April 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: rt5645: fixed section mismatch
by Sudip Mukherjee 28 Apr '15

28 Apr '15

while building as a module we are getting warning about section mismatch. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/codecs/rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 14570b3..0ed79ca 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2684,7 +2684,7 @@ static int strago_quirk_cb(const struct dmi_system_id *id) return 1; } -static struct dmi_system_id dmi_platform_intel_braswell[] __initdata = { +static struct dmi_system_id dmi_platform_intel_braswell[] = { { .ident = "Intel Strago", .callback = strago_quirk_cb, -- 1.8.1.2

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[alsa-devel] [PATCH] ALSA: emux: Fix mutex deadlock in OSS emulation
by Takashi Iwai 28 Apr '15

28 Apr '15

The OSS emulation in synth-emux helper has a potential AB/BA deadlock at the simultaneous closing and opening: close -> snd_seq_release() -> sne_seq_free_client() -> snd_seq_delete_all_ports(): takes client->ports_mutex -> port_delete() -> snd_emux_unuse(): takes emux->register_mutex open -> snd_seq_oss_open() -> snd_emux_open_seq_oss(): takes emux->register_mutex -> snd_seq_event_port_attach() -> snd_seq_create_port(): takes client->ports_mutex This patch addresses the deadlock by reducing the rance taking emux->register_mutex in snd_emux_open_seq_oss(). The lock is needed for the refcount handling, so move it locally. The calls in emux_seq.c are already with the mutex, thus they are replaced with the version without mutex lock/unlock. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/synth/emux/emux_oss.c | 11 +---------- sound/synth/emux/emux_seq.c | 27 +++++++++++++++++++++------ 2 files changed, 22 insertions(+), 16 deletions(-) diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index ab37add269ae..82e350e9501c 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -118,12 +118,8 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) if (snd_BUG_ON(!arg || !emu)) return -ENXIO; - mutex_lock(&emu->register_mutex); - - if (!snd_emux_inc_count(emu)) { - mutex_unlock(&emu->register_mutex); + if (!snd_emux_inc_count(emu)) return -EFAULT; - } memset(&callback, 0, sizeof(callback)); callback.owner = THIS_MODULE; @@ -135,7 +131,6 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) if (p == NULL) { snd_printk(KERN_ERR "can't create port\n"); snd_emux_dec_count(emu); - mutex_unlock(&emu->register_mutex); return -ENOMEM; } @@ -148,8 +143,6 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) reset_port_mode(p, arg->seq_mode); snd_emux_reset_port(p); - - mutex_unlock(&emu->register_mutex); return 0; } @@ -195,13 +188,11 @@ snd_emux_close_seq_oss(struct snd_seq_oss_arg *arg) if (snd_BUG_ON(!emu)) return -ENXIO; - mutex_lock(&emu->register_mutex); snd_emux_sounds_off_all(p); snd_soundfont_close_check(emu->sflist, SF_CLIENT_NO(p->chset.port)); snd_seq_event_port_detach(p->chset.client, p->chset.port); snd_emux_dec_count(emu); - mutex_unlock(&emu->register_mutex); return 0; } diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index 188fda0effb0..a0209204ae48 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -267,8 +267,8 @@ snd_emux_event_input(struct snd_seq_event *ev, int direct, void *private_data, /* * increment usage count */ -int -snd_emux_inc_count(struct snd_emux *emu) +static int +__snd_emux_inc_count(struct snd_emux *emu) { emu->used++; if (!try_module_get(emu->ops.owner)) @@ -282,12 +282,21 @@ snd_emux_inc_count(struct snd_emux *emu) return 1; } +int snd_emux_inc_count(struct snd_emux *emu) +{ + int ret; + + mutex_lock(&emu->register_mutex); + ret = __snd_emux_inc_count(emu); + mutex_unlock(&emu->register_mutex); + return ret; +} /* * decrease usage count */ -void -snd_emux_dec_count(struct snd_emux *emu) +static void +__snd_emux_dec_count(struct snd_emux *emu) { module_put(emu->card->module); emu->used--; @@ -296,6 +305,12 @@ snd_emux_dec_count(struct snd_emux *emu) module_put(emu->ops.owner); } +void snd_emux_dec_count(struct snd_emux *emu) +{ + mutex_lock(&emu->register_mutex); + __snd_emux_dec_count(emu); + mutex_unlock(&emu->register_mutex); +} /* * Routine that is called upon a first use of a particular port @@ -315,7 +330,7 @@ snd_emux_use(void *private_data, struct snd_seq_port_subscribe *info) mutex_lock(&emu->register_mutex); snd_emux_init_port(p); - snd_emux_inc_count(emu); + __snd_emux_inc_count(emu); mutex_unlock(&emu->register_mutex); return 0; } @@ -338,7 +353,7 @@ snd_emux_unuse(void *private_data, struct snd_seq_port_subscribe *info) mutex_lock(&emu->register_mutex); snd_emux_sounds_off_all(p); - snd_emux_dec_count(emu); + __snd_emux_dec_count(emu); mutex_unlock(&emu->register_mutex); return 0; } -- 2.3.5

