May 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/3] ALSA: opl4: Use IS_REACHABLE()
by Takashi Iwai 15 May '17

15 May '17

Rewrite the complex ifdef condition with IS_REACHABLE(). The ifdef in opl4_local.h was without defined(MODLE) check, but this is likely the oversight. Use IS_REACHABLE() here as well. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/drivers/opl4/opl4_lib.c | 4 ++-- sound/drivers/opl4/opl4_local.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index 89c7aa04b3bc..240656e54400 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -153,7 +153,7 @@ static int snd_opl4_detect(struct snd_opl4 *opl4) return 0; } -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_REACHABLE(CONFIG_SND_SEQUENCER) static void snd_opl4_seq_dev_free(struct snd_seq_device *seq_dev) { struct snd_opl4 *opl4 = seq_dev->private_data; @@ -249,7 +249,7 @@ int snd_opl4_create(struct snd_card *card, snd_opl4_create_mixer(opl4); snd_opl4_create_proc(opl4); -#if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) +#if IS_REACHABLE(CONFIG_SND_SEQUENCER) opl4->seq_client = -1; if (opl4->hardware < OPL3_HW_OPL4_ML) snd_opl4_create_seq_dev(opl4, seq_device); diff --git a/sound/drivers/opl4/opl4_local.h b/sound/drivers/opl4/opl4_local.h index 9a41bdebce6b..d5bac93f8245 100644 --- a/sound/drivers/opl4/opl4_local.h +++ b/sound/drivers/opl4/opl4_local.h @@ -184,7 +184,7 @@ struct snd_opl4 { #endif struct mutex access_mutex; -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_REACHABLE(CONFIG_SND_SEQUENCER) int used; int seq_dev_num; -- 2.12.2

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[alsa-devel] [PATCH v2] ASoC: rsnd: tidyup Sampling rate convert example on Document
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "renesas,rsrc-card" is exchanged to "simple-scu-audio-card". Let's update Document Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - simple-scu-card -> simple-scu-audio-card on git log .../devicetree/bindings/sound/renesas,rsnd.txt | 30 +++++++++++----------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 15a7316..3332910 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -83,11 +83,11 @@ SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes ** Asynchronous mode ------------------ -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * SRC Asynchronous mode setting @@ -97,12 +97,12 @@ example) * Inputed 48kHz data will be converted to * system specified Hz */ - convert-rate = <48000>; + simple-audio-card,convert-rate = <48000>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -141,23 +141,23 @@ For more detail information, see below ${LINUX}/sound/soc/sh/rcar/ctu.c - comment of header -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * CTU setting * All input data will be converted to 2ch * as output data */ - convert-channels = <2>; + simple-audio-card,convert-channels = <2>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -190,22 +190,22 @@ and these sounds will be merged by MIX. aplay -D plughw:0,0 xxxx.wav & aplay -D plughw:0,1 yyyy.wav -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. Ex) [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] | [MEM] -> [SRC2] -> [CTU03] -+ sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... - cpu@0 { + simple-audio-card,cpu@0 { sound-dai = <&rcar_sound 0>; }; - cpu@1 { + simple-audio-card,cpu@1 { sound-dai = <&rcar_sound 1>; }; - codec { + simple-audio-card,codec { ... }; }; -- 1.9.1

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[alsa-devel] [PATCH 1/2] ALSA: Use IS_ENABLED() in common headers
by Takashi Iwai 15 May '17

15 May '17

Simplify the ifdef conditions with IS_ENABLED() macro in the common sound headers. No functional changes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/core.h | 6 +++--- include/sound/mixer_oss.h | 2 +- include/sound/opl3.h | 4 ++-- include/sound/pcm.h | 10 +++++----- include/sound/rawmidi.h | 4 ++-- 5 files changed, 13 insertions(+), 13 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index f7d8c10c4c45..55385588eefa 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -142,7 +142,7 @@ struct snd_card { wait_queue_head_t power_sleep; #endif -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) struct snd_mixer_oss *mixer_oss; int mixer_oss_change_count; #endif @@ -243,7 +243,7 @@ int copy_from_user_toio(volatile void __iomem *dst, const void __user *src, size extern struct snd_card *snd_cards[SNDRV_CARDS]; int snd_card_locked(int card); -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) #define SND_MIXER_OSS_NOTIFY_REGISTER 0 #define SND_MIXER_OSS_NOTIFY_DISCONNECT 1 #define SND_MIXER_OSS_NOTIFY_FREE 2 @@ -394,7 +394,7 @@ static inline void snd_printdd(const char *format, ...) {} #define SNDRV_OSS_VERSION ((3<<16)|(8<<8)|(1<<4)|(0)) /* 3.8.1a */ /* for easier backward-porting */ -#if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE) +#if IS_ENABLED(CONFIG_GAMEPORT) #define gameport_set_dev_parent(gp,xdev) ((gp)->dev.parent = (xdev)) #define gameport_set_port_data(gp,r) ((gp)->port_data = (r)) #define gameport_get_port_data(gp) (gp)->port_data diff --git a/include/sound/mixer_oss.h b/include/sound/mixer_oss.h index 13cb0b430a1b..930da10fb65b 100644 --- a/include/sound/mixer_oss.h +++ b/include/sound/mixer_oss.h @@ -22,7 +22,7 @@ * */ -#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_MIXER_OSS) #define SNDRV_OSS_MAX_MIXERS 32 diff --git a/include/sound/opl3.h b/include/sound/opl3.h index 6ba670707831..5a87641c6ae4 100644 --- a/include/sound/opl3.h +++ b/include/sound/opl3.h @@ -321,7 +321,7 @@ struct snd_opl3 { unsigned char fm_mode; /* OPL mode, see SNDRV_DM_FM_MODE_XXX */ unsigned char rhythm; /* percussion mode flag */ unsigned char max_voices; /* max number of voices */ -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) #define SNDRV_OPL3_MODE_SYNTH 0 /* OSS - voices allocated by application */ #define SNDRV_OPL3_MODE_SEQ 1 /* ALSA - driver handles voice allocation */ int synth_mode; /* synth mode */ @@ -374,7 +374,7 @@ int snd_opl3_release(struct snd_hwdep * hw, struct file *file); void snd_opl3_reset(struct snd_opl3 * opl3); -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) long snd_opl3_write(struct snd_hwdep *hw, const char __user *buf, long count, loff_t *offset); int snd_opl3_load_patch(struct snd_opl3 *opl3, diff --git a/include/sound/pcm.h b/include/sound/pcm.h index eb16912d6046..3f084486868f 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -34,7 +34,7 @@ #define snd_pcm_substream_chip(substream) ((substream)->private_data) #define snd_pcm_chip(pcm) ((pcm)->private_data) -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) #include <sound/pcm_oss.h> #endif @@ -418,7 +418,7 @@ struct snd_pcm_runtime { struct snd_pcm_audio_tstamp_report audio_tstamp_report; struct timespec driver_tstamp; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; #endif @@ -464,7 +464,7 @@ struct snd_pcm_substream { unsigned int f_flags; void (*pcm_release)(struct snd_pcm_substream *); struct pid *pid; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) /* -- OSS things -- */ struct snd_pcm_oss_substream oss; #endif @@ -494,7 +494,7 @@ struct snd_pcm_str { unsigned int substream_count; unsigned int substream_opened; struct snd_pcm_substream *substream; -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) /* -- OSS things -- */ struct snd_pcm_oss_stream oss; #endif @@ -526,7 +526,7 @@ struct snd_pcm { void (*private_free) (struct snd_pcm *pcm); bool internal; /* pcm is for internal use only */ bool nonatomic; /* whole PCM operations are in non-atomic context */ -#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) +#if IS_ENABLED(CONFIG_SND_PCM_OSS) struct snd_pcm_oss oss; #endif }; diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 492a3ca7f17b..6665cb29e1a2 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -30,7 +30,7 @@ #include <linux/workqueue.h> #include <linux/device.h> -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) #include <sound/seq_device.h> #endif @@ -144,7 +144,7 @@ struct snd_rawmidi { struct snd_info_entry *proc_entry; -#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) +#if IS_ENABLED(CONFIG_SND_SEQUENCER) struct snd_seq_device *seq_dev; #endif }; -- 2.12.2

