entry has > not yet been completed. > yes > > I checked with HDA architects and they are saying that when the DMA > > completion interrupt occurs there can be slight delay before the DMA > > Position In Buffer Pointer gets updated. > > The driver's interrupt handler takes care to read the interrupt status > register first, so that any pending memory writes from the device are > flushed. > > Of course, this does not work if the device posts writes _after_ the > interrupt has been raised. This is a bug in the hardware. > > > So driver is expected to refill data based on whatever content is > > reflected in the DMA Position In Buffer. > > No, that's not how it works. The driver waits until all 4800 frames > are available, and the only way to wake it up is with an interrupt. > There is no other interrupt apart from the buffer completion interrupt to know the post write completion. The only option here would be to poll till the position is equal or more than the period boundary with timeout before the period elapsed reporting. I thought of avoiding this if possible but doesnt seem to be true. Thanks for your reply. I will make sure that the driver reports period ellapsed at or above the period boundary. > > It appears the driver must work around this bug by scheduling another > interrupt after the delay. Is there an upper limit (time or frames) for > this delay? The upper limit is not deterministic in this case. > > > Regards, > Clemens --

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Re: [alsa-devel] [PATCH] ALSA: usb: caiaq: audio: don't print error when allocating urb fails
by Takashi Iwai 11 Aug '16

11 Aug '16

On Thu, 11 Aug 2016 22:40:11 +0200, Wolfram Sang wrote: > > kmalloc will print enough information in case of failure. > > Signed-off-by: Wolfram Sang <wsa-dev(a)sang-engineering.com> Applied, thanks. Takashi > --- > sound/usb/caiaq/audio.c | 1 - > 1 file changed, 1 deletion(-) > > diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c > index 327f8642ca80e6..8f66ba730d69d9 100644 > --- a/sound/usb/caiaq/audio.c > +++ b/sound/usb/caiaq/audio.c > @@ -739,7 +739,6 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) > for (i = 0; i < N_URBS; i++) { > urbs[i] = usb_alloc_urb(FRAMES_PER_URB, GFP_KERNEL); > if (!urbs[i]) { > - dev_err(dev, "unable to usb_alloc_urb(), OOM!?\n"); > *ret = -ENOMEM; > return urbs; > } > -- > 2.8.1 > >

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[alsa-devel] [PATCH v3 0/4] ALSA: seq: obsolete change of address limit
by Takashi Sakamoto 11 Aug '16

11 Aug '16

Hi, ALSA sequencer core has two types of clients; application and kernel. For kernel clients, the core should handle data in kernel space, meanwhile this is achieved with a hack of change of address limit. This should be purged. This patchset is a revised version of my previous one: [alsa-devel] obsolete change of address limit http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/110937.html Changes in v2: - Use '_IOC_SIZE' macro to calculate the size of argument of ioctl command. - Keep redundant longer name. - Improve comments. - Merge some patches relevant to the same features. Changes in v3: - Improve series of patch - Use 'dir' field of ioctl command to detect the length of argument Regards Takashi Sakamoto (4): ALSA: seq: add documentation for snd_seq_kernel_client_ctl ALSA: seq: add an alternative way to handle ioctl requests ALSA: seq: change ioctl command operation to get data in kernel space ALSA: seq: obsolete change of address limit sound/core/seq/seq_clientmgr.c | 721 ++++++++++++++++++----------------------- sound/core/seq/seq_compat.c | 7 +- 2 files changed, 317 insertions(+), 411 deletions(-) -- 2.7.4

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Re: [alsa-devel] [PATCH v8 1/5] video: rmk's HDMI notification prototype
by Russell King - ARM Linux 11 Aug '16

11 Aug '16

On Thu, Aug 11, 2016 at 04:49:28PM +0200, Hans Verkuil wrote: > On 08/11/16 16:16, Russell King - ARM Linux wrote: > > On Thu, Aug 11, 2016 at 12:49:21PM +0200, Hans Verkuil wrote: > >> I thought this was the raw EDID data. But if you pass a struct edid around, > >> then why not make this const struct edid *edid? I fail to see why you would > >> want to use a void pointer here. > > > > struct edid is a DRM thing - it's not generic. Using struct edid here > > would force everyone to use the DRM structure, whereas other subsystems > > use, eg, unsigned char. > > So how the edid pointer should be interpreted depends on which subsystem > sends it? That doesn't sound right. It makes it really hard for the drivers > receiving the notifications. No, that's wrong... and I really don't see why you're trying to make a meal of this. In both cases, it's the EDID as received from the device. One subsystem happens to represent it internally as a "struct edid" in its drivers, another subsystem uses "unsigned char []". In order to _cleanly_ accept both styles, without having to resort to driver authors needing to litter their code with lots of idiotic casts, I'm accepting a void * pointer, so that we can accept _both_ a struct edid and an unsigned char array. Both are compatible with each other. Both have the same bytes at the same offset. There is no difference between them other than the C types that are used by each subsystem. > But I understand that currently the main purpose of the hdmi_event_new_edid > is to get the physical address from the EDID for CEC? Or do you have other > uses as well? That is one use, but if an audio driver wants to parse the raw EDID instead of the ELD, it can. What I want to avoid is having the video driver parse lots of different bits from the EDID and then send lots of events "just in case" something wants some of the parsed information - it's much more efficient to give drivers the raw information and allow them to parse what they need from the EDID. I'm not convinced that parsing out the physical address, and then passing that from the video side is a good idea. The reason I made an exception for the ELD is because (a) it's already passed between video and audio drivers, (b) the code we have present to convert an EDID to ELD is in DRM, and requires DRM data structures which won't be available in an audio driver. If you want a mechanism to pass just the physical address, we could add it, and require CEC drivers to make use of both the EDID or physical address events. That would mean situations where the physical address is known to be fixed before hand can be easily catered for. -- RMK's Patch system: http://www.armlinux.org.uk/developer/patches/ FTTC broadband for 0.8mile line: currently at 9.6Mbps down 400kbps up according to speedtest.net.

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Re: [alsa-devel] SND_PCM_STREAM_CAPTURE and SND_PCM_NONBLOCK producing strange artifacts
by Takashi Iwai 11 Aug '16

11 Aug '16

Please don't drop Cc to ML. Also avoid top-post. On Tue, 05 Jul 2016 09:02:12 +0200, Trent Reed wrote: > > Thanks Takashi, > > My implementation does only use the "hw:N" devices, unfortunately - I am > not using MMAP because I was under the impression that you must poll when > directly accessing the hardware. And yes, 48kHz is the exact rate I'm > attempting. :( > I do have an update, though - I'm able to reproduce the issue quite simply > by running the ALSA test application latency with the following command: > > `latency -P <playback-device> -C <capture-device> -r 48000` > > This, as far as I understand from the source code, should open R/W > interleaved streams on ("hw:2" is what I'm using for both, though the > problem persists in non-duplex mode, e.g. "hw:2" "hw:0") in non-blocking > mode without polling. > > What happens is the latency application does normalize around some value > (say, 16ms roundtrip time), but the audio is awful and broken, a lot of > this "static" I'm talking about. It seems to me that it's the same problem > that I have in my above-mentioned sample code. > > I would love to help more, so if I can in any way, please let me know! But > I feel at a loss for how to further debug. The latency program is too complex to analyze the issue. Check arecord with --nonblock --test-nowait options and with the hw device. Does the issue happen always? Or does it depend on the buffer or period size? Takashi > > Thanks, > - Trent Reed > > On Mon, Jul 4, 2016 at 11:52 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > On Sat, 02 Jul 2016 23:58:28 +0200, > > Trent Reed wrote: > > > > > > (Ping) > > > > > > I'm afraid I still don't understand why this is failing. I found that a > > > call to sdn_pcm_wait() when -EAGAIN is returned fixes the problem because > > > it waits for samples to be ready, but I don't understand why it is a > > > problem in the first place. Much of the sample code is written in a way > > > that suggests that the call to snd_pcm_wait() is optional, and only > > > required when directly reading/writing from the device (MMAP, I'm not > > doing > > > this). > > > > > > I'm just wondering what is wrong with basically looping over a call to > > > snd_pcm_readi() in non-blocking mode, reacting to errors, and capturing > > > frames. This produces awful static (which is basically garbage samples), > > is > > > it a requirement that I call snd_pcm_wait() when -EAGAIN is passed back > > to > > > wait for a full period to be ready? > > > > Well, in the API level, it should work as you expected. So there must > > be definitely some bug(s). But it's hard to spot out, as there are > > several layers behind the scene. > > > > As a first shot, try to reproduce without alsa-lib plugins, i.e. only > > with "hw" PCM device, if not done yet. Also, it's interesting to see > > whether it happens with or without mmap r/w. > > > > Another point is whether it depends on the parameter. Did you try > > 48kHz instead of 44.1kHz? > > > > > > Takashi > > > > > > > > > > Thanks, > > > - Trent Reed > > > > > > > > > On Tue, Jun 28, 2016 at 7:27 PM, Trent Reed <treed0803(a)gmail.com> wrote: > > > > > > > Hey All, > > > > > > > > I've been banging my head against the wall for about a week on this > > bug. > > > > The following gist shows my sample reproduction code: > > > > https://gist.github.com/TReed0803/985c5d5c3d295245e19006a27be447c3 > > > > > > > > I'm simply opening up a non-blocking PCM capture stream and writing the > > > > contents of the reads to stdout. > > > > (Originally, I was writing to the playback stream, but I was hearing > > this > > > > strange static occasionally!) > > > > > > > > It's the static I'm trying to debug. It doesn't happen every time. In > > > > fact, sometimes I'll go a few consecutive executions without hearing > > it. > > > > I was able to capture some of the bad data, and I loaded it up in > > Audacity > > > > for visualization: > > > > > > > > > > https://drive.google.com/file/d/0B-1aumGKQcQTcUJoZzIwRWhYSFE/view?usp=shari… > > > > > > > > It looks like the internal buffer occasionally is sending me more data > > > > than it actually captured, and I end up either reading old PCM data > > from > > > > the internal ring buffer, or (at the very beginning) a bunch of zeros. > > > > > > > > Can anyone help me understand what is going on? What could cause these > > > > definitely incorrect samples to be recorded? (I get the same effect > > > > regardless of hardware, but I will list hardware just in case.) > > > > > > > > I hope I have all the information you might need: > > > > Hardware: Samson Meteor Mic (USB-Audio, USB Mixer) [Though, it even > > > > happens with my built-in microphone] > > > > ALSA version: Advanced Linux Sound Architecture Driver Version > > > > k4.4.0-28-generic. > > > > apt-cache policy (installed alsa-lib version): 1.1.0-0ubuntu1 > > > > > > > > Thanks, > > > > - Trent Reed > > > > > > > _______________________________________________ > > > Alsa-devel mailing list > > > Alsa-devel(a)alsa-project.org > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > >

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Re: [alsa-devel] [PATCH v8 1/5] video: rmk's HDMI notification prototype
by Russell King - ARM Linux 11 Aug '16

11 Aug '16

On Thu, Aug 11, 2016 at 12:49:21PM +0200, Hans Verkuil wrote: > On 08/11/16 12:39, Russell King - ARM Linux wrote: > > On Thu, Aug 11, 2016 at 12:30:03PM +0200, Hans Verkuil wrote: > >> Hi Russell, > >> > >> I like this approach and it is something I will need for other CEC drivers > >> (not yet merged) as well. > >> > >> Just a few comments: > >> > >> On 08/11/16 11:20, Philipp Zabel wrote: > >>> This is Russell's HDMI notification prototype [1], currently waiting > >>> for the HDMI CEC situation to resolve. > >>> > >>> The use case for the notifications on MediaTek MT8173 is to let the > >>> (dis)connection notifications control an ALSA jack object. > >>> > >>> No Signed-off-by since this is not my code, and still up for discussion. > >>> > >>> [1] https://patchwork.kernel.org/patch/8351501/ > >>> --- > >>> drivers/video/Kconfig | 3 +++ > >>> drivers/video/Makefile | 1 + > >>> drivers/video/hdmi-notifier.c | 61 +++++++++++++++++++++++++++++++++++++++++++ > >>> include/linux/hdmi-notifier.h | 44 +++++++++++++++++++++++++++++++ > >>> 4 files changed, 109 insertions(+) > >>> create mode 100644 drivers/video/hdmi-notifier.c > >>> create mode 100644 include/linux/hdmi-notifier.h > >>> > >>> diff --git a/drivers/video/Kconfig b/drivers/video/Kconfig > >>> index 3c20af9..1ee7b9f 100644 > >>> --- a/drivers/video/Kconfig > >>> +++ b/drivers/video/Kconfig > >>> @@ -36,6 +36,9 @@ config VIDEOMODE_HELPERS > >>> config HDMI > >>> bool > >>> > >>> +config HDMI_NOTIFIERS > >>> + bool > >>> + > >>> if VT > >>> source "drivers/video/console/Kconfig" > >>> endif > >>> diff --git a/drivers/video/Makefile b/drivers/video/Makefile > >>> index 9ad3c17..65f5649 100644 > >>> --- a/drivers/video/Makefile > >>> +++ b/drivers/video/Makefile > >>> @@ -1,5 +1,6 @@ > >>> obj-$(CONFIG_VGASTATE) += vgastate.o > >>> obj-$(CONFIG_HDMI) += hdmi.o > >>> +obj-$(CONFIG_HDMI_NOTIFIERS) += hdmi-notifier.o > >>> > >>> obj-$(CONFIG_VT) += console/ > >>> obj-$(CONFIG_LOGO) += logo/ > >>> diff --git a/drivers/video/hdmi-notifier.c b/drivers/video/hdmi-notifier.c > >>> new file mode 100644 > >>> index 0000000..a233359 > >>> --- /dev/null > >>> +++ b/drivers/video/hdmi-notifier.c > >>> @@ -0,0 +1,61 @@ > >>> +#include <linux/export.h> > >>> +#include <linux/hdmi-notifier.h> > >>> +#include <linux/notifier.h> > >>> +#include <linux/string.h> > >>> + > >>> +static BLOCKING_NOTIFIER_HEAD(hdmi_notifier); > >>> + > >>> +int hdmi_register_notifier(struct notifier_block *nb) > >>> +{ > >>> + return blocking_notifier_chain_register(&hdmi_notifier, nb); > >>> +} > >>> +EXPORT_SYMBOL_GPL(hdmi_register_notifier); > >>> + > >>> +int hdmi_unregister_notifier(struct notifier_block *nb) > >>> +{ > >>> + return blocking_notifier_chain_unregister(&hdmi_notifier, nb); > >>> +} > >>> +EXPORT_SYMBOL_GPL(hdmi_unregister_notifier); > >>> + > >>> +void hdmi_event_connect(struct device *dev) > >>> +{ > >>> + struct hdmi_event_base base; > >>> + > >>> + base.source = dev; > >>> + > >>> + blocking_notifier_call_chain(&hdmi_notifier, HDMI_CONNECTED, &base); > >>> +} > >>> +EXPORT_SYMBOL_GPL(hdmi_event_connect); > >>> + > >>> +void hdmi_event_disconnect(struct device *dev) > >>> +{ > >>> + struct hdmi_event_base base; > >>> + > >>> + base.source = dev; > >>> + > >>> + blocking_notifier_call_chain(&hdmi_notifier, HDMI_DISCONNECTED, &base); > >>> +} > >>> +EXPORT_SYMBOL_GPL(hdmi_event_disconnect); > >>> + > >>> +void hdmi_event_new_edid(struct device *dev, const void *edid, size_t size) > >> > >> I would prefer const u8 *edid. It is after all just a u8 array and not some > >> opaque data structure. > > > > In DRM, the edid is of type "struct edid" (defined in > > include/drm/drm_edid.h). So, making it "const u8 *" means that all > > DRM drivers will have to explicitly cast it - not something I'm in > > favour of. > > I thought this was the raw EDID data. But if you pass a struct edid around, > then why not make this const struct edid *edid? I fail to see why you would > want to use a void pointer here. struct edid is a DRM thing - it's not generic. Using struct edid here would force everyone to use the DRM structure, whereas other subsystems use, eg, unsigned char. > Also struct edid is for the first EDID block only, whereas CEC drivers need > the CEA-861 block to get the physical address. That may be, but DRM parses the HDMI vendor block too while still using struct edid to represent it. See drm_find_edid_extension() in drivers/gpu/drm/drm_edid.c This is just how DRM is... I'm not sure what the design decisions were that came up with this, but it's what we have, and I'm not sure whether DRM would be open to changing it. > > The ELD stands for EDID-like data. See information on HDA039. It's a > > method of conveying the EDID data specific to audio drivers to them > > without having the complexities of parsing the full EDID each time. > > It's also exported to userland so that userland can determine the > > capabilities of the audio path, again, without having to have full > > EDID parsers and exporting the full EDID data block through audio > > drivers. > > > > OK. Since eld is an unsigned char array I would suggest changing void to > unsigned char here. > > Or perhaps const unsigned char eld[128]. Yes, I'll make that change. -- RMK's Patch system: http://www.armlinux.org.uk/developer/patches/ FTTC broadband for 0.8mile line: currently at 9.6Mbps down 400kbps up according to speedtest.net.

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[alsa-devel] [PATCH] ASoC: arizona: Connect ASRCs to both clock domains
by Charles Keepax 11 Aug '16

11 Aug '16

The two clock domains are enabled based on each input and output of the chip being connected to its respective clock domain. The ASRC however can bridge the two domains and as such can function as an input/output to either domain. The hardware also requires that both clocks are enabled before the ASRC is. Ensure these constraints by linking the ASRCs to both SYSCLK and ASYNCCLK. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs47l24.c | 10 ++++++++++ sound/soc/codecs/wm5102.c | 10 ++++++++++ sound/soc/codecs/wm5110.c | 10 ++++++++++ sound/soc/codecs/wm8998.c | 10 ++++++++++ 4 files changed, 40 insertions(+) diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 779a7bf..257b152 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -746,6 +746,16 @@ static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = { { "IN2L", NULL, "SYSCLK" }, { "IN2R", NULL, "SYSCLK" }, + { "ASRC1L", NULL, "SYSCLK" }, + { "ASRC1R", NULL, "SYSCLK" }, + { "ASRC2L", NULL, "SYSCLK" }, + { "ASRC2R", NULL, "SYSCLK" }, + + { "ASRC1L", NULL, "ASYNCCLK" }, + { "ASRC1R", NULL, "ASYNCCLK" }, + { "ASRC2L", NULL, "ASYNCCLK" }, + { "ASRC2R", NULL, "ASYNCCLK" }, + { "MICBIAS1", NULL, "MICVDD" }, { "MICBIAS2", NULL, "MICVDD" }, diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 1295978..2d6c19d 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1521,6 +1521,16 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "IN3L", NULL, "SYSCLK" }, { "IN3R", NULL, "SYSCLK" }, + { "ASRC1L", NULL, "SYSCLK" }, + { "ASRC1R", NULL, "SYSCLK" }, + { "ASRC2L", NULL, "SYSCLK" }, + { "ASRC2R", NULL, "SYSCLK" }, + + { "ASRC1L", NULL, "ASYNCCLK" }, + { "ASRC1R", NULL, "ASYNCCLK" }, + { "ASRC2L", NULL, "ASYNCCLK" }, + { "ASRC2R", NULL, "ASYNCCLK" }, + { "MICBIAS1", NULL, "MICVDD" }, { "MICBIAS2", NULL, "MICVDD" }, { "MICBIAS3", NULL, "MICVDD" }, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 3e7f873..fa35e19 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1745,6 +1745,16 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "IN4L", NULL, "SYSCLK" }, { "IN4R", NULL, "SYSCLK" }, + { "ASRC1L", NULL, "SYSCLK" }, + { "ASRC1R", NULL, "SYSCLK" }, + { "ASRC2L", NULL, "SYSCLK" }, + { "ASRC2R", NULL, "SYSCLK" }, + + { "ASRC1L", NULL, "ASYNCCLK" }, + { "ASRC1R", NULL, "ASYNCCLK" }, + { "ASRC2L", NULL, "ASYNCCLK" }, + { "ASRC2R", NULL, "ASYNCCLK" }, + { "MICBIAS1", NULL, "MICVDD" }, { "MICBIAS2", NULL, "MICVDD" }, { "MICBIAS3", NULL, "MICVDD" }, diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 315b23b..edaa4cf 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -966,6 +966,16 @@ static const struct snd_soc_dapm_route wm8998_dapm_routes[] = { { "IN2A", NULL, "SYSCLK" }, { "IN2B", NULL, "SYSCLK" }, + { "ASRC1L", NULL, "SYSCLK" }, + { "ASRC1R", NULL, "SYSCLK" }, + { "ASRC2L", NULL, "SYSCLK" }, + { "ASRC2R", NULL, "SYSCLK" }, + + { "ASRC1L", NULL, "ASYNCCLK" }, + { "ASRC1R", NULL, "ASYNCCLK" }, + { "ASRC2L", NULL, "ASYNCCLK" }, + { "ASRC2R", NULL, "ASYNCCLK" }, + { "SPD1", NULL, "SYSCLK" }, { "SPD1", NULL, "SPD1TX1" }, { "SPD1", NULL, "SPD1TX2" }, -- 2.1.4

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[alsa-devel] [PATCH] ASoC: wm2000: Fix return of uninitialised varible
by Charles Keepax 11 Aug '16

11 Aug '16

Anything that sets ret in wm2000_anc_transition will have immediately returned anyway as such we will always return an uninitialised ret at the bottom of the function. Simply replace the return with a return 0; Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 1ebaf99..199a3d8 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -581,7 +581,7 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, if (anc_transitions[i].dest == ANC_OFF) clk_disable_unprepare(wm2000->mclk); - return ret; + return 0; } static int wm2000_anc_set_mode(struct wm2000_priv *wm2000) -- 2.1.4

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[alsa-devel] [PATCH] ASoC: Fix leak of rtd in soc_bind_dai_link
by Charles Keepax 11 Aug '16

11 Aug '16

If we fail to find a platform we simply return EPROBE_DEFER, but we have allocated the rtd pointer. All error paths before soc_add_pcm_runtime need to call soc_free_pcm_runtime first to avoid leaking the rtd pointer. A suitable error path already exists and is used else where in the function so simply use that here as well. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 054d94f..84e3fa1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1056,7 +1056,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, if (!rtd->platform) { dev_err(card->dev, "ASoC: platform %s not registered\n", dai_link->platform_name); - return -EPROBE_DEFER; + goto _err_defer; } soc_add_pcm_runtime(card, rtd); -- 2.1.4

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[alsa-devel] [PATCH] ASoC: rt5659: Add mclk controls
by Nicolin Chen 11 Aug '16

11 Aug '16

The codec driver should control the mclk. So this patch adds this support. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5659.txt | 3 +++ sound/soc/codecs/rt5659.c | 24 ++++++++++++++++++++++ sound/soc/codecs/rt5659.h | 1 + 3 files changed, 28 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5659.txt b/Documentation/devicetree/bindings/sound/rt5659.txt index 5f79e7f..1766e05 100644 --- a/Documentation/devicetree/bindings/sound/rt5659.txt +++ b/Documentation/devicetree/bindings/sound/rt5659.txt @@ -12,6 +12,9 @@ Required properties: Optional properties: +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + - realtek,in1-differential - realtek,in3-differential - realtek,in4-differential diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 1b30914..d9ead96 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -9,6 +9,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -3565,7 +3566,9 @@ static int rt5659_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) static int rt5659_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct rt5659_priv *rt5659 = snd_soc_codec_get_drvdata(codec); + int ret; switch (level) { case SND_SOC_BIAS_PREPARE: @@ -3582,6 +3585,17 @@ static int rt5659_set_bias_level(struct snd_soc_codec *codec, RT5659_PWR_FV1 | RT5659_PWR_FV2); break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_OFF) { + ret = clk_prepare_enable(rt5659->mclk); + if (ret) { + dev_err(codec->dev, + "failed to enable MCLK: %d\n", ret); + return ret; + } + } + break; + case SND_SOC_BIAS_OFF: regmap_update_bits(rt5659->regmap, RT5659_PWR_DIG_1, RT5659_PWR_LDO, 0); @@ -3591,6 +3605,7 @@ static int rt5659_set_bias_level(struct snd_soc_codec *codec, RT5659_PWR_MB | RT5659_PWR_VREF2); regmap_update_bits(rt5659->regmap, RT5659_DIG_MISC, RT5659_DIG_GATE_CTRL, 0); + clk_disable_unprepare(rt5659->mclk); break; default: @@ -4020,6 +4035,15 @@ static int rt5659_i2c_probe(struct i2c_client *i2c, regmap_write(rt5659->regmap, RT5659_RESET, 0); + /* Check if MCLK provided */ + rt5659->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(rt5659->mclk)) { + if (PTR_ERR(rt5659->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + /* Otherwise mark the mclk pointer to NULL */ + rt5659->mclk = NULL; + } + rt5659_calibrate(rt5659); /* line in diff mode*/ diff --git a/sound/soc/codecs/rt5659.h b/sound/soc/codecs/rt5659.h index d31c9e5..d69b0eb 100644 --- a/sound/soc/codecs/rt5659.h +++ b/sound/soc/codecs/rt5659.h @@ -1796,6 +1796,7 @@ struct rt5659_priv { struct gpio_desc *gpiod_reset; struct snd_soc_jack *hs_jack; struct delayed_work jack_detect_work; + struct clk *mclk; int sysclk; int sysclk_src; -- 2.1.4

