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

[alsa-devel] [RFC][PATCH 0/6] ALSA: firewire: drop reusing incoming packet parameters for outgoing packet parameters
by Takashi Sakamoto 25 Apr '16

25 Apr '16

Hi, The purpose of this patchset is to remove a feature of firewire-lib module. I call it as 'full duplex streaming with timestamp synchronization'. This patchset should accumulated into my previous RFC patchset below: [alsa-devel] [RFC][PATCH 0/3 v2] firewire-lib: support Linux tracing framework to dump a part of packet data http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/106950.html The feature is based on a specification of IEC 61883-6. In the specification, an audio function block should reuse some parameters of incoming packets for outgoing packets. Current ALSA firewire stack partly implement it. However, actual audio and music units on IEEE 1394 bus don't expect drivers to perform the specification. Additionally to the fact, the feature brings inconveniences to handle payloads of isochronous packet in software interrupts. For these reasons, this patchset removes the feature from ALSA firewire stack. This decision may be a help for future improvement about dynamic interrupt interval of isochronous context, about which Clemens have mentioned[1]. [1][alsa-devel] For firewire drivers, reduce period time down to 2msec? http://mailman.alsa-project.org/pipermail/alsa-devel/2014-April/075696.html Regards Takashi Sakamoto (6): ALSA: bebob: drop reusing incoming packet parameter for outgoing packet parameter ALSA: fireworks: drop reusing incoming packet parameter for ougoing packet parameter ALSA: firewire-tascam: drop reusing incoming packet parameter for outgoing packet parameter ALSA: firewire-lib: handle IT/IR contexts in each software context ALSA: firewire-lib: code cleanup for incoming packet handling ALSA: firewire-lib: code cleanup for outgoing packet handling sound/firewire/amdtp-stream.c | 84 ++++++++---------------- sound/firewire/amdtp-stream.h | 35 ++-------- sound/firewire/bebob/bebob.h | 1 - sound/firewire/bebob/bebob_stream.c | 99 ++++++++--------------------- sound/firewire/fireworks/fireworks.h | 1 - sound/firewire/fireworks/fireworks_stream.c | 84 +++++------------------- sound/firewire/tascam/tascam-stream.c | 26 ++++---- 7 files changed, 87 insertions(+), 243 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 0/7] ALSA: firewire: apply deleyed registration to all drivers
by Takashi Sakamoto 25 Apr '16

25 Apr '16

Hi, Commit b59fb1900b4fee('ALSA: dice: postpone card registration') applied delayed registration[1] of sound card instance to ALSA dice driver. This idea is also nice for drivers in ALSA firewire stack because in IEEE 1394 bus adding units is corresponding to bus reset and the actual units tend not to work well during bus-reset state. This patchset adds a new function to firewire-lib module to schedule a delayed work for sound card registration. Currently, the work is commonly scheduled 2 seconds after time of the last bus-reset recorded by Linux firewire core. Then, take all drivers to use it as Dice driver does. Related to this patchset, I'd like to cancel a discussion about my proposed patchset to remove unpractical module parameters from ALSA bebob/fireworks[2]. I'm sorry for reviewers but my work priority to this patchset is prior to the former one. [1]: [alsa-devel] [PATCH 0/4 v5] ALSA: dice: improve card registration processing http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/102562.h… [2]: [alsa-devel] [PATCH 0/2] ALSA: bebob/fireworks: remove module parameters to reduce maintenance work http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/106219.html Regards Takashi Sakamoto (8): ALSA: firewire-lib: suppress kernel warnings when releasing uninitialized stream data ALSA: dice: simplify unit probe processing ALSA: firewire-lib: add new function to schedule a work for sound card registration ALSA: bebob: delayed registration of sound card ALSA: fireworks: delayed registration of sound card ALSA: oxfw: delayed registration of sound card ALSA: firewire-digi00x: delayed registration of sound card ALSA: firewire-tascam: deleyed registration of sound card sound/firewire/amdtp-stream.c | 4 + sound/firewire/bebob/bebob.c | 215 ++++++++++++++++----------- sound/firewire/bebob/bebob.h | 5 +- sound/firewire/dice/dice.c | 34 +---- sound/firewire/digi00x/digi00x-transaction.c | 7 +- sound/firewire/digi00x/digi00x.c | 107 +++++++++---- sound/firewire/digi00x/digi00x.h | 3 + sound/firewire/fireworks/fireworks.c | 147 ++++++++++++------ sound/firewire/fireworks/fireworks.h | 3 + sound/firewire/lib.c | 32 ++++ sound/firewire/lib.h | 3 + sound/firewire/oxfw/oxfw.c | 151 ++++++++++++------- sound/firewire/oxfw/oxfw.h | 3 + sound/firewire/tascam/tascam.c | 118 ++++++++++----- sound/firewire/tascam/tascam.h | 2 + 15 files changed, 551 insertions(+), 283 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 36/41] Documentation: sound: fix spelling mistakes
by Eric Engestrom 25 Apr '16

25 Apr '16

Signed-off-by: Eric Engestrom <eric(a)engestrom.ch> --- Documentation/sound/alsa/compress_offload.txt | 2 +- Documentation/sound/alsa/soc/dapm.txt | 2 +- Documentation/sound/alsa/soc/overview.txt | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 630c492..81476b4 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -149,7 +149,7 @@ Gapless Playback ================ When playing thru an album, the decoders have the ability to skip the encoder delay and padding and directly move from one track content to another. The end -user can perceive this as gapless playback as we dont have silence while +user can perceive this as gapless playback as we don't have silence while switching from one track to another Also, there might be low-intensity noises due to encoding. Perfect gapless is diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 6faab48..c45bd79 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -132,7 +132,7 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls, ARRAY_SIZE(wm8731_output_mixer_controls)), -If you dont want the mixer elements prefixed with the name of the mixer widget, +If you don't want the mixer elements prefixed with the name of the mixer widget, you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same as for SND_SOC_DAPM_MIXER. diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt index ff88f52..f3f28b7 100644 --- a/Documentation/sound/alsa/soc/overview.txt +++ b/Documentation/sound/alsa/soc/overview.txt @@ -63,7 +63,7 @@ multiple re-usable component drivers :- and any audio DSP drivers for that platform. * Machine class driver: The machine driver class acts as the glue that - decribes and binds the other component drivers together to form an ALSA + describes and binds the other component drivers together to form an ALSA "sound card device". It handles any machine specific controls and machine level audio events (e.g. turning on an amp at start of playback). -- 2.8.0