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[alsa-devel] [PATCH] ALSA: pcm: Build pcm notifier code conditionally
by Takashi Iwai 15 May '17

15 May '17

The PCM notifier code is used only by OSS emulation layer, so we can build it conditionally for reducing the size. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm.h | 15 ++++++++------- sound/core/pcm.c | 33 ++++++++++++++++++++------------- 2 files changed, 28 insertions(+), 20 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 361749e60799..eb16912d6046 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -531,13 +531,6 @@ struct snd_pcm { #endif }; -struct snd_pcm_notify { - int (*n_register) (struct snd_pcm * pcm); - int (*n_disconnect) (struct snd_pcm * pcm); - int (*n_unregister) (struct snd_pcm * pcm); - struct list_head list; -}; - /* * Registering */ @@ -552,7 +545,15 @@ int snd_pcm_new_internal(struct snd_card *card, const char *id, int device, struct snd_pcm **rpcm); int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count); +#if IS_ENABLED(CONFIG_SND_PCM_OSS) +struct snd_pcm_notify { + int (*n_register) (struct snd_pcm * pcm); + int (*n_disconnect) (struct snd_pcm * pcm); + int (*n_unregister) (struct snd_pcm * pcm); + struct list_head list; +}; int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree); +#endif /* * Native I/O diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8e980aa678d0..24acbfb5c531 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -36,8 +36,10 @@ MODULE_DESCRIPTION("Midlevel PCM code for ALSA."); MODULE_LICENSE("GPL"); static LIST_HEAD(snd_pcm_devices); -static LIST_HEAD(snd_pcm_notify_list); static DEFINE_MUTEX(register_mutex); +#if IS_ENABLED(CONFIG_SND_PCM_OSS) +static LIST_HEAD(snd_pcm_notify_list); +#endif static int snd_pcm_free(struct snd_pcm *pcm); static int snd_pcm_dev_free(struct snd_device *device); @@ -884,16 +886,23 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) put_device(&pstr->dev); } +#if IS_ENABLED(CONFIG_SND_PCM_OSS) +#define pcm_call_notify(pcm, call) \ + do { \ + struct snd_pcm_notify *_notify; \ + list_for_each_entry(_notify, &snd_pcm_notify_list, list) \ + _notify->call(pcm); \ + } while (0) +#else +#define pcm_call_notify(pcm, call) /* NOP */ +#endif + static int snd_pcm_free(struct snd_pcm *pcm) { - struct snd_pcm_notify *notify; - if (!pcm) return 0; - if (!pcm->internal) { - list_for_each_entry(notify, &snd_pcm_notify_list, list) - notify->n_unregister(pcm); - } + if (!pcm->internal) + pcm_call_notify(pcm, n_unregister); if (pcm->private_free) pcm->private_free(pcm); snd_pcm_lib_preallocate_free_for_all(pcm); @@ -1069,7 +1078,6 @@ static int snd_pcm_dev_register(struct snd_device *device) { int cidx, err; struct snd_pcm_substream *substream; - struct snd_pcm_notify *notify; struct snd_pcm *pcm; if (snd_BUG_ON(!device || !device->device_data)) @@ -1107,8 +1115,7 @@ static int snd_pcm_dev_register(struct snd_device *device) snd_pcm_timer_init(substream); } - list_for_each_entry(notify, &snd_pcm_notify_list, list) - notify->n_register(pcm); + pcm_call_notify(pcm, n_register); unlock: mutex_unlock(&register_mutex); @@ -1118,7 +1125,6 @@ static int snd_pcm_dev_register(struct snd_device *device) static int snd_pcm_dev_disconnect(struct snd_device *device) { struct snd_pcm *pcm = device->device_data; - struct snd_pcm_notify *notify; struct snd_pcm_substream *substream; int cidx; @@ -1138,8 +1144,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) } } if (!pcm->internal) { - list_for_each_entry(notify, &snd_pcm_notify_list, list) - notify->n_disconnect(pcm); + pcm_call_notify(pcm, n_disconnect); } for (cidx = 0; cidx < 2; cidx++) { if (!pcm->internal) @@ -1151,6 +1156,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) return 0; } +#if IS_ENABLED(CONFIG_SND_PCM_OSS) /** * snd_pcm_notify - Add/remove the notify list * @notify: PCM notify list @@ -1183,6 +1189,7 @@ int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree) return 0; } EXPORT_SYMBOL(snd_pcm_notify); +#endif /* CONFIG_SND_PCM_OSS */ #ifdef CONFIG_SND_PROC_FS /* -- 2.12.2

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[alsa-devel] [PATCH 1/3] ALSA: vx222: Use container_of()
by Takashi Iwai 15 May '17

15 May '17

The vx222 driver is using the explicit cast from the parent class pointer, but it'll be broken when the structure field randomization is applied. Use container_of() in a modern manner, instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/vx222/vx222.c | 4 ++-- sound/pci/vx222/vx222.h | 2 ++ sound/pci/vx222/vx222_ops.c | 26 +++++++++++++------------- 3 files changed, 17 insertions(+), 15 deletions(-) diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index ecbaf473fc1e..55861849d7df 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -116,7 +116,7 @@ static struct snd_vx_hardware vx222_mic_hw = { */ static int snd_vx222_free(struct vx_core *chip) { - struct snd_vx222 *vx = (struct snd_vx222 *)chip; + struct snd_vx222 *vx = to_vx222(chip); if (chip->irq >= 0) free_irq(chip->irq, (void*)chip); @@ -158,7 +158,7 @@ static int snd_vx222_create(struct snd_card *card, struct pci_dev *pci, pci_disable_device(pci); return -ENOMEM; } - vx = (struct snd_vx222 *)chip; + vx = to_vx222(chip); vx->pci = pci; if ((err = pci_request_regions(pci, CARD_NAME)) < 0) { diff --git a/sound/pci/vx222/vx222.h b/sound/pci/vx222/vx222.h index 2f0d78f609a6..cae355c8ed28 100644 --- a/sound/pci/vx222/vx222.h +++ b/sound/pci/vx222/vx222.h @@ -39,6 +39,8 @@ struct snd_vx222 { int mic_level; /* mic level for vx222 mic */ }; +#define to_vx222(x) container_of(x, struct snd_vx222, core) + /* we use a lookup table with 148 values, see vx_mixer.c */ #define VX2_AKM_LEVEL_MAX 0x93 diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index 7df1663ea510..d4298af6d3ee 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -86,7 +86,7 @@ static int vx2_reg_index[VX_REG_MAX] = { static inline unsigned long vx2_reg_addr(struct vx_core *_chip, int reg) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); return chip->port[vx2_reg_index[reg]] + vx2_reg_offset[reg]; } @@ -159,7 +159,7 @@ static void vx2_outl(struct vx_core *chip, int offset, unsigned int val) static void vx2_reset_dsp(struct vx_core *_chip) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); /* set the reset dsp bit to 0 */ vx_outl(chip, CDSP, chip->regCDSP & ~VX_CDSP_DSP_RESET_MASK); @@ -174,7 +174,7 @@ static void vx2_reset_dsp(struct vx_core *_chip) static int vx2_test_xilinx(struct vx_core *_chip) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); unsigned int data; dev_dbg(_chip->card->dev, "testing xilinx...\n"); @@ -479,7 +479,7 @@ static int vx2_test_and_ack(struct vx_core *chip) */ static void vx2_validate_irq(struct vx_core *_chip, int enable) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); /* Set the interrupt enable bit to 1 in CDSP register */ if (enable) { @@ -730,7 +730,7 @@ static void vx2_old_write_codec_bit(struct vx_core *chip, int codec, unsigned in */ static void vx2_reset_codec(struct vx_core *_chip) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); /* Set the reset CODEC bit to 0. */ vx_outl(chip, CDSP, chip->regCDSP &~ VX_CDSP_CODEC_RESET_MASK); @@ -772,7 +772,7 @@ static void vx2_reset_codec(struct vx_core *_chip) */ static void vx2_change_audio_source(struct vx_core *_chip, int src) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); switch (src) { case VX_AUDIO_SRC_DIGITAL: @@ -791,7 +791,7 @@ static void vx2_change_audio_source(struct vx_core *_chip, int src) */ static void vx2_set_clock_source(struct vx_core *_chip, int source) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); if (source == INTERNAL_QUARTZ) chip->regCFG &= ~VX_CFG_CLOCKIN_SEL_MASK; @@ -805,7 +805,7 @@ static void vx2_set_clock_source(struct vx_core *_chip, int source) */ static void vx2_reset_board(struct vx_core *_chip, int cold_reset) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); /* initialize the register values */ chip->regCDSP = VX_CDSP_CODEC_RESET_MASK | VX_CDSP_DSP_RESET_MASK ; @@ -878,7 +878,7 @@ static int vx_input_level_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ele static int vx_input_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); mutex_lock(&_chip->mixer_mutex); ucontrol->value.integer.value[0] = chip->input_level[0]; ucontrol->value.integer.value[1] = chip->input_level[1]; @@ -889,7 +889,7 @@ static int vx_input_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static int vx_input_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); if (ucontrol->value.integer.value[0] < 0 || ucontrol->value.integer.value[0] > MIC_LEVEL_MAX) return -EINVAL; @@ -922,7 +922,7 @@ static int vx_mic_level_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ static int vx_mic_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); ucontrol->value.integer.value[0] = chip->mic_level; return 0; } @@ -930,7 +930,7 @@ static int vx_mic_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); if (ucontrol->value.integer.value[0] < 0 || ucontrol->value.integer.value[0] > MIC_LEVEL_MAX) return -EINVAL; @@ -973,7 +973,7 @@ static const struct snd_kcontrol_new vx_control_mic_level = { static int vx2_add_mic_controls(struct vx_core *_chip) { - struct snd_vx222 *chip = (struct snd_vx222 *)_chip; + struct snd_vx222 *chip = to_vx222(_chip); int err; if (_chip->type != VX_TYPE_MIC) -- 2.12.2