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Re: [alsa-devel] [PATCH v8 1/5] video: rmk's HDMI notification prototype
by Russell King - ARM Linux 11 Aug '16

11 Aug '16

On Thu, Aug 11, 2016 at 12:30:03PM +0200, Hans Verkuil wrote: > Hi Russell, > > I like this approach and it is something I will need for other CEC drivers > (not yet merged) as well. > > Just a few comments: > > On 08/11/16 11:20, Philipp Zabel wrote: > > This is Russell's HDMI notification prototype [1], currently waiting > > for the HDMI CEC situation to resolve. > > > > The use case for the notifications on MediaTek MT8173 is to let the > > (dis)connection notifications control an ALSA jack object. > > > > No Signed-off-by since this is not my code, and still up for discussion. > > > > [1] https://patchwork.kernel.org/patch/8351501/ > > --- > > drivers/video/Kconfig | 3 +++ > > drivers/video/Makefile | 1 + > > drivers/video/hdmi-notifier.c | 61 +++++++++++++++++++++++++++++++++++++++++++ > > include/linux/hdmi-notifier.h | 44 +++++++++++++++++++++++++++++++ > > 4 files changed, 109 insertions(+) > > create mode 100644 drivers/video/hdmi-notifier.c > > create mode 100644 include/linux/hdmi-notifier.h > > > > diff --git a/drivers/video/Kconfig b/drivers/video/Kconfig > > index 3c20af9..1ee7b9f 100644 > > --- a/drivers/video/Kconfig > > +++ b/drivers/video/Kconfig > > @@ -36,6 +36,9 @@ config VIDEOMODE_HELPERS > > config HDMI > > bool > > > > +config HDMI_NOTIFIERS > > + bool > > + > > if VT > > source "drivers/video/console/Kconfig" > > endif > > diff --git a/drivers/video/Makefile b/drivers/video/Makefile > > index 9ad3c17..65f5649 100644 > > --- a/drivers/video/Makefile > > +++ b/drivers/video/Makefile > > @@ -1,5 +1,6 @@ > > obj-$(CONFIG_VGASTATE) += vgastate.o > > obj-$(CONFIG_HDMI) += hdmi.o > > +obj-$(CONFIG_HDMI_NOTIFIERS) += hdmi-notifier.o > > > > obj-$(CONFIG_VT) += console/ > > obj-$(CONFIG_LOGO) += logo/ > > diff --git a/drivers/video/hdmi-notifier.c b/drivers/video/hdmi-notifier.c > > new file mode 100644 > > index 0000000..a233359 > > --- /dev/null > > +++ b/drivers/video/hdmi-notifier.c > > @@ -0,0 +1,61 @@ > > +#include <linux/export.h> > > +#include <linux/hdmi-notifier.h> > > +#include <linux/notifier.h> > > +#include <linux/string.h> > > + > > +static BLOCKING_NOTIFIER_HEAD(hdmi_notifier); > > + > > +int hdmi_register_notifier(struct notifier_block *nb) > > +{ > > + return blocking_notifier_chain_register(&hdmi_notifier, nb); > > +} > > +EXPORT_SYMBOL_GPL(hdmi_register_notifier); > > + > > +int hdmi_unregister_notifier(struct notifier_block *nb) > > +{ > > + return blocking_notifier_chain_unregister(&hdmi_notifier, nb); > > +} > > +EXPORT_SYMBOL_GPL(hdmi_unregister_notifier); > > + > > +void hdmi_event_connect(struct device *dev) > > +{ > > + struct hdmi_event_base base; > > + > > + base.source = dev; > > + > > + blocking_notifier_call_chain(&hdmi_notifier, HDMI_CONNECTED, &base); > > +} > > +EXPORT_SYMBOL_GPL(hdmi_event_connect); > > + > > +void hdmi_event_disconnect(struct device *dev) > > +{ > > + struct hdmi_event_base base; > > + > > + base.source = dev; > > + > > + blocking_notifier_call_chain(&hdmi_notifier, HDMI_DISCONNECTED, &base); > > +} > > +EXPORT_SYMBOL_GPL(hdmi_event_disconnect); > > + > > +void hdmi_event_new_edid(struct device *dev, const void *edid, size_t size) > > I would prefer const u8 *edid. It is after all just a u8 array and not some > opaque data structure. In DRM, the edid is of type "struct edid" (defined in include/drm/drm_edid.h). So, making it "const u8 *" means that all DRM drivers will have to explicitly cast it - not something I'm in favour of. > > +{ > > + struct hdmi_event_new_edid new_edid; > > + > > + new_edid.base.source = dev; > > + new_edid.edid = edid; > > + new_edid.size = size; > > + > > + blocking_notifier_call_chain(&hdmi_notifier, HDMI_NEW_EDID, &new_edid); > > +} > > +EXPORT_SYMBOL_GPL(hdmi_event_new_edid); > > + > > +void hdmi_event_new_eld(struct device *dev, const void *eld) > > Stupid question: what does ELD stand for? I've no idea... > > And is void the right choice here or should it also be u8? The ELD stands for EDID-like data. See information on HDA039. It's a method of conveying the EDID data specific to audio drivers to them without having the complexities of parsing the full EDID each time. It's also exported to userland so that userland can determine the capabilities of the audio path, again, without having to have full EDID parsers and exporting the full EDID data block through audio drivers. -- RMK's Patch system: http://www.armlinux.org.uk/developer/patches/ FTTC broadband for 0.8mile line: currently at 9.6Mbps down 400kbps up according to speedtest.net.

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[alsa-devel] [PATCH 0/3] ASoC: Intel: Skylake: Refine manifest module info
by Vinod Koul 10 Aug '16

10 Aug '16

The module query uses a local structure dfw_config and then populates module skl_module_cfg. Instead we can pass skl_module_cfg as argument and get it updated in snd_skl_get_module_info() Also we can optimize runtime processing by using driver list to populate module info after init. That makes runtime query introduced earlier redundant so remove that while keeping error messages. Dharageswari R (1): ASoC: Intel: Skylake: modify snd_skl_get_module_info args Vinod Koul (2): ASoC: Intel: Skylake: Populate modules after loading ASoC: Intel: Skylake: remove module id query at runtime sound/soc/intel/skylake/skl-pcm.c | 27 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-dsp.h | 7 ++++--- sound/soc/intel/skylake/skl-sst-utils.c | 10 +++++----- sound/soc/intel/skylake/skl-topology.c | 22 ++++------------------ 4 files changed, 40 insertions(+), 26 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/6] ASoC: simple-card: add canonicalize_cpu / clean_reference
by Kuninori Morimoto 10 Aug '16

10 Aug '16

Hi Mark These add canonicalize_cpu / clean_reference to simple utiles. These are part of simple-card cleanup Kuninori Morimoto (6): ASoC: simple-card-utils: add asoc_simple_card_init_dai() ASoC: simple-card: use asoc_simple_card_init_dai() ASoC: rsrc-card: use asoc_simple_card_init_dai() ASoC: simple-card-utils: add asoc_simple_card_canonicalize_dailink() ASoC: simple-card: use asoc_simple_card_canonicalize_dailink() ASoC: rsrc-card: use asoc_simple_card_canonicalize_dailink() include/sound/simple_card_utils.h | 5 +++++ sound/soc/generic/simple-card-utils.c | 42 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 47 +++++------------------------------------------ sound/soc/sh/rcar/rsrc-card.c | 32 +++++--------------------------- 4 files changed, 57 insertions(+), 69 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 00/49] ASoC: add simple-card-core and standardize "simple" card
by Kuninori Morimoto 10 Aug '16

10 Aug '16

Hi Mark These are very big patch-set, but basically just cleanup for simple-card driver. Now, we have simple-card driver, but it is not simple today. I would like to cleanup this. And Renesas already have rsrc-card driver which is supporting DPCM, and it is based on simple-card. This measn rsrc-card and simple-card are very similar. Thus, I would like to share these code between rsrc-card and simple-card. This rsrc-card is only used from Renesas (actually, there is no upstream user, current user is only me, by local). But this is not Renesas special driver, I can say it is DPCM version of simple-card. I would like to open it for all of ALSA people. But, not merge simple-card <-> rsrc-card, to avoid diffcult future maintenance. Thus, rsrc-card will be simple-dpcm-audio-card, and we can maintenance both simple-card, and simple-dpcm-card. I'm not sure this is good nameing. And finally, we will need HDMI sound support which uses graph base probing. I know it needs more discussion with V4L2 people. But in my understanding, basically, we can reuse simple-card base feature/function on graph base driver somehow. I would like to standardize simple card style, and use/share it with graph driver too. It will be simple-graph-card ? I'm not sure. But I think we can share same code, and can avoid development from 0. These patch are for it. Basically, it pickups common function/feature from simple-card and rsrc-card, and standardize each feature/parse/init etc in simple-card-core. Because of this patch-set, "simple-card" become "simple" again :) I tested these (simple-card, rsrc-card) on my lager, Salvator boards. But I couldn't test some feature. I hope test it on many environment. 1) - 4) cleanup for these patch-set 5) - 19) pickuped common function/feature and put it to simple-card-core 20) - 32) use simple-card-core on simple-card 33) - 46) use simple-card-core on rsrc-card 47) - 49) rename rsrc-card to simple-dpcm-audio-card Kuninori Morimoto (49): 1) ASoC: simple-card: remove duplicate header 2) ASoC: rsrc-card: remove duplicate header 3) ASoC: rsrc-card: remove unused dai_num 4) ASoC: simple-card: platform also uses asoc_simple_card_sub_parse_of() 5) ASoC: add new simple-card-core.c 6) ASoC: simple-card-core: add asoc_simple_card_parse_tdm() 7) ASoC: simple-card-core: add asoc_simple_card_parse_dailink_name() 8) ASoC: simple-card-core: add asoc_simple_card_parse_card_name() 9) ASoC: simple-card-core: add asoc_simple_card_parse_card_prefix() 10) ASoC: simple-card-core: add asoc_simple_card_parse_card_widgets() 11) ASoC: simple-card-core: add asoc_simple_card_parse_card_widgets() 12) ASoC: simple-card-core: add asoc_simple_card_parse_clk() 13) ASoC: simple-card-core: add asoc_simple_card_parse_endpoint() 14) ASoC: simple-card-core: add asoc_simple_card_parse_dpcm() 15) ASoC: simple-card-core: add asoc_simple_card_init_jack() 16) ASoC: simple-card-core: add asoc_simple_card_init_dai() 17) ASoC: simple-card-core: add asoc_simple_card_canonicalize_dailink() 18) ASoC: simple-card-core: add asoc_simple_card_canonicalize_cpu() 19) ASoC: simple-card-core: add asoc_simple_card_clean_reference() 20) ASoC: simple-card: use asoc_simple_card_parse_daifmt() 21) ASoC: simple-card: use asoc_simple_card_parse_clk() 22) ASoC: simple-card: use asoc_simple_card_parse_endpoint() 23) ASoC: simple-card: use asoc_simple_card_parse_tdm() 24) ASoC: simple-card: use asoc_simple_card_parse_card_name() 25) ASoC: simple-card: use asoc_simple_card_parse_card_route() 26) ASoC: simple-card: use asoc_simple_card_parse_card_widgets() 27) ASoC: simple-card: use asoc_simple_card_parse_dailink_name() 28) ASoC: simple-card: use asoc_simple_card_init_jack() 29) ASoC: simple-card: use asoc_simple_card_init_dai() 30) ASoC: simple-card: use asoc_simple_card_canonicalize_dailink() 31) ASoC: simple-card: use asoc_simple_card_canonicalize_cpu() 32) ASoC: simple-card: use asoc_simple_card_clean_reference() 33) ASoC: rsrc-card: use asoc_simple_card_parse_daifmt() 34) ASoC: rsrc-card: use asoc_simple_card_parse_dailink_name() 35) ASoC: rsrc-card: use asoc_simple_dai instead of rsrc_card_dai 36) ASoC: rsrc-card: use asoc_simple_card_parse_clk() 37) ASoC: rsrc-card: use asoc_simple_card_parse_endpoint() 38) ASoC: rsrc-card: use asoc_simple_card_parse_card_name() 39) ASoC: rsrc-card: use asoc_simple_card_parse_card_prefix() 40) ASoC: rsrc-card: use asoc_simple_card_parse_dpcm() 41) ASoC: rsrc-card: use asoc_simple_card_parse_card_route() 42) ASoC: rsrc-card: use asoc_simple_card_parse_tdm() 43) ASoC: rsrc-card: use asoc_simple_card_init_dai() 44) ASoC: rsrc-card: use asoc_simple_card_canonicalize_cpu() 45) ASoC: rsrc-card: use asoc_simple_card_canonicalize_dailink() 46) ASoC: rsrc-card: use asoc_simple_card_clean_reference() 47) ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase1 48) ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase2 49) ASoC: rsrc-card: rename rsrc-card to simple-dpcm-card phase3 Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 75 -------------- Documentation/devicetree/bindings/sound/simple-dpcm-card.txt | 79 +++++++++++++++ include/sound/simple_card.h | 11 +-- include/sound/simple_card_core.h | 106 ++++++++++++++++++++ sound/soc/generic/Kconfig | 10 ++ sound/soc/generic/Makefile | 4 + sound/soc/generic/simple-card-core.c | 409 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 359 ++++++++++++------------------------------------------------------ sound/soc/generic/simple-dpcm-card.c | 330 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/Kconfig | 5 - sound/soc/sh/rcar/Makefile | 3 - sound/soc/sh/rcar/rsrc-card.c | 529 -------------------------------------------------------------------------------------------------- 12 files changed, 999 insertions(+), 921 deletions(-)

2 72

[alsa-devel] [PATCH v4 2/4] mfd: Add Samsung Exynos Low Power Audio Subsystem driver
by Sylwester Nawrocki 10 Aug '16

10 Aug '16

This patch adds common driver for the Top block of the Samsung Exynos SoC Low Power Audio Subsystem. This is a minimal driver which prepares resources for IP blocks like I2S, audio DMA and UART and exposes a regmap for the Top block registers. Also system power ops are added to ensure the Audio Subsystem is operational after system suspend/resume cycle. Signed-off-by: Inha Song <ideal.song(a)samsung.com> Signed-off-by: Beomho Seo <beomho.seo(a)samsung.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes since v3: - moved from sound/soc/samsung and rewritten as a MFD driver, - PMU register definitions moved to include/linux/mfd/syscon/exynos5-pmu.h, - added regmap for LPASS Top SFR region, - cleaned up register bit field defintions. Changes since v2: - move misplaced SND_SAMSUNG_AUDSS Kconfig symbol addition to this patch. --- drivers/mfd/Kconfig | 8 ++ drivers/mfd/Makefile | 1 + drivers/mfd/exynos-lpass.c | 185 +++++++++++++++++++++++++++++++++ include/linux/mfd/syscon/exynos5-pmu.h | 4 +- 4 files changed, 197 insertions(+), 1 deletion(-) create mode 100644 drivers/mfd/exynos-lpass.c diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 1bcf601..fc2dffb 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -270,6 +270,14 @@ config MFD_DLN2 etc. must be enabled in order to use the functionality of the device. +config MFD_EXYNOS_LPASS + tristate "Samsung Exynos SoC Low Power Audio Subsystem" + select MFD_CORE + select REGMAP_MMIO + help + Select this option to enable support for Samsung Exynos Low Power + Audio Subsystem. + config MFD_MC13XXX tristate depends on (SPI_MASTER || I2C) diff --git a/drivers/mfd/Makefile b/drivers/mfd/Makefile index 42a66e1..5c4be31 100644 --- a/drivers/mfd/Makefile +++ b/drivers/mfd/Makefile @@ -13,6 +13,7 @@ obj-$(CONFIG_MFD_BCM590XX) += bcm590xx.o obj-$(CONFIG_MFD_CROS_EC) += cros_ec.o obj-$(CONFIG_MFD_CROS_EC_I2C) += cros_ec_i2c.o obj-$(CONFIG_MFD_CROS_EC_SPI) += cros_ec_spi.o +obj-$(CONFIG_MFD_EXYNOS_LPASS) += exynos-lpass.o rtsx_pci-objs := rtsx_pcr.o rts5209.o rts5229.o rtl8411.o rts5227.o rts5249.o obj-$(CONFIG_MFD_RTSX_PCI) += rtsx_pci.o diff --git a/drivers/mfd/exynos-lpass.c b/drivers/mfd/exynos-lpass.c new file mode 100644 index 0000000..956ce1f --- /dev/null +++ b/drivers/mfd/exynos-lpass.c @@ -0,0 +1,185 @@ +/* + * Copyright (C) 2015 - 2016 Samsung Electronics Co., Ltd. + * + * Authors: Inha Song <ideal.song(a)samsung.com> + * Sylwester Nawrocki <s.nawrocki(a)samsung.com> + * + * Samsung Exynos SoC series Low Power Audio Subsystem driver. + * + * This module provides regmap for the Top SFR region and instantiates + * devices for IP blocks like DMAC, I2S, UART. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + */ + +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/mfd/syscon.h> +#include <linux/mfd/syscon/exynos5-pmu.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <linux/types.h> + +/* LPASS Top register definitions */ +#define SFR_LPASS_CORE_SW_RESET (0x08) +#define LPASS_SB_SW_RESET (1 << 11) +#define LPASS_UART_SW_RESET (1 << 10) +#define LPASS_PCM_SW_RESET (1 << 9) +#define LPASS_I2S_SW_RESET (1 << 8) +#define LPASS_WDT1_SW_RESET (1 << 4) +#define LPASS_WDT0_SW_RESET (1 << 3) +#define LPASS_TIMER_SW_RESET (1 << 2) +#define LPASS_MEM_SW_RESET (1 << 1) +#define LPASS_DMA_SW_RESET (1 << 0) + +#define SFR_LPASS_INTR_CA5_MASK (0x48) +#define SFR_LPASS_INTR_CPU_MASK (0x58) +#define LPASS_INTR_APM (1 << 9) +#define LPASS_INTR_MIF (1 << 8) +#define LPASS_INTR_TIMER (1 << 7) +#define LPASS_INTR_DMA (1 << 6) +#define LPASS_INTR_GPIO (1 << 5) +#define LPASS_INTR_I2S (1 << 4) +#define LPASS_INTR_PCM (1 << 3) +#define LPASS_INTR_SLIMBUS (1 << 2) +#define LPASS_INTR_UART (1 << 1) +#define LPASS_INTR_SFR (1 << 0) + +struct exynos_lpass { + struct regmap *pmu; + struct regmap *top; +}; + +static void exynos_lpass_core_sw_reset(struct exynos_lpass *lpass, int mask) +{ + unsigned int val = 0; + + regmap_read(lpass->top, SFR_LPASS_CORE_SW_RESET, &val); + + val &= ~mask; + regmap_write(lpass->top, SFR_LPASS_CORE_SW_RESET, val); + + usleep_range(100, 150); + + val |= mask; + regmap_write(lpass->top, SFR_LPASS_CORE_SW_RESET, val); +} + +static void exynos_lpass_enable(struct exynos_lpass *lpass) +{ + /* Unmask SFR, DMA and I2S interrupt */ + regmap_write(lpass->top, SFR_LPASS_INTR_CA5_MASK, + LPASS_INTR_SFR | LPASS_INTR_DMA | LPASS_INTR_I2S); + + regmap_write(lpass->top, SFR_LPASS_INTR_CPU_MASK, + LPASS_INTR_SFR | LPASS_INTR_DMA | LPASS_INTR_I2S); + + /* Activate related PADs from retention state */ + regmap_write(lpass->pmu, EXYNOS5433_PAD_RETENTION_AUD_OPTION, + EXYNOS5433_PAD_INITIATE_WAKEUP_FROM_LOWPWR); + + exynos_lpass_core_sw_reset(lpass, LPASS_I2S_SW_RESET); + exynos_lpass_core_sw_reset(lpass, LPASS_DMA_SW_RESET); + exynos_lpass_core_sw_reset(lpass, LPASS_MEM_SW_RESET); +} + +static void exynos_lpass_disable(struct exynos_lpass *lpass) +{ + /* Mask any unmasked IP interrupt sources */ + regmap_write(lpass->top, SFR_LPASS_INTR_CPU_MASK, 0); + regmap_write(lpass->top, SFR_LPASS_INTR_CA5_MASK, 0); + + /* Deactivate related PADs from retention state */ + regmap_write(lpass->pmu, EXYNOS5433_PAD_RETENTION_AUD_OPTION, 0); +} + +static const struct regmap_config exynos_lpass_reg_conf = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = 0xfc, + .fast_io = true, +}; + +static int exynos_lpass_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct exynos_lpass *lpass; + void __iomem *base_top; + struct resource *res; + + if (!dev->of_node) + return -ENODEV; + + lpass = devm_kzalloc(dev, sizeof(*lpass), GFP_KERNEL); + if (!lpass) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + base_top = devm_ioremap_resource(dev, res); + if (IS_ERR(base_top)) + return PTR_ERR(base_top); + + lpass->top = regmap_init_mmio(dev, base_top, + &exynos_lpass_reg_conf); + if (IS_ERR(lpass->top)) { + dev_err(dev, "LPASS top regmap initialization failed\n"); + return PTR_ERR(lpass->top); + } + + lpass->pmu = syscon_regmap_lookup_by_phandle(dev->of_node, + "samsung,pmu-syscon"); + if (IS_ERR(lpass->pmu)) { + dev_err(dev, "Failed to lookup PMU regmap\n"); + return PTR_ERR(lpass->pmu); + } + + platform_set_drvdata(pdev, lpass); + exynos_lpass_enable(lpass); + + return of_platform_populate(dev->of_node, NULL, NULL, dev); +} + +static int exynos_lpass_suspend(struct device *dev) +{ + struct exynos_lpass *lpass = dev_get_drvdata(dev); + + exynos_lpass_disable(lpass); + return 0; +} + +static int exynos_lpass_resume(struct device *dev) +{ + struct exynos_lpass *lpass = dev_get_drvdata(dev); + + exynos_lpass_enable(lpass); + return 0; +} + +static const struct of_device_id exynos_lpass_of_match[] = { + { .compatible = "samsung,exynos5433-lpass" }, + { }, +}; +MODULE_DEVICE_TABLE(of, exynos_lpass_of_match); + +static SIMPLE_DEV_PM_OPS(lpass_pm_ops, exynos_lpass_suspend, + exynos_lpass_resume); + +static struct platform_driver exynos_lpass_driver = { + .driver = { + .name = "exynos-lpass", + .pm = &lpass_pm_ops, + .of_match_table = exynos_lpass_of_match, + }, + .probe = exynos_lpass_probe, +}; +module_platform_driver(exynos_lpass_driver); + +MODULE_DESCRIPTION("Samsung Low Power Audio Subsystem driver"); +MODULE_LICENSE("GPL v2"); diff --git a/include/linux/mfd/syscon/exynos5-pmu.h b/include/linux/mfd/syscon/exynos5-pmu.h index 76f30f9..c28ff21 100644 --- a/include/linux/mfd/syscon/exynos5-pmu.h +++ b/include/linux/mfd/syscon/exynos5-pmu.h @@ -43,8 +43,10 @@ #define EXYNOS5433_MIPI_PHY2_CONTROL (0x718) #define EXYNOS5_PHY_ENABLE BIT(0) - #define EXYNOS5_MIPI_PHY_S_RESETN BIT(1) #define EXYNOS5_MIPI_PHY_M_RESETN BIT(2) +#define EXYNOS5433_PAD_RETENTION_AUD_OPTION (0x3028) +#define EXYNOS5433_PAD_INITIATE_WAKEUP_FROM_LOWPWR BIT(28) + #endif /* _LINUX_MFD_SYSCON_PMU_EXYNOS5_H_ */ -- 1.9.1