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[alsa-devel] [PATCH] ALSA: timer: remove legacy rtctimer
by Alexandre Belloni 25 Apr '16

25 Apr '16

There are no users of rtctimer left. Remove its code as this is the in-kernel user of the legacy PC RTC driver that will hopefully be removed at some point. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- Note that I've kept the definition of SNDRV_TIMER_GLOBAL_RTC as it is exposed to userspace. USerspace will still properly get -ENODEV when trying to use it (as if it was compiled out of the kernel anyway). include/uapi/sound/asound.h | 2 +- sound/core/Kconfig | 29 ------- sound/core/Makefile | 1 - sound/core/rtctimer.c | 187 -------------------------------------------- sound/core/seq/seq.c | 2 - sound/core/timer.c | 2 - 6 files changed, 1 insertion(+), 222 deletions(-) delete mode 100644 sound/core/rtctimer.c diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 67bf49d8c944..609cadb8739d 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -672,7 +672,7 @@ enum { /* global timers (device member) */ #define SNDRV_TIMER_GLOBAL_SYSTEM 0 -#define SNDRV_TIMER_GLOBAL_RTC 1 +#define SNDRV_TIMER_GLOBAL_RTC 1 /* unused */ #define SNDRV_TIMER_GLOBAL_HPET 2 #define SNDRV_TIMER_GLOBAL_HRTIMER 3 diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 6d12ca9bcb80..9749f9e8b45c 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -141,35 +141,6 @@ config SND_SEQ_HRTIMER_DEFAULT Say Y here to use the HR-timer backend as the default sequencer timer. -config SND_RTCTIMER - tristate "RTC Timer support" - depends on RTC - select SND_TIMER - help - Say Y here to enable RTC timer support for ALSA. ALSA uses - the RTC timer as a precise timing source and maps the RTC - timer to ALSA's timer interface. The ALSA sequencer code also - can use this timing source. - - To compile this driver as a module, choose M here: the module - will be called snd-rtctimer. - - Note that this option is exclusive with the new RTC drivers - (CONFIG_RTC_CLASS) since this requires the old API. - -config SND_SEQ_RTCTIMER_DEFAULT - bool "Use RTC as default sequencer timer" - depends on SND_RTCTIMER && SND_SEQUENCER - depends on !SND_SEQ_HRTIMER_DEFAULT - default y - help - Say Y here to use the RTC timer as the default sequencer - timer. This is strongly recommended because it ensures - precise MIDI timing even when the system timer runs at less - than 1000 Hz. - - If in doubt, say Y. - config SND_DYNAMIC_MINORS bool "Dynamic device file minor numbers" help diff --git a/sound/core/Makefile b/sound/core/Makefile index 48ab4b8f8279..e85d9dd12c2d 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -37,7 +37,6 @@ obj-$(CONFIG_SND) += snd.o obj-$(CONFIG_SND_HWDEP) += snd-hwdep.o obj-$(CONFIG_SND_TIMER) += snd-timer.o obj-$(CONFIG_SND_HRTIMER) += snd-hrtimer.o -obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o obj-$(CONFIG_SND_PCM) += snd-pcm.o obj-$(CONFIG_SND_DMAENGINE_PCM) += snd-pcm-dmaengine.o obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c deleted file mode 100644 index f3420d11a12f..000000000000 --- a/sound/core/rtctimer.c +++ /dev/null @@ -1,187 +0,0 @@ -/* - * RTC based high-frequency timer - * - * Copyright (C) 2000 Takashi Iwai - * based on rtctimer.c by Steve Ratcliffe - * - * 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 - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include <linux/init.h> -#include <linux/interrupt.h> -#include <linux/module.h> -#include <linux/log2.h> -#include <sound/core.h> -#include <sound/timer.h> - -#if IS_ENABLED(CONFIG_RTC) - -#include <linux/mc146818rtc.h> - -#define RTC_FREQ 1024 /* default frequency */ -#define NANO_SEC 1000000000L /* 10^9 in sec */ - -/* - * prototypes - */ -static int rtctimer_open(struct snd_timer *t); -static int rtctimer_close(struct snd_timer *t); -static int rtctimer_start(struct snd_timer *t); -static int rtctimer_stop(struct snd_timer *t); - - -/* - * The hardware dependent description for this timer. - */ -static struct snd_timer_hardware rtc_hw = { - .flags = SNDRV_TIMER_HW_AUTO | - SNDRV_TIMER_HW_FIRST | - SNDRV_TIMER_HW_TASKLET, - .ticks = 100000000L, /* FIXME: XXX */ - .open = rtctimer_open, - .close = rtctimer_close, - .start = rtctimer_start, - .stop = rtctimer_stop, -}; - -static int rtctimer_freq = RTC_FREQ; /* frequency */ -static struct snd_timer *rtctimer; -static struct tasklet_struct rtc_tasklet; -static rtc_task_t rtc_task; - - -static int -rtctimer_open(struct snd_timer *t) -{ - int err; - - err = rtc_register(&rtc_task); - if (err < 0) - return err; - t->private_data = &rtc_task; - return 0; -} - -static int -rtctimer_close(struct snd_timer *t) -{ - rtc_task_t *rtc = t->private_data; - if (rtc) { - rtc_unregister(rtc); - tasklet_kill(&rtc_tasklet); - t->private_data = NULL; - } - return 0; -} - -static int -rtctimer_start(struct snd_timer *timer) -{ - rtc_task_t *rtc = timer->private_data; - if (snd_BUG_ON(!rtc)) - return -EINVAL; - rtc_control(rtc, RTC_IRQP_SET, rtctimer_freq); - rtc_control(rtc, RTC_PIE_ON, 0); - return 0; -} - -static int -rtctimer_stop(struct snd_timer *timer) -{ - rtc_task_t *rtc = timer->private_data; - if (snd_BUG_ON(!rtc)) - return -EINVAL; - rtc_control(rtc, RTC_PIE_OFF, 0); - return 0; -} - -static void rtctimer_tasklet(unsigned long data) -{ - snd_timer_interrupt((struct snd_timer *)data, 1); -} - -/* - * interrupt - */ -static void rtctimer_interrupt(void *private_data) -{ - tasklet_schedule(private_data); -} - - -/* - * ENTRY functions - */ -static int __init rtctimer_init(void) -{ - int err; - struct snd_timer *timer; - - if (rtctimer_freq < 2 || rtctimer_freq > 8192 || - !is_power_of_2(rtctimer_freq)) { - pr_err("ALSA: rtctimer: invalid frequency %d\n", rtctimer_freq); - return -EINVAL; - } - - /* Create a new timer and set up the fields */ - err = snd_timer_global_new("rtc", SNDRV_TIMER_GLOBAL_RTC, &timer); - if (err < 0) - return err; - - timer->module = THIS_MODULE; - strcpy(timer->name, "RTC timer"); - timer->hw = rtc_hw; - timer->hw.resolution = NANO_SEC / rtctimer_freq; - - tasklet_init(&rtc_tasklet, rtctimer_tasklet, (unsigned long)timer); - - /* set up RTC callback */ - rtc_task.func = rtctimer_interrupt; - rtc_task.private_data = &rtc_tasklet; - - err = snd_timer_global_register(timer); - if (err < 0) { - snd_timer_global_free(timer); - return err; - } - rtctimer = timer; /* remember this */ - - return 0; -} - -static void __exit rtctimer_exit(void) -{ - if (rtctimer) { - snd_timer_global_free(rtctimer); - rtctimer = NULL; - } -} - - -/* - * exported stuff - */ -module_init(rtctimer_init) -module_exit(rtctimer_exit) - -module_param(rtctimer_freq, int, 0444); -MODULE_PARM_DESC(rtctimer_freq, "timer frequency in Hz"); - -MODULE_LICENSE("GPL"); - -MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC)); - -#endif /* IS_ENABLED(CONFIG_RTC) */ diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 7e0aabb808a6..639544b4fb04 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -47,8 +47,6 @@ int seq_default_timer_card = -1; int seq_default_timer_device = #ifdef CONFIG_SND_SEQ_HRTIMER_DEFAULT SNDRV_TIMER_GLOBAL_HRTIMER -#elif defined(CONFIG_SND_SEQ_RTCTIMER_DEFAULT) - SNDRV_TIMER_GLOBAL_RTC #else SNDRV_TIMER_GLOBAL_SYSTEM #endif diff --git a/sound/core/timer.c b/sound/core/timer.c index 6469bedda2f3..0cfc028c1193 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -37,8 +37,6 @@ #if IS_ENABLED(CONFIG_SND_HRTIMER) #define DEFAULT_TIMER_LIMIT 4 -#elif IS_ENABLED(CONFIG_SND_RTCTIMER) -#define DEFAULT_TIMER_LIMIT 2 #else #define DEFAULT_TIMER_LIMIT 1 #endif -- 2.8.0.rc3