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[alsa-devel] Headphones not detected when using Dell Dock WD15
by Sergio Villar Senin 15 May '17

15 May '17

Hi, I reported this bug to pulseaudio[1] but after some investigation it looks like the alsa mixer is not working for the dock. I'll be happy to provide any additional info which might be required (attaching alsa-info.sh BTW). Please add me to the Cc as I am not subscribed to this list BR [1] https://bugs.freedesktop.org/show_bug.cgi?id=100711

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[alsa-devel] [PATCH 0/4] ALSA/ASoC: fix after merging ASoC PR for v4.12-rc1
by Takashi Sakamoto 15 May '17

15 May '17

Hi, This patchset is for current 'for-linus' remote branch[1], after merging PR from ALSA SoC part for v4.12-rc1[2]. The first two patch includes bug fixes introduced at former kernel releases, the rest is to suppress sparse. [1] https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?h=f… [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120308.html Takashi Sakamoto (4): ASoC: codecs: da7213: fix invalid usage of bitwise operation in loop condition ASoC: codecs: msm8916: fix invalid cast to bool type ASoC: hisilicon: localize functions without external linkage ASoC: intel: atom: localize variable without external linkage sound/soc/codecs/da7213.c | 14 +++++--------- sound/soc/codecs/msm8916-wcd-analog.c | 6 +++--- sound/soc/hisilicon/hi6210-i2s.c | 11 ++++++----- sound/soc/intel/atom/sst/sst.c | 2 +- 4 files changed, 15 insertions(+), 18 deletions(-) -- 2.9.3

4 12

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Support for one channel (mono) audio
by Peter Ujfalusi 15 May '17

15 May '17

Mono audio can be achieved by configuring McASP to transmit/receive only during one timeslot. McASP will still going to generate clocks for the other slot(s), but will only use the single slot to transmit/receive. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5e02c7a15a83..31b4eac87aa2 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -650,7 +650,7 @@ static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, int stream, if (mcasp->tdm_mask[stream]) slots = hweight32(mcasp->tdm_mask[stream]); - for (i = 2; i <= slots; i++) + for (i = 1; i <= slots; i++) list[count++] = i; for (i = 2; i <= serializers; i++) @@ -1318,7 +1318,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_CHANNELS, - 2, max_channels); + 0, max_channels); snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, @@ -1480,13 +1480,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .suspend = davinci_mcasp_suspend, .resume = davinci_mcasp_resume, .playback = { - .channels_min = 2, + .channels_min = 1, .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 32 * 16, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, @@ -1992,12 +1992,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) */ mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list = devm_kzalloc(mcasp->dev, sizeof(unsigned int) * - (32 + mcasp->num_serializer - 2), + (32 + mcasp->num_serializer - 1), GFP_KERNEL); mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list = devm_kzalloc(mcasp->dev, sizeof(unsigned int) * - (32 + mcasp->num_serializer - 2), + (32 + mcasp->num_serializer - 1), GFP_KERNEL); if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list || -- 2.12.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5514: fix gcc-7 warning
by Arnd Bergmann 15 May '17

15 May '17

gcc-7 warns that there is a duplicate 'const' specifier in some variables that are declared using the SOC_ENUM_SINGLE_DECL macro: sound/soc/codecs/rt5514.c:398:14: error: duplicate 'const' declaration specifier [-Werror=duplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL( sound/soc/codecs/rt5514.c:405:14: error: duplicate 'const' declaration specifier [-Werror=duplicate-decl-specifier] static const SOC_ENUM_SINGLE_DECL( This removes one to fix the warning. Fixes: 4a6180ea7399 ("ASoC: rt5514: add rt5514 codec driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/rt5514.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index f91221b1ddf0..28ab9e2bb051 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -395,14 +395,14 @@ static const char * const rt5514_dmic_src[] = { "DMIC1", "DMIC2" }; -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5514_stereo1_dmic_enum, RT5514_DIG_SOURCE_CTRL, RT5514_AD0_DMIC_INPUT_SEL_SFT, rt5514_dmic_src); static const struct snd_kcontrol_new rt5514_sto1_dmic_mux = SOC_DAPM_ENUM("Stereo1 DMIC Source", rt5514_stereo1_dmic_enum); -static const SOC_ENUM_SINGLE_DECL( +static SOC_ENUM_SINGLE_DECL( rt5514_stereo2_dmic_enum, RT5514_DIG_SOURCE_CTRL, RT5514_AD1_DMIC_INPUT_SEL_SFT, rt5514_dmic_src); -- 2.9.0

2 3

[alsa-devel] [PATCH v2] ASoC: ak4613: disable asymmetric audio interface format
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Asymmetric audio interface formats exist in ak4613 by same register settings. Capture Playback 24bit LEFT_J 16bit RIGHT_J 24bit LEFT_J 20bit RIGHT_J 24bit LEFT_J 24bit RIGHT_J 24bit LEFT_J 24bit LEFT_J 24bit I2S 24bit I2S These asymmetric formats makes driver / behavior difficult. It is not HW limitation, but SW limitation. To makes code reading easy, this patch removes asymmetric format support. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - just remove code instead of comment out sound/soc/codecs/ak4613.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index b2dfdde..557ac16 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -139,9 +139,7 @@ struct ak4613_priv { #define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } static const struct ak4613_interface ak4613_iface[] = { /* capture */ /* playback */ - [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, - [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, - [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, + /* [0] - [2] are not supported */ [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) }, [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) }, }; @@ -262,11 +260,9 @@ static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) fmt &= SND_SOC_DAIFMT_FORMAT_MASK; switch (fmt) { - case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_I2S: priv->fmt = fmt; - break; default: return -EINVAL; @@ -286,13 +282,8 @@ static bool ak4613_dai_fmt_matching(const struct ak4613_interface *iface, if (fmts->fmt != fmt) return false; - if (fmt == SND_SOC_DAIFMT_RIGHT_J) { - if (fmts->width != width) - return false; - } else { - if (fmts->width < width) - return false; - } + if (fmts->width != width) + return false; return true; } @@ -420,8 +411,7 @@ static int ak4613_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_96000 |\ SNDRV_PCM_RATE_176400 |\ SNDRV_PCM_RATE_192000) -#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) +#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_dai_driver ak4613_dai = { .name = "ak4613-hifi", -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: Skylake: Move i915 registration to worker thread
by Subhransu S. Prusty 15 May '17