2 7

[alsa-devel] [PATCH] ucm: Add ucm files for DB410c board.
by Riku Voipio 10 Aug '16

10 Aug '16

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> DB410c board has support for both Digital and Analog audio. Digital audio is over HDMI and analog is over codec chip integrated inside the APQ8016 SOC. It can support: - 3 Microphones: Primary Mic(Handset mic), Headset Mic and Secondary - 2 Digital Microphones. - Earpiece. - Headset. - Loud Speaker. - HDMI. [Riku: squashed Srinivas's patches together and converted spaces to tabs] Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Riku Voipio <riku.voipio(a)linaro.org> Cc: nicolas.dechesne(a)linaro.org --- configure.ac | 1 + src/conf/ucm/DB410c/DB410c.conf | 8 ++ src/conf/ucm/DB410c/HDMI | 31 +++++++ src/conf/ucm/DB410c/HiFi | 185 ++++++++++++++++++++++++++++++++++++++++ src/conf/ucm/DB410c/Makefile.am | 4 + src/conf/ucm/Makefile.am | 2 +- 6 files changed, 230 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/DB410c/DB410c.conf create mode 100644 src/conf/ucm/DB410c/HDMI create mode 100644 src/conf/ucm/DB410c/HiFi create mode 100644 src/conf/ucm/DB410c/Makefile.am diff --git a/configure.ac b/configure.ac index f592e8b..014af5f 100644 --- a/configure.ac +++ b/configure.ac @@ -675,6 +675,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/broadwell-rt286/Makefile \ src/conf/ucm/VEYRON-I2S/Makefile \ src/conf/ucm/chtrt5645/Makefile \ + src/conf/ucm/DB410c/Makefile \ src/conf/topology/Makefile \ src/conf/topology/broadwell/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ diff --git a/src/conf/ucm/DB410c/DB410c.conf b/src/conf/ucm/DB410c/DB410c.conf new file mode 100644 index 0000000..590278f --- /dev/null +++ b/src/conf/ucm/DB410c/DB410c.conf @@ -0,0 +1,8 @@ +SectionUseCase."HiFi" { + File "HiFi" + Comment "Play HiFi quality Music." +} +SectionUseCase."HDMI" { + File "HDMI" + Comment "HDMI output." +} diff --git a/src/conf/ucm/DB410c/HDMI b/src/conf/ucm/DB410c/HDMI new file mode 100644 index 0000000..8112bc0 --- /dev/null +++ b/src/conf/ucm/DB410c/HDMI @@ -0,0 +1,31 @@ +# Use case configuration for DB410c board. +# Author: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +SectionVerb { + EnableSequence [ + ] + + DisableSequence [ + ] + Value { + TQ "HiFi" + PlaybackPCM "plughw:0,0" + } +} + +SectionDevice."HDMI-stereo" { + #Name "HDMI-stereo" + Comment "HDMI Digital Stereo Output" + + EnableSequence [ + cdev "hw:0" + ] + + DisableSequence [ + cdev "hw:0" + ] + + Value { + PlaybackChannels "2" + } +} diff --git a/src/conf/ucm/DB410c/HiFi b/src/conf/ucm/DB410c/HiFi new file mode 100644 index 0000000..f9cbcf0 --- /dev/null +++ b/src/conf/ucm/DB410c/HiFi @@ -0,0 +1,185 @@ +# Use case configuration for DB410c board. +# Author: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +SectionVerb { + + EnableSequence [ + ] + + DisableSequence [ + ] + + # ALSA PCM + Value { + # ALSA PCM device for HiFi + PlaybackPCM "plughw:0,1" + CapturePCM "plughw:0,2" + } +} + + +SectionDevice."Speaker" { + Comment "Speaker playback" + + ConflictingDevice [ + "Headphones" + "Earpiece" + ] + + EnableSequence [ + cdev "hw:0" + cset "name='SPK DAC Switch' 1" + cset "name='RX3 MIX1 INP1' RX1" + ## gain to 0dB + cset "name='RX3 Digital Volume' 128" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='SPK DAC Switch' 0" + cset "name='RX3 MIX1 INP1' ZERO" + ] + + Value { + PlaybackChannels "2" + } +} + + +SectionDevice."Headphones" { + Comment "Headphones playback" + + ConflictingDevice [ + "Speaker" + "Earpiece" + ] + EnableSequence [ + cdev "hw:0" + cset "name='RX1 MIX1 INP1' RX1" + cset "name='RX2 MIX1 INP1' RX2" + cset "name='RDAC2 MUX' RX2" + cset "name='HPHL' 1" + cset "name='HPHR' 1" + ## gain to 0dB + cset "name='RX1 Digital Volume' 128" + ## gain to 0dB + cset "name='RX2 Digital Volume' 128" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='RX1 Digital Volume' 0" + cset "name='RX2 Digital Volume' 0" + cset "name='HPHL' 0" + cset "name='HPHR' 0" + cset "name='RDAC2 MUX' ZERO" + cset "name='RX1 MIX1 INP1' ZERO" + cset "name='RX2 MIX1 INP1' ZERO" + ] + + Value { + PlaybackChannels "2" + } +} + +SectionDevice."Earpiece" { + Comment "Earpiece playback" + + ConflictingDevice [ + "Speaker" + "Headphones" + ] + EnableSequence [ + cdev "hw:0" + ] + + DisableSequence [ + cdev "hw:0" + ] + + Value { + PlaybackChannels "2" + } +} + +SectionDevice."Handset" { + Comment "Headset Microphone" + + EnableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ADC2" + cset "name='ADC2 Volume' 8" + cset "name='ADC2 MUX' INP2" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='ADC2 MUX' ZERO" + cset "name='ADC2 Volume' 0" + cset "name='DEC1 MUX' ZERO" + ] + + Value { + CaptureChannels "2" + } +} + +SectionDevice."Primarymic" { + Comment "Primary Microphone" + + EnableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ADC1" + cset "name='ADC1 Volume' 8" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ZERO" + cset "name='ADC1 Volume' 0" + ] + + Value { + CaptureChannels "2" + } +} + +SectionDevice."Secondarymic" { + Comment "Secondary Microphone" + + EnableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ADC2" + cset "name='ADC2 Volume' 8" + cset "name='ADC2 MUX' INP2" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ZERO" + cset "name='ADC2 Volume' 0" + cset "name='ADC2 MUX' ZERO" + ] + + Value { + CaptureChannels "2" + } +} + +SectionDevice."DMIC" { + Comment "Digital Microphone" + + EnableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' DMIC1" + ] + + DisableSequence [ + cdev "hw:0" + cset "name='DEC1 MUX' ZERO" + ] + + Value { + CaptureChannels "2" + } +} diff --git a/src/conf/ucm/DB410c/Makefile.am b/src/conf/ucm/DB410c/Makefile.am new file mode 100644 index 0000000..e10a136 --- /dev/null +++ b/src/conf/ucm/DB410c/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/DB410c +ucm_DATA = DB410c.conf HDMI HiFi +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 8865770..7bf432e 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 VEYRON-I2S chtrt5645 +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 VEYRON-I2S chtrt5645 DB410c -- 2.1.4

2 1

[alsa-devel] [PATCH] sound: soc: atmel: Don't unconditionally reset SSC on stream startup
by Peter Meerwald 10 Aug '16

10 Aug '16

From: Christoph Huber <c.huber(a)bct-electronic.com> commit cbaadf0f90d6 (ASoC: atmel_ssc_dai: refactor the startup and shutdown) refactored code such that the SSC is reset on every startup; this breaks duplex audio (e.g. first start audio playback, then start record, causing the playback to stop/hang) Signed-off-by: Christoph Huber <c.huber(a)bct-electronic.com> Signed-off-by: Peter Meerwald-Stadler <p.meerwald(a)bct-electronic.com> Cc: Songjun Wu <songjun.wu(a)microchip.com> Cc: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 841d059..b3e9f0c 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -298,8 +298,9 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, clk_enable(ssc_p->ssc->clk); ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); - /* Reset the SSC to keep it at a clean status */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Reset the SSC if it is not initialized to keep it at a clean status */ + if(!ssc_p->initialized) + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dir = 0; -- 2.1.4

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[alsa-devel] [PATCH v5 1/2] mfd: Add DT bindings documentation for Samsung Exynos LPASS
by Sylwester Nawrocki 10 Aug '16

10 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 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

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[alsa-devel] Underrun with 2 periods on skylake HDA driver
by Dharageswari R 10 Aug '16

10 Aug '16

Hello, While doing the validation of Skylake HDA driver using aplay and following parameters, we noticed underrun issues. aplay -Dhw:0,0,0 -vv s_48K_16Bit_Sunshine.wav -F 100000 -B 200000 -d 20 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 9600 period_size : 4800 period_time : 100000 tstamp_mode : NONE period_step : 1 avail_min : 4800 period_event : 0 start_threshold : 9600 stop_threshold : 9600 silence_threshold: 0 silence_size : 0 boundary : 5404319552844595200 appl_ptr : 0 hw_ptr : 0 After doing further debugging, what I found is 1.DMA complete interrupt occurs for the first period. 2.Driver reads DMA position in buffer pointer and gets value of 4784. 3.Driver calls snd_pcm_period_elapsed() function to report period elapsed interrupt to application with DMA position = 4784. 4.snd_pcm_lib_write1() is waiting for “avail_min” free space to be available. In this case the avail_min = 4800 and at this interrupt the avail = 4874 and so it does not write any data. 5.DMA complete interrupt occurs for the second period. 6.Driver reads DMA position in the buffer pointer and gets value of 9600. 7.Driver calls snd_pcm_period_elapsed() function to report period elapsed interrupt. 8.Now the snd_pcm_lib_write1() gets avail > avail_min but it’s too late because DMA has already started playing data from first period, which is not yet written by the application. 9.This causes underrun. Logs: 857.870974: hwptr: pcmC0D0p/sub0: IRQ: pos=4800, old=19200, base=19200, period=4800, buf=9600 857.871237: hwptr: pcmC0D0p/sub0: POS: pos=4800, old=24000, base=19200, period=4800, buf=9600 857.970972: hwptr: pcmC0D0p/sub0: IRQ: pos=0, old=24000, base=19200, period=4800, buf=9600 857.971101: hwptr: pcmC0D0p/sub0: POS: pos=0, old=28800, base=28800, period=4800, buf=9600 858.070971: hwptr: pcmC0D0p/sub0: IRQ: pos=4784, old=28800, base=28800, period=4800, buf=9600 858.170968: hwptr: pcmC0D0p/sub0: IRQ: pos=0, old=33584, base=28800, period=4800, buf=9600 858.170988: xrun: pcmC0D0p/sub0: XRUN: old=38400, base=38400, period=4800, buf=9600 858.172240: hwptr: pcmC0D0p/sub0: POS: pos=96, old=0, base=0, period=4800, buf=9600 858.269867: hwptr: pcmC0D0p/sub0: IRQ: pos=4798, old=96, base=0, period=4800, buf=9600 858.369902: hwptr: pcmC0D0p/sub0: IRQ: pos=0, old=4798, base=0, period=4800, buf=9600 I checked with HDA architects and they are saying that when the DMA completion interrupt occurs there can be slight delay before the DMA Position In Buffer Pointer gets updated. So driver is expected to refill data based on whatever content is reflected in the DMA Position In Buffer. What is the right way to fix this issue ? Thanks, Dhara.R --

2 1

[alsa-devel] [PATCH v4 0/5] ALSA - hda: Add support for link audio time reporting
by Vinod Koul 10 Aug '16

10 Aug '16

Skylake onwards we support GTS capability and thus can report audio link time. This series adds support for parsing the capability and then reports link time. The parsing of caps is made common and moved up from extended hda to hdac. The SKL driver and hdac extended core has been updated for this. Mark, The update of the driver needs to update Skl driver, so if you can ACK the 2nd patch, it can go along with extended audio change thru ALSA tree. Changes in v4: - fix code style - use hdac_readl/writel - add cpufeatures header Guneshwor Singh (2): ALSA - hda: Add support for parsing new HDA capabilities ALSA - hda: Add support for link audio time reporting Vinod Koul (3): ALSA: hda - move bus_parse_capabilities to core ALSA: convert users to core bus_parse_capabilities ALSA - Ext hda: remove bus_parse_capabilities include/sound/hda_register.h | 36 ++++++ include/sound/hdaudio.h | 13 +++ include/sound/hdaudio_ext.h | 12 -- sound/hda/ext/hdac_ext_controller.c | 91 ++------------- sound/hda/ext/hdac_ext_stream.c | 46 ++++---- sound/hda/hdac_controller.c | 75 ++++++++++++ sound/pci/hda/hda_controller.c | 203 ++++++++++++++++++++++++++++++++- sound/pci/hda/hda_controller.h | 3 + sound/pci/hda/hda_intel.c | 17 +++ sound/soc/intel/skylake/skl-messages.c | 4 +- sound/soc/intel/skylake/skl-pcm.c | 8 +- sound/soc/intel/skylake/skl.c | 6 +- 12 files changed, 386 insertions(+), 128 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: core: fix shift used for second item in snd_soc_get_enum_double
by jjassal 09 Aug '16

09 Aug '16

Incorrect shift value was being used to extract the second item. Signed-off-by: jjassal <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] [PATCH 00/39] seq: obsolete change of address limit
by Takashi Sakamoto 09 Aug '16

09 Aug '16

Hi, ALSA sequencer core has two types of client; user application and kernel driver. The core allows both types of client to do relevant operations, thus it's required for the core to handle data in both user and kernel spaces. Currently, this is achieved by changing address limit of running task. This is a well-known technique to suppress address check, while it's just a suppression and unfriendly to readers or static code parsers. This patchset obsoletes the usage. In process context, data in user space is once copied to kernel space, then operated and copied to user space. As a result, actual operations for each ioctl command handle data in kernel space, and '__user' qualifier is useless. In this series, patch 1-8 just apply above design. The rest changes each operation following to the design. Therefore, static code parser like sparce generates warnings temporarily in a way to apply these patches. There's a concern of this solution. The data for ioctl is always copied to kernel space or to kernel space even when it's read-only or write-only. But this brings no severe issue as long as I read ALSA sequencer core. (This patchset is a part of my work to introduce EPIPE into ALSA rawmidi core.) Takashi Sakamoto (39): ALSA: seq: add const qualifier to table of functions for ioctl ALSA: seq: apply shorter name for file local functions ALSA: seq: fulfill callback entry for ioctl ALSA: seq: copy ioctl data from user space to kernel stack ALSA: seq: add documentation for snd_seq_kernel_client_ctl ALSA: seq: obsolete address mode in compatibility layer ALSA: seq: obsolete change of address limit in in-kernel path for ioctl ALSA: seq: obsolete address limit helper ALSA: seq: optimize pversion function to new design ALSA: seq: optimize client_id function to new design ALSA: seq: optimize system_info function to new design ALSA: seq: optimize running mode function to new design ALSA: seq: optimize client_info function to new design ALSA: seq: optimize set_client_info function to new design ALSA: seq: optimize create_port function to new design ALSA: seq: optimize delete_port function to new design ALSA: seq: optimize get_port_info function to new design ALSA: seq: optimize seq_port_info function to new design ALSA: seq: optimize subscribe_port function to new design ALSA: seq: optimize unsubscribe_port function to new design ALSA: seq: optimize create_queue function to new design ALSA: seq: optimize delete_queue function to new design ALSA: seq: optimize get_queue_info function to new design ALSA: seq: optimize seq_queue_info function to new design ALSA: seq: optimize get_named_queue function to new design ALSA: seq: optimize get_queue_status function to new design ALSA: seq: optimize get_queue_tempo function to new design ALSA: seq: optimize set_queue_tempo function to new design ALSA: seq: optimize get_queue_timer function to new design ALSA: seq: optimize seq_queue_timer function to new design ALSA: seq: optimize get_queue_client function to new design ALSA: seq: optimize set_queue_client function to new design ALSA: seq: optimize get_client_pool function to new design ALSA: seq: optimize seq_client_pool function to new design ALSA: seq: optimize remove_events function to new design ALSA: seq: optimize get_subscription function to new design ALSA: seq: optimize query_subs function to new design ALSA: seq: optimize query_next_client function to new design ALSA: seq: optimize query_next_port function to new design sound/core/seq/seq_clientmgr.c | 854 +++++++++++++++++++---------------------- sound/core/seq/seq_compat.c | 26 +- 2 files changed, 407 insertions(+), 473 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v3 0/5] ALSA - hda: Add support for link audio time reporting
by Vinod Koul 09 Aug '16

09 Aug '16

Skylake onwards we support GTS capability and thus can report audio link time. This series adds support for parsing the capability and then reports link time. The parsing of caps is made common and moved up from extended hda to hdac. The SKL driver and hdac extended core has been updated for this. Mark, The update of the driver needs to update Skl driver, so if you can ACK the 2nd patch, it can go along with extended audio change thru ALSA tree. Changes in v3: moved extended hda to hdac update SKL drv to use new hdac caps update patchset for HDA drv to use new caps parsing Guneshwor Singh (2): ALSA - hda: Add support for parsing new HDA capabilities ALSA - hda: Add support for link audio time reporting Vinod Koul (3): ALSA: hda - move bus_parse_capabilities to core ALSA: convert users to core bus_parse_capabilities ALSA - Ext hda: remove bus_parse_capabilities include/sound/hda_register.h | 36 ++++++ include/sound/hdaudio.h | 13 +++ include/sound/hdaudio_ext.h | 12 -- sound/hda/ext/hdac_ext_controller.c | 91 ++------------- sound/hda/ext/hdac_ext_stream.c | 46 ++++---- sound/hda/hdac_controller.c | 76 +++++++++++++ sound/pci/hda/hda_controller.c | 200 ++++++++++++++++++++++++++++++++- sound/pci/hda/hda_controller.h | 3 + sound/pci/hda/hda_intel.c | 16 +++ sound/soc/intel/skylake/skl-messages.c | 4 +- sound/soc/intel/skylake/skl-pcm.c | 8 +- sound/soc/intel/skylake/skl.c | 6 +- 12 files changed, 383 insertions(+), 128 deletions(-) -- 1.9.1

3 15

[alsa-devel] [PATCH 0/3] ASoC: remove snd_soc_pcm_set/get_drvdata()
by Kuninori Morimoto 09 Aug '16

09 Aug '16

Hi Mark snd_soc_pcm_set_drvdata() will set driver data to rtd->dev, but driver data of rtd->dev is already used as "rtd" on soc_post_component_init(). static int soc_post_component_init(xxx) { ... dev_set_drvdata(rtd->dev, rtd); ... } But, ASoC has snd_soc_pcm_set/get_drvdata() to set rtd->dev. It will overwrite existing data and will break system. This patch tries to solve this issue. Intel haswell and qcom is using these functions. ** Note ** I don't have these boards, thus, I did compile test only. Kuninori Morimoto (3): ASoC: sst-haswell-pcm: don't use snd_soc_pcm_set_drvdata() ASoC: lpass-platform: don't use snd_soc_pcm_set_drvdata() ASoC: remove snd_soc_pcm_set/get_drvdata() include/sound/soc.h | 11 ----------- sound/soc/intel/haswell/sst-haswell-pcm.c | 1 - sound/soc/qcom/lpass-platform.c | 14 +++++++------- sound/soc/qcom/lpass.h | 2 ++ 4 files changed, 9 insertions(+), 19 deletions(-)

2 7

[alsa-devel] [PATCH 0/6] ASoC: simple-card: add init_dai / canonicalize_dailink
by Kuninori Morimoto 09 Aug '16

09 Aug '16

Hi Mark These add init_dai / canonicalize_dailink to simple utiles. These are part of simple-card cleanup Kuninori Morimoto (6): ASoC: simple-card-utils: add asoc_simple_card_init_dai() ASoC: simple-card: use asoc_simple_card_init_dai() ASoC: rsrc-card: use asoc_simple_card_init_dai() ASoC: simple-card-utils: add asoc_simple_card_canonicalize_dailink() ASoC: simple-card: use asoc_simple_card_canonicalize_dailink() ASoC: rsrc-card: use asoc_simple_card_canonicalize_dailink() include/sound/simple_card_utils.h | 5 +++++ sound/soc/generic/simple-card-utils.c | 42 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 47 +++++------------------------------------------ sound/soc/sh/rcar/rsrc-card.c | 32 +++++--------------------------- 4 files changed, 57 insertions(+), 69 deletions(-) -- 1.9.1

2 10

[alsa-devel] [PATCH 0/2] alsabat: test scripts and reference amixer config files
by focus.luo＠linux.intel.com 09 Aug '16

09 Aug '16

From: Focus Luo <focus.luo(a)linux.intel.com> The patch set supplies some automated test scripts and reference amixer config files for linux audio driver testing by using alsa-bat as test tool on Intel platforms. The first patch includes automated test scripts for linux audio driver based on alsa-lib interface by using alsabat as test tool. It supports analog audio and display audio(HDMI/DP) interface. The second patch supplies some amixer reference config file on Intel platforms. Focus Luo (2): alsabat: automation test scripts alsabat: add amixer config files bat/tests/README | 32 ++ bat/tests/alsabat_main.sh | 178 ++++++++++ bat/tests/analog_audio_playback_and_capture.sh | 84 +++++ bat/tests/asound_state/asound.state.Broadwell | 439 +++++++++++++++++++++++++ bat/tests/asound_state/asound.state.Haswell | 342 +++++++++++++++++++ bat/tests/asound_state/asound.state.Skylake | 437 ++++++++++++++++++++++++ bat/tests/dp_audio_playback.sh | 83 +++++ bat/tests/dp_audio_subdevice_number.sh | 68 ++++ bat/tests/hdmi_audio_playback.sh | 81 +++++ bat/tests/hdmi_audio_subdevice_number.sh | 65 ++++ bat/tests/map_test_case | 8 + 11 files changed, 1817 insertions(+) create mode 100644 bat/tests/README create mode 100755 bat/tests/alsabat_main.sh create mode 100755 bat/tests/analog_audio_playback_and_capture.sh create mode 100644 bat/tests/asound_state/asound.state.Broadwell create mode 100644 bat/tests/asound_state/asound.state.Haswell create mode 100644 bat/tests/asound_state/asound.state.Skylake create mode 100755 bat/tests/dp_audio_playback.sh create mode 100755 bat/tests/dp_audio_subdevice_number.sh create mode 100755 bat/tests/hdmi_audio_playback.sh create mode 100755 bat/tests/hdmi_audio_subdevice_number.sh create mode 100644 bat/tests/map_test_case -- 1.9.1

5 8

[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for ELP HD USB Camera
by Vittorio Gambaletta (VittGam) 09 Aug '16

09 Aug '16

The ELP HD USB Camera (05a3:9420) needs this quirk for suppressing the unsupported sample rate inquiry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481 Cc: <stable(a)vger.kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Vittorio Gambaletta <linuxbugs(a)vittgam.net> --- --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1138,6 +1138,7 @@ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ + case USB_ID(0x05A3, 0x9420): /* ELP HD USB Camera */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x1de7, 0x0013): /* Phoenix Audio MT202exe */ case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */

2 1

[alsa-devel] [PATCH] ASoC: simple-card: fix module load "unknown symbol" issue
by Kevin Hilman 09 Aug '16

09 Aug '16

An ASoC driver using simple-card, when built as a module gets the following error at module load time: [ 23.571873] simple_card_utils: Unknown symbol snd_soc_of_parse_daifmt (err 0) [ 23.712726] simple_card_utils: Unknown symbol snd_soc_of_parse_card_name (err 0) fix this by including simple-card-utils.o symbols in the module build its symols are resolved correctly. Tested on Davinci DA850-EVM using snd_soc_edma, snc_soc_davinci_mcasp, with all of ASoC built as modules. Cc: Sekhar Nori <nsekhar(a)ti.com> Signed-off-by: Kevin Hilman <khilman(a)baylibre.com> --- sound/soc/generic/Makefile | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 45602ca8536e..ff0c55eee4f7 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -2,4 +2,8 @@ obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) := simple-card-utils.o snd-soc-simple-card-objs := simple-card.o +ifneq ($(CONFIG_SND_SIMPLE_CARD_UTILS),) +snd-soc-simple-card-objs += simple-card-utils.o +endif + obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o -- 2.9.0