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[alsa-devel] [PATCH 0/3] Add support simple-card parse jack detection via external codec
by Xing Zheng 22 Apr '16

22 Apr '16

Hi, In most cases, many codecs already supports jack detection, previouslly, we need to create a customized machine driver every time. Hence, the simple-card need to support use them dynamically via parse dts file for better flexibility. Xing Zheng (3): ASoC: jack: Add a jack detect callback via codec ASoC: simple-card: Add support jack detection via codec ASoC: da7219: Add detect_jack callback in the snd_soc_codec_driver .../devicetree/bindings/sound/simple-card.txt | 17 +++ include/sound/soc.h | 6 + sound/soc/codecs/da7219.c | 2 + sound/soc/generic/simple-card.c | 126 +++++++++++++++++++- sound/soc/soc-jack.c | 8 ++ 5 files changed, 156 insertions(+), 3 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: da7218: Update PLL ranges and dividers to improve locking
by Adam Thomson 22 Apr '16

22 Apr '16

The expected MCLK frequency ranges and the associated dividers are updated to improve PLL locking in a corner scenario, with low MCLK frequency near an input divider change boundary. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- This patch is based on v4.6-rc1 kernel version sound/soc/codecs/da7218.c | 32 ++++++++++++++++---------------- sound/soc/codecs/da7218.h | 21 ++++++++++++--------- 2 files changed, 28 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 93575f2..99ce23e 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -1868,27 +1868,27 @@ static int da7218_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */ if (da7218->mclk_rate == 32768) { - indiv_bits = DA7218_PLL_INDIV_2_5_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; + indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL; } else if (da7218->mclk_rate < 2000000) { dev_err(codec->dev, "PLL input clock %d below valid range\n", da7218->mclk_rate); return -EINVAL; - } else if (da7218->mclk_rate <= 5000000) { - indiv_bits = DA7218_PLL_INDIV_2_5_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; - } else if (da7218->mclk_rate <= 10000000) { - indiv_bits = DA7218_PLL_INDIV_5_10_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; - } else if (da7218->mclk_rate <= 20000000) { - indiv_bits = DA7218_PLL_INDIV_10_20_MHZ; - indiv = DA7218_PLL_INDIV_10_20_MHZ_VAL; - } else if (da7218->mclk_rate <= 40000000) { - indiv_bits = DA7218_PLL_INDIV_20_40_MHZ; - indiv = DA7218_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7218->mclk_rate <= 4500000) { + indiv_bits = DA7218_PLL_INDIV_2_TO_4_5_MHZ; + indiv = DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL; + } else if (da7218->mclk_rate <= 9000000) { + indiv_bits = DA7218_PLL_INDIV_4_5_TO_9_MHZ; + indiv = DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL; + } else if (da7218->mclk_rate <= 18000000) { + indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL; + } else if (da7218->mclk_rate <= 36000000) { + indiv_bits = DA7218_PLL_INDIV_18_TO_36_MHZ; + indiv = DA7218_PLL_INDIV_18_TO_36_MHZ_VAL; } else if (da7218->mclk_rate <= 54000000) { - indiv_bits = DA7218_PLL_INDIV_40_54_MHZ; - indiv = DA7218_PLL_INDIV_40_54_MHZ_VAL; + indiv_bits = DA7218_PLL_INDIV_36_TO_54_MHZ; + indiv = DA7218_PLL_INDIV_36_TO_54_MHZ_VAL; } else { dev_err(codec->dev, "PLL input clock %d above valid range\n", da7218->mclk_rate); diff --git a/sound/soc/codecs/da7218.h b/sound/soc/codecs/da7218.h index c2c5904..477cd37 100644 --- a/sound/soc/codecs/da7218.h +++ b/sound/soc/codecs/da7218.h @@ -876,15 +876,11 @@ /* DA7218_PLL_CTRL = 0x91 */ #define DA7218_PLL_INDIV_SHIFT 0 #define DA7218_PLL_INDIV_MASK (0x7 << 0) -#define DA7218_PLL_INDIV_2_5_MHZ (0x0 << 0) -#define DA7218_PLL_INDIV_5_10_MHZ (0x1 << 0) -#define DA7218_PLL_INDIV_10_20_MHZ (0x2 << 0) -#define DA7218_PLL_INDIV_20_40_MHZ (0x3 << 0) -#define DA7218_PLL_INDIV_40_54_MHZ (0x4 << 0) -#define DA7218_PLL_INDIV_2_10_MHZ_VAL 2 -#define DA7218_PLL_INDIV_10_20_MHZ_VAL 4 -#define DA7218_PLL_INDIV_20_40_MHZ_VAL 8 -#define DA7218_PLL_INDIV_40_54_MHZ_VAL 16 +#define DA7218_PLL_INDIV_2_TO_4_5_MHZ (0x0 << 0) +#define DA7218_PLL_INDIV_4_5_TO_9_MHZ (0x1 << 0) +#define DA7218_PLL_INDIV_9_TO_18_MHZ (0x2 << 0) +#define DA7218_PLL_INDIV_18_TO_36_MHZ (0x3 << 0) +#define DA7218_PLL_INDIV_36_TO_54_MHZ (0x4 << 0) #define DA7218_PLL_MCLK_SQR_EN_SHIFT 4 #define DA7218_PLL_MCLK_SQR_EN_MASK (0x1 << 4) #define DA7218_PLL_MODE_SHIFT 6 @@ -1336,6 +1332,13 @@ #define DA7218_PLL_FREQ_OUT_90316 90316800 #define DA7218_PLL_FREQ_OUT_98304 98304000 +/* PLL Frequency Dividers */ +#define DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL 1 +#define DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL 2 +#define DA7218_PLL_INDIV_9_TO_18_MHZ_VAL 4 +#define DA7218_PLL_INDIV_18_TO_36_MHZ_VAL 8 +#define DA7218_PLL_INDIV_36_TO_54_MHZ_VAL 16 + /* ALC Calibration */ #define DA7218_ALC_CALIB_DELAY_MIN 2500 #define DA7218_ALC_CALIB_DELAY_MAX 5000 -- 1.9.3