15 May '17

From: Vinod Koul <vinod.koul(a)intel.com> The i915 component framework expects the caller to be invoking snd_hdac_i915_init() from a thread context. Otherwise it results in lockups on drm side. So move the registering of component interface and probing of codecs on this bus to a worker thread. init_failed in skl structure is not used currently, so renamed to init_done and used to track the initialization done in worker thread. Reported-by: Imre Deak <imre.deak(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Sodhi, VunnyX <vunnyx.sodhi(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- sound/soc/intel/skylake/skl.c | 162 +++++++++++++++++++++++------------------- sound/soc/intel/skylake/skl.h | 4 +- 2 files changed, 91 insertions(+), 75 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 6df3b317a476..4c9b5781282b 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -410,7 +410,7 @@ static int skl_free(struct hdac_ext_bus *ebus) struct skl *skl = ebus_to_skl(ebus); struct hdac_bus *bus = ebus_to_hbus(ebus); - skl->init_failed = 1; /* to be sure */ + skl->init_done = 0; /* to be sure */ snd_hdac_ext_stop_streams(ebus); @@ -428,8 +428,10 @@ static int skl_free(struct hdac_ext_bus *ebus) snd_hdac_ext_bus_exit(ebus); + cancel_work_sync(&skl->probe_work); if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) snd_hdac_i915_exit(&ebus->bus); + return 0; } @@ -566,6 +568,84 @@ static int skl_codec_create(struct hdac_ext_bus *ebus) .get_response = snd_hdac_bus_get_response, }; +static int skl_i915_init(struct hdac_bus *bus) +{ + int err; + + /* + * The HDMI codec is in GPU so we need to ensure that it is powered + * up and ready for probe + */ + err = snd_hdac_i915_init(bus); + if (err < 0) + return err; + + err = snd_hdac_display_power(bus, true); + if (err < 0) + dev_err(bus->dev, "Cannot turn on display power on i915\n"); + + return err; +} + +static void skl_probe_work(struct work_struct *work) +{ + struct skl *skl = container_of(work, struct skl, probe_work); + struct hdac_ext_bus *ebus = &skl->ebus; + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_ext_link *hlink = NULL; + int err; + + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + err = skl_i915_init(bus); + if (err < 0) + return; + } + + err = skl_init_chip(bus, true); + if (err < 0) { + dev_err(bus->dev, "Init chip failed with err: %d\n", err); + goto out_err; + } + + /* codec detection */ + if (!bus->codec_mask) + dev_info(bus->dev, "no hda codecs found!\n"); + + /* create codec instances */ + err = skl_codec_create(ebus); + if (err < 0) + goto out_err; + + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + err = snd_hdac_display_power(bus, false); + if (err < 0) { + dev_err(bus->dev, "Cannot turn off display power on i915\n"); + return; + } + } + + /* register platform dai and controls */ + err = skl_platform_register(bus->dev); + if (err < 0) + return; + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) + snd_hdac_ext_bus_link_put(ebus, hlink); + + /* configure PM */ + pm_runtime_put_noidle(bus->dev); + pm_runtime_allow(bus->dev); + skl->init_done = 1; + + return; + +out_err: + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) + err = snd_hdac_display_power(bus, false); +} + /* * constructor */ @@ -593,6 +673,7 @@ static int skl_create(struct pci_dev *pci, snd_hdac_ext_bus_init(ebus, &pci->dev, &bus_core_ops, io_ops); ebus->bus.use_posbuf = 1; skl->pci = pci; + INIT_WORK(&skl->probe_work, skl_probe_work); ebus->bus.bdl_pos_adj = 0; @@ -601,27 +682,6 @@ static int skl_create(struct pci_dev *pci, return 0; } -static int skl_i915_init(struct hdac_bus *bus) -{ - int err; - - /* - * The HDMI codec is in GPU so we need to ensure that it is powered - * up and ready for probe - */ - err = snd_hdac_i915_init(bus); - if (err < 0) - return err; - - err = snd_hdac_display_power(bus, true); - if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); - return err; - } - - return err; -} - static int skl_first_init(struct hdac_ext_bus *ebus) { struct skl *skl = ebus_to_skl(ebus); @@ -684,20 +744,7 @@ static int skl_first_init(struct hdac_ext_bus *ebus) /* initialize chip */ skl_init_pci(skl); - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = skl_i915_init(bus); - if (err < 0) - return err; - } - - skl_init_chip(bus, true); - - /* codec detection */ - if (!bus->codec_mask) { - dev_info(bus->dev, "no hda codecs found!\n"); - } - - return 0; + return skl_init_chip(bus, true); } static int skl_probe(struct pci_dev *pci, @@ -706,7 +753,6 @@ static int skl_probe(struct pci_dev *pci, struct skl *skl; struct hdac_ext_bus *ebus = NULL; struct hdac_bus *bus = NULL; - struct hdac_ext_link *hlink = NULL; int err; /* we use ext core ops, so provide NULL for ops here */ @@ -729,7 +775,7 @@ static int skl_probe(struct pci_dev *pci, if (skl->nhlt == NULL) { err = -ENODEV; - goto out_display_power_off; + goto out_free; } err = skl_nhlt_create_sysfs(skl); @@ -760,56 +806,24 @@ static int skl_probe(struct pci_dev *pci, if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(ebus); + snd_hdac_bus_stop_chip(bus); + /* create device for soc dmic */ err = skl_dmic_device_register(skl); if (err < 0) goto out_dsp_free; - /* register platform dai and controls */ - err = skl_platform_register(bus->dev); - if (err < 0) - goto out_dmic_free; - - /* create codec instances */ - err = skl_codec_create(ebus); - if (err < 0) - goto out_unregister; - - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(bus->dev, "Cannot turn off display power on i915\n"); - return err; - } - } - - /* - * we are done probling so decrement link counts - */ - list_for_each_entry(hlink, &ebus->hlink_list, list) - snd_hdac_ext_bus_link_put(ebus, hlink); - - /* configure PM */ - pm_runtime_put_noidle(bus->dev); - pm_runtime_allow(bus->dev); + schedule_work(&skl->probe_work); return 0; -out_unregister: - skl_platform_unregister(bus->dev); -out_dmic_free: - skl_dmic_device_unregister(skl); out_dsp_free: skl_free_dsp(skl); out_mach_free: skl_machine_device_unregister(skl); out_nhlt_free: skl_nhlt_free(skl->nhlt); -out_display_power_off: - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, false); out_free: - skl->init_failed = 1; skl_free(ebus); return err; @@ -828,7 +842,7 @@ static void skl_shutdown(struct pci_dev *pci) skl = ebus_to_skl(ebus); - if (skl->init_failed) + if (!skl->init_done) return; snd_hdac_ext_stop_streams(ebus); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 5b3fa4b91691..c1968a06174a 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -67,7 +67,7 @@ struct skl { struct hdac_ext_bus ebus; struct pci_dev *pci; - unsigned int init_failed:1; /* delayed init failed */ + unsigned int init_done:1; /* delayed init status */ struct platform_device *dmic_dev; struct platform_device *i2s_dev; struct snd_soc_platform *platform; @@ -85,6 +85,8 @@ struct skl { const struct firmware *tplg; int supend_active; + + struct work_struct probe_work; }; #define skl_to_ebus(s) (&(s)->ebus) -- 1.9.1

3 5

[alsa-devel] [PATCH v2 1/2] ASoC: Intel: Skylake: Move i915 registration to worker thread
by Subhransu S. Prusty 15 May '17