4 9

[alsa-devel] [PATCH] ASoC: nau8810: Add driver for Nuvoton codec chip NAU88C10
by John Hsu 09 Aug '16

09 Aug '16

The driver is for codec NAU88C10 of Nuvoton Technology Corporation. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8810.txt | 16 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8810.c | 833 +++++++++++++++++++++ sound/soc/codecs/nau8810.h | 272 +++++++ 5 files changed, 1128 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/nau8810.txt create mode 100755 sound/soc/codecs/nau8810.c create mode 100755 sound/soc/codecs/nau8810.h diff --git a/Documentation/devicetree/bindings/sound/nau8810.txt b/Documentation/devicetree/bindings/sound/nau8810.txt new file mode 100755 index 0000000..05830e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8810.txt @@ -0,0 +1,16 @@ +NAU8810 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "nuvoton,nau8810" + + - reg : the I2C address of the device. + +Example: + +codec: nau8810@1a { + compatible = "nuvoton,nau8810"; + reg = <0x1a>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb57f8f..bbd7324 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -91,6 +91,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C @@ -1046,6 +1047,10 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8810 + tristate "Nuvoton Technology Corporation NAU88C10 CODEC" + depends on I2C + config SND_SOC_NAU8825 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 550a174..3c15bc5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -84,6 +84,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8810-objs := nau8810.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o snd-soc-pcm1681-objs := pcm1681.o @@ -301,6 +302,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c new file mode 100755 index 0000000..d03e9c76 --- /dev/null +++ b/sound/soc/codecs/nau8810.c @@ -0,0 +1,833 @@ +/* + * nau8810.c -- NAU8810 ALSA Soc Audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * + * Author: David Lin <ctlin0(a)nuvoton.com> + * + * Based on WM8974.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "nau8810.h" + + +static const int nau8810_mclk_scaler[] = { 10, 15, 20, 30, 40, 60, 80, 120 }; + +static const struct reg_default nau8810_reg_defaults[] = { + { NAU8810_REG_POWER1, 0x0000 }, + { NAU8810_REG_POWER2, 0x0000 }, + { NAU8810_REG_POWER3, 0x0000 }, + { NAU8810_REG_IFACE, 0x0050 }, + { NAU8810_REG_COMP, 0x0000 }, + { NAU8810_REG_CLOCK, 0x0140 }, + { NAU8810_REG_SMPLR, 0x0000 }, + { NAU8810_REG_DAC, 0x0000 }, + { NAU8810_REG_DACGAIN, 0x00FF }, + { NAU8810_REG_ADC, 0x0100 }, + { NAU8810_REG_ADCGAIN, 0x00FF }, + { NAU8810_REG_EQ1, 0x012C }, + { NAU8810_REG_EQ2, 0x002C }, + { NAU8810_REG_EQ3, 0x002C }, + { NAU8810_REG_EQ4, 0x002C }, + { NAU8810_REG_EQ5, 0x002C }, + { NAU8810_REG_DACLIM1, 0x0032 }, + { NAU8810_REG_DACLIM2, 0x0000 }, + { NAU8810_REG_NOTCH1, 0x0000 }, + { NAU8810_REG_NOTCH2, 0x0000 }, + { NAU8810_REG_NOTCH3, 0x0000 }, + { NAU8810_REG_NOTCH4, 0x0000 }, + { NAU8810_REG_ALC1, 0x0038 }, + { NAU8810_REG_ALC2, 0x000B }, + { NAU8810_REG_ALC3, 0x0032 }, + { NAU8810_REG_NOISEGATE, 0x0000 }, + { NAU8810_REG_PLLN, 0x0008 }, + { NAU8810_REG_PLLK1, 0x000C }, + { NAU8810_REG_PLLK2, 0x0093 }, + { NAU8810_REG_PLLK3, 0x00E9 }, + { NAU8810_REG_ATTEN, 0x0000 }, + { NAU8810_REG_INPUT_SIGNAL, 0x0003 }, + { NAU8810_REG_PGAGAIN, 0x0010 }, + { NAU8810_REG_ADCBOOST, 0x0100 }, + { NAU8810_REG_OUTPUT, 0x0002 }, + { NAU8810_REG_SPKMIX, 0x0001 }, + { NAU8810_REG_SPKGAIN, 0x0039 }, + { NAU8810_REG_MONOMIX, 0x0001 }, + { NAU8810_REG_POWER4, 0x0000 }, + { NAU8810_REG_TSLOTCTL1, 0x0000 }, + { NAU8810_REG_TSLOTCTL2, 0x0020 }, + { NAU8810_REG_DEVICE_REVID, 0x00EF }, + { NAU8810_REG_I2C_DEVICEID, 0x001A }, + { NAU8810_REG_ADDITIONID, 0x00CA }, + { NAU8810_REG_RESERVE, 0x0124 }, + { NAU8810_REG_OUTCTL, 0x0001 }, + { NAU8810_REG_ALC1ENHAN1, 0x0000 }, + { NAU8810_REG_ALC1ENHAN2, 0x0039 }, + { NAU8810_REG_MISCCTL, 0x0000 }, + { NAU8810_REG_OUTTIEOFF, 0x0000 }, + { NAU8810_REG_AGCP2POUT, 0x0000 }, + { NAU8810_REG_AGCPOUT, 0x0000 }, + { NAU8810_REG_AMTCTL, 0x0000 }, + { NAU8810_REG_OUTTIEOFFMAN, 0x0000 }, +}; + +static bool nau8810_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET ... NAU8810_REG_SMPLR: + case NAU8810_REG_DAC ... NAU8810_REG_DACGAIN: + case NAU8810_REG_ADC ... NAU8810_REG_ADCGAIN: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_DACLIM1 ... NAU8810_REG_DACLIM2: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + case NAU8810_REG_ALC1 ... NAU8810_REG_ATTEN: + case NAU8810_REG_INPUT_SIGNAL ... NAU8810_REG_PGAGAIN: + case NAU8810_REG_ADCBOOST: + case NAU8810_REG_OUTPUT ... NAU8810_REG_SPKMIX: + case NAU8810_REG_SPKGAIN: + case NAU8810_REG_MONOMIX: + case NAU8810_REG_POWER4 ... NAU8810_REG_TSLOTCTL2: + case NAU8810_REG_DEVICE_REVID ... NAU8810_REG_RESERVE: + case NAU8810_REG_ALC1ENHAN1 ... NAU8810_REG_ALC1ENHAN2: + case NAU8810_REG_MISCCTL: + case NAU8810_REG_OUTTIEOFF ... NAU8810_REG_OUTTIEOFFMAN: + return true; + default: + return false; + } +} + +static bool nau8810_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET ... NAU8810_REG_SMPLR: + case NAU8810_REG_DAC ... NAU8810_REG_DACGAIN: + case NAU8810_REG_ADC ... NAU8810_REG_ADCGAIN: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_DACLIM1 ... NAU8810_REG_DACLIM2: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + case NAU8810_REG_ALC1 ... NAU8810_REG_ATTEN: + case NAU8810_REG_INPUT_SIGNAL ... NAU8810_REG_PGAGAIN: + case NAU8810_REG_ADCBOOST: + case NAU8810_REG_OUTPUT ... NAU8810_REG_SPKMIX: + case NAU8810_REG_SPKGAIN: + case NAU8810_REG_MONOMIX: + case NAU8810_REG_POWER4 ... NAU8810_REG_TSLOTCTL2: + case NAU8810_REG_ALC1ENHAN1 ... NAU8810_REG_ALC1ENHAN2: + case NAU8810_REG_MISCCTL: + case NAU8810_REG_OUTTIEOFF ... NAU8810_REG_OUTTIEOFFMAN: + return true; + default: + return false; + } +} + +static bool nau8810_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8810_REG_RESET: + case NAU8810_REG_EQ1 ... NAU8810_REG_EQ5: + case NAU8810_REG_NOTCH1 ... NAU8810_REG_NOTCH4: + return true; + default: + return false; + } +} + +static int nau8810_reg_write(void *context, unsigned int reg, + unsigned int value) +{ + struct i2c_client *client = context; + uint8_t buf[2]; + __be16 *out = (void *)buf; + int ret; + + *out = cpu_to_be16((reg << 9) | value); + ret = i2c_master_send(client, buf, sizeof(buf)); + if (ret == sizeof(buf)) + return 0; + else if (ret < 0) + return ret; + else + return -EIO; +} + +static int nau8810_reg_read(void *context, unsigned int reg, + unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + uint8_t reg_buf; + uint16_t val_buf; + int ret; + + reg_buf = (uint8_t)(reg << 1); + xfer[0].addr = client->addr; + xfer[0].len = sizeof(reg_buf); + xfer[0].buf = &reg_buf; + xfer[0].flags = 0; + + xfer[1].addr = client->addr; + xfer[1].len = sizeof(val_buf); + xfer[1].buf = (uint8_t *)&val_buf; + xfer[1].flags = I2C_M_RD; + + ret = i2c_transfer(client->adapter, xfer, ARRAY_SIZE(xfer)); + if (ret < 0) + return ret; + else if (ret != ARRAY_SIZE(xfer)) + return -EIO; + + *value = be16_to_cpu(val_buf); + + return 0; +} + +static const char * const nau8810_companding[] = { + "Off", "NC", "u-law", "A-law" +}; +static const char * const nau8810_deemp[] = { + "None", "32kHz", "44.1kHz", "48kHz" +}; +static const char * const nau8810_eqmode[] = {"Capture", "Playback" }; +static const char * const nau8810_bw[] = {"Narrow", "Wide" }; +static const char * const nau8810_eq1[] = { + "80Hz", "105Hz", "135Hz", "175Hz" +}; +static const char * const nau8810_eq2[] = { + "230Hz", "300Hz", "385Hz", "500Hz" +}; +static const char * const nau8810_eq3[] = { + "650Hz", "850Hz", "1.1kHz", "1.4kHz" +}; +static const char * const nau8810_eq4[] = { + "1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" +}; +static const char * const nau8810_eq5[] = { + "5.3kHz", "6.9kHz", "9kHz", "11.7kHz" +}; +static const char * const nau8810_alc[] = {"Normal", "Limiter" }; + +static const struct soc_enum nau8810_enum[] = { + SOC_ENUM_SINGLE(NAU8810_REG_COMP, NAU8810_ADCCM_SFT, + ARRAY_SIZE(nau8810_companding), nau8810_companding), + SOC_ENUM_SINGLE(NAU8810_REG_COMP, NAU8810_DACCM_SFT, + ARRAY_SIZE(nau8810_companding), nau8810_companding), + SOC_ENUM_SINGLE(NAU8810_REG_DAC, NAU8810_DEEMP_SFT, + ARRAY_SIZE(nau8810_deemp), nau8810_deemp), + SOC_ENUM_SINGLE(NAU8810_REG_EQ1, NAU8810_EQM_SFT, + ARRAY_SIZE(nau8810_eqmode), nau8810_eqmode), + + SOC_ENUM_SINGLE(NAU8810_REG_EQ1, NAU8810_EQ1CF_SFT, + ARRAY_SIZE(nau8810_eq1), nau8810_eq1), + SOC_ENUM_SINGLE(NAU8810_REG_EQ2, NAU8810_EQ2BW_SFT, + ARRAY_SIZE(nau8810_bw), nau8810_bw), + SOC_ENUM_SINGLE(NAU8810_REG_EQ2, NAU8810_EQ2CF_SFT, + ARRAY_SIZE(nau8810_eq2), nau8810_eq2), + SOC_ENUM_SINGLE(NAU8810_REG_EQ3, NAU8810_EQ3BW_SFT, + ARRAY_SIZE(nau8810_bw), nau8810_bw), + + SOC_ENUM_SINGLE(NAU8810_REG_EQ3, NAU8810_EQ3CF_SFT, + ARRAY_SIZE(nau8810_eq3), nau8810_eq3), + SOC_ENUM_SINGLE(NAU8810_REG_EQ4, NAU8810_EQ4BW_SFT, + ARRAY_SIZE(nau8810_bw), nau8810_bw), + SOC_ENUM_SINGLE(NAU8810_REG_EQ4, NAU8810_EQ4CF_SFT, + ARRAY_SIZE(nau8810_eq4), nau8810_eq4), + SOC_ENUM_SINGLE(NAU8810_REG_EQ5, NAU8810_EQ5CF_SFT, + ARRAY_SIZE(nau8810_eq5), nau8810_eq5), + + SOC_ENUM_SINGLE(NAU8810_REG_ALC3, NAU8810_ALCM_SFT, + ARRAY_SIZE(nau8810_alc), nau8810_alc), +}; + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); + +static const struct snd_kcontrol_new nau8810_snd_controls[] = { + + SOC_ENUM("ADC Companding", nau8810_enum[0]), + SOC_ENUM("DAC Companding", nau8810_enum[1]), + SOC_ENUM("DAC De-emphasis", nau8810_enum[2]), + SOC_ENUM("EQ Function", nau8810_enum[3]), + SOC_ENUM("EQ1 Cut Off", nau8810_enum[4]), + SOC_ENUM("EQ2 Bandwidth", nau8810_enum[5]), + SOC_ENUM("EQ2 Cut Off", nau8810_enum[6]), + SOC_ENUM("EQ3 Bandwidth", nau8810_enum[7]), + SOC_ENUM("EQ3 Cut Off", nau8810_enum[8]), + SOC_ENUM("EQ4 Bandwidth", nau8810_enum[9]), + SOC_ENUM("EQ4 Cut Off", nau8810_enum[10]), + SOC_ENUM("EQ5 Cut Off", nau8810_enum[11]), + SOC_ENUM("ALC Mode", nau8810_enum[12]), + + SOC_SINGLE("Digital Loopback Switch", NAU8810_REG_COMP, + NAU8810_ADDAP_SFT, 1, 0), + + SOC_SINGLE("DAC Inversion Switch", NAU8810_REG_DAC, + NAU8810_DACPL_SFT, 1, 0), + SOC_SINGLE_TLV("Playback Gain", NAU8810_REG_DACGAIN, + NAU8810_DACGAIN_SFT, 0xff, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", NAU8810_REG_ADC, + NAU8810_HPFEN_SFT, 1, 0), + SOC_SINGLE("High Pass Cut Off", NAU8810_REG_ADC, + NAU8810_HPF_SFT, 0x7, 0), + + SOC_SINGLE("ADC Inversion Switch", NAU8810_REG_ADC, + NAU8810_ADCPL_SFT, 1, 0), + SOC_SINGLE_TLV("Capture Gain", NAU8810_REG_ADCGAIN, + NAU8810_ADCGAIN_SFT, 0xff, 0, digital_tlv), + + SOC_SINGLE_TLV("EQ1 Gain", NAU8810_REG_EQ1, + NAU8810_EQ1GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ2 Gain", NAU8810_REG_EQ2, + NAU8810_EQ2GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ3 Gain", NAU8810_REG_EQ3, + NAU8810_EQ3GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ4 Gain", NAU8810_REG_EQ4, + NAU8810_EQ4GC_SFT, 0x18, 1, eq_tlv), + SOC_SINGLE_TLV("EQ5 Gain", NAU8810_REG_EQ5, + NAU8810_EQ5GC_SFT, 0x18, 1, eq_tlv), + + SOC_SINGLE("DAC Limiter Switch", NAU8810_REG_DACLIM1, + NAU8810_DACLIMEN_SFT, 1, 0), + SOC_SINGLE("DAC Limiter Decay", NAU8810_REG_DACLIM1, + NAU8810_DACLIMDCY_SFT, 0xf, 0), + SOC_SINGLE("DAC Limiter Attack", NAU8810_REG_DACLIM1, + NAU8810_DACLIMATK_SFT, 0xf, 0), + + SOC_SINGLE("DAC Limiter Threshold", NAU8810_REG_DACLIM2, + NAU8810_DACLIMTHL_SFT, 0x7, 0), + SOC_SINGLE("DAC Limiter Boost", NAU8810_REG_DACLIM2, + NAU8810_DACLIMBST_SFT, 0xf, 0), + + SOC_SINGLE("ALC Enable Switch", NAU8810_REG_ALC1, + NAU8810_ALCEN_SFT, 1, 0), + SOC_SINGLE("ALC Max Gain", NAU8810_REG_ALC1, + NAU8810_ALCMXGAIN_SFT, 0x7, 0), + SOC_SINGLE("ALC Min Gain", NAU8810_REG_ALC1, + NAU8810_ALCMINGAIN_SFT, 0x7, 0), + + SOC_SINGLE("ALC ZC Switch", NAU8810_REG_ALC2, + NAU8810_ALCZC_SFT, 1, 0), + SOC_SINGLE("ALC Hold", NAU8810_REG_ALC2, + NAU8810_ALCHT_SFT, 0xf, 0), + SOC_SINGLE("ALC Target", NAU8810_REG_ALC2, + NAU8810_ALCSL_SFT, 0xf, 0), + + SOC_SINGLE("ALC Decay", NAU8810_REG_ALC3, + NAU8810_ALCDCY_SFT, 0xf, 0), + SOC_SINGLE("ALC Attack", NAU8810_REG_ALC3, + NAU8810_ALCATK_SFT, 0xf, 0), + + SOC_SINGLE("ALC Noise Gate Switch", NAU8810_REG_NOISEGATE, + NAU8810_ALCNEN_SFT, 1, 0), + SOC_SINGLE("ALC Noise Gate Threshold", NAU8810_REG_NOISEGATE, + NAU8810_ALCNTH_SFT, 0x7, 0), + + SOC_SINGLE("PGA ZC Switch", NAU8810_REG_PGAGAIN, + NAU8810_PGAZC_SFT, 1, 0), + SOC_SINGLE_TLV("PGA Volume", NAU8810_REG_PGAGAIN, + NAU8810_PGAGAIN_SFT, 0x3f, 0, inpga_tlv), + + SOC_SINGLE("Speaker ZC Switch", NAU8810_REG_SPKGAIN, + NAU8810_SPKZC_SFT, 1, 0), + SOC_SINGLE("Speaker Mute Switch", NAU8810_REG_SPKGAIN, + NAU8810_SPKMT_SFT, 1, 0), + SOC_SINGLE_TLV("Speaker Volume", NAU8810_REG_SPKGAIN, + NAU8810_SPKGAIN_SFT, 0x3f, 0, spk_tlv), + + SOC_SINGLE("Capture Boost(+20dB)", NAU8810_REG_ADCBOOST, + NAU8810_PGABST_SFT, 1, 0), + SOC_SINGLE("Mono Mute Switch", NAU8810_REG_MONOMIX, + NAU8810_MOUTMXMT_SFT, 1, 0), + + SOC_SINGLE("DAC Oversampling Rate(128x) Switch", NAU8810_REG_DAC, + NAU8810_DACOS_SFT, 1, 0), + SOC_SINGLE("ADC Oversampling Rate(128x) Switch", NAU8810_REG_ADC, + NAU8810_ADCOS_SFT, 1, 0), +}; + +/* Speaker Output Mixer */ +static const struct snd_kcontrol_new nau8810_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_SPKMIX, + NAU8810_BYPSPK_SFT, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_SPKMIX, + NAU8810_DACSPK_SFT, 1, 0), +}; + +/* Mono Output Mixer */ +static const struct snd_kcontrol_new nau8810_mono_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", NAU8810_REG_MONOMIX, + NAU8810_BYPMOUT_SFT, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", NAU8810_REG_MONOMIX, + NAU8810_DACMOUT_SFT, 1, 0), +}; + +/* PGA Mute */ +static const struct snd_kcontrol_new nau8810_inpga_mute[] = { + SOC_DAPM_SINGLE("PGA Mute Switch", NAU8810_REG_PGAGAIN, + NAU8810_PGAMT_SFT, 1, 0), +}; + +/* Input PGA */ +static const struct snd_kcontrol_new nau8810_inpga[] = { + SOC_DAPM_SINGLE("MicN Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_NMICPGA_SFT, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", NAU8810_REG_INPUT_SIGNAL, + NAU8810_PMICPGA_SFT, 1, 0), +}; + +/* Mic Input boost vol */ +static const struct snd_kcontrol_new nau8810_mic_boost_controls = + SOC_DAPM_SINGLE("Mic Volume", NAU8810_REG_ADCBOOST, + NAU8810_PMICBSTGAIN_SFT, 0x7, 0); + +static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Speaker Mixer", NAU8810_REG_POWER3, + NAU8810_SPKMX_EN_SFT, 0, &nau8810_speaker_mixer_controls[0], + ARRAY_SIZE(nau8810_speaker_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", NAU8810_REG_POWER3, + NAU8810_MOUTMX_EN_SFT, 0, &nau8810_mono_mixer_controls[0], + ARRAY_SIZE(nau8810_mono_mixer_controls)), + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", NAU8810_REG_POWER3, + NAU8810_DAC_EN_SFT, 0), + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", NAU8810_REG_POWER2, + NAU8810_ADC_EN_SFT, 0), + SND_SOC_DAPM_PGA("SpkN Out", NAU8810_REG_POWER3, + NAU8810_NSPK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SpkP Out", NAU8810_REG_POWER3, + NAU8810_PSPK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", NAU8810_REG_POWER3, + NAU8810_MOUT_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Input PGA", NAU8810_REG_POWER2, + NAU8810_PGA_EN_SFT, 0, nau8810_inpga, + ARRAY_SIZE(nau8810_inpga)), + SND_SOC_DAPM_MIXER("Input Boost Stage", NAU8810_REG_POWER2, + NAU8810_BST_EN_SFT, 0, nau8810_inpga_mute, + ARRAY_SIZE(nau8810_inpga_mute)), + + SND_SOC_DAPM_SUPPLY("Mic Bias", NAU8810_REG_POWER1, + NAU8810_MICBIAS_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("MICN"), + SND_SOC_DAPM_INPUT("MICP"), + SND_SOC_DAPM_OUTPUT("MONOOUT"), + SND_SOC_DAPM_OUTPUT("SPKOUTP"), + SND_SOC_DAPM_OUTPUT("SPKOUTN"), +}; + +static const struct snd_soc_dapm_route nau8810_dapm_routes[] = { + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Line Bypass Switch", "Input Boost Stage"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Line Bypass Switch", "Input Boost Stage"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Input Boost Stage */ + {"ADC", NULL, "Input Boost Stage"}, + {"Input Boost Stage", NULL, "Input PGA"}, + {"Input Boost Stage", NULL, "MICP"}, + + /* Input PGA */ + {"Input PGA", "MicN Switch", "MICN"}, + {"Input PGA", "MicP Switch", "MICP"}, +}; + +static int nau8810_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + + nau8810->sysclk = freq; + dev_dbg(nau8810->dev, "master sysclk %dHz\n", nau8810->sysclk); + + return 0; +} + +static int nau8810_config_clkdiv(struct nau8810 *nau8810, int div, int rate) +{ + struct regmap *regmap = nau8810->regmap; + int i, sclk, imclk; + + switch (nau8810->div_id) { + case NAU8810_MCLK_DIV_PLL: + /* master clock from PLL and enable PLL */ + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_MCLKSEL_MASK, (div << NAU8810_MCLKSEL_SFT)); + regmap_update_bits(regmap, NAU8810_REG_POWER1, + NAU8810_PLL_EN, NAU8810_PLL_EN); + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_PLL); + break; + + case NAU8810_MCLK_DIV_MCLK: + /* Configure the master clock prescaler div to make system + * clock to approximate the internal master clock (IMCLK); + * and large or equal to IMCLK. + */ + div = 0; + imclk = rate * 256; + for (i = 1; i < ARRAY_SIZE(nau8810_mclk_scaler); i++) { + sclk = (nau8810->sysclk * 10) / + nau8810_mclk_scaler[i]; + if (sclk < imclk) + break; + div = i; + } + dev_dbg(nau8810->dev, + "master clock prescaler %x for fs %d\n", div, rate); + + /* master clock from MCLK and disable PLL */ + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_MCLKSEL_MASK, (div << NAU8810_MCLKSEL_SFT)); + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_CLKM_MASK, NAU8810_CLKM_MCLK); + regmap_update_bits(regmap, NAU8810_REG_POWER1, + NAU8810_PLL_EN, 0); + break; + + case NAU8810_BCLK_DIV: + regmap_update_bits(regmap, NAU8810_REG_CLOCK, + NAU8810_BCLKSEL_MASK, (div << NAU8810_BCLKSEL_SFT)); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int nau8810_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + nau8810->div_id = div_id; + if (div_id != NAU8810_MCLK_DIV_MCLK) + /* Defer the master clock prescaler configuration to DAI + * hardware parameter if master clock from MCLK because + * it needs runtime fs information to get the proper div. + */ + ret = nau8810_config_clkdiv(nau8810, div, 0); + + return ret; +} + +static int nau8810_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + u16 ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8810_CLKIO_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8810_AIFMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8810_AIFMT_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8810_AIFMT_PCM_A; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1_val |= NAU8810_BCLKP_IB | NAU8810_FSP_IF; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8810_BCLKP_IB; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1_val |= NAU8810_FSP_IF; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8810->regmap, NAU8810_REG_IFACE, + NAU8810_AIFMT_MASK | NAU8810_FSP_IF | + NAU8810_BCLKP_IB, ctrl1_val); + regmap_update_bits(nau8810->regmap, NAU8810_REG_CLOCK, + NAU8810_CLKIO_MASK, ctrl2_val); + + return 0; +} + +static int nau8810_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + int val_len = 0, val_rate = 0; + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= NAU8810_WLEN_20; + break; + case 24: + val_len |= NAU8810_WLEN_24; + break; + case 32: + val_len |= NAU8810_WLEN_32; + break; + } + + switch (params_rate(params)) { + case 8000: + val_rate |= NAU8810_SMPLR_8K; + break; + case 11025: + val_rate |= NAU8810_SMPLR_12K; + break; + case 16000: + val_rate |= NAU8810_SMPLR_16K; + break; + case 22050: + val_rate |= NAU8810_SMPLR_24K; + break; + case 32000: + val_rate |= NAU8810_SMPLR_32K; + break; + case 44100: + case 48000: + break; + } + + regmap_update_bits(nau8810->regmap, NAU8810_REG_IFACE, + NAU8810_WLEN_MASK, val_len); + regmap_update_bits(nau8810->regmap, NAU8810_REG_SMPLR, + NAU8810_SMPLR_MASK, val_rate); + + /* If the master clock is from MCLK, provide the runtime FS for driver + * to get the master clock prescaler configuration. + */ + if (nau8810->div_id == NAU8810_MCLK_DIV_MCLK) + nau8810_config_clkdiv(nau8810, 0, params_rate(params)); + + return 0; +} + +static int nau8810_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + struct regmap *map = nau8810->regmap; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_80K); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_IOBUF_EN | NAU8810_ABIAS_EN, + NAU8810_IOBUF_EN | NAU8810_ABIAS_EN); + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + regcache_sync(map); + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_3K); + mdelay(100); + } + regmap_update_bits(map, NAU8810_REG_POWER1, + NAU8810_REFIMP_MASK, NAU8810_REFIMP_300K); + break; + + case SND_SOC_BIAS_OFF: + regmap_write(map, NAU8810_REG_POWER1, 0); + regmap_write(map, NAU8810_REG_POWER2, 0); + regmap_write(map, NAU8810_REG_POWER3, 0); + break; + } + + return 0; +} + + +#define NAU8810_RATES (SNDRV_PCM_RATE_8000_48000) + +#define NAU8810_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8810_ops = { + .hw_params = nau8810_pcm_hw_params, + .set_fmt = nau8810_set_dai_fmt, + .set_sysclk = nau8810_set_sysclk, + .set_clkdiv = nau8810_set_clkdiv, +}; + +static struct snd_soc_dai_driver nau8810_dai = { + .name = "nau8810-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = NAU8810_RATES, + .formats = NAU8810_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = NAU8810_RATES, + .formats = NAU8810_FORMATS, + }, + .ops = &nau8810_ops, + .symmetric_rates = 1, +}; + +static const struct regmap_config nau8810_regmap_config = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = NAU8810_REG_MAX, + .readable_reg = nau8810_readable_reg, + .writeable_reg = nau8810_writeable_reg, + .volatile_reg = nau8810_volatile_reg, + .reg_read = nau8810_reg_read, + .reg_write = nau8810_reg_write, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8810_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8810_reg_defaults), +}; + +static int nau8810_probe(struct snd_soc_codec *codec) +{ + struct nau8810 *nau8810 = snd_soc_codec_get_drvdata(codec); + + regmap_write(nau8810->regmap, NAU8810_REG_RESET, 0x00); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_nau8810 = { + .probe = nau8810_probe, + .set_bias_level = nau8810_set_bias_level, + .suspend_bias_off = true, + + .controls = nau8810_snd_controls, + .num_controls = ARRAY_SIZE(nau8810_snd_controls), + .dapm_widgets = nau8810_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8810_dapm_widgets), + .dapm_routes = nau8810_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8810_dapm_routes), +}; + +static int nau8810_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8810 *nau8810 = dev_get_platdata(dev); + + if (!nau8810) { + nau8810 = devm_kzalloc(dev, sizeof(*nau8810), GFP_KERNEL); + if (!nau8810) + return -ENOMEM; + } + i2c_set_clientdata(i2c, nau8810); + + nau8810->regmap = devm_regmap_init(dev, NULL, + i2c, &nau8810_regmap_config); + if (IS_ERR(nau8810->regmap)) + return PTR_ERR(nau8810->regmap); + + nau8810->dev = dev; + + return snd_soc_register_codec(dev, + &soc_codec_dev_nau8810, &nau8810_dai, 1); +} + +static int nau8810_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id nau8810_i2c_id[] = { + { "nau8810", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8810_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id nau8810_of_match[] = { + { .compatible = "nuvoton,nau8810", }, + { } +}; +MODULE_DEVICE_TABLE(of, nau8810_of_match); +#endif + +static struct i2c_driver nau8810_i2c_driver = { + .driver = { + .name = "nau8810", + .of_match_table = of_match_ptr(nau8810_of_match), + }, + .probe = nau8810_i2c_probe, + .remove = nau8810_i2c_remove, + .id_table = nau8810_i2c_id, +}; + +module_i2c_driver(nau8810_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8810 driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8810.h b/sound/soc/codecs/nau8810.h new file mode 100755 index 0000000..174d26e --- /dev/null +++ b/sound/soc/codecs/nau8810.h @@ -0,0 +1,272 @@ +/* + * NAU8810 ALSA SoC audio driver + * + * Copyright 2016 Nuvoton Technology Corp. + * Author: David Lin <ctlin0(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8810_H__ +#define __NAU8810_H__ + +#define NAU8810_REG_RESET 0x00 +#define NAU8810_REG_POWER1 0x01 +#define NAU8810_REG_POWER2 0x02 +#define NAU8810_REG_POWER3 0x03 +#define NAU8810_REG_IFACE 0x04 +#define NAU8810_REG_COMP 0x05 +#define NAU8810_REG_CLOCK 0x06 +#define NAU8810_REG_SMPLR 0x07 +#define NAU8810_REG_DAC 0x0A +#define NAU8810_REG_DACGAIN 0x0B +#define NAU8810_REG_ADC 0x0E +#define NAU8810_REG_ADCGAIN 0x0F +#define NAU8810_REG_EQ1 0x12 +#define NAU8810_REG_EQ2 0x13 +#define NAU8810_REG_EQ3 0x14 +#define NAU8810_REG_EQ4 0x15 +#define NAU8810_REG_EQ5 0x16 +#define NAU8810_REG_DACLIM1 0x18 +#define NAU8810_REG_DACLIM2 0x19 +#define NAU8810_REG_NOTCH1 0x1B +#define NAU8810_REG_NOTCH2 0x1C +#define NAU8810_REG_NOTCH3 0x1D +#define NAU8810_REG_NOTCH4 0x1E +#define NAU8810_REG_ALC1 0x20 +#define NAU8810_REG_ALC2 0x21 +#define NAU8810_REG_ALC3 0x22 +#define NAU8810_REG_NOISEGATE 0x23 +#define NAU8810_REG_PLLN 0x24 +#define NAU8810_REG_PLLK1 0x25 +#define NAU8810_REG_PLLK2 0x26 +#define NAU8810_REG_PLLK3 0x27 +#define NAU8810_REG_ATTEN 0x28 +#define NAU8810_REG_INPUT_SIGNAL 0x2C +#define NAU8810_REG_PGAGAIN 0x2D +#define NAU8810_REG_ADCBOOST 0x2F +#define NAU8810_REG_OUTPUT 0x31 +#define NAU8810_REG_SPKMIX 0x32 +#define NAU8810_REG_SPKGAIN 0x36 +#define NAU8810_REG_MONOMIX 0x38 +#define NAU8810_REG_POWER4 0x3A +#define NAU8810_REG_TSLOTCTL1 0x3B +#define NAU8810_REG_TSLOTCTL2 0x3C +#define NAU8810_REG_DEVICE_REVID 0x3E +#define NAU8810_REG_I2C_DEVICEID 0x3F +#define NAU8810_REG_ADDITIONID 0x40 +#define NAU8810_REG_RESERVE 0x41 +#define NAU8810_REG_OUTCTL 0x45 +#define NAU8810_REG_ALC1ENHAN1 0x46 +#define NAU8810_REG_ALC1ENHAN2 0x47 +#define NAU8810_REG_MISCCTL 0x49 +#define NAU8810_REG_OUTTIEOFF 0x4B +#define NAU8810_REG_AGCP2POUT 0x4C +#define NAU8810_REG_AGCPOUT 0x4D +#define NAU8810_REG_AMTCTL 0x4E +#define NAU8810_REG_OUTTIEOFFMAN 0x4F +#define NAU8810_REG_MAX NAU8810_REG_OUTTIEOFFMAN + + +/* NAU8810_REG_POWER1 (0x1) */ +#define NAU8810_DCBUF_EN (0x1 << 8) +#define NAU8810_PLL_EN (0x1 << 5) +#define NAU8810_MICBIAS_EN_SFT 4 +#define NAU8810_ABIAS_EN (0x1 << 3) +#define NAU8810_IOBUF_EN (0x1 << 2) +#define NAU8810_REFIMP_MASK 0x3 +#define NAU8810_REFIMP_DIS 0x0 +#define NAU8810_REFIMP_80K 0x1 +#define NAU8810_REFIMP_300K 0x2 +#define NAU8810_REFIMP_3K 0x3 + +/* NAU8810_REG_POWER2 (0x2) */ +#define NAU8810_BST_EN_SFT 4 +#define NAU8810_PGA_EN_SFT 2 +#define NAU8810_ADC_EN_SFT 0 + +/* NAU8810_REG_POWER3 (0x3) */ +#define NAU8810_DAC_EN_SFT 0 +#define NAU8810_SPKMX_EN_SFT 2 +#define NAU8810_MOUTMX_EN_SFT 3 +#define NAU8810_PSPK_EN_SFT 5 +#define NAU8810_NSPK_EN_SFT 6 +#define NAU8810_MOUT_EN_SFT 7 + +/* NAU8810_REG_IFACE (0x4) */ +#define NAU8810_AIFMT_SFT 3 +#define NAU8810_AIFMT_MASK (0x3 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_RIGHT (0x0 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_LEFT (0x1 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_I2S (0x2 << NAU8810_AIFMT_SFT) +#define NAU8810_AIFMT_PCM_A (0x3 << NAU8810_AIFMT_SFT) +#define NAU8810_WLEN_SFT 5 +#define NAU8810_WLEN_MASK (0x3 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_16 (0x0 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_20 (0x1 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_24 (0x2 << NAU8810_WLEN_SFT) +#define NAU8810_WLEN_32 (0x3 << NAU8810_WLEN_SFT) +#define NAU8810_FSP_IF (0x1 << 7) +#define NAU8810_BCLKP_IB (0x1 << 8) + +/* NAU8810_REG_COMP (0x5) */ +#define NAU8810_ADDAP_SFT 0 +#define NAU8810_ADCCM_SFT 1 +#define NAU8810_DACCM_SFT 3 + +/* NAU8810_REG_CLOCK (0x6) */ +#define NAU8810_CLKIO_MASK 0x1 +#define NAU8810_CLKIO_SLAVE 0x0 +#define NAU8810_CLKIO_MASTER 0x1 +#define NAU8810_BCLKSEL_SFT 2 +#define NAU8810_BCLKSEL_MASK (0x7 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_1 (0x0 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_2 (0x1 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_4 (0x2 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_8 (0x3 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_16 (0x4 << NAU8810_BCLKSEL_SFT) +#define NAU8810_BCLKDIV_32 (0x5 << NAU8810_BCLKSEL_SFT) +#define NAU8810_MCLKSEL_SFT 5 +#define NAU8810_MCLKSEL_MASK (0x7 << NAU8810_MCLKSEL_SFT) +#define NAU8810_CLKM_SFT 8 +#define NAU8810_CLKM_MASK (0x1 << NAU8810_CLKM_SFT) +#define NAU8810_CLKM_MCLK (0x0 << NAU8810_CLKM_SFT) +#define NAU8810_CLKM_PLL (0x1 << NAU8810_CLKM_SFT) + +/* NAU8810_REG_SMPLR (0x7) */ +#define NAU8810_SMPLR_SFT 1 +#define NAU8810_SMPLR_MASK (0x7 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_48K (0x0 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_32K (0x1 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_24K (0x2 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_16K (0x3 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_12K (0x4 << NAU8810_SMPLR_SFT) +#define NAU8810_SMPLR_8K (0x5 << NAU8810_SMPLR_SFT) + +/* NAU8810_REG_DAC (0xA) */ +#define NAU8810_DACPL_SFT 0 +#define NAU8810_DACOS_SFT 3 +#define NAU8810_DEEMP_SFT 4 + +/* NAU8810_REG_DACGAIN (0xB) */ +#define NAU8810_DACGAIN_SFT 0 + +/* NAU8810_REG_ADC (0xE) */ +#define NAU8810_ADCPL_SFT 0 +#define NAU8810_ADCOS_SFT 3 +#define NAU8810_HPF_SFT 4 +#define NAU8810_HPFEN_SFT 8 + +/* NAU8810_REG_ADCGAIN (0xF) */ +#define NAU8810_ADCGAIN_SFT 0 + +/* NAU8810_REG_EQ1 (0x12) */ +#define NAU8810_EQ1GC_SFT 0 +#define NAU8810_EQ1CF_SFT 5 +#define NAU8810_EQM_SFT 8 + +/* NAU8810_REG_EQ2 (0x13) */ +#define NAU8810_EQ2GC_SFT 0 +#define NAU8810_EQ2CF_SFT 5 +#define NAU8810_EQ2BW_SFT 8 + +/* NAU8810_REG_EQ3 (0x14) */ +#define NAU8810_EQ3GC_SFT 0 +#define NAU8810_EQ3CF_SFT 5 +#define NAU8810_EQ3BW_SFT 8 + +/* NAU8810_REG_EQ4 (0x15) */ +#define NAU8810_EQ4GC_SFT 0 +#define NAU8810_EQ4CF_SFT 5 +#define NAU8810_EQ4BW_SFT 8 + +/* NAU8810_REG_EQ5 (0x16) */ +#define NAU8810_EQ5GC_SFT 0 +#define NAU8810_EQ5CF_SFT 5 + +/* NAU8810_REG_DACLIM1 (0x18) */ +#define NAU8810_DACLIMATK_SFT 0 +#define NAU8810_DACLIMDCY_SFT 4 +#define NAU8810_DACLIMEN_SFT 8 + +/* NAU8810_REG_DACLIM2 (0x19) */ +#define NAU8810_DACLIMBST_SFT 0 +#define NAU8810_DACLIMTHL_SFT 4 + +/* NAU8810_REG_ALC1 (0x20) */ +#define NAU8810_ALCMINGAIN_SFT 0 +#define NAU8810_ALCMXGAIN_SFT 3 +#define NAU8810_ALCEN_SFT 8 + +/* NAU8810_REG_ALC2 (0x21) */ +#define NAU8810_ALCSL_SFT 0 +#define NAU8810_ALCHT_SFT 4 +#define NAU8810_ALCZC_SFT 8 + +/* NAU8810_REG_ALC3 (0x22) */ +#define NAU8810_ALCATK_SFT 0 +#define NAU8810_ALCDCY_SFT 4 +#define NAU8810_ALCM_SFT 8 + +/* NAU8810_REG_NOISEGATE (0x23) */ +#define NAU8810_ALCNTH_SFT 0 +#define NAU8810_ALCNEN_SFT 3 + +/* NAU8810_REG_PLLN (0x24) */ +#define NAU8810_PLLN_MASK 0xF +#define NAU8810_PLLMCLK_DIV2 (0x1 << 4) + +/* NAU8810_REG_PLLK1 (0x25) */ +#define NAU8810_PLLK1_MASK 0x3F + +/* NAU8810_REG_PLLK2 (0x26) */ +#define NAU8810_PLLK2_MASK 0x1FF + +/* NAU8810_REG_PLLK3 (0x27) */ +#define NAU8810_PLLK3_MASK 0x1FF + +/* NAU8810_REG_INPUT_SIGNAL (0x2C) */ +#define NAU8810_PMICPGA_SFT 0 +#define NAU8810_NMICPGA_SFT 1 + +/* NAU8810_REG_PGAGAIN (0x2D) */ +#define NAU8810_PGAGAIN_SFT 0 +#define NAU8810_PGAMT_SFT 6 +#define NAU8810_PGAZC_SFT 7 + +/* NAU8810_REG_ADCBOOST (0x2F) */ +#define NAU8810_PMICBSTGAIN_SFT 4 +#define NAU8810_PGABST_SFT 8 + +/* NAU8810_REG_SPKMIX (0x32) */ +#define NAU8810_DACSPK_SFT 0 +#define NAU8810_BYPSPK_SFT 1 + +/* NAU8810_REG_SPKGAIN (0x36) */ +#define NAU8810_SPKGAIN_SFT 0 +#define NAU8810_SPKMT_SFT 6 +#define NAU8810_SPKZC_SFT 7 + +/* NAU8810_REG_MONOMIX (0x38) */ +#define NAU8810_DACMOUT_SFT 0 +#define NAU8810_BYPMOUT_SFT 1 +#define NAU8810_MOUTMXMT_SFT 6 + + +/* Clock divider Id's */ +enum { + NAU8810_MCLK_DIV_PLL, + NAU8810_MCLK_DIV_MCLK, + NAU8810_BCLK_DIV, +}; + +struct nau8810 { + struct device *dev; + struct regmap *regmap; + int sysclk; + int div_id; +}; + +#endif -- 2.6.4