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[alsa-devel] [PATCH 0/4] ASoC: simple-card/davinci-mcasp: System clock configuration support
by Peter Ujfalusi 22 Apr '16

22 Apr '16

Hi, Currently davinci-mcasp is hardwired for two configuration regarding to reference clock: McASP is master: AHCLKX is selected as source of the reference clock used to generate the FS and BCLK McASP is slave: AHCLKX is selected as output and the default AUXCLK is routed there. This works in most of the boards, but not all configuration can be supported with this limitation. McASP uses it's internal AUXCLK as a reference clock. AUXCLK can be sourced from the external AHCLKX pin or from internal functional clock. If the board uses McASP as master and AHCLKX is not receiving clock, we need to be able to select the internal clock source to be able to generate the FS and BCLK. Since we try to move all of our boards to simple-card instead of the davinci-evm custom machine driver, I have included Jyri's two patch to add the needed support for it to be able to select clocks and directions. I remember the discussion regarding to clock IDs, and to try to move to common clock framework. I have taken a look at that path and I'm not convinced that by trying to add ASoC level common clock handling will simplify or makes things cleaner/simpler. Regards, Peter --- Jyri Sarha (2): ASoC: simple-card: Add system-clock-direction DT parameter to dai nodes ASoC: simple-card: Support for selecting system clocks by ID Peter Ujfalusi (2): ASoC: davinci-mcasp: Use defines for clkdiv IDs via DT binding header ASoC: davinci-mcasp: Improve the sysclk selection .../devicetree/bindings/sound/simple-card.txt | 3 ++ include/dt-bindings/sound/ti-mcasp.h | 13 +++++++ include/sound/simple_card.h | 2 ++ sound/soc/davinci/davinci-mcasp.c | 42 ++++++++++++++++------ sound/soc/generic/simple-card.c | 17 ++++++++- 5 files changed, 66 insertions(+), 11 deletions(-) create mode 100644 include/dt-bindings/sound/ti-mcasp.h -- 2.7.1

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[alsa-devel] [PATCH] ASoC: Intel: Atom: fix boot warning
by Vinod Koul 22 Apr '16

22 Apr '16

Users have reported seeing this false warning on atom driver [ 5.647469] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 0 doesn't have DAPM widget!!! [ 5.661612] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 1 doesn't have DAPM widget!!! [ 5.661646] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 2 doesn't have DAPM widget!!! [ 5.661681] sst-mfld-platform sst-mfld-platform: Slot control: codec_out tx interleaver slot 3 doesn't have DAPM widget!!! [ 5.661708] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_0 doesn't have DAPM widget!!! [ 5.661738] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in0_1 doesn't have DAPM widget!!! [ 5.661771] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_0 doesn't have DAPM widget!!! [ 5.661807] sst-mfld-platform sst-mfld-platform: Slot control: codec_in rx deinterleaver codec_in1_1 doesn't have DAPM widget!!! This is caused when check for control is not being associated with a dapm widget, but the check is wrong as the else case maybe triggered when widget is not powered up, so we should check if widget is associated before printing this message. Tested-by: Sandeep Tayal <sandeepx.tayal(a)intel.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/atom/sst-atom-controls.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index b97e6adcf1b2..98720a93de8a 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -195,7 +195,7 @@ static int sst_check_and_send_slot_map(struct sst_data *drv, struct snd_kcontrol if (e->w && e->w->power) ret = sst_send_slot_map(drv); - else + else if (!e->w) dev_err(&drv->pdev->dev, "Slot control: %s doesn't have DAPM widget!!!\n", kcontrol->id.name); return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: tidyup alphabetical order for SND_SOC_Bxx
by Kuninori Morimoto 22 Apr '16

22 Apr '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/Kconfig | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 907c804..7d2d994 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK5386 select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C + select SND_SOC_BT_SCO select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C select SND_SOC_CS42L51_I2C if I2C @@ -64,7 +65,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DMIC - select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_GTM601 @@ -365,6 +365,9 @@ config SND_SOC_ALC5623 config SND_SOC_ALC5632 tristate +config SND_SOC_BT_SCO + tristate + config SND_SOC_CQ0093VC tristate @@ -471,9 +474,6 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_BT_SCO - tristate - config SND_SOC_DMIC tristate -- 1.9.1