15 May '17

From: Vinod Koul <vinod.koul(a)intel.com> The i915 component framework expects the caller to be invoking snd_hdac_i915_init() from a thread context. Otherwise it results in lockups on drm side. So move the registering of component interface and probing of codecs on this bus to a worker thread. init_failed in skl structure is not used currently, so renamed to init_done and used to track the initialization done in worker thread. Reported-by: Imre Deak <imre.deak(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Sodhi, VunnyX <vunnyx.sodhi(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> --- v1->v2 - no change sound/soc/intel/skylake/skl.c | 162 +++++++++++++++++++++++------------------- sound/soc/intel/skylake/skl.h | 4 +- 2 files changed, 91 insertions(+), 75 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 6df3b317a476..4c9b5781282b 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -410,7 +410,7 @@ static int skl_free(struct hdac_ext_bus *ebus) struct skl *skl = ebus_to_skl(ebus); struct hdac_bus *bus = ebus_to_hbus(ebus); - skl->init_failed = 1; /* to be sure */ + skl->init_done = 0; /* to be sure */ snd_hdac_ext_stop_streams(ebus); @@ -428,8 +428,10 @@ static int skl_free(struct hdac_ext_bus *ebus) snd_hdac_ext_bus_exit(ebus); + cancel_work_sync(&skl->probe_work); if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) snd_hdac_i915_exit(&ebus->bus); + return 0; } @@ -566,6 +568,84 @@ static int skl_codec_create(struct hdac_ext_bus *ebus) .get_response = snd_hdac_bus_get_response, }; +static int skl_i915_init(struct hdac_bus *bus) +{ + int err; + + /* + * The HDMI codec is in GPU so we need to ensure that it is powered + * up and ready for probe + */ + err = snd_hdac_i915_init(bus); + if (err < 0) + return err; + + err = snd_hdac_display_power(bus, true); + if (err < 0) + dev_err(bus->dev, "Cannot turn on display power on i915\n"); + + return err; +} + +static void skl_probe_work(struct work_struct *work) +{ + struct skl *skl = container_of(work, struct skl, probe_work); + struct hdac_ext_bus *ebus = &skl->ebus; + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_ext_link *hlink = NULL; + int err; + + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + err = skl_i915_init(bus); + if (err < 0) + return; + } + + err = skl_init_chip(bus, true); + if (err < 0) { + dev_err(bus->dev, "Init chip failed with err: %d\n", err); + goto out_err; + } + + /* codec detection */ + if (!bus->codec_mask) + dev_info(bus->dev, "no hda codecs found!\n"); + + /* create codec instances */ + err = skl_codec_create(ebus); + if (err < 0) + goto out_err; + + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + err = snd_hdac_display_power(bus, false); + if (err < 0) { + dev_err(bus->dev, "Cannot turn off display power on i915\n"); + return; + } + } + + /* register platform dai and controls */ + err = skl_platform_register(bus->dev); + if (err < 0) + return; + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) + snd_hdac_ext_bus_link_put(ebus, hlink); + + /* configure PM */ + pm_runtime_put_noidle(bus->dev); + pm_runtime_allow(bus->dev); + skl->init_done = 1; + + return; + +out_err: + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) + err = snd_hdac_display_power(bus, false); +} + /* * constructor */ @@ -593,6 +673,7 @@ static int skl_create(struct pci_dev *pci, snd_hdac_ext_bus_init(ebus, &pci->dev, &bus_core_ops, io_ops); ebus->bus.use_posbuf = 1; skl->pci = pci; + INIT_WORK(&skl->probe_work, skl_probe_work); ebus->bus.bdl_pos_adj = 0; @@ -601,27 +682,6 @@ static int skl_create(struct pci_dev *pci, return 0; } -static int skl_i915_init(struct hdac_bus *bus) -{ - int err; - - /* - * The HDMI codec is in GPU so we need to ensure that it is powered - * up and ready for probe - */ - err = snd_hdac_i915_init(bus); - if (err < 0) - return err; - - err = snd_hdac_display_power(bus, true); - if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); - return err; - } - - return err; -} - static int skl_first_init(struct hdac_ext_bus *ebus) { struct skl *skl = ebus_to_skl(ebus); @@ -684,20 +744,7 @@ static int skl_first_init(struct hdac_ext_bus *ebus) /* initialize chip */ skl_init_pci(skl); - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = skl_i915_init(bus); - if (err < 0) - return err; - } - - skl_init_chip(bus, true); - - /* codec detection */ - if (!bus->codec_mask) { - dev_info(bus->dev, "no hda codecs found!\n"); - } - - return 0; + return skl_init_chip(bus, true); } static int skl_probe(struct pci_dev *pci, @@ -706,7 +753,6 @@ static int skl_probe(struct pci_dev *pci, struct skl *skl; struct hdac_ext_bus *ebus = NULL; struct hdac_bus *bus = NULL; - struct hdac_ext_link *hlink = NULL; int err; /* we use ext core ops, so provide NULL for ops here */ @@ -729,7 +775,7 @@ static int skl_probe(struct pci_dev *pci, if (skl->nhlt == NULL) { err = -ENODEV; - goto out_display_power_off; + goto out_free; } err = skl_nhlt_create_sysfs(skl); @@ -760,56 +806,24 @@ static int skl_probe(struct pci_dev *pci, if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(ebus); + snd_hdac_bus_stop_chip(bus); + /* create device for soc dmic */ err = skl_dmic_device_register(skl); if (err < 0) goto out_dsp_free; - /* register platform dai and controls */ - err = skl_platform_register(bus->dev); - if (err < 0) - goto out_dmic_free; - - /* create codec instances */ - err = skl_codec_create(ebus); - if (err < 0) - goto out_unregister; - - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(bus->dev, "Cannot turn off display power on i915\n"); - return err; - } - } - - /* - * we are done probling so decrement link counts - */ - list_for_each_entry(hlink, &ebus->hlink_list, list) - snd_hdac_ext_bus_link_put(ebus, hlink); - - /* configure PM */ - pm_runtime_put_noidle(bus->dev); - pm_runtime_allow(bus->dev); + schedule_work(&skl->probe_work); return 0; -out_unregister: - skl_platform_unregister(bus->dev); -out_dmic_free: - skl_dmic_device_unregister(skl); out_dsp_free: skl_free_dsp(skl); out_mach_free: skl_machine_device_unregister(skl); out_nhlt_free: skl_nhlt_free(skl->nhlt); -out_display_power_off: - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, false); out_free: - skl->init_failed = 1; skl_free(ebus); return err; @@ -828,7 +842,7 @@ static void skl_shutdown(struct pci_dev *pci) skl = ebus_to_skl(ebus); - if (skl->init_failed) + if (!skl->init_done) return; snd_hdac_ext_stop_streams(ebus); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 5b3fa4b91691..c1968a06174a 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -67,7 +67,7 @@ struct skl { struct hdac_ext_bus ebus; struct pci_dev *pci; - unsigned int init_failed:1; /* delayed init failed */ + unsigned int init_done:1; /* delayed init status */ struct platform_device *dmic_dev; struct platform_device *i2s_dev; struct snd_soc_platform *platform; @@ -85,6 +85,8 @@ struct skl { const struct firmware *tplg; int supend_active; + + struct work_struct probe_work; }; #define skl_to_ebus(s) (&(s)->ebus) -- 1.9.1

3 3

[alsa-devel] [PATCH] rt286: add Thinkpad Helix 2 to force_combo_jack_table
by Yifeng Li 15 May '17

15 May '17

Thinkpad Helix 2 is a tablet PC, the audio is powered by Core M broadwell-audio and rt286 codec. For all versions of Linux kernel, the stereo output doesn't work properly when earphones are plugged in, the sound was coming out from both channels even if the audio contains only the left or right channel. Furthermore, if a music recorded in stereo is played, the two channels cancle out each other out, as a result, no voice but only distorted background music can be heard, like a sound card with builtin a Karaoke sount effect. Apparently this tablet uses a combo jack with polarity incorrectly set by rt286 driver. This patch adds DMI information of Thinkpad Helix 2 to force_combo_jack_table[] and the issue is resolved. The microphone input doesn't work regardless to the presence of this patch and still needs help from other developers to investigate. This is my first patch to LKML directly, sorry for CC-ing too many people here. Link: https://bugzilla.kernel.org/show_bug.cgi?id=93841 Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Keyon <yang.jie(a)linux.intel.com> Cc: Jinliang Wang <jinliangx.wang(a)intel.com> Cc: stable(a)vger.kernel.org Signed-off-by: Yifeng Li <tomli(a)tomli.me> --- sound/soc/codecs/rt286.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9c365a7f758d..7899a2cdeb42 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1108,6 +1108,13 @@ static const struct dmi_system_id force_combo_jack_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Kabylake Client platform") } }, + { + .ident = "Thinkpad Helix 2nd", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_VERSION, "ThinkPad Helix 2nd") + } + }, { } }; -- 2.12.1