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[alsa-devel] [PATCH 00/27 v2] ALSA: seq: obsolete change of address limit
by Takashi Sakamoto 08 Aug '16

08 Aug '16

Hi, For purpose of this patchset, please refer to my previous post: [alsa-devel] obsolete change of address limit http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/110937.html Changes in v2: - Use '_IOC_SIZE' macro to calculate the size of argument of ioctl command. - Keep redundant longer name. - Improve comments. - Merge some patches relevant to the same features. I give up using 'dir' field in each ioctl command to decide copying, because some ioctl commands for ALSA sequencer core includes bugs against actual implementations. Takashi Sakamoto (27): ALSA: seq: add const qualifier to table of functions for ioctl ALSA: seq: fulfill callback entry for ioctl ALSA: seq: copy ioctl data from user space to kernel stack ALSA: seq: add documentation for snd_seq_kernel_client_ctl ALSA: seq: obsolete address mode in compatibility layer ALSA: seq: obsolete change of address limit in in-kernel path for ioctl ALSA: seq: obsolete address limit helper ALSA: seq: optimize pversion function to new design ALSA: seq: optimize client_id function to new design ALSA: seq: optimize system_info function to new design ALSA: seq: optimize running mode function to new design ALSA: seq: optimize get/set_client_info function to new design ALSA: seq: optimize create/delete_port function to new design ALSA: seq: optimize get/set_port_info function to new design ALSA: seq: optimize subscribe_port function to new design ALSA: seq: optimize create/delete_queue function to new design ALSA: seq: optimize get/set_queue_info function to new design ALSA: seq: optimize get/set_queue_info function to new design ALSA: seq: optimize get_named_queue function to new design ALSA: seq: optimize get_queue_status function to new design ALSA: seq: optimize get/set_queue_tempo function to new design ALSA: seq: optimize get/set_queue_timer function to new design ALSA: seq: optimize get/set_queue_client function to new design ALSA: seq: optimize get/set_client_pool function to new design ALSA: seq: optimize remove_events function to new design ALSA: seq: optimize get_subscription function to new design ALSA: seq: optimize query functions for subscribes, next client and port to new design sound/core/seq/seq_clientmgr.c | 750 +++++++++++++++++++---------------------- sound/core/seq/seq_compat.c | 26 +- 2 files changed, 353 insertions(+), 423 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 0/2] ASoC: da7218: Small clocking related updates for driver
by Adam Thomson 08 Aug '16

08 Aug '16

This patch set contains two updates relating to clocking. The first removes the 32KHz PLL feature as this is no longer supported, and the second provides MCLK effciency improvements to avoid unnecessary MCLK enabling. Changes are based on v4.8-rc1 Linux kernel. Adam Thomson (2): ASoC: da7218: Remove 32KHz PLL mode from driver ASoC: da7218: Improve driver efficiency with regards to MCLK usage sound/soc/codecs/da7218.c | 34 +++++++++++++++------------------- sound/soc/codecs/da7218.h | 2 -- 2 files changed, 15 insertions(+), 21 deletions(-) -- 1.9.3

1 2

[alsa-devel] [PATCH 0/3] drm/i2c: tda998x ASoC hdmi-codec support + BeagleBoneBlack audio support
by Jyri Sarha 08 Aug '16

08 Aug '16

These patches are just a rebase of the same patches from my "[PATCH v4 0/7] Implement generic ASoC HDMI codec and use it in tda998x" -patch series. I have only made updates required by efc9194bcff8 ("ASoC: hdmi-codec: callback function will be called with private data") -patch. The first patch changes tda998x pdata, so if there are any out of tree users of tda998x-driver the out of tree code needs to be updated. 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 | 276 ++++++++++++++++++--- include/drm/i2c/tda998x.h | 24 +- include/dt-bindings/display/tda998x.h | 7 + 6 files changed, 342 insertions(+), 55 deletions(-) create mode 100644 include/dt-bindings/display/tda998x.h -- 1.9.1

3 15

[alsa-devel] [PATCH 000/159] ASoC: codec cleanup - codec duplicate functions
by Kuninori Morimoto 08 Aug '16

08 Aug '16

Hi Mark These are codec duplicate functions cleanup patches - v1 We would like to switch to "component" style base ASoC in the future instead of current "CPU/Codec/Platform/Card" style. Current "component" already has callback functions, and "codec" is based on "component", but "codec" is still requesting/using its original callback. This is bacause of historical reason, but this is the time to cleanup. Lars's opinion is that this kind of cleanup can be final stage, but I think these are needed for "component" base style switching These are very big patch-set, but it doesn't add new features. [001/159] - [158/159] move callback functions from codec to component, and last [159/159] will remove codec side callback. We need this kind of cleanup for .probe/.remove too, and on Platform too. This cares codec duplicate callback function only as 1st step. I will post codec .probe/.remove cleanup if these are accepted, and will post platform side cleanup too. I build-tested these patches with make allyesconfig on x86, and tested on Lager board. Best regards --- Kuninori Morimoto

3 294

[alsa-devel] How to post cleanup patches ?
by Kuninori Morimoto 08 Aug '16

08 Aug '16

Hi Mark Now codec driver and component driver has duplicated callback function, and snd_soc_register_codec() just copied it to component side. int snd_soc_register_codec(xxx) { ... if (codec_drv->controls) { codec->component.controls = codec_drv->controls; codec->component.num_controls = codec_drv->num_controls; } if (codec_drv->dapm_widgets) { codec->component.dapm_widgets = codec_drv->dapm_widgets; codec->component.num_dapm_widgets = codec_drv->num_dapm_widgets; } if (codec_drv->dapm_routes) { codec->component.dapm_routes = codec_drv->dapm_routes; codec->component.num_dapm_routes = codec_drv->num_dapm_routes; } ... } I think we can cleanup this duplicated function, similar things happen on .probe, .remove, and platform side too. I would like to post cleanup patches for these. As 1st step, remove codec duplicate callback. But my concern is that you hate big-patch-set, and it is almost 70 patches. Can I post these patches to ML ? or should I use git pull request ? or I shouldn't post ? I will post main patches as sample patch Best regards --- Kuninori Morimoto

3 9

[alsa-devel] [PATCH 0/7][REBASE] ASoC: simple-card: add parse_clk / parse_dai
by Kuninori Morimoto 08 Aug '16