2 3

[alsa-devel] [PATCH] ASoC: Add kerneldoc comments for snd_soc_find_dai
by mengdong.lin＠linux.intel.com 22 Apr '16

22 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> snd_soc_find_dai() has been exported and so add the kerneldoc comments for it. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 07663de..16369ca 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -930,6 +930,17 @@ static struct snd_soc_component *soc_find_component( return NULL; } +/** + * snd_soc_find_dai - Find a registered DAI + * + * @dlc: name of the DAI and optional component info to match + * + * This function will search all regsitered components and their DAIs to + * find the DAI of the same name. The component's of_node and name + * should also match if being specified. + * + * Return: pointer of DAI, or NULL if not found. + */ struct snd_soc_dai *snd_soc_find_dai( const struct snd_soc_dai_link_component *dlc) { -- 2.5.0

2 1

[alsa-devel] [PATCH v3 0/7] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 22 Apr '16

22 Apr '16

This series of patches adds devicetree support for the davinci McBSP audio interface. Changes in version 3 of the patch: * remove the index note from the "mpu" property in the binding * describe exactly the supported platforms in compatible string * remove unused platform data from davinci-i2s driver and the related EVK board drivers

3 10

[alsa-devel] [PATCH v3 0/3] conf: topology: Add the conf for SKL-rt286
by Vinod Koul 22 Apr '16

22 Apr '16

This series adds SKL-i2c conf for simple PCM topology graph Changes in v3: - remove bin blobs, now they are generated in install step - hook the tool compilation to alsa-lib and run it on install - add u8 in alsa type_compat.h Shreyas NC (3): conf: topology: Add Skylake i2s conf Add u8 in type_compat.h conf: topology: Generate Private data binary blobs configure.ac | 2 + include/sound/type_compat.h | 2 + src/conf/topology/Makefile.am | 2 +- src/conf/topology/sklrt286/Makefile.am | 11 + src/conf/topology/sklrt286/data/Makefile.am | 6 + src/conf/topology/sklrt286/data/pvt.c | 1815 ++++++++++++++++++++ src/conf/topology/sklrt286/data/pvt_data.c | 90 + src/conf/topology/sklrt286/data/pvt_local.h | 9 + .../topology/sklrt286/data/skl-tplg-interface.h | 232 +++ src/conf/topology/sklrt286/skl_i2s.conf | 342 ++++ 10 files changed, 2510 insertions(+), 1 deletion(-) create mode 100644 src/conf/topology/sklrt286/Makefile.am create mode 100644 src/conf/topology/sklrt286/data/Makefile.am create mode 100644 src/conf/topology/sklrt286/data/pvt.c create mode 100644 src/conf/topology/sklrt286/data/pvt_data.c create mode 100644 src/conf/topology/sklrt286/data/pvt_local.h create mode 100644 src/conf/topology/sklrt286/data/skl-tplg-interface.h create mode 100644 src/conf/topology/sklrt286/skl_i2s.conf -- 1.9.1

2 13

[alsa-devel] [PATCH v2 0/2] ASoC: Intel: Skylake: Add support for firmware manifest
by Vinod Koul 22 Apr '16

22 Apr '16

In the DSP firmware, module params like module_id and loadable flags can be modified. So, it is required to read these parameters in the driver from the firmware manifest. Changes in v2: - remove strings and use UUID in bin format Shreyas NC (2): ASoC: Intel: Skylake: Use UUID in binary format ASoC: Intel: Skylake: Add DSP firmware manifest parsing sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/skl-dsp-parse.c | 105 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-dsp-parse.h | 107 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-dsp.h | 2 +- sound/soc/intel/skylake/skl-sst-ipc.h | 4 + sound/soc/intel/skylake/skl-sst.c | 20 ++++- sound/soc/intel/skylake/skl-topology.c | 11 ++- sound/soc/intel/skylake/skl-topology.h | 2 +- sound/soc/intel/skylake/skl-tplg-interface.h | 2 +- 9 files changed, 243 insertions(+), 12 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.c create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.h -- 1.9.1

2 5

[alsa-devel] A&H Zed R16 not completely working (DICE Jr)
by Allan Klinbail 21 Apr '16

21 Apr '16

Hi, I thought I would try the DICE chipset driver from ALSA, as it appears that this will eventually take over from ffado.. It appears that it doesn't work very well for my Allen & Heath Zed -R16 which works very well under FFADO Using kernel 4.4.0 from Ubuntu The ALSA driver limited the device to only 16 inputs and outputs, however it has a total capability of 26 This is comprises of 16 inputs and outputs on the desk channels, a stereo main pair of outputs and inputs for the master bus 8 additional ins and outs to support the ADAT ports on the back (availoable in 44.1 and 48 khz modes, but not 88.2 or 96 Khz Similiarly I was not able to set the device to run at a latency lower than 256 frames in jack, however in FFADO I could go as low as 64 frames. Please let me know if I can help with any further testing to get this to the same level as the FFADO driver. Regards Allan

2 11

[alsa-devel] [PATCH v6 00/10] ASoC: Add mediatek HDMI codec support
by Philipp Zabel 21 Apr '16