2 1

[alsa-devel] [PATCH] ASoC: da7213: Fix incorrect usage of bitwise '&' operator for SRM check
by Adam Thomson 15 May '17

15 May '17

In the SRM lock check section of code the '&' bitwise operator is used as part of checking lock status. Functionally the code works as intended, but the conditional statement is a boolean comparison so should really use '&&' logical operator instead. This commit rectifies this discrepancy. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7213.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 12da558..d114f33 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -771,7 +771,7 @@ static int da7213_dai_event(struct snd_soc_dapm_widget *w, ++i; msleep(50); } - } while ((i < DA7213_SRM_CHECK_RETRIES) & (!srm_lock)); + } while ((i < DA7213_SRM_CHECK_RETRIES) && (!srm_lock)); if (!srm_lock) dev_warn(codec->dev, "SRM failed to lock\n"); -- 1.9.1

2 1

[alsa-devel] Applied "ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx codec" to the asoc tree
by Mark Brown 15 May '17

15 May '17

The patch ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx codec has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 51827c41c9ce07293b094691673e6ec23dfdc5e8 Mon Sep 17 00:00:00 2001 From: Tomas Vilda <tomas.vilda(a)gmail.com> Date: Sat, 13 May 2017 00:29:37 +0300 Subject: [PATCH] ASoC: tlv320dac31xx: Fix mistype in tlv320dac31xx codec Signed-off-by: Tomas Vilda <tomas.vilda(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index f8a90ba8cd71..d7d03c92cb8a 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1210,7 +1210,7 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = { static struct snd_soc_dai_driver dac31xx_dai_driver[] = { { - .name = "tlv32dac31xx-hifi", + .name = "tlv320dac31xx-hifi", .playback = { .stream_name = "Playback", .channels_min = 2, -- 2.11.0

1 0

[alsa-devel] [PATCH][V2] ASoC: Intel: sst: fix spelling mistake: "allocationf" -> "allocation"
by Colin King 15 May '17

15 May '17

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message. Also replace "fail" with "failure". Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 21cac1c8dd4c..e74119113713 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -690,7 +690,7 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { - dev_err(rtd->dev, "dma buffer allocationf fail\n"); + dev_err(rtd->dev, "dma buffer allocation failure\n"); return retval; } } -- 2.11.0

3 2

[alsa-devel] [PATCH] ASoC: nau8824: TDM support
by John Hsu 15 May '17

15 May '17

Support TDM format for NAU88L24. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: John Hsu <supercraig0719(a)gmail.com> --- sound/soc/codecs/nau8824.c | 52 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8824.h | 12 +++++++++++ 2 files changed, 64 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index cca974d..3a309b1 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1125,6 +1125,57 @@ static int nau8824_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /** + * nau8824_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @tx_mask: Bitmask representing active TX slots. Ex. + * 0xf for normal 4 channel TDM. + * 0xf0 for shifted 4 channel TDM + * @rx_mask: Bitmask [0:1] representing active DACR RX slots. + * Bitmask [2:3] representing active DACL RX slots. + * 00=CH0,01=CH1,10=CH2,11=CH3. Ex. + * 0xf for DACL/R selecting TDM CH3. + * 0xf0 for DACL/R selecting shifted TDM CH3. + * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. + * + * Configures a DAI for TDM operation. Only support 4 slots TDM. + */ +static int nau8824_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8824 *nau8824 = snd_soc_codec_get_drvdata(codec); + unsigned int tslot_l = 0, ctrl_val = 0; + + if (slots > 4 || ((tx_mask & 0xf0) && (tx_mask & 0xf)) || + ((rx_mask & 0xf0) && (rx_mask & 0xf)) || + ((rx_mask & 0xf0) && (tx_mask & 0xf)) || + ((rx_mask & 0xf) && (tx_mask & 0xf0))) + return -EINVAL; + + ctrl_val |= (NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN); + if (tx_mask & 0xf0) { + tslot_l = 4 * slot_width; + ctrl_val |= (tx_mask >> 4); + } else { + ctrl_val |= tx_mask; + } + if (rx_mask & 0xf0) + ctrl_val |= ((rx_mask >> 4) << NAU8824_TDM_DACR_RX_SFT); + else + ctrl_val |= (rx_mask << NAU8824_TDM_DACR_RX_SFT); + + regmap_update_bits(nau8824->regmap, NAU8824_REG_TDM_CTRL, + NAU8824_TDM_MODE | NAU8824_TDM_OFFSET_EN | + NAU8824_TDM_DACL_RX_MASK | NAU8824_TDM_DACR_RX_MASK | + NAU8824_TDM_TX_MASK, ctrl_val); + regmap_update_bits(nau8824->regmap, NAU8824_REG_PORT0_LEFT_TIME_SLOT, + NAU8824_TSLOT_L_MASK, tslot_l); + + return 0; +} + +/** * nau8824_calc_fll_param - Calculate FLL parameters. * @fll_in: external clock provided to codec. * @fs: sampling rate. @@ -1440,6 +1491,7 @@ static struct snd_soc_codec_driver nau8824_codec_driver = { static const struct snd_soc_dai_ops nau8824_dai_ops = { .hw_params = nau8824_hw_params, .set_fmt = nau8824_set_fmt, + .set_tdm_slot = nau8824_set_tdm_slot, }; #define NAU8824_RATES SNDRV_PCM_RATE_8000_192000 diff --git a/sound/soc/codecs/nau8824.h b/sound/soc/codecs/nau8824.h index 87ac9a3..21eae24 100644 --- a/sound/soc/codecs/nau8824.h +++ b/sound/soc/codecs/nau8824.h @@ -258,6 +258,18 @@ #define NAU8824_I2S_MS_SLAVE (0 << NAU8824_I2S_MS_SFT) #define NAU8824_I2S_BLK_DIV_MASK 0x7 +/* PORT0_LEFT_TIME_SLOT (0x1E) */ +#define NAU8824_TSLOT_L_MASK 0x3ff + +/* TDM_CTRL (0x20) */ +#define NAU8824_TDM_MODE (0x1 << 15) +#define NAU8824_TDM_OFFSET_EN (0x1 << 14) +#define NAU8824_TDM_DACL_RX_SFT 6 +#define NAU8824_TDM_DACL_RX_MASK (0x3 << NAU8824_TDM_DACL_RX_SFT) +#define NAU8824_TDM_DACR_RX_SFT 4 +#define NAU8824_TDM_DACR_RX_MASK (0x3 << NAU8824_TDM_DACR_RX_SFT) +#define NAU8824_TDM_TX_MASK 0xf + /* ADC_FILTER_CTRL (0x24) */ #define NAU8824_ADC_SYNC_DOWN_MASK 0x3 #define NAU8824_ADC_SYNC_DOWN_32 0 -- 2.6.4

2 1

[alsa-devel] [PATCH] ASoC: rt5514: Add more width and channels support in the TDM mode
by Oder Chiou 15 May '17