08 Aug '16

Hi Mark These are rebase version patches for parse_clk / parse_dai on simple utiles. I think today's topic/simple is missing [1/7] patch, but have [2/7] [3/7]. Because of this, it has bulid error, and you can't apply [4/7]. These patches add new functions on utils.c, and I noticed that it will have conflict with fix/simple branch. Because of these reasons, I think you want to redo topic/simple branch (?). These patches are based on fix/simple branch. Kuninori Morimoto (7): ASoC: simple-card-utils: add asoc_simple_card_parse_clk() ASoC: simple-card: use asoc_simple_card_parse_clk() ASoC: rsrc-card: use asoc_simple_card_parse_clk() ASoC: simple-card-utils: add asoc_simple_card_parse_dai() ASoC: simple-card: use asoc_simple_card_parse_dai() ASoC: rsrc-card: use asoc_simple_card_parse_dai() ASoC: simple-card: remove asoc_simple_card_sub_parse_of() include/sound/simple_card_utils.h | 25 ++++++++ sound/soc/generic/simple-card-utils.c | 69 ++++++++++++++++++++ sound/soc/generic/simple-card.c | 116 ++++++++++------------------------ sound/soc/sh/rcar/rsrc-card.c | 93 ++++++--------------------- 4 files changed, 146 insertions(+), 157 deletions(-) -- 1.9.1

2 10

[alsa-devel] [PATCH 0/7] ASoC: simple-card: add parse_clk / parse_dai
by Kuninori Morimoto 08 Aug '16

08 Aug '16

Hi Mark These add parse_clk / parse_dai to simple utiles. I know that we want to use "devm_" version of of_clk_get(), and now I'm posting its patch to kernel ML, but it still under discussion. Thus, this patch is using current style (with small fixup). I will replace it to devm version if it was accepted. Kuninori Morimoto (7): ASoC: simple-card-utils: add asoc_simple_card_parse_clk() ASoC: simple-card: use asoc_simple_card_parse_clk() ASoC: rsrc-card: use asoc_simple_card_parse_clk() ASoC: simple-card-utils: add asoc_simple_card_parse_dai() ASoC: simple-card: use asoc_simple_card_parse_dai() ASoC: rsrc-card: use asoc_simple_card_parse_dai() ASoC: simple-card: remove asoc_simple_card_sub_parse_of() include/sound/simple_card_utils.h | 25 ++++++++ sound/soc/generic/simple-card-utils.c | 69 ++++++++++++++++++++ sound/soc/generic/simple-card.c | 116 ++++++++++------------------------ sound/soc/sh/rcar/rsrc-card.c | 93 ++++++--------------------- 4 files changed, 146 insertions(+), 157 deletions(-) -- 1.9.1

2 14

[alsa-devel] [PATCH 0/4] ASoC: da7213: Device clocking updates and fixes
by Adam Thomson 08 Aug '16

08 Aug '16

This patch set contains a series of patches relating to device clocking. The changes are as follows: 1) Set correct default BCLKs per WCLK to achieve all DAI formats. Currently only 16-bit formats would operate correctly. 2) Only use MCLK as required, when an audio stream is active. Can be disabled all other times. 3) Refactoring of sysclk() and pll() functions to make them cleaner, removing need for unnecessary private flags, and resolve incorrect 32Khz mode configuration. 4) Improve 32KHz PLL locking with some small register updates when configuring PLL and when DAI is enabled. Patches are based on v4.7 Linux kernel Adam Thomson (4): ASoC: da7213: Default to 64 BCLKs per WCLK to support all formats ASoC: da7213: Improve driver efficiency with regards to MCLK usage ASoC: da7213: Refactor sysclk(), pll() functions to improve handling ASoC: da7213: Improve 32KHz mode PLL locking sound/soc/codecs/da7213.c | 134 +++++++++++++++++++++++++++++----------------- sound/soc/codecs/da7213.h | 12 +++-- 2 files changed, 94 insertions(+), 52 deletions(-) -- 1.9.3

3 10

[alsa-devel] Audible Pop noise with CS4271 codec
by Vishal Thanki 08 Aug '16

08 Aug '16

Hi all, On a custom built am335x based board with CS4271 codec, I am observing the audible pop/click sound during playback start and playback stop (using aplay). The pop sound is observed with kernel 4.0 as well as on kernel 4.4. I have not tried the recent kernels but since there is no change in the CS4271 codec driver, I am assuming the problem can be reproduced in the latest kernel as well. I have experimented a bit with conditional "soft reset" (enable_soft_reset flag) which is added in driver and which seems to have little influence in pop sound. My experiment is like following: 1) Start the board with "soft reset" enabled in CS4271 driver 2) Play sine wave tone for 5 seconds using aplay --> The pop sound is present during starting and stopping of the playback 3) Play the silence file (no audio data - filled with zero) using aplay --> The pop sound is present only during starting of the playback 4) Play the silence file again using aplay --> The pop sound is present only during starting of the playback Now disable the "soft reset" in driver and 1) Play sine wave tone for 5 seconds using aplay --> The pop sound is present during starting and stopping of the playback 2) Play the silence file using aplay --> The pop sound is present only during starting of the playback 3) Play the silence file again using aplay --> The pop sound is gone 4) Play the silence file successively multiple times --> The pop sound not observed 5) Now play the sine wave tone again --> The pop sound is present during startup and stop. 6) Now play the silent file again --> The pop sound is present during startup, and no pop sound during stop. The driver handles the muting and unmuting of DAC in mute_stream DAI callback which looks okay to me. So I have added delays in mute_stream callback after "unmute" of DACs and noticed that unmuting of DAC is not causing the pop noise, but it is heard as soon as the audio data starts flowing (after .trigger() ?). I could also not figure out what problem the "soft reset" is addressing and how it is influencing the pop noise in successive silence file playback. Any help in further debugging would be greatly appreciated. Thanks, Vishal

1 0

[alsa-devel] [PATCH] alsa-python: alsaseq: added binding for delete_port
by Maurizio Berti 07 Aug '16

07 Aug '16

Hello, I was going to send this patch via git, but I've troubles setting it up against gmail, then I hope I'm not doing something wrong by posting this directly here. I found out that pyalsa doesn't provide delete_port, then I added it; please, note, I'm not a programmer, I just do this for fun and I actually have no experience in C programming: I just understand (most of) it by reading the code, so I just used some knowledge and logic to add this binding. So far, it looks fine to me and works without any problem. Signed-off-by: MaurizioB <maurizio.berti(a)gmail.com> diff --git a/pyalsa/alsaseq.c b/pyalsa/alsaseq.c index ce332ed..2ad4d55 100644 --- a/pyalsa/alsaseq.c +++ b/pyalsa/alsaseq.c @@ -2343,6 +2343,44 @@ Sequencer_create_simple_port(SequencerObject *self, return PyInt_FromLong(port); } + +/** alsaseq.Sequencer delete_port() method: __doc__ */ +PyDoc_STRVAR(Sequencer_delete_port__doc__, + "delete_port(port_id) -> int" + "\n" + "Delete an existing port.\n" + "\n" + "Parameters:\n" + " port_id -- id of the port\n" + "Raises:\n" + " SequencerError: if ALSA can't delete the port" +); + + +/** alsaseq.Sequencer delete_port() method */ +static PyObject * +Sequencer_delete_port(SequencerObject *self, + PyObject *args, + PyObject *kwds) { + int port_id = 0; + int ret; + char *kwlist[] = { "port_id", NULL}; + + if (!PyArg_ParseTupleAndKeywords(args, kwds, "|i", kwlist, + &port_id)) { + return NULL; + } + + ret = snd_seq_delete_port(self->handle, port_id); + if (ret < 0) { + RAISESND(ret, "Failed to delete port"); + return NULL; + } + + Py_RETURN_NONE; +} + + /** alsaseq.Sequencer connection_list() method: __doc__ */ PyDoc_STRVAR(Sequencer_connection_list__doc__, "connection_list() -> list\n" @@ -3248,6 +3286,10 @@ static PyMethodDef Sequencer_methods[] = { (PyCFunction) Sequencer_create_simple_port, METH_VARARGS | METH_KEYWORDS, Sequencer_create_simple_port__doc__ }, + {"delete_port", + (PyCFunction) Sequencer_delete_port, + METH_VARARGS | METH_KEYWORDS, + Sequencer_delete_port__doc__ }, {"connection_list", (PyCFunction) Sequencer_connection_list, METH_VARARGS, -- 2.4.10 -- È difficile avere una convinzione precisa quando si parla delle ragioni del cuore. - "Sostiene Pereira", Antonio Tabucchi http://www.jidesk.net

1 0

[alsa-devel] Alsa Application using the "default" plugin
by Hauke Krüger 07 Aug '16

07 Aug '16

Hi everyone, I have created an ALSA audio application which works fine for any kind of hardware device (all devices addressed by "hw:n,m"). Today, I planned to implement the functionality to also address plugins. My test case is the "default" plugin. When using this, I see significant differences: 1) Function "snd_pcm_hw_params_any" returns a value of 1. For hardware devices it returns 0. There is no documentation of the return type from that function. What does the 1 say? 2) I want to simultaneously capture and playback. Therefore I call "snd_pcm_link". This function fails with the error saying that this function is not implemented. Is that the intended behavior and how would I synchronize input and output? What I do to fix this is to start input and output ("snd_pcm_start") independently if "snd_pcm_link" fails. 3) The program stops working on the second call to "snd_pcm_writei": The function call simply never returns. I guess that the pcm output interface does not run at this moment and the output buffer has been completely filled before. All other functions calls work as expected but the described behavior prevents the application from working properly. And - to repeat - the same code works for all hardware soundcards "hw:n,m" which I tested. What is the recommended way to address the "default" plugin pcm device? Best regards Hauke

1 0

[alsa-devel] ASoC: sun4i-i2s: Restricting the MCLK
by Thomas Niederprüm 05 Aug '16

05 Aug '16

I am using the new sun4i-i2s driver in linux-next to run a sta326 codec via the pin header on a cubietruck (this involves some soldering as described in [1]). The codec can only be configured as slave and needs a master clock for its internal PLL. It is therefore great that, unlike many other SOCs, the a20 provides a MCLK that can be used to sync the codec. However, in my case there is a restriction imposed on the MCLK by the codec. As stated in the datasheet [2], the master clock should be at maximum 12.288 MHz/11.2896 MHz or otherwise the codec will not work (confirmed by experiment!). Since the mod clock of the a20's dai block is running at twice this frequency, I need to assure that the MLCK divider is at least 2. Actually it is rather straight forward to achieve this by changing the breaking condition in the divider calculation, as shown by the following one-line patch: diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 687a8f8..8920844 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -206,7 +206,7 @@ static int sun4i_i2s_set_clk_rate(struct sun4i_i2s *i2s, clk_rate, rate); - if ((bclk_div >= 0) && (mclk_div >= 0)) + if ((bclk_div >= 0) && (mclk_div >= 1)) break; } Of course, this is just a hack to fix my use case and I am wondering what would be the best way to handle my corner case more generally? I see one option in introducing a DT property mlck_mindiv, or similar, that defaults to 1 and can be overwritten at the board level. I'm willing to prepare a proper patch to tackle this problem but right now I'm wondering if there a major objections to this approach or whether there is any better solution to the problem. Best regards, Thomas [1] https://hifiduino.wordpress.com/2014/03/07/cubieboard-for-audio/ [2] www.st.com/resource/en/datasheet/sta326.pdf

2 2

[alsa-devel] [PATCH v4 1/4] mfd: Add DT bindings documentation for Samsung Exynos LPASS
by Sylwester Nawrocki 05 Aug '16

05 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> --- There were few rather significant additions since v3 and I had to drop tags: Acked-by: Rob Herring <robh(a)kernel.org> Acked-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> 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 | 21 +++++++++++++++++++++ 1 file changed, 21 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..7e97c0d --- /dev/null +++ b/Documentation/devicetree/bindings/mfd/samsung,exynos5433-lpass.txt @@ -0,0 +1,21 @@ + +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, WDT 0...1 +devices. + +Bindings of the sub-nodes are described in: + Documentation/devicetree/bindings/serial/samsung_uart.txt + Documentation/devicetree/bindings/sound/samsung-i2s.txt + Documentation/devicetree/bindings/dma/arm-pl330.txt -- 1.9.1

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[alsa-devel] Stale data interleaved in capture driver
by Rob Nertney 05 Aug '16

05 Aug '16

My DMA is configured with a debug mode in which it sends an a simple increasing counter value in the format I wish to receive: an IRQ every 32Bytes (half-frame). it transfers 32KBytes of data, and then restarts from it's original offset. This is tested in bare-metal mode as working. I wrote my handler as described in the high-frequency interrupts described here: http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ch05s07.html My data is valid until 1/4 buffer full, then it interleaves stale (lower numbers) in, replacing the desired numbers. It's sporadic, and it appears to repeat, as if ALSA never returns to the 0-offset of dma_area. I've dumped a snip below. If the formatting is lost, I can post a google drive or an image somewhere. Thanks, Rob 00004000: 8182 8183 8184 8185 00004008: 8186 8187 8188 8189 00004010: 8190 8191 8192 1 00004018: 2 3 4 5 00004020: 6 7 8 8201 00004028: 8202 8203 8204 8205 00004030: 8206 8207 8208 8209 00004038: 8210 8211 8212 8213 00004040: 8214 8215 8216 8217 00004048: 8218 8219 8220 8221 00004050: 8222 8223 8224 8225 00004058: 8226 8227 8228 8229 00004060: 8230 8231 8232 8233 00004068: 8234 8235 8236 8237 00004070: 8238 8239 8240 49 00004078: 50 51 52 53 00004080: 54 55 56 8249 00004088: 8250 8251 8252 8253 00004090: 8254 8255 8256 65 00004098: 66 67 68 69 000040a0: 70 71 72 73 000040a8: 74 75 76 77 000040b0: 78 79 80 8273 000040b8: 8274 8275 8276 8277 000040c0: 8278 8279 8280 8281 000040c8: 8282 8283 8284 8285 000040d0: 8286 8287 8288 8289 000040d8: 8290 8291 8292 8293 000040e0: 8294 8295 8296 8297 000040e8: 8298 8299 8300 8301 000040f0: 8302 8303 8304 8305 000040f8: 8306 8307 8308 8309 00004100: 8310 8311 8312 121

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[alsa-devel] [GIT PULL] Skylake firmware update
by Vinod Koul 05 Aug '16

05 Aug '16

Hi, Please PULL to recive updated Skylake audio firmware. The following changes since commit 6bc2c609988823744d1a45d33ee370c8310ec305: amdgpu: update stoney UVD firmware (2016-07-29 14:40:03 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to c170c8d95794d6aedbaeea44674daaa96baf04f7: linux-firmware: intel: Update Skylake audio firmware (2016-08-04 16:09:21 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Update Skylake audio firmware WHENCE | 4 ++-- intel/dsp_fw_release.bin | 2 +- intel/dsp_fw_release_v958.bin | Bin 0 -> 221184 bytes 3 files changed, 3 insertions(+), 3 deletions(-) create mode 100644 intel/dsp_fw_release_v958.bin Thanks -- ~Vinod

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[alsa-devel] [asoc:topic/simple 20/20] sound/soc/sh/rcar/rsrc-card.c:193:9: error: implicit declaration of function 'asoc_simple_card_parse_clk_cpu'
by kbuild test robot 05 Aug '16

05 Aug '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/simple head: c9a235da8a61cf9af7653fca341ded0881a64eca commit: c9a235da8a61cf9af7653fca341ded0881a64eca [20/20] ASoC: rsrc-card: use asoc_simple_card_parse_clk() config: arm-shmobile_defconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 5.4.0-6) 5.4.0 20160609 reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c9a235da8a61cf9af7653fca341ded0881a64eca # save the attached .config to linux build tree make.cross ARCH=arm All errors (new ones prefixed by >>): sound/soc/sh/rcar/rsrc-card.c: In function 'rsrc_card_parse_links': >> sound/soc/sh/rcar/rsrc-card.c:193:9: error: implicit declaration of function 'asoc_simple_card_parse_clk_cpu' [-Werror=implicit-function-declaration] ret = asoc_simple_card_parse_clk_cpu(np, dai_link, dai_props); ^ >> sound/soc/sh/rcar/rsrc-card.c:232:9: error: implicit declaration of function 'asoc_simple_card_parse_clk_codec' [-Werror=implicit-function-declaration] ret = asoc_simple_card_parse_clk_codec(np, dai_link, dai_props); ^ cc1: some warnings being treated as errors vim +/asoc_simple_card_parse_clk_cpu +193 sound/soc/sh/rcar/rsrc-card.c 187 dai_link->dpcm_merged_format = 1; 188 dai_link->cpu_of_node = args.np; 189 ret = snd_soc_of_get_dai_name(np, &dai_link->cpu_dai_name); 190 if (ret < 0) 191 return ret; 192 > 193 ret = asoc_simple_card_parse_clk_cpu(np, dai_link, dai_props); 194 if (ret < 0) 195 return ret; 196 197 ret = asoc_simple_card_set_dailink_name(dev, dai_link, 198 "fe.%s", 199 dai_link->cpu_dai_name); 200 if (ret < 0) 201 return ret; 202 203 /* 204 * In soc_bind_dai_link() will check cpu name after 205 * of_node matching if dai_link has cpu_dai_name. 206 * but, it will never match if name was created by 207 * fmt_single_name() remove cpu_dai_name if cpu_args 208 * was 0. See: 209 * fmt_single_name() 210 * fmt_multiple_name() 211 */ 212 if (!args.args_count) 213 dai_link->cpu_dai_name = NULL; 214 } else { 215 const struct rsrc_card_of_data *of_data; 216 217 of_data = of_device_get_match_data(dev); 218 219 /* FE is dummy */ 220 dai_link->cpu_of_node = NULL; 221 dai_link->cpu_dai_name = "snd-soc-dummy-dai"; 222 dai_link->cpu_name = "snd-soc-dummy"; 223 224 /* BE settings */ 225 dai_link->no_pcm = 1; 226 dai_link->be_hw_params_fixup = rsrc_card_be_hw_params_fixup; 227 dai_link->codec_of_node = args.np; 228 ret = snd_soc_of_get_dai_name(np, &dai_link->codec_dai_name); 229 if (ret < 0) 230 return ret; 231 > 232 ret = asoc_simple_card_parse_clk_codec(np, dai_link, dai_props); 233 if (ret < 0) 234 return ret; 235 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/simple 19/20] sound/soc/generic/simple-card.c:350:8: error: implicit declaration of function 'asoc_simple_card_parse_clk_cpu'
by kbuild test robot 05 Aug '16

05 Aug '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/simple head: c9a235da8a61cf9af7653fca341ded0881a64eca commit: 28abd99b6e40741cd0e75be20817f23a3044d338 [19/20] ASoC: simple-card: use asoc_simple_card_parse_clk() config: i386-randconfig-s0-08050951 (attached as .config) compiler: gcc-6 (Debian 6.1.1-9) 6.1.1 20160705 reproduce: git checkout 28abd99b6e40741cd0e75be20817f23a3044d338 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/generic/simple-card.c: In function 'asoc_simple_card_dai_link_of': >> sound/soc/generic/simple-card.c:350:8: error: implicit declaration of function 'asoc_simple_card_parse_clk_cpu' [-Werror=implicit-function-declaration] ret = asoc_simple_card_parse_clk_cpu(cpu, dai_link, cpu_dai); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ >> sound/soc/generic/simple-card.c:354:8: error: implicit declaration of function 'asoc_simple_card_parse_clk_codec' [-Werror=implicit-function-declaration] ret = asoc_simple_card_parse_clk_codec(codec, dai_link, codec_dai); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors -- /kbuild/src/rand/sound/soc/generic/simple-card.c: In function 'asoc_simple_card_dai_link_of': >> /kbuild/src/rand/sound/soc/generic/simple-card.c:350:8: error: implicit declaration of function 'asoc_simple_card_parse_clk_cpu' [-Werror=implicit-function-declaration] ret = asoc_simple_card_parse_clk_cpu(cpu, dai_link, cpu_dai); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ >> /kbuild/src/rand/sound/soc/generic/simple-card.c:354:8: error: implicit declaration of function 'asoc_simple_card_parse_clk_codec' [-Werror=implicit-function-declaration] ret = asoc_simple_card_parse_clk_codec(codec, dai_link, codec_dai); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/asoc_simple_card_parse_clk_cpu +350 sound/soc/generic/simple-card.c 344 ret = asoc_simple_card_sub_parse_of(plat, NULL, 345 &dai_link->platform_of_node, 346 NULL, NULL); 347 if (ret < 0) 348 goto dai_link_of_err; 349 > 350 ret = asoc_simple_card_parse_clk_cpu(cpu, dai_link, cpu_dai); 351 if (ret < 0) 352 goto dai_link_of_err; 353 > 354 ret = asoc_simple_card_parse_clk_codec(codec, dai_link, codec_dai); 355 if (ret < 0) 356 goto dai_link_of_err; 357 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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Re: [alsa-devel] [PATCH 10/11] soc: ti: knav_qmss_queue: use of_property_read_bool
by Julia Lawall 05 Aug '16

05 Aug '16

On Fri, 5 Aug 2016, Robin Murphy wrote: > Hi Julia, > > On 05/08/16 09:56, Julia Lawall wrote: > > Use of_property_read_bool to check for the existence of a property. > > This caught my eye since Rob told me off for doing the same recently[1]. > > > The semantic patch that makes this change is as follows: > > (http://coccinelle.lip6.fr/) > > > > // <smpl> > > @@ > > expression e1,e2; > > statement S2,S1; > > @@ > > - if (of_get_property(e1,e2,NULL)) > > + if (of_property_read_bool(e1,e2)) > > S1 else S2 > > // </smpl> > > > > Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> > > > > --- > > drivers/soc/ti/knav_qmss_queue.c | 2 +- > > 1 file changed, 1 insertion(+), 1 deletion(-) > > > > diff --git a/drivers/soc/ti/knav_qmss_queue.c b/drivers/soc/ti/knav_qmss_queue.c > > index b73e353..56b5d7c 100644 > > --- a/drivers/soc/ti/knav_qmss_queue.c > > +++ b/drivers/soc/ti/knav_qmss_queue.c > > @@ -1240,7 +1240,7 @@ static int knav_setup_queue_range(struct knav_device *kdev, > > if (of_get_property(node, "qalloc-by-id", NULL)) > > According to the binding, "qalloc-by-id" _is_ a boolean property, so > this one really does deserve to be of_property_read_bool()... > > > range->flags |= RANGE_RESERVED; > > > > - if (of_get_property(node, "accumulator", NULL)) { > > + if (of_property_read_bool(node, "accumulator")) { > > ...whereas "accumulator" must have a value, so this isn't technically > appropriate. In general, most of these "if the property exists, read the > property and do stuff" checks are probably a sign of code that could be > simplified by refactoring the "do stuff" step to just specifically > handle the "read the property" step returning -EINVAL when it's not present. Thanks for the very helpful feedback. I will rethink the patch set in light of this information. julia > Robin. > > [1]:https://www.mail-archive.com/iommu@lists.linux-foundation.org/msg13375.h… > > > ret = knav_init_acc_range(kdev, node, range); > > if (ret < 0) { > > devm_kfree(dev, range); > > > > > > _______________________________________________ > > linux-arm-kernel mailing list > > linux-arm-kernel(a)lists.infradead.org > > http://lists.infradead.org/mailman/listinfo/linux-arm-kernel > > > > -- > To unsubscribe from this list: send the line "unsubscribe kernel-janitors" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html >

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[alsa-devel] [PATCH 00/11] use of_property_read_bool
by Julia Lawall 05 Aug '16

05 Aug '16

Use of_property_read_bool to check for the existence of a property. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e1,e2; statement S2,S1; @@ - if (of_get_property(e1,e2,NULL)) + if (of_property_read_bool(e1,e2)) S1 else S2 // </smpl> --- arch/powerpc/sysdev/mpic.c | 2 +- drivers/i2c/busses/i2c-mpc.c | 2 +- drivers/mmc/host/sdhci-of-esdhc.c | 2 +- drivers/net/ethernet/freescale/xgmac_mdio.c | 3 +-- drivers/phy/phy-qcom-ufs.c | 2 +- drivers/pinctrl/nomadik/pinctrl-nomadik.c | 2 +- drivers/soc/fsl/qe/qe_tdm.c | 2 +- drivers/soc/ti/knav_qmss_queue.c | 2 +- drivers/tty/serial/atmel_serial.c | 8 ++++---- drivers/usb/host/fsl-mph-dr-of.c | 6 +++--- sound/soc/codecs/ab8500-codec.c | 10 +++++----- sound/soc/sh/rcar/ssi.c | 2 +- sound/soc/soc-core.c | 2 +- 13 files changed, 22 insertions(+), 23 deletions(-)