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[alsa-devel] [PATCH V2] Update email-id of author
by Rajeev Kumar 28 Apr '15

28 Apr '15

rajeev-dlh.kumar(a)st.com email-id doesn't exist anymore as I have left the company. Replace ST's id with Rajeev Kumar <rajeevkumar.linux(a)gmail.com> Signed-off-by: Rajeev Kumar <rajeevkumar.linux(a)gmail.com> --- include/sound/designware_i2s.h | 2 +- include/sound/spear_dma.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index 26f406e..3a8fca9 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -1,5 +1,5 @@ /* - * Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar(a)st.com) + * Copyright (ST) 2012 Rajeev Kumar (rajeevkumar.linux(a)gmail.com) * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by diff --git a/include/sound/spear_dma.h b/include/sound/spear_dma.h index 65aca51..e290de4 100644 --- a/include/sound/spear_dma.h +++ b/include/sound/spear_dma.h @@ -1,7 +1,7 @@ /* * linux/spear_dma.h * -* Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar(a)st.com) +* Copyright (ST) 2012 Rajeev Kumar (rajeevkumar.linux(a)gmail.com) * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by -- 1.7.9.5

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[alsa-devel] [RFC PATCH 1/2] ALSA: hda - divide controller and codec dependency on i915 gfx power well
by mengdong.lin＠intel.com 28 Apr '15