21 Apr '16

these patches add an interface for Jyri's generic hdmi-codec driver [1] to the mediatek hdmi bridge driver. This version fixes the hdmi-codec notification patch and splits the jack support out of the ASoC machine driver patches. Changes since v5: - select SND_SOC_HDMI_CODEC and HDMI_NOTIFIERS where appropriate - split out jack support from machine driver patches - fix include of hdmi-notifier.h in hdmi-codec notification support - Drop applied patch: commit 05f5afd3f0d9 ("ASoC: mediatek: address dai link array entries by enum") The mediatek drm patches apply on top of https://patchwork.kernel.org/patch/8534251/ ("drm/mediatek: Add HDMI support"), and need this patch: [2] https://patchwork.kernel.org/patch/8091531/ ("video: hdmi: add helper function for N and CTS") The asoc patches depend on these two patches: https://patchwork.kernel.org/patch/7215121/ ("ALSA: pcm: add IEC958 channel status helper for hw_params") [1] https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") regards Philipp Koro Chen (1): ASoC: mediatek: Add HDMI dai-links in the mt8173-rt5650-rt5676 machine driver PC Liao (1): ASoC: mediatek: Add HDMI dai-links to the mt8173-rt5650 machine driver Philipp Zabel (8): drm/mediatek: hdmi: Add audio interface to the hdmi-codec driver video: rmk's HDMI notification prototype drm/mediatek: hdmi: issue notifications ASoC: hdmi-codec: Use HDMI notifications to add jack support ASoC: hdmi-codec: Add ELD control drm/mediatek: hdmi: use helper function for N and CTS calculation ASoC: mediatek: Add jack detection support to mt8173-rt5650-rt5676 machine driver ASoC: mediatek: Add jack detection support to the mt8173-rt5650 machine driver .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 +- .../devicetree/bindings/sound/mt8173-rt5650.txt | 5 +- drivers/gpu/drm/mediatek/Kconfig | 1 + drivers/gpu/drm/mediatek/mtk_cec.c | 11 ++ drivers/gpu/drm/mediatek/mtk_drm_hdmi_drv.c | 148 +++++++++++++++++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 89 +++++----- drivers/gpu/drm/mediatek/mtk_hdmi.h | 10 +- drivers/gpu/drm/mediatek/mtk_hdmi_hw.c | 179 ++++----------------- drivers/gpu/drm/mediatek/mtk_hdmi_hw.h | 4 +- drivers/video/Kconfig | 3 + drivers/video/Makefile | 1 + drivers/video/hdmi-notifier.c | 61 +++++++ include/linux/hdmi-notifier.h | 44 +++++ include/sound/hdmi-codec.h | 6 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdmi-codec.c | 123 +++++++++++++- sound/soc/mediatek/Kconfig | 2 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 48 ++++++ sound/soc/mediatek/mt8173-rt5650.c | 47 ++++++ 19 files changed, 576 insertions(+), 212 deletions(-) create mode 100644 drivers/video/hdmi-notifier.c create mode 100644 include/linux/hdmi-notifier.h -- 2.7.0

2 11

[alsa-devel] [PATCH 0/2] ASoC: tlv320aic32x4: Better model the input mixers
by Jeremy McDermond 21 Apr '16

21 Apr '16

The input mixers have resistors on all the input lines as well as an additional line that crosses between right and left pins. This patch enables those routings within the driver to provide more flexibility for different board configurations. These settings are documented on pages 135 and 136 of TI's SLAA557 document available at http://www.ti.com/lit/ml/slaa557/slaa557.pdf. This patch might introduce some breakage for current users because it changes some mixer routing settings from a straight switch to an enum for the different resistor values instead. I'm not sure there's anything we can do in code to mitigate these effects, but am open to requests for changes if there's something I'm not aware of. Jeremy McDermond (2): ASoC: tlv320aic32x4: Implement resistors on input pins ASoC: tlv320aic32x4: Add additional input pins sound/soc/codecs/tlv320aic32x4.c | 38 ++++++++++++++++++++++++++++++++------ 1 file changed, 32 insertions(+), 6 deletions(-) -- 2.5.0

2 4

[alsa-devel] [PATCH] ASoC: arizona: Prefer lower FRATIO in pseudo-fractional mode
by Richard Fitzgerald 21 Apr '16

21 Apr '16

When setting up an FLL in pseudo-fractional mode it is preferred to use a lower FRATIO if possible to give a higher reference clock frequency. This patch swaps the two loops in arizona_calc_fratio() so that lower FRATIOs are tried first. The decrementing loop is also changed to start from init_ratio because the original settings might already give a fractional value for N.K Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0cd27e0..d1e1127 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2039,7 +2039,21 @@ static int arizona_calc_fratio(struct arizona_fll *fll, init_ratio, Fref, refdiv); while (div <= ARIZONA_FLL_MAX_REFDIV) { - for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; + /* start from init_ratio because this may already give a + * fractional N.K + */ + for (ratio = init_ratio; ratio > 0; ratio--) { + if (target % (ratio * Fref)) { + cfg->refdiv = refdiv; + cfg->fratio = ratio - 1; + arizona_fll_dbg(fll, + "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", + Fref, refdiv, div, ratio); + return ratio; + } + } + + for (ratio = init_ratio + 1; ratio <= ARIZONA_FLL_MAX_FRATIO; ratio++) { if ((ARIZONA_FLL_VCO_CORNER / 2) / (fll->vco_mult * ratio) < Fref) { @@ -2065,17 +2079,6 @@ static int arizona_calc_fratio(struct arizona_fll *fll, } } - for (ratio = init_ratio - 1; ratio > 0; ratio--) { - if (target % (ratio * Fref)) { - cfg->refdiv = refdiv; - cfg->fratio = ratio - 1; - arizona_fll_dbg(fll, - "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", - Fref, refdiv, div, ratio); - return ratio; - } - } - div *= 2; Fref /= 2; refdiv++; -- 1.9.1

2 1

[alsa-devel] [PATCH v8] ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver
by PC Liao 21 Apr '16

21 Apr '16

This patch adds second I2S connection to rt5650 codec for capture path on mt8173-rt5650 machine driver. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- Changes since v7: Change the code for compatibility Changes since v6: Use subnode to determine the capture path --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 10 ++++ sound/soc/mediatek/mt8173-rt5650.c | 50 ++++++++++++++++++-- 2 files changed, 57 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index fe5a5ef..5bfa6b6 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -5,11 +5,21 @@ Required properties: - mediatek,audio-codec: the phandles of rt5650 codecs - mediatek,platform: the phandle of MT8173 ASoC platform +Optional subnodes: +- codec-capture : the subnode of rt5650 codec capture +Required codec-capture subnode properties: +- sound-dai: audio codec dai name on capture path + <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) + <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + codec-capture { + sound-dai = <&rt5650 1>; + }; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1..a27a667 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -85,12 +85,29 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + const char *codec_capture_dai = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, + RT5645_DA_STEREO_FILTER, RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_capture_dai, "rt5645-aif1")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_capture_dai, "rt5645-aif2")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else { + dev_warn(card->dev, + "Only one dai codec found in DTS, enabled rt5645 AD filter\n"); + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -110,6 +127,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { + /* Playback */ + .dai_name = "rt5645-aif1", + }, + { + /* Capture */ .dai_name = "rt5645-aif1", }, }; @@ -149,7 +171,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .cpu_dai_name = "I2S", .no_pcm = 1, .codecs = mt8173_rt5650_codecs, - .num_codecs = 1, + .num_codecs = 2, .init = mt8173_rt5650_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -177,6 +199,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_card; struct device_node *platform_node; + struct device_node *np; + const char *codec_capture_dai; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +223,26 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } + mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; + + if (of_find_node_by_name(platform_node, "codec-capture")) { + np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); + if (!np) { + dev_err(&pdev->dev, + "%s: Can't find codec-capture DT node\n", + __func__); + return -EINVAL; + } + ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); + if (ret < 0) { + dev_err(&pdev->dev, + "%s codec_capture_dai name fail %d\n", + __func__, ret); + return ret; + } + mt8173_rt5650_codecs[1].dai_name = codec_capture_dai; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

