November 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: atmel_ssc_dai: Match the CMR divider only in full duplex.
by Peter Rosin 03 Nov '14

03 Nov '14

>From 86be84c4de4e7b21cfda9656a02a902c543210af Mon Sep 17 00:00:00 2001 From: Peter Rosin <peda(a)axentia.se> Date: Wed, 22 Oct 2014 16:45:29 +0200 Subject: [PATCH v2] ASoC: atmel_ssc_dai: Match the CMR divider only in full duplex. The CMR divider register is shared by playback and capture. The SSC driver therefore tries to enforce rules so that the needed register content do not conflict during simultaneous playback/capture. However, the implementation also prevents changing the register content in half-duplex scenarios, which is needed when using the OSS API. Thus, only lock the divider if there is a stream in the other direction. Fixes the below program to not fail with the atmel ssc dai in master mode. #include <sys/ioctl.h> #include <unistd.h> #include <fcntl.h> #include <sys/soundcard.h> int main(void) { int fd; int format; int channels; int speed; if ((fd = open("/dev/dsp", O_WRONLY, 0)) == -1) { perror("open"); return 1; } format = AFMT_S16_LE; if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1) { perror("SNDCTL_DSP_SETFMT"); return 1; } channels = 2; if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) { perror("SNDCTL_DSP_CHANNELS"); return 1; } speed = 22025; if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { perror("SNDCTL_DSP_SPEED"); return 1; } return 0; } Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index de433cfd..9ae8475 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -310,7 +310,10 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, * transmit and receive, so if a value has already * been set, it must match this value. */ - if (ssc_p->cmr_div == 0) + if (ssc_p->dir_mask != + (SSC_DIR_MASK_PLAYBACK | SSC_DIR_MASK_CAPTURE)) + ssc_p->cmr_div = div; + else if (ssc_p->cmr_div == 0) ssc_p->cmr_div = div; else if (div != ssc_p->cmr_div) -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: simple-card: add "invert" property for detect GPIOs
by Jianqun 03 Nov '14

03 Nov '14

Since hardware may invert detect GPIO of headphone or mic, add one property to support software invert. Signed-off-by: Jianqun <jay.xu(a)rock-chips.com> --- sound/soc/generic/simple-card.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d1b7293..f1bb537 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -29,7 +29,9 @@ struct simple_card_data { } *dai_props; unsigned int mclk_fs; int gpio_hp_det; + int gpio_hp_det_invert; int gpio_mic_det; + int gpio_mic_det_invert; struct snd_soc_dai_link dai_link[]; /* dynamically allocated */ }; @@ -148,6 +150,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) simple_card_hp_jack_pins); simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det; + simple_card_hp_jack_gpio.invert = priv->gpio_hp_det_invert; snd_soc_jack_add_gpios(&simple_card_hp_jack, 1, &simple_card_hp_jack_gpio); } @@ -159,6 +162,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(simple_card_mic_jack_pins), simple_card_mic_jack_pins); simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det; + simple_card_mic_jack_gpio.invert = priv->gpio_mic_det_invert; snd_soc_jack_add_gpios(&simple_card_mic_jack, 1, &simple_card_mic_jack_gpio); } @@ -381,6 +385,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, struct simple_card_data *priv) { struct device *dev = simple_priv_to_dev(priv); + enum of_gpio_flags flags; u32 val; int ret; @@ -436,13 +441,15 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } - priv->gpio_hp_det = of_get_named_gpio(node, - "simple-audio-card,hp-det-gpio", 0); + priv->gpio_hp_det = of_get_named_gpio_flags(node, + "simple-audio-card,hp-det-gpio", 0, &flags); + priv->gpio_hp_det_invert = !!(flags & OF_GPIO_ACTIVE_LOW); if (priv->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; - priv->gpio_mic_det = of_get_named_gpio(node, - "simple-audio-card,mic-det-gpio", 0); + priv->gpio_mic_det = of_get_named_gpio_flags(node, + "simple-audio-card,mic-det-gpio", 0, &flags); + priv->gpio_mic_det_invert = !!(flags & OF_GPIO_ACTIVE_LOW); if (priv->gpio_mic_det == -EPROBE_DEFER) return -EPROBE_DEFER; -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: Remove snd_soc_platform_driver suspend/resume callbacks
by Lars-Peter Clausen 03 Nov '14