28 Apr '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch can improve power saving for Intel platforms on which only the display audio codec is in the shared i915 power well: - divide the controller and codec dependency on i915 power well by adding flag "need_i915_power" for the chip (controller) and codec respectively. - If the controller does not need the power well, the driver will release the power after probe. And if only the codec needs the power, its runtime PM ops will request/release the power. Background: - For Haswell/Broadwell, which has a separate HD-A controller for display audio, both the controller and the display codec are in the i915 power well. - For Baytrail/Braswell, the display and analog audio share the same HDA controller and link, and only the display codec is in the i915 power well. - For Skylake, the display and analog audio share the same HDA controller but use separate links. Only the display codec is in the i915 power well. And in legacy mode we take the two links as one. So it can follow Baytrail/Braswell. Open questions: - Move the hda_i915 to sound/hda at first? So both legacy and new drivers can use it as Takashi suggested? The PCI ID check in haswell_set_bclk will no longer be needed, since it's only called when chip's "need_i915_power" flag is set and so only called for Haswell & Broadwell. - move flag "need_i915_power" from struct hda_codec to hdac_device, for the same reason as above? - If the audio component moves from struct hda_intel to hdac_bus, how can the controller access the component? Need to add "bus" to struct azx? - implement codec resume/suspend ops to get/put i915 power if only the codec needs the power? Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 2a8aa9d..baa01a0 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -169,6 +169,8 @@ struct hdac_bus_ops { /* get a response from the last command */ int (*get_response)(struct hdac_bus *bus, unsigned int addr, unsigned int *res); + /* request/release display power */ + int (*display_power)(struct hdac_bus *bus, bool enable); }; #define HDA_UNSOL_QUEUE_SIZE 64 @@ -222,6 +224,11 @@ static inline void snd_hdac_codec_link_down(struct hdac_device *codec) clear_bit(codec->addr, &codec->bus->codec_powered); } +static inline int snd_hdac_display_power(struct hdac_device *codec, bool enable) +{ + return codec->bus->ops->display_power(codec->bus, enable); +} + /* * generic array helpers */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c old mode 100755 new mode 100644 index 18a021a..034488e --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -524,9 +524,21 @@ static int _hda_bus_get_response(struct hdac_bus *_bus, unsigned int addr, return bus->rirb_error ? -EIO : 0; } +static int _hda_bus_display_power(struct hdac_bus *_bus, bool enable) +{ + struct hda_bus *bus = container_of(_bus, struct hda_bus, core); + + if (bus->ops.display_power) + return bus->ops.display_power(bus, enable); + else + return -EINVAL; + +} + static const struct hdac_bus_ops bus_ops = { .command = _hda_bus_command, .get_response = _hda_bus_get_response, + .display_power = _hda_bus_display_power, }; /** @@ -949,6 +961,10 @@ void snd_hda_codec_register(struct hda_codec *codec) return; if (device_is_registered(hda_codec_dev(codec))) { snd_hda_register_beep_device(codec); + + if (codec->need_i915_power) + snd_hdac_display_power(&codec->core, true); + pm_runtime_enable(hda_codec_dev(codec)); /* it was powered up in snd_hda_codec_new(), now all done */ snd_hda_power_down(codec); @@ -3196,6 +3212,9 @@ static int hda_codec_runtime_suspend(struct device *dev) if (codec_has_clkstop(codec) && codec_has_epss(codec) && (state & AC_PWRST_CLK_STOP_OK)) snd_hdac_codec_link_down(&codec->core); + + if (codec->need_i915_power) + snd_hdac_display_power(&codec->core, false); return 0; } @@ -3205,6 +3224,9 @@ static int hda_codec_runtime_resume(struct device *dev) printk("amanda: hda_codec_runtime_resume, codec addr %d \n", codec->addr); + if (codec->need_i915_power) + snd_hdac_display_power(&codec->core, true); + snd_hdac_codec_link_up(&codec->core); hda_call_codec_resume(codec); pm_runtime_mark_last_busy(dev); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9075ac2..32d7fa7 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -64,6 +64,7 @@ struct hda_bus_ops { void (*load_dsp_cleanup)(struct hda_bus *bus, struct snd_dma_buffer *dmab); #endif + int (*display_power)(struct hda_bus *bus, bool enable); }; /* @@ -281,6 +282,7 @@ struct hda_codec { unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ unsigned int power_save_node:1; /* advanced PM for each widget */ + unsigned int need_i915_power:1; /* only codec needs i915 power */ #ifdef CONFIG_PM unsigned long power_on_acct; unsigned long power_off_acct; diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 26ce990..b7eaf2d 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1354,6 +1354,17 @@ static unsigned int azx_get_response(struct hda_bus *bus, return azx_rirb_get_response(bus, addr); } + +static unsigned int azx_display_power(struct hda_bus *bus, bool enable) +{ + struct azx *chip = bus->private_data; + + if (chip->ops->display_power) + return chip->ops->display_power(chip, enable); + else + return -EINVAL; +} + #ifdef CONFIG_SND_HDA_DSP_LOADER /* * DSP loading code (e.g. for CA0132) @@ -1819,6 +1830,7 @@ static struct hda_bus_ops bus_ops = { .load_dsp_trigger = azx_load_dsp_trigger, .load_dsp_cleanup = azx_load_dsp_cleanup, #endif + .display_power = azx_display_power, }; /* HD-audio bus initialization */ diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index be1b7de..2d6c6c2 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -277,6 +277,7 @@ struct hda_controller_ops { struct vm_area_struct *area); /* Check if current position is acceptable */ int (*position_check)(struct azx *chip, struct azx_dev *azx_dev); + int (*display_power)(struct azx *chip, bool enable); }; struct azx_pcm { @@ -358,6 +359,7 @@ struct azx { unsigned int align_buffer_size:1; unsigned int region_requested:1; unsigned int disabled:1; /* disabled by VGA-switcher */ + unsigned int need_i915_power:1; /* both controller & display codec needs i915 power */ /* for debugging */ unsigned int last_cmd[AZX_MAX_CODECS]; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 34040d2..f6c3e37 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -543,6 +543,13 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) return 0; } +static int azx_intel_display_power(struct azx *chip, bool enable) +{ + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + + return hda_display_power(hda, enable); +} + /* * Check whether the current DMA position is acceptable for updating * periods. Returns non-zero if it's OK. @@ -796,7 +803,8 @@ static int azx_suspend(struct device *dev) if (chip->msi) pci_disable_msi(chip->pci); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL + && chip->need_i915_power) hda_display_power(hda, false); return 0; } @@ -816,7 +824,8 @@ static int azx_resume(struct device *dev) if (chip->disabled || hda->init_failed) return 0; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL + && chip->need_i915_power) { hda_display_power(hda, true); haswell_set_bclk(hda); } @@ -859,7 +868,8 @@ static int azx_runtime_suspend(struct device *dev) azx_stop_chip(chip); azx_enter_link_reset(chip); azx_clear_irq_pending(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL + && chip->need_i915_power) hda_display_power(hda, false); return 0; @@ -885,7 +895,8 @@ static int azx_runtime_resume(struct device *dev) if (!azx_has_pm_runtime(chip)) return 0; - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL + && chip->need_i915_power) { hda_display_power(hda, true); haswell_set_bclk(hda); } @@ -1105,7 +1116,8 @@ static int azx_free(struct azx *chip) release_firmware(chip->fw); #endif if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - hda_display_power(hda, false); + if (chip->need_i915_power) + hda_display_power(hda, false); hda_i915_exit(hda); } kfree(hda); @@ -1757,6 +1769,7 @@ static const struct hda_controller_ops pci_hda_ops = { .substream_free_pages = substream_free_pages, .pcm_mmap_prepare = pcm_mmap_prepare, .position_check = azx_position_check, + .display_power = azx_intel_display_power, }; static int azx_probe(struct pci_dev *pci, @@ -1855,12 +1868,12 @@ static int azx_probe_continue(struct azx *chip) #ifdef CONFIG_SND_HDA_I915 err = hda_i915_init(hda); if (err < 0) - goto out_free; + goto i915_power_fail; err = hda_display_power(hda, true); if (err < 0) { dev_err(chip->card->dev, "Cannot turn on display power on i915\n"); - goto out_free; + goto i915_power_fail; } #endif } @@ -1911,6 +1924,10 @@ static int azx_probe_continue(struct azx *chip) pm_runtime_put_noidle(&pci->dev); out_free: + if (!chip->need_i915_power) + hda_display_power(hda, false); + +i915_power_fail: if (err < 0) hda->init_failed = 1; complete_all(&hda->probe_wait); -- 1.9.1