15 May '17

This patch adds more width and channels support in the TDM mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 23 +++++++++++++++++++++-- sound/soc/codecs/rt5514.h | 6 ++++++ 2 files changed, 27 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index f91221b..ff97360 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -906,9 +906,23 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, if (rx_mask || tx_mask) val |= RT5514_TDM_MODE; - if (slots == 4) + switch (slots) { + case 4: val |= RT5514_TDMSLOT_SEL_RX_4CH | RT5514_TDMSLOT_SEL_TX_4CH; + break; + + case 6: + val |= RT5514_TDMSLOT_SEL_RX_6CH | RT5514_TDMSLOT_SEL_TX_6CH; + break; + + case 8: + val |= RT5514_TDMSLOT_SEL_RX_8CH | RT5514_TDMSLOT_SEL_TX_8CH; + break; + case 2: + default: + break; + } switch (slot_width) { case 20: @@ -919,6 +933,10 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, val |= RT5514_CH_LEN_RX_24 | RT5514_CH_LEN_TX_24; break; + case 25: + val |= RT5514_TDM_MODE2; + break; + case 32: val |= RT5514_CH_LEN_RX_32 | RT5514_CH_LEN_TX_32; break; @@ -930,7 +948,8 @@ static int rt5514_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(rt5514->regmap, RT5514_I2S_CTRL1, RT5514_TDM_MODE | RT5514_TDMSLOT_SEL_RX_MASK | RT5514_TDMSLOT_SEL_TX_MASK | - RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK, val); + RT5514_CH_LEN_RX_MASK | RT5514_CH_LEN_TX_MASK | + RT5514_TDM_MODE2, val); return 0; } diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 5d343fb..02bc212 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -117,6 +117,8 @@ #define RT5514_POW_ADCFEDL_BIT 0 /* RT5514_I2S_CTRL1 (0x2010) */ +#define RT5514_TDM_MODE2 (0x1 << 30) +#define RT5514_TDM_MODE2_SFT 30 #define RT5514_TDM_MODE (0x1 << 28) #define RT5514_TDM_MODE_SFT 28 #define RT5514_I2S_LR_MASK (0x1 << 26) @@ -136,6 +138,8 @@ #define RT5514_TDMSLOT_SEL_RX_MASK (0x3 << 10) #define RT5514_TDMSLOT_SEL_RX_SFT 10 #define RT5514_TDMSLOT_SEL_RX_4CH (0x1 << 10) +#define RT5514_TDMSLOT_SEL_RX_6CH (0x2 << 10) +#define RT5514_TDMSLOT_SEL_RX_8CH (0x3 << 10) #define RT5514_CH_LEN_RX_MASK (0x3 << 8) #define RT5514_CH_LEN_RX_SFT 8 #define RT5514_CH_LEN_RX_16 (0x0 << 8) @@ -145,6 +149,8 @@ #define RT5514_TDMSLOT_SEL_TX_MASK (0x3 << 6) #define RT5514_TDMSLOT_SEL_TX_SFT 6 #define RT5514_TDMSLOT_SEL_TX_4CH (0x1 << 6) +#define RT5514_TDMSLOT_SEL_TX_6CH (0x2 << 6) +#define RT5514_TDMSLOT_SEL_TX_8CH (0x3 << 6) #define RT5514_CH_LEN_TX_MASK (0x3 << 4) #define RT5514_CH_LEN_TX_SFT 4 #define RT5514_CH_LEN_TX_16 (0x0 << 4) -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rt5665: add ADC STO2 ASRC support
by Bard Liao 15 May '17

15 May '17

"ADC Stereo2 Filter" is with ASRC supported. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 8cd2230..ece77b8 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -2684,6 +2684,8 @@ static const struct snd_soc_dapm_widget rt5665_dapm_widgets[] = { RT5665_DAC_MONO_R_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5665_ASRC_1, RT5665_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5665_ASRC_1, + RT5665_ADC_STO2_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC Mono L ASRC", 1, RT5665_ASRC_1, RT5665_ADC_MONO_L_ASRC_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC Mono R ASRC", 1, RT5665_ASRC_1, @@ -3227,6 +3229,7 @@ static const struct snd_soc_dapm_route rt5665_dapm_routes[] = { /*ASRC*/ {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc}, {"ADC Mono Left Filter", NULL, "ADC Mono L ASRC", is_using_asrc}, {"ADC Mono Right Filter", NULL, "ADC Mono R ASRC", is_using_asrc}, {"DAC Mono Left Filter", NULL, "DAC Mono L ASRC", is_using_asrc}, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup Sampling rate convert example on Document
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> "renesas,rsrc-card" is exchanged to "simple-scu-card". Let's update Document Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/renesas,rsnd.txt | 30 +++++++++++----------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 15a7316..3332910 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -83,11 +83,11 @@ SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes ** Asynchronous mode ------------------ -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * SRC Asynchronous mode setting @@ -97,12 +97,12 @@ example) * Inputed 48kHz data will be converted to * system specified Hz */ - convert-rate = <48000>; + simple-audio-card,convert-rate = <48000>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -141,23 +141,23 @@ For more detail information, see below ${LINUX}/sound/soc/sh/rcar/ctu.c - comment of header -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. example) sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... /* * CTU setting * All input data will be converted to 2ch * as output data */ - convert-channels = <2>; + simple-audio-card,convert-channels = <2>; ... - cpu { + simple-audio-card,cpu { sound-dai = <&rcar_sound>; }; - codec { + simple-audio-card,codec { ... }; }; @@ -190,22 +190,22 @@ and these sounds will be merged by MIX. aplay -D plughw:0,0 xxxx.wav & aplay -D plughw:0,1 yyyy.wav -You need to use "renesas,rsrc-card" sound card for it. +You need to use "simple-scu-audio-card" sound card for it. Ex) [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] | [MEM] -> [SRC2] -> [CTU03] -+ sound { - compatible = "renesas,rsrc-card"; + compatible = "simple-scu-audio-card"; ... - cpu@0 { + simple-audio-card,cpu@0 { sound-dai = <&rcar_sound 0>; }; - cpu@1 { + simple-audio-card,cpu@1 { sound-dai = <&rcar_sound 1>; }; - codec { + simple-audio-card,codec { ... }; }; -- 1.9.1

2 2

[alsa-devel] [PATCH][RESEND] ASoC: rsnd: add missing clocks/clock-names on Document
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 3332910..13cb21d5 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -368,6 +368,10 @@ Required properties: see below for detail. - #sound-dai-cells : it must be 0 if your system is using single DAI it must be 1 if your system is using multi DAI +- clocks : References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. +- clock-names : List of necessary clock names. + "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X", + "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i" Optional properties: - #clock-cells : it must be 0 if your system has audio_clkout -- 1.9.1

1 0

[alsa-devel] [PATCH v2][RESEND] ASoC: rsnd: control kctrl items acceptance anytime/runtime
by Kuninori Morimoto 15 May '17