03 Nov '14

Those are unused and new drivers should use device driver suspend/resume. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 4 ---- sound/soc/soc-core.c | 10 ---------- 2 files changed, 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ad47e96..edbb07b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -857,8 +857,6 @@ struct snd_soc_platform_driver { int (*probe)(struct snd_soc_platform *); int (*remove)(struct snd_soc_platform *); - int (*suspend)(struct snd_soc_dai *dai); - int (*resume)(struct snd_soc_dai *dai); struct snd_soc_component_driver component_driver; /* pcm creation and destruction */ @@ -891,8 +889,6 @@ struct snd_soc_platform { struct device *dev; const struct snd_soc_platform_driver *driver; - unsigned int suspended:1; /* platform is suspended */ - struct list_head list; struct snd_soc_component component; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dd8234d..00e944e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -592,17 +592,12 @@ int snd_soc_suspend(struct device *dev) for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_platform *platform = card->rtd[i].platform; if (card->rtd[i].dai_link->ignore_suspend) continue; if (cpu_dai->driver->suspend && !cpu_dai->driver->ac97_control) cpu_dai->driver->suspend(cpu_dai); - if (platform->driver->suspend && !platform->suspended) { - platform->driver->suspend(cpu_dai); - platform->suspended = 1; - } } /* close any waiting streams and save state */ @@ -775,17 +770,12 @@ static void soc_resume_deferred(struct work_struct *work) for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_platform *platform = card->rtd[i].platform; if (card->rtd[i].dai_link->ignore_suspend) continue; if (cpu_dai->driver->resume && !cpu_dai->driver->ac97_control) cpu_dai->driver->resume(cpu_dai); - if (platform->driver->resume && platform->suspended) { - platform->driver->resume(cpu_dai); - platform->suspended = 0; - } } if (card->resume_post) -- 1.8.0

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[alsa-devel] [PATCH] ASoC: rockchip-i2s: add rockchip-max98090 driver support
by Jianqun 03 Nov '14

03 Nov '14

This patch is to add support for rockchip board with max98090 codec. It also needs sound node in dt file. Tested on rockchip board based on rk3288. Signed-off-by: Jianqun <jay.xu(a)rock-chips.com> --- sound/soc/rockchip/Kconfig | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index 78fc159..482cbba 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -9,3 +9,16 @@ config SND_SOC_ROCKCHIP config SND_SOC_ROCKCHIP_I2S tristate + depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP + help + Say Y or M if you want to add support for I2S driver for Rockchip + I2S device. + +config SND_SOC_ROCKCHIP_MAX98090 + tristate "ASoC support Rockchip board with max98090" + depends on I2C && GPIOLIB + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98090 + help + Say Y or M here if you want to add support for Rockchio board using + a max98090 codec. -- 1.9.1

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[alsa-devel] [PATCH] ASoC: tlv320aic23: add dependencies on I2C/SPI_MASTER
by Max Filippov 03 Nov '14

03 Nov '14

This fixes build errors in configurations with I2C/SPI master disabled. Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- sound/soc/codecs/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7dad805..b0bcd0d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -553,10 +553,12 @@ config SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC23_I2C tristate "Texas Instruments TLV320AIC23 audio CODEC - I2C" + depends on I2C select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC23_SPI tristate "Texas Instruments TLV320AIC23 audio CODEC - SPI" + depends on SPI_MASTER select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC26 -- 1.8.1.4

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[alsa-devel] [PATCH] [RESEND] snd-es18xx: Add GPO controls
by Ondrej Zary 03 Nov '14