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[alsa-devel] Audio crackles with 4.1-rc1
by Jonathan McDowell 28 Apr '15

28 Apr '15

Having upgraded to 4.1-rc1 from 4.0 I'm now hearing audio crackles at regular intervals. I'm fairly sure this is due to the HDA power save as once audio is playing things are fine, it's just when starting to play audio that I hear the crackle. System is a Dell Latitude E7240. I haven't tried a bisect yet but will attempt to find some time to do so in the next few days. It looks like there have been some changes in sound/hda/ between 4.0 + 4.1-rc1 so I'll concentrate on those first. Hopefully relevant dmesg output for the device: [ 23.551075] snd_hda_intel 0000:00:03.0: enabling device (0000 -> 0002) [ 23.551279] snd_hda_intel 0000:00:03.0: bound 0000:00:02.0 (ops 0xffffffff81891a00) [ 23.575852] snd_hda_codec_realtek hdaudioC1D0: autoconfig for ALC3226: line_outs=1 (0x16/0x0/0x0/0x0/0x0) type:line [ 23.575855] snd_hda_codec_realtek hdaudioC1D0: speaker_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 23.575857] snd_hda_codec_realtek hdaudioC1D0: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 23.575858] snd_hda_codec_realtek hdaudioC1D0: mono: mono_out=0x0 [ 23.575859] snd_hda_codec_realtek hdaudioC1D0: inputs: [ 23.575861] snd_hda_codec_realtek hdaudioC1D0: Dock Mic=0x19 [ 23.575863] snd_hda_codec_realtek hdaudioC1D0: Headset Mic=0x1a [ 23.575870] snd_hda_codec_realtek hdaudioC1D0: Internal Mic=0x12 [ 23.582324] input: HDA Intel HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/sound/card0/input12 [ 23.582397] input: HDA Intel HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/sound/card0/input13 [ 23.582747] input: HDA Intel HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/sound/card0/input14 [ 23.593267] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card1/input9 [ 23.594102] input: HDA Intel PCH Dock Mic as /devices/pci0000:00/0000:00:1b.0/sound/card1/input15 [ 23.594268] input: HDA Intel PCH Dock Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card1/input16 [ 23.594346] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card1/input17 I have: CONFIG_SND_HDA_POWER_SAVE_DEFAULT=600 set in my kernel config. There's no dmesg output around the times when the crackling is occurring. J. -- Friends are God's apology for relations.