15 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current SRC/DVC/CTU adds kctrl for each device, and SRC can adjust its sampling rate during playback, thus, this feature should be enabled only *during* playback. This patch controls it more clearly Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 17 +++++++++++++++++ sound/soc/sh/rcar/ctu.c | 6 ++++++ sound/soc/sh/rcar/dvc.c | 5 +++++ sound/soc/sh/rcar/rsnd.h | 16 ++++++++++------ sound/soc/sh/rcar/src.c | 2 ++ 5 files changed, 40 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1744015..d3c9078 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -982,6 +982,9 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); int i, change = 0; + if (!cfg->accept(cfg->io)) + return 0; + for (i = 0; i < cfg->size; i++) { if (cfg->texts) { change |= (uc->value.enumerated.item[i] != cfg->val[i]); @@ -998,6 +1001,18 @@ static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, return change; } +int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io) +{ + return 1; +} + +int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io) +{ + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + + return !!runtime; +} + struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg) { cfg->cfg.val = cfg->val; @@ -1016,6 +1031,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, const unsigned char *name, + int (*accept)(struct rsnd_dai_stream *io), void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod), struct rsnd_kctrl_cfg *cfg, @@ -1050,6 +1066,7 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, cfg->texts = texts; cfg->max = max; cfg->size = size; + cfg->accept = accept; cfg->update = update; cfg->card = card; cfg->kctrl = kctrl; diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 9dcc1f9..4ba8f2f 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -279,12 +279,14 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* CTU Pass */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU Pass", + rsnd_kctrl_accept_anytime, NULL, &ctu->pass, RSND_MAX_CHANNELS, 0xC); /* ROW0 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV0", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv0, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -293,6 +295,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW1 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV1", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv1, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -301,6 +304,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW2 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV2", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv2, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -309,6 +313,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* ROW3 */ ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV3", + rsnd_kctrl_accept_anytime, NULL, &ctu->sv3, RSND_MAX_CHANNELS, 0x00FFFFFF); @@ -317,6 +322,7 @@ static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, /* Reset */ ret = rsnd_kctrl_new_s(mod, io, rtd, "CTU Reset", + rsnd_kctrl_accept_anytime, rsnd_ctu_value_reset, &ctu->reset, 1); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 463de83..75af6e7 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -257,6 +257,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? "DVC Out Playback Volume" : "DVC In Capture Volume", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->volume, slots, 0x00800000 - 1); @@ -267,6 +268,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_m(mod, io, rtd, is_play ? "DVC Out Mute Switch" : "DVC In Mute Switch", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->mute, slots, 1); @@ -277,6 +279,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_s(mod, io, rtd, is_play ? "DVC Out Ramp Switch" : "DVC In Ramp Switch", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->ren, 1); if (ret < 0) @@ -285,6 +288,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_e(mod, io, rtd, is_play ? "DVC Out Ramp Up Rate" : "DVC In Ramp Up Rate", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->rup, dvc_ramp_rate); @@ -294,6 +298,7 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, ret = rsnd_kctrl_new_e(mod, io, rtd, is_play ? "DVC Out Ramp Down Rate" : "DVC In Ramp Down Rate", + rsnd_kctrl_accept_anytime, rsnd_dvc_volume_update, &dvc->rdown, dvc_ramp_rate); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index dbf4163..f68aecb 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -594,6 +594,7 @@ struct rsnd_kctrl_cfg { unsigned int size; u32 *val; const char * const *texts; + int (*accept)(struct rsnd_dai_stream *io); void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod); struct rsnd_dai_stream *io; struct snd_card *card; @@ -611,12 +612,15 @@ struct rsnd_kctrl_cfg_s { u32 val; }; +int rsnd_kctrl_accept_anytime(struct rsnd_dai_stream *io); +int rsnd_kctrl_accept_runtime(struct rsnd_dai_stream *io); struct rsnd_kctrl_cfg *rsnd_kctrl_init_m(struct rsnd_kctrl_cfg_m *cfg); struct rsnd_kctrl_cfg *rsnd_kctrl_init_s(struct rsnd_kctrl_cfg_s *cfg); int rsnd_kctrl_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd, const unsigned char *name, + int (*accept)(struct rsnd_dai_stream *io), void (*update)(struct rsnd_dai_stream *io, struct rsnd_mod *mod), struct rsnd_kctrl_cfg *cfg, @@ -624,16 +628,16 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, int size, u32 max); -#define rsnd_kctrl_new_m(mod, io, rtd, name, update, cfg, size, max) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_m(cfg), \ +#define rsnd_kctrl_new_m(mod, io, rtd, name, accept, update, cfg, size, max) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_m(cfg), \ NULL, size, max) -#define rsnd_kctrl_new_s(mod, io, rtd, name, update, cfg, max) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_s(cfg), \ +#define rsnd_kctrl_new_s(mod, io, rtd, name, accept, update, cfg, max) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_s(cfg), \ NULL, 1, max) -#define rsnd_kctrl_new_e(mod, io, rtd, name, update, cfg, texts) \ - rsnd_kctrl_new(mod, io, rtd, name, update, rsnd_kctrl_init_s(cfg), \ +#define rsnd_kctrl_new_e(mod, io, rtd, name, accept, update, cfg, texts) \ + rsnd_kctrl_new(mod, io, rtd, name, accept, update, rsnd_kctrl_init_s(cfg), \ texts, 1, ARRAY_SIZE(texts)) /* diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index e4bde0c..e395fcb 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -489,6 +489,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate Switch" : "SRC In Rate Switch", + rsnd_kctrl_accept_anytime, rsnd_src_set_convert_rate, &src->sen, 1); if (ret < 0) @@ -498,6 +499,7 @@ static int rsnd_src_pcm_new(struct rsnd_mod *mod, rsnd_io_is_play(io) ? "SRC Out Rate" : "SRC In Rate", + rsnd_kctrl_accept_runtime, rsnd_src_set_convert_rate, &src->sync, 192000); -- 1.9.1

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[alsa-devel] linux-4.12-rc1/sound/soc/codecs/cs35l35.c: false comparison ?
by David Binderman 14 May '17

14 May '17

hello there, [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:495] -> [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:498]: (style) Mismatching assignment and comparison, comparison 'errata_chk==1' is always false. [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:495] -> [linux-4.12-rc1/sound/soc/codecs/cs35l35.c:498]: (style) Mismatching assignment and comparison, comparison 'errata_chk==3' is always false. Source code is errata_chk = clk_ctl & CS35L35_SP_RATE_MASK; if (classh->classh_wk_fet_disable == 0x00 && (errata_chk == 0x01 || errata_chk == 0x03)) { but ./sound/soc/codecs/cs35l35.h:#define CS35L35_SP_RATE_MASK 0xC0 Regards David Binderman

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[alsa-devel] [sound:topic/for-4.13 22/30] sound/core/pcm.c:905:3: error: implicit declaration of function 'pcm_call_notify'
by kbuild test robot 12 May '17

12 May '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/for-4.13 head: c56407e6d958efda0d2a587ed89e02a251975e06 commit: e9a0fa885872c92e6ada08fb1cf3433945d235a3 [22/30] ALSA: pcm: Build pcm notifier code conditionally config: x86_64-kexec (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout e9a0fa885872c92e6ada08fb1cf3433945d235a3 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/core/pcm.c: In function 'snd_pcm_free': >> sound/core/pcm.c:905:3: error: implicit declaration of function 'pcm_call_notify' [-Werror=implicit-function-declaration] pcm_call_notify(pcm, n_unregister); ^~~~~~~~~~~~~~~ >> sound/core/pcm.c:905:24: error: 'n_unregister' undeclared (first use in this function) pcm_call_notify(pcm, n_unregister); ^~~~~~~~~~~~ sound/core/pcm.c:905:24: note: each undeclared identifier is reported only once for each function it appears in sound/core/pcm.c: In function 'snd_pcm_dev_register': >> sound/core/pcm.c:1118:23: error: 'n_register' undeclared (first use in this function) pcm_call_notify(pcm, n_register); ^~~~~~~~~~ sound/core/pcm.c: In function 'snd_pcm_dev_disconnect': >> sound/core/pcm.c:1147:24: error: 'n_disconnect' undeclared (first use in this function) pcm_call_notify(pcm, n_disconnect); ^~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/pcm_call_notify +905 sound/core/pcm.c 899 900 static int snd_pcm_free(struct snd_pcm *pcm) 901 { 902 if (!pcm) 903 return 0; 904 if (!pcm->internal) > 905 pcm_call_notify(pcm, n_unregister); 906 if (pcm->private_free) 907 pcm->private_free(pcm); 908 snd_pcm_lib_preallocate_free_for_all(pcm); --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 0/3] ALSA: USB-descriptor endianess fixes
by Johan Hovold 12 May '17

12 May '17

These patches add missing endianness conversions when accessing the USB device-descriptor fields. In the process, clean up the us122l driver which had product-id conditionals sprinkled throughout. The final patch drops the Kconfig dependency on x86 since it's a USB driver that compiles just fine on non-x86. Perhaps this should be a (X86 || COMPILE_TEST) dependency instead in case there are external dependencies that warrants this limitation. Note that these patches have only been compile-tested. Johan Johan Hovold (3): ALSA: usb-audio: fix Amanero Combo384 quirk on big-endian hosts ALSA: us122l: clean up US144 handling ALSA: us122l: drop x86 dependency sound/usb/Kconfig | 1 - sound/usb/quirks.c | 2 +- sound/usb/usx2y/us122l.c | 36 ++++++++++++++++++------------------ sound/usb/usx2y/us122l.h | 2 ++ 4 files changed, 21 insertions(+), 20 deletions(-) -- 2.13.0

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