03 Nov '14

Add GPO0 and GPO1 (General Purpose Outputs) controls to mixer. These can be used on some cards to control amplifier mute (seen in ES1868 datasheet) or additional onboard chips such as QX2130 QXpander processor. These GPOs are present on ES1868, ES1869, ES1887 and ES1888 chips. Tested on ES1868 with QX2130. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/isa/es18xx.c | 49 +++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 39 insertions(+), 10 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 6faaac6..8966ed2 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -156,6 +156,7 @@ struct snd_es18xx { #define ES18XX_I2S 0x0200 /* I2S mixer control */ #define ES18XX_MUTEREC 0x0400 /* Record source can be muted */ #define ES18XX_CONTROL 0x0800 /* Has control ports */ +#define ES18XX_GPO_2BIT 0x1000 /* GPO0,1 controlled by PM port */ /* Power Management */ #define ES18XX_PM 0x07 @@ -1136,11 +1137,14 @@ static int snd_es18xx_reg_read(struct snd_es18xx *chip, unsigned char reg) return snd_es18xx_read(chip, reg); } -#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, invert) \ +#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, flags) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_es18xx_info_single, \ .get = snd_es18xx_get_single, .put = snd_es18xx_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } + .private_value = reg | (shift << 8) | (mask << 16) | (flags << 24) } + +#define ES18XX_FL_INVERT (1 << 0) +#define ES18XX_FL_PMPORT (1 << 1) static int snd_es18xx_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1159,10 +1163,14 @@ static int snd_es18xx_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; int val; - - val = snd_es18xx_reg_read(chip, reg); + + if (pm_port) + val = inb(chip->port + ES18XX_PM); + else + val = snd_es18xx_reg_read(chip, reg); ucontrol->value.integer.value[0] = (val >> shift) & mask; if (invert) ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; @@ -1175,7 +1183,8 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; unsigned char val; val = (ucontrol->value.integer.value[0] & mask); @@ -1183,6 +1192,12 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e val = mask - val; mask <<= shift; val <<= shift; + if (pm_port) { + val = (inb(chip->port + ES18XX_PM) & ~mask) | val; + outb(val, chip->port + ES18XX_PM); + return 0; + } + return snd_es18xx_reg_bits(chip, reg, mask, val) != val; } @@ -1304,7 +1319,7 @@ static struct snd_kcontrol_new snd_es18xx_opt_speaker = ES18XX_SINGLE("Beep Playback Volume", 0, 0x3c, 0, 7, 0); static struct snd_kcontrol_new snd_es18xx_opt_1869[] = { -ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), +ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, ES18XX_FL_INVERT), ES18XX_SINGLE("Video Playback Switch", 0, 0x7f, 0, 1, 0), ES18XX_DOUBLE("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0), ES18XX_DOUBLE("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0) @@ -1363,6 +1378,11 @@ static struct snd_kcontrol_new snd_es18xx_hw_volume_controls[] = { ES18XX_SINGLE("Hardware Master Volume Split", 0, 0x64, 7, 1, 0), }; +static struct snd_kcontrol_new snd_es18xx_opt_gpo_2bit[] = { +ES18XX_SINGLE("GPO0 Switch", 0, ES18XX_PM, 0, 1, ES18XX_FL_PMPORT), +ES18XX_SINGLE("GPO1 Switch", 0, ES18XX_PM, 1, 1, ES18XX_FL_PMPORT), +}; + static int snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { int data; @@ -1629,10 +1649,10 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, switch (chip->version) { case 0x1868: - chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL; + chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL | ES18XX_GPO_2BIT; break; case 0x1869: - chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV; + chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV | ES18XX_GPO_2BIT; break; case 0x1878: chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_I2S | ES18XX_CONTROL; @@ -1642,7 +1662,7 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, break; case 0x1887: case 0x1888: - chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; + chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME | ES18XX_GPO_2BIT; break; default: snd_printk(KERN_ERR "[0x%lx] unsupported chip ES%x\n", @@ -1944,6 +1964,15 @@ static int snd_es18xx_mixer(struct snd_card *card) return err; } } + if (chip->caps & ES18XX_GPO_2BIT) { + for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_opt_gpo_2bit); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_es18xx_opt_gpo_2bit[idx], + chip)); + if (err < 0) + return err; + } + } return 0; } -- Ondrej Zary

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[alsa-devel] [PATCH 1/2] ALSA: echoaudio: add reference of struct echoaudio
by Sudip Mukherjee 03 Nov '14