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[alsa-devel] M-Audio Midisport 2x2 Anniversary edition, latency problem
by Ilya Dikariev 28 Apr '15

28 Apr '15

Hi all, I bought a MIDI-interface von M-Audio (see subject), but there are strange things with it on my desktop (Arch, latest kernel). The latency is about 5 seconds. If I reconnect cabels, the latency can change from 5 sec to 0.5 sec, or to ~0sec, but in 90% of all the tries, the latency stay at 3-5 sec. I tried it on my older laptop (kernel 2.6.32 from Ubuntu 10.04) and it works! I think it's a regression. How can I help to fix this bug? best regards, Ilya.

2 1

[alsa-devel] [PATCH] conf/ucm: broadwell-rt286: change to use the correct jack kcontrol name
by Jie Yang 28 Apr '15

28 Apr '15

The headset jack has two kctls: "Headphone Jack" and "Mic Jack", we need switch speaker output according to the former JackControl. Here correct it. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- src/conf/ucm/broadwell-rt286/HiFi | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/conf/ucm/broadwell-rt286/HiFi b/src/conf/ucm/broadwell-rt286/HiFi index 08f5c50..58983ea 100644 --- a/src/conf/ucm/broadwell-rt286/HiFi +++ b/src/conf/ucm/broadwell-rt286/HiFi @@ -43,7 +43,7 @@ SectionDevice."Headphones" { Value { PlaybackChannels "2" JackDev "rt286-jack" - JackControl "Headset Jack" + JackControl "Headphone Jack" JackHWMute "Speaker" } } -- 1.9.1

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[alsa-devel] [PATCH 03/12] trivial: ASoC: pxa: pxa-ssp.c: indent with TABs, not spaces
by Antonio Ospite 28 Apr '15

28 Apr '15

Signed-off-by: Antonio Ospite <ao2(a)ao2.it> Cc: Daniel Mack <daniel(a)zonque.org> Cc: Mark Brown <broonie(a)kernel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/pxa/pxa-ssp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index fbe2e93..eb181ab 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -586,7 +586,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sspsp = pxa_ssp_read_reg(ssp, SSPSP); + sspsp = pxa_ssp_read_reg(ssp, SSPSP); if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs -- 2.1.4

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[alsa-devel] [PATCH v10 0/9] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 28 Apr '15

28 Apr '15

Based on broonie/ASoC v10: - add the generic dt-card - define the audio ports from a DT graph of ports (Russell King) - reuse HDMI constants (Andrew Jackson - Jyri Sarha) - alloc rate_constraints in codec (Jyri Sarha) - fix bad number of channels (Jyri Sarha) - correct codec generation from config (Russell King - Jyri Sarha) - no module init/exit (Russell King) v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (9): ASoC: kirkwood: dynamically build the DAI array ASoC: kirkwood: check the DAI type from the DAI name ASoC: kirkwood: accept the DAI definitions from a graph of ports drm/i2c: tda998x: Add support of a DT graph of ports drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter drm/i2c: tda998x: set cts_n according to the sample width ASoC: core: export snd_soc_get_dai_name ASoC: add generic dt-card support .../devicetree/bindings/drm/i2c/tda998x.txt | 51 ++++ .../devicetree/bindings/sound/mvebu-audio.txt | 30 +++ drivers/gpu/drm/i2c/tda998x_drv.c | 237 +++++++++++++++++- include/sound/soc.h | 2 + include/sound/tda998x.h | 32 +++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 175 +++++++++++++ sound/soc/generic/Kconfig | 2 + sound/soc/generic/Makefile | 2 + sound/soc/generic/dt-card.c | 275 +++++++++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 141 ++++++----- sound/soc/kirkwood/kirkwood.h | 4 +- sound/soc/soc-core.c | 5 +- 14 files changed, 877 insertions(+), 86 deletions(-) create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/tda998x.c create mode 100644 sound/soc/generic/dt-card.c -- 2.1.4