2 1

[alsa-devel] [sound:topic/hdmi 34/34] include/sound/hda_i915.h:29:1: error: expected identifier or '(' before '{' token
by kbuild test robot 21 Apr '16

21 Apr '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/hdmi head: e8de8fa72b640de1e30dcf62bab1a7b012d48cc2 commit: e8de8fa72b640de1e30dcf62bab1a7b012d48cc2 [34/34] ALSA: hda - Move i915 HSW/BDW BCLK setup to a common function config: i386-randconfig-i0-201616 (attached as .config) reproduce: git checkout e8de8fa72b640de1e30dcf62bab1a7b012d48cc2 # save the attached .config to linux build tree make ARCH=i386 All error/warnings (new ones prefixed by >>): In file included from sound/pci/hda/patch_hdmi.c:41:0: >> include/sound/hda_i915.h:29:1: error: expected identifier or '(' before '{' token { ^ >> include/sound/hda_i915.h:28:20: warning: 'snd_hdac_i915_set_bclk' used but never defined static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus); ^ vim +29 include/sound/hda_i915.h 98d8fc6c Mengdong Lin 2015-05-19 22 return 0; 98d8fc6c Mengdong Lin 2015-05-19 23 } 98d8fc6c Mengdong Lin 2015-05-19 24 static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable) 98d8fc6c Mengdong Lin 2015-05-19 25 { 98d8fc6c Mengdong Lin 2015-05-19 26 return 0; 98d8fc6c Mengdong Lin 2015-05-19 27 } e8de8fa7 Takashi Iwai 2016-04-21 @28 static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus); 98d8fc6c Mengdong Lin 2015-05-19 @29 { 98d8fc6c Mengdong Lin 2015-05-19 30 } d745f5e7 Takashi Iwai 2016-03-21 31 static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec, d745f5e7 Takashi Iwai 2016-03-21 32 hda_nid_t nid, int rate) :::::: The code at line 29 was first introduced by commit :::::: 98d8fc6c5d3652e91c61d78941e0fa6f94771d67 ALSA: hda - Move hda_i915.c from sound/pci/hda to sound/hda :::::: TO: Mengdong Lin <mengdong.lin(a)intel.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix possible race on regmap bypass flip
by Takashi Iwai 21 Apr '16

21 Apr '16

HD-audio driver uses regmap cache bypass feature for reading a raw value without the cache. But this is racy since both the cached and the uncached reads may occur concurrently. The former is done via the normal control API access while the latter comes from the proc file read. Even though the regmap itself has the protection against the concurrent accesses, the flag set/reset is done without the protection, so it may lead to inconsistent state of bypass flag that doesn't match with the current read and occasionally result in a kernel WARNING like: WARNING: CPU: 3 PID: 2731 at drivers/base/regmap/regcache.c:499 regcache_cache_only+0x78/0x93 One way to work around such a problem is to wrap with a mutex. But in this case, the solution is simpler: for the uncached read, we just skip the regmap and directly calls its accessor. The verb execution there is protected by itself, so basically it's safe to call individually. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=116171 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_regmap.h | 2 ++ sound/hda/hdac_device.c | 10 ++++------ sound/hda/hdac_regmap.c | 40 ++++++++++++++++++++++++++++------------ 3 files changed, 34 insertions(+), 18 deletions(-) diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index 2767c55a641e..ca64f0f50b45 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -17,6 +17,8 @@ int snd_hdac_regmap_add_vendor_verb(struct hdac_device *codec, unsigned int verb); int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg, unsigned int *val); +int snd_hdac_regmap_read_raw_uncached(struct hdac_device *codec, + unsigned int reg, unsigned int *val); int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, unsigned int val); int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg, diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index d1a4d6973330..03c9872c31cf 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -299,13 +299,11 @@ EXPORT_SYMBOL_GPL(_snd_hdac_read_parm); int snd_hdac_read_parm_uncached(struct hdac_device *codec, hda_nid_t nid, int parm) { - int val; + unsigned int cmd, val; - if (codec->regmap) - regcache_cache_bypass(codec->regmap, true); - val = snd_hdac_read_parm(codec, nid, parm); - if (codec->regmap) - regcache_cache_bypass(codec->regmap, false); + cmd = snd_hdac_regmap_encode_verb(nid, AC_VERB_PARAMETERS) | parm; + if (snd_hdac_regmap_read_raw_uncached(codec, cmd, &val) < 0) + return -1; return val; } EXPORT_SYMBOL_GPL(snd_hdac_read_parm_uncached); diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index bdbcd6b75ff6..87041ddd29cb 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -453,14 +453,30 @@ int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, EXPORT_SYMBOL_GPL(snd_hdac_regmap_write_raw); static int reg_raw_read(struct hdac_device *codec, unsigned int reg, - unsigned int *val) + unsigned int *val, bool uncached) { - if (!codec->regmap) + if (uncached || !codec->regmap) return hda_reg_read(codec, reg, val); else return regmap_read(codec->regmap, reg, val); } +static int __snd_hdac_regmap_read_raw(struct hdac_device *codec, + unsigned int reg, unsigned int *val, + bool uncached) +{ + int err; + + err = reg_raw_read(codec, reg, val, uncached); + if (err == -EAGAIN) { + err = snd_hdac_power_up_pm(codec); + if (!err) + err = reg_raw_read(codec, reg, val, uncached); + snd_hdac_power_down_pm(codec); + } + return err; +} + /** * snd_hdac_regmap_read_raw - read a pseudo register with power mgmt * @codec: the codec object @@ -472,19 +488,19 @@ static int reg_raw_read(struct hdac_device *codec, unsigned int reg, int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg, unsigned int *val) { - int err; - - err = reg_raw_read(codec, reg, val); - if (err == -EAGAIN) { - err = snd_hdac_power_up_pm(codec); - if (!err) - err = reg_raw_read(codec, reg, val); - snd_hdac_power_down_pm(codec); - } - return err; + return __snd_hdac_regmap_read_raw(codec, reg, val, false); } EXPORT_SYMBOL_GPL(snd_hdac_regmap_read_raw); +/* Works like snd_hdac_regmap_read_raw(), but this doesn't read from the + * cache but always via hda verbs. + */ +int snd_hdac_regmap_read_raw_uncached(struct hdac_device *codec, + unsigned int reg, unsigned int *val) +{ + return __snd_hdac_regmap_read_raw(codec, reg, val, true); +} + /** * snd_hdac_regmap_update_raw - update a pseudo register with power mgmt * @codec: the codec object -- 2.8.1