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[alsa-devel] [PATCH 0/2 v2] DesignWare HDMI I2S suport
by Kuninori Morimoto 05 Aug '16

05 Aug '16

Hi Thierry, Russell, Mark These are v2 of DesignWare HDMI I2S support patches. It will use ALSA SoC hdmi-codec driver, but we can't use it as-is at this point. 1) patch tidyup hdmi-codec driver to enable dw-hdmi I2S support. 2) patch is based on it. Difficult is that these 2 patches are under different maintainer. Kuninori Morimoto (2): 1) ASoC: hdmi-codec: enable multi probe for same device 2) drm: bridge: add DesignWare HDMI I2S audio support drivers/gpu/drm/bridge/Kconfig | 8 ++++++++ drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw-hdmi-audio.h | 7 +++++++ drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c | 130 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw-hdmi.c | 22 ++++++++++++++++++-- drivers/gpu/drm/bridge/dw-hdmi.h | 21 +++++++++++++++++++ sound/soc/codecs/hdmi-codec.c | 67 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++--- 7 files changed, 251 insertions(+), 5 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v7] ASoC: rockchip: Add machine driver for RK3399 GRU Boards
by Xing Zheng 05 Aug '16

05 Aug '16

Because we need to support the multiple codecs (MAX98357A/RT5514/DA7219) on the RK3399 GRU boards, this patch can help us to support these codecs. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes in v7: - add default MCLK and PLL settings for the da7219_init() so that the accessory detection works as expected Changes in v6: - remove enable/disable dapm pins, and to switch codecs via user space - fix "Speaker" dapm route for max98357a - add SND_JACK_LINEOUT flag for da7219 jack detection Changes in v5: - fix the build warning for missing I2C/CLKDEV_LOOKUP dependency Changes in v4: - the compatible name needs to be based on a single reference design, and the name of the board is called "GRU" - use one i2sX card "rockchip,cpu = <&i2s0>" instead of array that is based on GRU - rename DOC to rockchip,rk3399-gru-sound.txt - rename compatible to rockchip,rk3399-gru-sound - rename source code to rk3399_gru_sound.c Changes in v3: - rename DOC to rockchip,rk3399-max98357a-rt5514-da7219.txt - rename compatible to rockchip,rk3399-max98357a-rt5514-da7219 - rename source code to rk3399_max98357a_rt5514_da7219.c Changes in v2: - use the FS 256 to set mclks of the max98357a and rt5514 danamically - add more sample rate for da7219 .../bindings/sound/rockchip,rk3399-gru-sound.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 337 ++++++++++++++++++++ 4 files changed, 365 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt create mode 100644 sound/soc/rockchip/rk3399_gru_sound.c diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt new file mode 100644 index 0000000..f19b6c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs on GRU boards + +Required properties: +- compatible: "rockchip,rk3399-gru-sound" +- rockchip,cpu: The phandle of the Rockchip I2S controller that's + connected to the codecs +- rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs + +Example: + +sound { + compatible = "rockchip,rk3399-gru-sound"; + rockchip,cpu = <&i2s0>; + rockchip,codec = <&max98357a &rt5514 &da7219>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..6d39032 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -41,3 +41,14 @@ config SND_SOC_ROCKCHIP_RT5645 help Say Y or M here if you want to add support for SoC audio on Rockchip boards using the RT5645/RT5650 codec, such as Veyron. + +config SND_SOC_RK3399_GRU_SOUND + tristate "ASoC support multiple codecs for Rockchip RK3399 GRU boards" + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98357A + select SND_SOC_RT5514 + select SND_SOC_DA7219 + help + Say Y or M here if you want to add support multiple codecs for SoC + audio on Rockchip RK3399 GRU boards. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..84e5c7c 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rk3399-gru-sound-objs := rk3399_gru_sound.o obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o +obj-$(CONFIG_SND_SOC_RK3399_GRU_SOUND) += snd-soc-rk3399-gru-sound.o diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c new file mode 100644 index 0000000..9933703 --- /dev/null +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -0,0 +1,337 @@ +/* + * Rockchip machine ASoC driver for boards using MAX98357A/RT5514/DA7219 + * + * Copyright (c) 2016, ROCKCHIP CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/delay.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "rockchip_i2s.h" +#include "../codecs/da7219.h" +#include "../codecs/da7219-aad.h" +#include "../codecs/rt5514.h" + +#define DRV_NAME "rk3399-gru-sound" + +#define SOUND_FS 256 + +static struct snd_soc_jack rockchip_sound_jack; + +static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { + /* Input Lines */ + {"MIC", NULL, "Headset Mic"}, + {"DMIC1L", NULL, "Int Mic"}, + {"DMIC1R", NULL, "Int Mic"}, + + /* Output Lines */ + {"Headphones", NULL, "HPL"}, + {"Headphones", NULL, "HPR"}, + {"Speakers", NULL, "Speaker"}, +}; + +static const struct snd_kcontrol_new rockchip_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), +}; + +static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int mclk; + int ret; + + /* max98357a supports these sample rates */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = params_rate(params) * SOUND_FS; + break; + default: + dev_err(rtd->card->dev, "%s() doesn't support this sample rate: %d\n", + __func__, params_rate(params)); + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, mclk, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int mclk; + int ret; + + mclk = params_rate(params) * SOUND_FS; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(rtd->card->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5514_SCLK_S_MCLK, + mclk, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, params_rate(params) * 512, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int mclk, ret; + + /* in bypass mode, the mclk has to be one of the frequencies below */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* We need default MCLK and PLL settings for the accessory detection */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12288000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Init can't set codec clock in %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Init can't set pll sysclk mclk %d\n", ret); + return ret; + } + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &rockchip_sound_jack, NULL, 0); + + if (ret) { + dev_err(rtd->card->dev, "New Headset Jack failed! (%d)\n", ret); + return ret; + } + + da7219_aad_jack_det(codec, &rockchip_sound_jack); + + return 0; +} + +static struct snd_soc_ops rockchip_sound_max98357a_ops = { + .hw_params = rockchip_sound_max98357a_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_rt5514_ops = { + .hw_params = rockchip_sound_rt5514_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_da7219_ops = { + .hw_params = rockchip_sound_da7219_hw_params, +}; + +enum { + DAILINK_MAX98357A, + DAILINK_RT5514, + DAILINK_DA7219, +}; + +static struct snd_soc_dai_link rockchip_dailinks[] = { + [DAILINK_MAX98357A] = { + .name = "MAX98357A", + .stream_name = "MAX98357A PCM", + .codec_dai_name = "HiFi", + .ops = &rockchip_sound_max98357a_ops, + /* set max98357a as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_RT5514] = { + .name = "RT5514", + .stream_name = "RT5514 PCM", + .codec_dai_name = "rt5514-aif1", + .ops = &rockchip_sound_rt5514_ops, + /* set rt5514 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_DA7219] = { + .name = "DA7219", + .stream_name = "DA7219 PCM", + .codec_dai_name = "da7219-hifi", + .init = rockchip_sound_da7219_init, + .ops = &rockchip_sound_da7219_ops, + /* set da7219 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card rockchip_sound_card = { + .name = "rk3399-gru-sound", + .owner = THIS_MODULE, + .dai_link = rockchip_dailinks, + .num_links = ARRAY_SIZE(rockchip_dailinks), + .dapm_widgets = rockchip_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets), + .dapm_routes = rockchip_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes), + .controls = rockchip_controls, + .num_controls = ARRAY_SIZE(rockchip_controls), +}; + +static int rockchip_sound_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &rockchip_sound_card; + struct device_node *cpu_node; + int i, ret; + + cpu_node = of_parse_phandle(pdev->dev.of_node, "rockchip,cpu", 0); + if (!cpu_node) { + dev_err(&pdev->dev, "Property 'rockchip,cpu' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + rockchip_dailinks[i].platform_of_node = cpu_node; + rockchip_dailinks[i].cpu_of_node = cpu_node; + + rockchip_dailinks[i].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "rockchip,codec", i); + if (!rockchip_dailinks[i].codec_of_node) { + dev_err(&pdev->dev, + "Property[%d] 'rockchip,codec' missing or invalid\n", i); + return -EINVAL; + } + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + + return ret; +} + +static const struct of_device_id rockchip_sound_of_match[] = { + { .compatible = "rockchip,rk3399-gru-sound", }, + {}, +}; + +static struct platform_driver rockchip_sound_driver = { + .probe = rockchip_sound_probe, + .driver = { + .name = DRV_NAME, + .of_match_table = rockchip_sound_of_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, +}; + +module_platform_driver(rockchip_sound_driver); + +MODULE_AUTHOR("Xing Zheng <zhengxing(a)rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_sound_of_match); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: dwc: Drop DMA channel names assignment
by Sylwester Nawrocki 05 Aug '16

05 Aug '16

The dw_configure_dai_by_dt() function and further dev->{play,capture} _dma_data.dt data structures seem to be used in this driver only in case of a system using devicetree, thus chan_name assignments have no effect since they will be ignored in dmaengine_pcm_request_chan_of() call and will be substituted with values taken form dmaengine_pcm_dma_channel_names[] table ("tx", "rx"). Also there is no any "TX", "RX" dma-names entries in arch/arm/boot/dts, only lower case "tx", "rx" seem to be used. Lastly, this driver doesn't set SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag when registering a dmaengine PCM to indicate the chan_name should be used. My intention is to eventually remove the struct snd_dmaengine_dai_dma_data chan_name field as there is also a chan_names[] field in struct snd_dmaengine_pcm_config which can be used for same purpose. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/dwc/designware_i2s.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index dc97f43..2998954 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -577,7 +577,6 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, dev->capability |= DWC_I2S_PLAY; dev->play_dma_data.dt.addr = res->start + I2S_TXDMA; dev->play_dma_data.dt.addr_width = bus_widths[idx]; - dev->play_dma_data.dt.chan_name = "TX"; dev->play_dma_data.dt.fifo_size = fifo_depth * (fifo_width[idx2]) >> 8; dev->play_dma_data.dt.maxburst = 16; @@ -588,7 +587,6 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, dev->capability |= DWC_I2S_RECORD; dev->capture_dma_data.dt.addr = res->start + I2S_RXDMA; dev->capture_dma_data.dt.addr_width = bus_widths[idx]; - dev->capture_dma_data.dt.chan_name = "RX"; dev->capture_dma_data.dt.fifo_size = fifo_depth * (fifo_width[idx2] >> 8); dev->capture_dma_data.dt.maxburst = 16; -- 1.9.1

2 1

[alsa-devel] [PATCH 0/3] ASoC: topology: Add support for BE DAIs
by mengdong.lin＠linux.intel.com 05 Aug '16

05 Aug '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is topology ABI udpate in this series. Topology core can check size of each ABI objects to detect version mismatch between user space and kernel. This series adds support for BE (Back End) DAIs: - Define the type and ABI struct for Backend DAIs. - The topology core can configure an existing BE DAI or create new BE DAIs. The user space series "topology: Add support for BE DAIs" is also submitted. Is it possible to meet the timeline of v4.8? There will be 2 pairs of kernel & user space series for PCM & BE links later. Mengdong Lin (3): ASoC: topology: ABI - Add the types for BE DAI ASoC: topology: Add support for configuring existing BE DAIs ASoC: topology: Able to create BE DAIs include/sound/soc-topology.h | 1 + include/uapi/sound/asoc.h | 34 +++++++++- sound/soc/soc-core.c | 3 +- sound/soc/soc-topology.c | 156 ++++++++++++++++++++++++++++++++++++++++++- 4 files changed, 189 insertions(+), 5 deletions(-) -- 2.5.0

2 5

[alsa-devel] [PATCH RESEND 7/8] ASoC: samsung: s3c24xx_uda134x: debug/error trace cleanup
by Sylwester Nawrocki 05 Aug '16

05 Aug '16

Switch from pr_* to dev_* macros and drop some debug traces. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Rebased onto topic/samsung branch. sound/soc/samsung/s3c24xx_uda134x.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 13e52fb..1619e76 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -62,18 +62,18 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) #endif mutex_lock(&clk_lock); - pr_debug("%s %d\n", __func__, clk_users); + if (clk_users == 0) { xtal = clk_get(rtd->dev, "xtal"); if (IS_ERR(xtal)) { - printk(KERN_ERR "%s cannot get xtal\n", __func__); + dev_err(rtd->dev, "%s cannot get xtal\n", __func__); ret = PTR_ERR(xtal); } else { pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, "pclk"); if (IS_ERR(pclk)) { - printk(KERN_ERR "%s cannot get pclk\n", - __func__); + dev_err(rtd->dev, "%s cannot get pclk\n", + __func__); clk_put(xtal); ret = PTR_ERR(pclk); } @@ -99,8 +99,8 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); if (ret < 0) - printk(KERN_ERR "%s cannot set constraints\n", - __func__); + dev_err(rtd->dev, "%s cannot set constraints\n", + __func__); #endif } return ret; @@ -109,7 +109,6 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) { mutex_lock(&clk_lock); - pr_debug("%s %d\n", __func__, clk_users); clk_users -= 1; if (clk_users == 0) { clk_put(xtal); @@ -156,18 +155,19 @@ static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, clk_source = S3C24XX_CLKSRC_PCLK; div = bi % 33; } - pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi); + + dev_dbg(rtd->dev, "%s desired rate %lu, %d\n", __func__, rate, bi); clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; - pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__, - fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", - clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", - div, clk, err); + + dev_dbg(rtd->dev, "%s will use: %s %s %d sysclk %d err %ld\n", __func__, + fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", + clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", + div, clk, err); if ((err * 100 / rate) > 5) { - printk(KERN_ERR "S3C24XX_UDA134X: effective frequency " - "too different from desired (%ld%%)\n", - err * 100 / rate); + dev_err(rtd->dev, "effective frequency too different " + "from desired (%ld%%)\n", err * 100 / rate); return -EINVAL; } -- 1.9.1

2 4

[alsa-devel] [PATCH 00/12] ASoC: Intel: Skylake: Add support for library load
by Vinod Koul 05 Aug '16

05 Aug '16

This patch series adds support for library loading. A module maybe part of base firmware or in a different library. A library may have one or more modules. The names of library to be loaded are passed as topology manifest data. The library list is loaded after base FW during dsp init. So we add support library load IPC and the usage in broxton ops for loading libs. To load a list of lib(s) coming from topology, we need to split DSP init, into object creation and fw_init. The fw_init is invoked from soc-probe after topology is initialized. Now fw is initialized later, so module UUID query was move to runtime as the firmware is not available at topology initialization. While at it, also fix the parsing UUIDs only once for SKL and BXT ops and one comment code style. Lastly this series adds support for specifying processing domain for modules. Kranthi G (1): ASoC: Intel: Skylake: add support for tplg manifest load Ramesh Babu (2): ASoC: Intel: Skylake: Add library loading IPCs ASoC: Intel: Skylake: Add library loading support Senthilnathan Veppur (2): ASoC: Intel: Skylake: add additional args to module parsing ASoC: Intel: Skylake: Add module processing domain support Vinod Koul (7): ASoC: Intel: Skylake: Check list empty while getting module info ASoC: Intel: Skylake: Move modules query to runtime ASoC: Intel: Skylake: modify skl_get_dsp_ops() ASoC: Intel: Skylake: split fw and dsp initialization ASoC: Intel: Skylake: Parse UUIDs once ASoC: Intel: Bxt: Parse UUIDs once ASoC: Intel: Skylake: Fix a comment style sound/soc/intel/skylake/bxt-sst.c | 114 ++++++++++++++++++++++++--- sound/soc/intel/skylake/skl-messages.c | 39 +++++---- sound/soc/intel/skylake/skl-pcm.c | 20 +++++ sound/soc/intel/skylake/skl-sst-dsp.h | 7 +- sound/soc/intel/skylake/skl-sst-ipc.c | 29 +++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 12 ++- sound/soc/intel/skylake/skl-sst-utils.c | 14 +++- sound/soc/intel/skylake/skl-sst.c | 43 ++++++---- sound/soc/intel/skylake/skl-topology.c | 55 +++++++++++-- sound/soc/intel/skylake/skl-topology.h | 2 +- sound/soc/intel/skylake/skl-tplg-interface.h | 15 +++- sound/soc/intel/skylake/skl.h | 2 + 12 files changed, 297 insertions(+), 55 deletions(-) -- 1.9.1

2 24

[alsa-devel] [PATCH 0/3] DesignWare HDMI I2S suport
by Kuninori Morimoto 05 Aug '16

05 Aug '16

Hi Mark, Thierry, Russell These are DesignWare HDMI I2S support patches. It will use ALSA SoC hdmi-codec driver, but we can't use it as-is. So, 2), 3) patches modify hdmi-codec style. Kuninori Morimoto (3): 1) drm: bridge: add DesignWare HDMI I2S audio support 2) ASoC: hdmi-codec: callback function will be called with private data 3) ASoC: hdmi-codec: enable multi probe for same device drivers/gpu/drm/bridge/Kconfig | 8 ++ drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw-hdmi-audio.h | 7 ++ drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c | 123 +++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw-hdmi.c | 22 +++++- drivers/gpu/drm/bridge/dw-hdmi.h | 21 +++++ include/sound/hdmi-codec.h | 13 +-- sound/soc/codecs/hdmi-codec.c | 81 ++++++++++++++++--- 8 files changed, 259 insertions(+), 17 deletions(-) create mode 100644 drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c -- 1.9.1

5 16

Re: [alsa-devel] [PATCH] drm/i915/dp: DP audio API changes for MST
by Ville Syrjälä 05 Aug '16