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[alsa-devel] [PATCH v10 0/7] ALSA: jack: Refactoring for jack kctls
by Jie Yang 28 Apr '15

28 Apr '15

This patch series aligns jack handling between the HDA and ASoC subsystems. The HDA subsystem currently creates a kcontrol for each HDA jack registered. ASoC subsystem creates input device jacks for each jack registered. This series aligns both jack APIs to create kcontrol jacks. This implementation keeps compatibility with existing HDA jacks so no userspace changes are required. Changes in v10: 1. rebase it on top of tiwai/sound.git for-linus branch; 2. Liam helps fix some typos and syntax errors in comments and descriptions. Changes in v9: 1. move struct snd_jack_kctl definition to jack.c, to hides the internal implementation details; 2. some small code fixing; 3. update documentation file. Changes in v8: 1. remove snd_jack_kctl from HDA part, change snd_jack_new param to use bool for creating kcontrol at jack creating stage; 2. create jack for phantom jack, with no input device; 3. remove HDA jack get_unique_index() and index related code. Changes in v7: 1. move name generating and index getting part into ctljack.c; 2. remove exposing private_data and destructor to HDA, then jack core can handle kctls by itself totally. 3. some small code fixing. Changes in v6: 1. refine kctl name generating function; 2. integrate HDA jack kctl more deeply, support phantom jack in snd_jack_new(); 3. add documentation for Jack kcontrols to explain how to use it. Changes in v5: 1. remove SND_KCTL_JACK config item, we need jack kctl once CONFIG_SND_JACK is selected. Changes in v4: 1. use snd_ctl_find_id() to get avaliable index; 2. add initial kctl for snd_jack_new(compatible for HDA); 3. add struct snd_jack_kctl * field to struct hda_jack_tbl; 4. new kctls for soc jack during jack pins creating. 5. add a patch to remove exporting snd_kctl_jack_new(). Changes in v3: 1. replace bit index with bit mask in jack_kctl; 2. add exception for SND_JACK_HEADSET and SND_JACK_AVOUT, only create one jack kctl for these two combo jacks, respectively. 3. add NULL check, mem kfree, and fix some potential risk. Change in v2: 1. define jack_kctl struct, and put jack kctl related stuff there; 2. add a patch to remove the existing controls for HDA jack. Jie Yang (7): ALSA: jack: implement kctl creating for jack devices ALSA: Jack: handle jack embedded kcontrol creating within ctljack ALSA: jack: extend snd_jack_new to support phantom jack ALSA: hda - Update to use the new jack kctls method ASoC: jack: create kctls according to jack pins info ALSA: jack: remove exporting ctljack functions ALSA: Docs: Add documentation for Jack kcontrols Documentation/sound/alsa/Jack-Controls.txt | 43 ++++++++++ include/sound/control.h | 2 +- include/sound/jack.h | 13 ++- sound/core/Kconfig | 3 - sound/core/Makefile | 3 +- sound/core/ctljack.c | 41 +++++++-- sound/core/jack.c | 133 ++++++++++++++++++++++++++--- sound/pci/hda/Kconfig | 2 +- sound/pci/hda/hda_jack.c | 90 ++++++------------- sound/pci/hda/hda_jack.h | 5 +- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 3 +- 13 files changed, 242 insertions(+), 100 deletions(-) create mode 100644 Documentation/sound/alsa/Jack-Controls.txt -- 1.9.1

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