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[alsa-devel] [PATCH] ALSA: pcxhr: Fix missing mutex unlock
by Takashi Iwai 21 Apr '16

21 Apr '16

The commit [9bef72bdb26e: ALSA: pcxhr: Use nonatomic PCM ops] converted to non-atomic PCM ops, but shamelessly with an unbalanced mutex locking, which leads to the hangup easily. Fix it. Fixes: 9bef72bdb26e ('ALSA: pcxhr: Use nonatomic PCM ops') Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=116441 Cc: <stable(a)vger.kernel.org> # 3.18+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/pcxhr/pcxhr_core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index c5194f5b150a..d7e71f309299 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -1341,5 +1341,6 @@ irqreturn_t pcxhr_threaded_irq(int irq, void *dev_id) } pcxhr_msg_thread(mgr); + mutex_unlock(&mgr->lock); return IRQ_HANDLED; } -- 2.8.1

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[alsa-devel] [PATCH v2 0/6] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 21 Apr '16

21 Apr '16

This series of patches adds devicetree support for the davinci McBSP audio interface. Changes in version 2 of the patch: * add missing TC channel in dmas properties (for compatibility with the new EDMA3 binding) * remove "-audio" postfix from the compatible string * remove "channel-combine" property * of_match_table renamed consistently with the rest of the file (davinci_i2s_match) * devicetree DMA configuration in probe simplified

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[alsa-devel] [PATCH 0/2] Yet more hardening of ALSA hrtimer races
by Takashi Iwai 21 Apr '16

21 Apr '16

Hi, this is a resubmission of patches for hardening the ALSA hrtimer backend against open/close races. Takashi === Takashi Iwai (2): ALSA: timer: Allow backend disabling start/stop from handler ALSA: hrtimer: Use manual start/stop in callback include/sound/timer.h | 12 +++++++++++- sound/core/hrtimer.c | 51 ++++++++++++++++++++++++++++++++++++++------------- sound/core/timer.c | 24 ++++++++++++++++++------ 3 files changed, 67 insertions(+), 20 deletions(-) -- 2.8.1

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[alsa-devel] [PATCH v3 0/2] sti: add audio interface to the hdmi driver
by Arnaud Pouliquen 21 Apr '16

21 Apr '16

This patchset implements audio interface in HDMI drm driver. Implementation is based on ASoC generic hdmi codec driver( https://patchwork.kernel.org/patch/8713141/). It also proposes helper functions to compute N and CTS parameters according to HDMI 1.4b specification. V3: - video: hdmi: add helper function for N and CTS Also used on Mediatek platform (https://patchwork.kernel.org/patch/8887341) delta vs V2: - typo fixes - if/else code optimisation - drm: sti: Add ASoC generic hdmi codec support. - typo fixes - add audio registers in debugfs information V2: RFC https://patchwork.kernel.org/patch/8091531/("video: hdmi: add helper function for N and CTS") https://patchwork.kernel.org/patch/8091561/("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") - patch: video: hdmi: add helper function for N and CTS Fixes based on Russel King remarks - Duplicate function to have a separte treatment for coherent and non-coherent clocks - Add ratio field for alternate CTS value - Clock frequency in Hz for TMDS and audio clocks - Add information concerning clocks and CTS calculation. V1: This RFC is the implementation of audio HDMI on sti platform based on generic hdmi-codec driver: https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") https://patchwork.kernel.org/patch/8062611/ ("video: hdmi: add helper function for N and CTS") Arnaud Pouliquen (2): drm: sti: Add ASoC generic hdmi codec support. video: hdmi: add helper function for N and CTS drivers/gpu/drm/sti/Kconfig | 1 + drivers/gpu/drm/sti/sti_hdmi.c | 248 ++++++++++++++++++++++++++++++++++++++--- drivers/gpu/drm/sti/sti_hdmi.h | 13 +++ drivers/video/hdmi.c | 202 +++++++++++++++++++++++++++++++++ include/linux/hdmi.h | 22 ++++ 5 files changed, 469 insertions(+), 17 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v5 0/4] Propagate errors out from compressed streams
by Charles Keepax 21 Apr '16

21 Apr '16

If the DSP suffers an unrecoverable error, the driver likely knows about this, however the framework may not get informed because errors returned from pointer requests are ignored within the framework. Things work out fine if user-space is doing a read as reads return error status back to user-space so the user can find out that things have gone bad. However, if user-space is doing an avail request there is no path for the error to come back up to user-space. The pointer request returns zero available data, so a read never happens and we basically just end up sitting waiting for data on a stream that we know full well has died. This patch set attempts to address this and ensure that errors are fully propagated to user-space and we don't ever end up wait for data that will never come. Changes since v4: - Fix issue noticed by Vinod whereby the error state was being overwritten when snd_compr_stop was called It might be easiest to put the ASoC changes through Mark's tree as there are some merge conflicts otherwise, as such I have based this series on Mark's tree. Thanks, Charles Charles Keepax (4): ALSA: compress: Replace complex if statement with switch ALSA: compress: Add function to indicate the stream has gone bad ASoC: wm_adsp: Use new snd_compr_stop_error to signal stream failure ASoC: compress: Pass error out of soc_compr_pointer include/sound/compress_driver.h | 5 +++ sound/core/compress_offload.c | 79 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 11 ++++-- sound/soc/soc-compress.c | 5 +-- 4 files changed, 91 insertions(+), 9 deletions(-) -- 2.1.4

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