05 Aug '16

On Thu, Aug 04, 2016 at 10:18:52AM -0700, Jim Bride wrote: > On Wed, Aug 03, 2016 at 10:08:12PM +0300, Ville Syrjälä wrote: > > On Tue, Aug 02, 2016 at 07:14:30PM -0700, Dhinakaran Pandiyan wrote: > > > DP MST provides the capability to send multiple video and audio streams via > > > one single port. This requires the API's between i915 and audio drivers to > > > distinguish between audio capable displays connected to a port. This patch > > > adds this support via an additional parameter 'int dev_id'. The existing > > > parameter 'port' does not change it's meaning. > > > > > > dev_id = > > > MST : pipe that the stream originates from > > > Non-MST : -1 > > > > > > Affected APIs: > > > struct i915_audio_component_ops > > > - int (*sync_audio_rate)(struct device *, int port, int rate); > > > + int (*sync_audio_rate)(struct device *, int port, int dev_id, > > > > Does the term 'dev_id' have some special meaning on the audio side? On > > the i915 side things would be less confusing if we just called it > > 'pipe'. > > Yeah, it does. All of the documentation on the audio side is written > in terms of device ID, so they asked for that nomenclature. And is the device ID always the same as the pipe? Until now we've made due with passing the port instead of the pipe, so either the audio side didn't use the device ID, or its meaning changes based on how we drive things, or they dug it out from somewhere else based on the port? > > Jim > > > > > + int rate); > > > > > > - int (*get_eld)(struct device *, int port, bool *enabled, > > > - unsigned char *buf, int max_bytes); > > > + int (*get_eld)(struct device *, int port, int dev_id, > > > + bool *enabled, unsigned char *buf, int max_bytes); > > > > > > struct i915_audio_component_audio_ops > > > - void (*pin_eld_notify)(void *audio_ptr, int port); > > > + void (*pin_eld_notify)(void *audio_ptr, int port, int dev_id); > > > > > > This patch makes dummy changes in the audio drivers for build to succeed. > > > > > > Signed-off-by: Dhinakaran Pandiyan <dhinakaran.pandiyan(a)intel.com> > > > --- > > > drivers/gpu/drm/i915/i915_drv.h | 2 +- > > > drivers/gpu/drm/i915/intel_audio.c | 82 +++++++++++++++++++++++++++++--------- > > > include/drm/i915_component.h | 6 +-- > > > include/sound/hda_i915.h | 11 ++--- > > > sound/hda/hdac_i915.c | 9 +++-- > > > sound/pci/hda/patch_hdmi.c | 7 ++-- > > > 6 files changed, 82 insertions(+), 35 deletions(-) > > > > > > diff --git a/drivers/gpu/drm/i915/i915_drv.h b/drivers/gpu/drm/i915/i915_drv.h > > > index 65ada5d..8e4c8c8 100644 > > > --- a/drivers/gpu/drm/i915/i915_drv.h > > > +++ b/drivers/gpu/drm/i915/i915_drv.h > > > @@ -2054,7 +2054,7 @@ struct drm_i915_private { > > > /* perform PHY state sanity checks? */ > > > bool chv_phy_assert[2]; > > > > > > - struct intel_encoder *dig_port_map[I915_MAX_PORTS]; > > > + struct intel_encoder *av_enc_map[I915_MAX_PIPES]; > > > > > > /* > > > * NOTE: This is the dri1/ums dungeon, don't add stuff here. Your patch > > > diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c > > > index 8c493de..cbe44c8 100644 > > > --- a/drivers/gpu/drm/i915/intel_audio.c > > > +++ b/drivers/gpu/drm/i915/intel_audio.c > > > @@ -500,6 +500,9 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) > > > struct i915_audio_component *acomp = dev_priv->audio_component; > > > struct intel_digital_port *intel_dig_port = enc_to_dig_port(encoder); > > > enum port port = intel_dig_port->port; > > > + enum pipe pipe = crtc->pipe; > > > + int dev_id = -1; > > > + > > > > > > connector = drm_select_eld(encoder); > > > if (!connector) > > > @@ -522,14 +525,19 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) > > > dev_priv->display.audio_codec_enable(connector, intel_encoder, > > > adjusted_mode); > > > > > > + if (intel_encoder->type == INTEL_OUTPUT_DP_MST) > > > + dev_id = pipe; > > > + > > > mutex_lock(&dev_priv->av_mutex); > > > intel_encoder->audio_connector = connector; > > > + > > > /* referred in audio callbacks */ > > > - dev_priv->dig_port_map[port] = intel_encoder; > > > + dev_priv->av_enc_map[pipe] = intel_encoder; > > > mutex_unlock(&dev_priv->av_mutex); > > > > > > if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) > > > - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port); > > > + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, > > > + (int) port, dev_id); > > > } > > > > > > /** > > > @@ -542,22 +550,29 @@ void intel_audio_codec_enable(struct intel_encoder *intel_encoder) > > > void intel_audio_codec_disable(struct intel_encoder *intel_encoder) > > > { > > > struct drm_encoder *encoder = &intel_encoder->base; > > > + struct intel_crtc *crtc = to_intel_crtc(encoder->crtc); > > > struct drm_device *dev = encoder->dev; > > > struct drm_i915_private *dev_priv = to_i915(dev); > > > struct i915_audio_component *acomp = dev_priv->audio_component; > > > struct intel_digital_port *intel_dig_port = enc_to_dig_port(encoder); > > > enum port port = intel_dig_port->port; > > > + enum pipe pipe = crtc->pipe; > > > + int dev_id = -1; > > > + > > > + if (intel_encoder->type == INTEL_OUTPUT_DP_MST) > > > + dev_id = pipe; > > > > > > if (dev_priv->display.audio_codec_disable) > > > dev_priv->display.audio_codec_disable(intel_encoder); > > > > > > mutex_lock(&dev_priv->av_mutex); > > > intel_encoder->audio_connector = NULL; > > > - dev_priv->dig_port_map[port] = NULL; > > > + dev_priv->av_enc_map[pipe] = NULL; > > > mutex_unlock(&dev_priv->av_mutex); > > > > > > if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) > > > - acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, (int) port); > > > + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, > > > + (int) port, dev_id); > > > } > > > > > > /** > > > @@ -628,8 +643,32 @@ static int i915_audio_component_get_cdclk_freq(struct device *dev) > > > return dev_priv->cdclk_freq; > > > } > > > > > > -static int i915_audio_component_sync_audio_rate(struct device *dev, > > > - int port, int rate) > > > +static struct intel_encoder *get_saved_encoder(struct intel_encoder *av_enc_map[], > > > + int port, int dev_id) > > > +{ > > > + > > > + enum pipe pipe; > > > + struct drm_encoder *encoder; > > > + > > > + if (dev_id >= 0 && dev_id < I915_MAX_PIPES) > > > + return av_enc_map[dev_id]; > > > + > > > + for (pipe = PIPE_A; pipe < I915_MAX_PIPES; pipe++) { > > > + > > > + if (!av_enc_map[pipe]) > > > + continue; > > > + > > > + encoder = &av_enc_map[pipe]->base; > > > + if (port == enc_to_dig_port(encoder)->port) > > > + return av_enc_map[pipe]; > > > + } > > > + > > > + return NULL; > > > + > > > +} > > > + > > > +static int i915_audio_component_sync_audio_rate(struct device *dev, int port, > > > + int dev_id, int rate) > > > { > > > struct drm_i915_private *dev_priv = dev_to_i915(dev); > > > struct intel_encoder *intel_encoder; > > > @@ -649,15 +688,16 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, > > > return 0; > > > > > > mutex_lock(&dev_priv->av_mutex); > > > + > > > /* 1. get the pipe */ > > > - intel_encoder = dev_priv->dig_port_map[port]; > > > - /* intel_encoder might be NULL for DP MST */ > > > + intel_encoder = get_saved_encoder(dev_priv->av_enc_map, port, dev_id); > > > if (!intel_encoder || !intel_encoder->base.crtc || > > > intel_encoder->type != INTEL_OUTPUT_HDMI) { > > > DRM_DEBUG_KMS("no valid port %c\n", port_name(port)); > > > err = -ENODEV; > > > goto unlock; > > > } > > > + > > > crtc = to_intel_crtc(intel_encoder->base.crtc); > > > pipe = crtc->pipe; > > > if (pipe == INVALID_PIPE) { > > > @@ -702,7 +742,7 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, > > > } > > > > > > static int i915_audio_component_get_eld(struct device *dev, int port, > > > - bool *enabled, > > > + int dev_id, bool *enabled, > > > unsigned char *buf, int max_bytes) > > > { > > > struct drm_i915_private *dev_priv = dev_to_i915(dev); > > > @@ -710,17 +750,21 @@ static int i915_audio_component_get_eld(struct device *dev, int port, > > > const u8 *eld; > > > int ret = -EINVAL; > > > > > > + > > > mutex_lock(&dev_priv->av_mutex); > > > - intel_encoder = dev_priv->dig_port_map[port]; > > > - /* intel_encoder might be NULL for DP MST */ > > > - if (intel_encoder) { > > > - ret = 0; > > > - *enabled = intel_encoder->audio_connector != NULL; > > > - if (*enabled) { > > > - eld = intel_encoder->audio_connector->eld; > > > - ret = drm_eld_size(eld); > > > - memcpy(buf, eld, min(max_bytes, ret)); > > > - } > > > + > > > + intel_encoder = get_saved_encoder(dev_priv->av_enc_map, port, dev_id); > > > + if (!intel_encoder) { > > > + mutex_unlock(&dev_priv->av_mutex); > > > + return ret; > > > + } > > > + > > > + ret = 0; > > > + *enabled = intel_encoder->audio_connector != NULL; > > > + if (*enabled) { > > > + eld = intel_encoder->audio_connector->eld; > > > + ret = drm_eld_size(eld); > > > + memcpy(buf, eld, min(max_bytes, ret)); > > > } > > > > > > mutex_unlock(&dev_priv->av_mutex); > > > diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h > > > index b46fa0e..1419c98 100644 > > > --- a/include/drm/i915_component.h > > > +++ b/include/drm/i915_component.h > > > @@ -64,7 +64,7 @@ struct i915_audio_component_ops { > > > * Called from audio driver. After audio driver sets the > > > * sample rate, it will call this function to set n/cts > > > */ > > > - int (*sync_audio_rate)(struct device *, int port, int rate); > > > + int (*sync_audio_rate)(struct device *, int port, int dev_id, int rate); > > > /** > > > * @get_eld: fill the audio state and ELD bytes for the given port > > > * > > > @@ -77,7 +77,7 @@ struct i915_audio_component_ops { > > > * Note that the returned size may be over @max_bytes. Then it > > > * implies that only a part of ELD has been copied to the buffer. > > > */ > > > - int (*get_eld)(struct device *, int port, bool *enabled, > > > + int (*get_eld)(struct device *, int port, int dev_id, bool *enabled, > > > unsigned char *buf, int max_bytes); > > > }; > > > > > > @@ -97,7 +97,7 @@ struct i915_audio_component_audio_ops { > > > * status accordingly (even when the HDA controller is in power save > > > * mode). > > > */ > > > - void (*pin_eld_notify)(void *audio_ptr, int port); > > > + void (*pin_eld_notify)(void *audio_ptr, int port, int dev_id); > > > }; > > > > > > /** > > > diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h > > > index 796cabf..5ab972e 100644 > > > --- a/include/sound/hda_i915.h > > > +++ b/include/sound/hda_i915.h > > > @@ -10,8 +10,9 @@ > > > int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable); > > > int snd_hdac_display_power(struct hdac_bus *bus, bool enable); > > > void snd_hdac_i915_set_bclk(struct hdac_bus *bus); > > > -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate); > > > -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, > > > +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, > > > + int dev_id, int rate); > > > +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, > > > bool *audio_enabled, char *buffer, int max_bytes); > > > int snd_hdac_i915_init(struct hdac_bus *bus); > > > int snd_hdac_i915_exit(struct hdac_bus *bus); > > > @@ -29,13 +30,13 @@ static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus) > > > { > > > } > > > static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec, > > > - hda_nid_t nid, int rate) > > > + hda_nid_t nid, int dev_id, int rate) > > > { > > > return 0; > > > } > > > static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, > > > - bool *audio_enabled, char *buffer, > > > - int max_bytes) > > > + int dev_id, bool *audio_enabled, > > > + char *buffer, int max_bytes) > > > { > > > return -ENODEV; > > > } > > > diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c > > > index c9af022..6ea63ac 100644 > > > --- a/sound/hda/hdac_i915.c > > > +++ b/sound/hda/hdac_i915.c > > > @@ -201,7 +201,8 @@ static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) > > > * This function sets N/CTS value based on the given sample rate. > > > * Returns zero for success, or a negative error code. > > > */ > > > -int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate) > > > +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, > > > + int dev_id, int rate) > > > { > > > struct hdac_bus *bus = codec->bus; > > > struct i915_audio_component *acomp = bus->audio_component; > > > @@ -212,7 +213,7 @@ int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate) > > > port = pin2port(codec, nid); > > > if (port < 0) > > > return -EINVAL; > > > - return acomp->ops->sync_audio_rate(acomp->dev, port, rate); > > > + return acomp->ops->sync_audio_rate(acomp->dev, port, dev_id, rate); > > > } > > > EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); > > > > > > @@ -236,7 +237,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); > > > * thus it may be over @max_bytes. If it's over @max_bytes, it implies > > > * that only a part of ELD bytes have been fetched. > > > */ > > > -int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, > > > +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, int dev_id, > > > bool *audio_enabled, char *buffer, int max_bytes) > > > { > > > struct hdac_bus *bus = codec->bus; > > > @@ -249,7 +250,7 @@ int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, > > > port = pin2port(codec, nid); > > > if (port < 0) > > > return -EINVAL; > > > - return acomp->ops->get_eld(acomp->dev, port, audio_enabled, > > > + return acomp->ops->get_eld(acomp->dev, port, dev_id, audio_enabled, > > > buffer, max_bytes); > > > } > > > EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); > > > diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c > > > index d0d5ad8..077d48a 100644 > > > --- a/sound/pci/hda/patch_hdmi.c > > > +++ b/sound/pci/hda/patch_hdmi.c > > > @@ -1485,7 +1485,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, > > > > > > mutex_lock(&per_pin->lock); > > > eld->monitor_present = false; > > > - size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, > > > + size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, -1, > > > &eld->monitor_present, eld->eld_buffer, > > > ELD_MAX_SIZE); > > > if (size > 0) { > > > @@ -1739,7 +1739,8 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, > > > /* Call sync_audio_rate to set the N/CTS/M manually if necessary */ > > > /* Todo: add DP1.2 MST audio support later */ > > > if (codec_has_acomp(codec)) > > > - snd_hdac_sync_audio_rate(&codec->core, pin_nid, runtime->rate); > > > + snd_hdac_sync_audio_rate(&codec->core, pin_nid, -1, > > > + runtime->rate); > > > > > > non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); > > > mutex_lock(&per_pin->lock); > > > @@ -2285,7 +2286,7 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, > > > snd_hda_codec_set_power_to_all(codec, fg, power_state); > > > } > > > > > > -static void intel_pin_eld_notify(void *audio_ptr, int port) > > > +static void intel_pin_eld_notify(void *audio_ptr, int port, int dev_id) > > > { > > > struct hda_codec *codec = audio_ptr; > > > int pin_nid; > > > -- > > > 2.5.0 > > > > -- > > Ville Syrjälä > > Intel OTC -- Ville Syrjälä Intel OTC

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[alsa-devel] Question about struct snd_soc_dai() :: cpu_dai->codec
by Kuninori Morimoto 05 Aug '16

05 Aug '16

Hi ALSA SoC My current headache is ALSA SoC's each modules (= Card/Codec/CPU/Platform) doesn't care about "unbind/rmmod". For example, if someone unbinded/rmmoded "Codec", Card or other modules doesn't know about it. Thus, user can continue to use this sound card, and kernel will be Oops. So, I would like to solve this issue, and for this purpose, now I'm investigating about ALSA SoC structures. It is now super complex, and I noticed that it is having duplicated struct members. If possible, I would like to cleanup this issue. During this investigating, I noticed 1 strange operation ${LINUX}/sound/soc/soc-core.c :: snd_soc_runtime_set_dai_fmt int snd_soc_runtime_set_dai_fmt(xxx) { ... /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */ if (cpu_dai->codec) { ... } ... } struct snd_soc_dai { ... /* parent platform/codec */ struct snd_soc_codec *codec; struct snd_soc_component *component; ... } According to snd_soc_dai, *codec is its "parent". I wonder does "cpu_dai" really can have "codec" parent ?? I think this is not correct, and we can remove this operation ?

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[alsa-devel] DP audio API changes for identifying displays connected to a port
by Dhinakaran Pandiyan 05 Aug '16

05 Aug '16

An additional parameter has been added to the API's between i915 and audio drivers to identify display devices that can be connected to the same port. Currently only the port identity is being shared, but this is not sufficient as multiple displays can be driven from the same port (E.g. DP MST). This patch makes the API changes with dummy changes in the audio driver (Thanks Libin). Implementation to fully enable DP MST audio will follow later. Dhinakaran Pandiyan(1): drm/i915/dp: DP audio API changes for MST

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[alsa-devel] [PATCH 1/2] ASoC: nau8825: fix bug in playback when suspend
by John Hsu 04 Aug '16

04 Aug '16

In chromium, the following steps will make codec function fail. \1. plug in headphones, Play music \2. run "powerd_dbus_suspend" \3. resume from S3 After resume, the jack detection will restart and make configuration for the headset. Meanwhile, the playback prepares and starts to work. The two sequences will conflict and make wrong register configuration. Originally, the driver adds protection for the case when it finds the playback is active. But the "powerd_dbus_suspend" command will close the pcm stream before suspend. Therefore, the driver can't detect the playback after resume, and the protection not works. For the issue, the driver raises protection every time after resume. The protection will release after jack detection and configuration completes, and then the playback just will goes on. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 51 +++++++++++----------------------------------- 1 file changed, 12 insertions(+), 39 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 5c9707a..20d2f15 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -212,31 +212,6 @@ static const unsigned short logtable[256] = { 0xfa2f, 0xfaea, 0xfba5, 0xfc60, 0xfd1a, 0xfdd4, 0xfe8e, 0xff47 }; -static struct snd_soc_dai *nau8825_get_codec_dai(struct nau8825 *nau8825) -{ - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(nau8825->dapm); - struct snd_soc_component *component = &codec->component; - struct snd_soc_dai *codec_dai, *_dai; - - list_for_each_entry_safe(codec_dai, _dai, &component->dai_list, list) { - if (!strncmp(codec_dai->name, NUVOTON_CODEC_DAI, - strlen(NUVOTON_CODEC_DAI))) - return codec_dai; - } - return NULL; -} - -static bool nau8825_dai_is_active(struct nau8825 *nau8825) -{ - struct snd_soc_dai *codec_dai = nau8825_get_codec_dai(nau8825); - - if (codec_dai) { - if (codec_dai->playback_active || codec_dai->capture_active) - return true; - } - return false; -} - /** * nau8825_sema_acquire - acquire the semaphore of nau88l25 * @nau8825: component to register the codec private data with @@ -1205,6 +1180,8 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); unsigned int val_len = 0; + nau8825_sema_acquire(nau8825, 2 * HZ); + switch (params_width(params)) { case 16: val_len |= NAU8825_I2S_DL_16; @@ -1225,6 +1202,9 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, NAU8825_I2S_DL_MASK, val_len); + /* Release the semaphone. */ + nau8825_sema_release(nau8825); + return 0; } @@ -1234,6 +1214,8 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); unsigned int ctrl1_val = 0, ctrl2_val = 0; + nau8825_sema_acquire(nau8825, 2 * HZ); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: ctrl2_val |= NAU8825_I2S_MS_MASTER; @@ -1282,6 +1264,9 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, NAU8825_I2S_MS_MASK, ctrl2_val); + /* Release the semaphone. */ + nau8825_sema_release(nau8825); + return 0; } @@ -2241,20 +2226,8 @@ static int __maybe_unused nau8825_resume(struct snd_soc_codec *codec) regcache_cache_only(nau8825->regmap, false); regcache_sync(nau8825->regmap); - if (nau8825_is_jack_inserted(nau8825->regmap)) { - /* If the jack is inserted, we need to check whether the play- - * back is active before suspend. If active, the driver has to - * raise the protection for cross talk function to avoid the - * playback recovers before cross talk process finish. Other- - * wise, the playback will be interfered by cross talk func- - * tion. It is better to apply hardware related parameters - * before starting playback or record. - */ - if (nau8825_dai_is_active(nau8825)) { - nau8825->xtalk_protect = true; - nau8825_sema_acquire(nau8825, 0); - } - } + nau8825->xtalk_protect = true; + nau8825_sema_acquire(nau8825, 0); enable_irq(nau8825->irq); return 0; -- 2.6.4

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[alsa-devel] [PATCH] ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card
by Sylwester Nawrocki 04 Aug '16

04 Aug '16

There is no "pclk" alias in the s3c2440 clk driver for "soc-audio" device so related clk_get() fails, which prevents any operation of the S3C24XX_UDA134X sound card. Instead we get the clock on behalf of the I2S device, i.e. we use the I2S block gate clock which has PCLK is its parent clock. Without this patch there is an error like: s3c24xx_uda134x_startup cannot get pclk ASoC: UDA134X startup failed: -2 Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/s3c24xx_uda134x.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 50849e1..92e88bc 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -58,10 +58,12 @@ static struct platform_device *s3c24xx_uda134x_snd_device; static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) { - int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; #ifdef ENFORCE_RATES struct snd_pcm_runtime *runtime = substream->runtime; #endif + int ret = 0; mutex_lock(&clk_lock); pr_debug("%s %d\n", __func__, clk_users); @@ -71,8 +73,7 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) printk(KERN_ERR "%s cannot get xtal\n", __func__); ret = PTR_ERR(xtal); } else { - pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, - "pclk"); + pclk = clk_get(cpu_dai->dev, "iis"); if (IS_ERR(pclk)) { printk(KERN_ERR "%s cannot get pclk\n", __func__); -- 1.9.1

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[alsa-devel] [PATCH 0/7] ASoC: samsung: DMA parameters handling cleanup
by Sylwester Nawrocki 04 Aug '16

04 Aug '16

The motivation for this series is to remove the s3c_dma_params data structure and use struct snd_dmaengine_dai_dma_data instead, which removes some kind of indirection and makes the code easier to follow. Sylwester Nawrocki (7): ASoC: samsung: Drop usage of struct s3c_dma_params from i2s.c ASoC: samsung: Drop usage of struct s3c_dma_params from s3c24xx-i2s.c ASoC: samsung: Drop usage of struct s3c_dma_params from s3c2412-i2s.c ASoC: samsung: Drop usage of struct s3c_dma_params from ac97.c ASoC: samsung: Drop usage of struct s3c_dma_params from spdif.c ASoC: samsung: Drop usage of of struct s3c_dma_params from pcm.c ASoC: samsung: Remove unused now unused struct s3c_dma_params sound/soc/samsung/ac97.c | 28 +++++++++++----------- sound/soc/samsung/dma.h | 20 +++------------- sound/soc/samsung/dmaengine.c | 31 ------------------------ sound/soc/samsung/i2s.c | 48 +++++++++++++++++++------------------- sound/soc/samsung/idma.c | 1 - sound/soc/samsung/pcm.c | 26 ++++++++++----------- sound/soc/samsung/s3c-i2s-v2.c | 3 +-- sound/soc/samsung/s3c-i2s-v2.h | 4 ++-- sound/soc/samsung/s3c2412-i2s.c | 24 +++++++++---------- sound/soc/samsung/s3c24xx-i2s.c | 24 +++++++++---------- sound/soc/samsung/smdk_wm8580pcm.c | 1 - sound/soc/samsung/smdk_wm8994pcm.c | 1 - sound/soc/samsung/spdif.c | 12 +++++----- 13 files changed, 86 insertions(+), 137 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Fix headset mic detection problem for two dell machines
by Hui Wang 04 Aug '16

04 Aug '16

One of the machines has ALC255 on it, another one has ALC298 on it. On the machine with the codec ALC298, it also has the speaker volume problem, so we add the fixup chained to ALC298_FIXUP_SPK_VOLUME rather than adding a group of pin definition in the pin quirk table, since the speak volume problem does not happen on other machines yet. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ce5f1ba..574b1b4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5513,6 +5513,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC298_FIXUP_SPK_VOLUME] = { .type = HDA_FIXUP_FUNC, .v.func = alc298_fixup_speaker_volume, + .chained = true, + .chain_id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, }, }; @@ -5836,6 +5838,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1b, 0x01014020}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x14, 0x90170130}, + {0x1b, 0x02011020}, + {0x21, 0x0221103f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x14, 0x90170150}, {0x1b, 0x02011020}, {0x21, 0x0221105f}), -- 1.9.1

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[alsa-devel] atmel_ssc_dai, duplex audio support?
by Peter Meerwald-Stadler 03 Aug '16

03 Aug '16

Hello, it appears that atmel_ssc_dai does not support duplex audio, i.e. when first starting a playback stream and then starting a record stream, the playback stops problem seems to be in atmel_ssc_startup(): /* Reset the SSC to keep it at a clean status */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); which is performed unconditionally when a stream starts code was introduced with ASoC: atmel_ssc_dai: refactor the startup and shutdown, commit cbaadf0f90d6 tested with the 4.1 branch of linux-at91 git thanks, regards, p. -- Peter Meerwald-Stadler +43-664-2444418 (mobile)

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[alsa-devel] [PATCH] ALSA: hda: Fix krealloc() with __GFP_ZERO usage
by Takashi Iwai 03 Aug '16

03 Aug '16

krealloc() doesn't work always properly with __GFP_ZERO flag as expected. For clearing the reallocated area, we need to clear explicitly instead. Reported-by: Joe Perches <joe(a)perches.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/array.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/hda/array.c b/sound/hda/array.c index 516795baa7db..5dfa610e4471 100644 --- a/sound/hda/array.c +++ b/sound/hda/array.c @@ -21,13 +21,15 @@ void *snd_array_new(struct snd_array *array) return NULL; if (array->used >= array->alloced) { int num = array->alloced + array->alloc_align; + int oldsize = array->alloced * array->elem_size; int size = (num + 1) * array->elem_size; void *nlist; if (snd_BUG_ON(num >= 4096)) return NULL; - nlist = krealloc(array->list, size, GFP_KERNEL | __GFP_ZERO); + nlist = krealloc(array->list, size, GFP_KERNEL); if (!nlist) return NULL; + memset(nlist + oldsize, 0, size - oldsize); array->list = nlist; array->alloced = num; } -- 2.9.2

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[alsa-devel] [PATCH] ALSA: hda: add AMD Bonaire AZ PCI ID with proper driver caps
by Maruthi Srinivas Bayyavarapu 03 Aug '16

03 Aug '16

This commit fixes garbled audio on Bonaire HDMI Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6f8ea13..89dacf9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2265,6 +2265,8 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, /* ATI HDMI */ + { PCI_DEVICE(0x1002, 0x0002), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x1308), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0x157a), -- 1.9.1

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[alsa-devel] [PATCH] sound: Fix empty menuconfig SND_ARM
by Eugeniu Rosca 03 Aug '16

03 Aug '16

commit 3c8f7710c1c4 ("ASoC: fix broken pxa SoC support") has removed the SND_ARM dependency from SND_PXA2XX_LIB and SND_PXA2XX_LIB_AC97, by moving these config entries outside of the "if SND_ARM ... endif" construct. However, by placing these 2 symbols right between the SND_ARM menuconfig definition and the first SND_ARM menu entry, the side effect is that the SND_ARM menu becomes empty and all the config entries caught between "if SND_ARM ... endif" no more belong to menuconfig SND_ARM, but to its parent (menuconfig SND). Fix this. Signed-off-by: Eugeniu Rosca <rosca.eugeniu(a)gmail.com> --- sound/arm/Kconfig | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index e040621..65171f6 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -9,14 +9,6 @@ menuconfig SND_ARM Drivers that are implemented on ASoC can be found in "ALSA for SoC audio support" section. -config SND_PXA2XX_LIB - tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 - select SND_DMAENGINE_PCM - -config SND_PXA2XX_LIB_AC97 - bool - if SND_ARM config SND_ARMAACI @@ -42,3 +34,10 @@ config SND_PXA2XX_AC97 endif # SND_ARM +config SND_PXA2XX_LIB + tristate + select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 + select SND_DMAENGINE_PCM + +config SND_PXA2XX_LIB_AC97 + bool -- 2.9.2

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[alsa-devel] [PATCH v2] ASoC: simple-card-utils: add missing MODULE_xxx()
by Kuninori Morimoto 03 Aug '16

03 Aug '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> simple-card-utils might be used as module, but MODULE_xxx() information was missed. This patch adds it. Otherwise, we will have below error, and can't use it. Specil thanks to Kevin. > insmod simple-card-utils.ko simple_card_utils: module license 'unspecified' taints kernel. Disabling lock debugging due to kernel taint simple_card_utils: Unknown symbol snd_soc_of_parse_daifmt (err 0) simple_card_utils: Unknown symbol snd_soc_of_parse_card_name (err 0) insmod: can't insert 'simple-card-utils.ko': \ unknown symbol in module, or unknown parameter Reported-by: Kevin Hilman <khilman(a)baylibre.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - remove MODULE_ALIAS() sound/soc/generic/Makefile | 6 +++--- sound/soc/generic/simple-card-utils.c | 6 ++++++ 2 files changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 45602ca..2d53c8d 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,5 +1,5 @@ -obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) := simple-card-utils.o - +snd-soc-simple-card-utils-objs := simple-card-utils.o snd-soc-simple-card-objs := simple-card.o -obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o +obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) += snd-soc-simple-card-utils.o +obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index d89a9a1..9599de6 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -7,6 +7,7 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ +#include <linux/module.h> #include <linux/of.h> #include <sound/simple_card_utils.h> @@ -95,3 +96,8 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, return 0; } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); + +/* Module information */ +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); +MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); +MODULE_LICENSE("GPL v2"); -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: cs42l73: Constify cs42l73_mclk_coeffs and cs42l73_mclkx_coeffs tables
by Axel Lin 02 Aug '16

02 Aug '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l73.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 42a8fd4..8524e22 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -794,7 +794,7 @@ struct cs42l73_mclk_div { u8 mmcc; }; -static struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { +static const struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { /* MCLK, Sample Rate, xMMCC[5:0] */ {5644800, 11025, 0x30}, {5644800, 22050, 0x20}, @@ -844,7 +844,7 @@ struct cs42l73_mclkx_div { u8 mclkdiv; }; -static struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { +static const struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { {5644800, 1, 0}, /* 5644800 */ {6000000, 1, 0}, /* 6000000 */ {6144000, 1, 0}, /* 6144000 */ -- 2.7.4

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