03 Nov '14

added reference of struct echoaudio to free_firmware function. this structure will be later used to get a reference of the card when converting snd_printk to dev_* in the next patch of the series. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/echoaudio/echoaudio.c | 3 ++- sound/pci/echoaudio/echoaudio.h | 3 ++- sound/pci/echoaudio/echoaudio_dsp.c | 10 +++++----- 3 files changed, 9 insertions(+), 7 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index d82321f..db1b247 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -69,7 +69,8 @@ static int get_firmware(const struct firmware **fw_entry, -static void free_firmware(const struct firmware *fw_entry) +static void free_firmware(const struct firmware *fw_entry, + struct echoaudio *chip) { #ifdef CONFIG_PM_SLEEP DE_ACT(("firmware not released (kept in cache)\n")); diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index b86b88d..a4f112a 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -468,7 +468,8 @@ static int wait_handshake(struct echoaudio *chip); static int send_vector(struct echoaudio *chip, u32 command); static int get_firmware(const struct firmware **fw_entry, struct echoaudio *chip, const short fw_index); -static void free_firmware(const struct firmware *fw_entry); +static void free_firmware(const struct firmware *fw_entry, + struct echoaudio *chip); #ifdef ECHOCARD_HAS_MIDI static int enable_midi_input(struct echoaudio *chip, char enable); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 5a6a217..977b2bd 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -206,12 +206,12 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) } DE_INIT(("ASIC loaded\n")); - free_firmware(fw); + free_firmware(fw, chip); return 0; la_error: DE_INIT(("failed on write_dsp\n")); - free_firmware(fw); + free_firmware(fw, chip); return -EIO; } @@ -317,11 +317,11 @@ static int install_resident_loader(struct echoaudio *chip) } DE_INIT(("Resident loader successfully installed\n")); - free_firmware(fw); + free_firmware(fw, chip); return 0; irl_error: - free_firmware(fw); + free_firmware(fw, chip); return -EIO; } @@ -491,7 +491,7 @@ static int load_firmware(struct echoaudio *chip) if (err < 0) return err; err = load_dsp(chip, (u16 *)fw->data); - free_firmware(fw); + free_firmware(fw, chip); if (err < 0) return err; -- 1.8.1.2

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[alsa-devel] [PATCH 0/3] Kill the rest snd_print*() usages
by Takashi Iwai 03 Nov '14

03 Nov '14

Hi, it's a small series to get rid of snd_print*() usages in a few PCI sound drivers. There remain still a few, I know, but this is just a matter of cleanup, so let them be there... [PATCH 1/3] ALSA: ctxfi: Kill the rest snd_print*() [PATCH 2/3] ALSA: pcxhr: Kill the rest snd_print*() [PATCH 3/3] ALSA: au88x0: Kill the rest snd_print*() Takashi

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Re: [alsa-devel] ASoC: Intel: mrfld: add DSP core controls
by Dan Carpenter 03 Nov '14

03 Nov '14

Hello Vinod Koul, The patch 24c8d14192cc: "ASoC: Intel: mrfld: add DSP core controls" from Oct 15, 2014, leads to the following static checker warning: sound/soc/intel/sst-atom-controls.c:1354 sst_map_modules_to_pipe() warn: variable dereferenced before check 'platform' (see line 1353) sound/soc/intel/sst-atom-controls.c 1348 static int sst_map_modules_to_pipe(struct snd_soc_platform *platform) 1349 { 1350 struct snd_soc_dapm_widget *w; 1351 int ret = 0; 1352 1353 list_for_each_entry(w, &platform->component.card->widgets, list) { ^^^^^^^^ New dereference. 1354 if (platform && is_sst_dapm_widget(w) && (w->priv)) { ^^^^^^^^ New check. 1355 struct sst_ids *ids = w->priv; 1356 1357 dev_dbg(platform->dev, "widget type=%d name=%s\n", 1358 w->id, w->name); 1359 INIT_LIST_HEAD(&ids->algo_list); 1360 INIT_LIST_HEAD(&ids->gain_list); 1361 ret = sst_fill_widget_module_info(w, platform); 1362 1363 if (ret < 0) 1364 return ret; 1365 1366 /* fill linked widgets */ 1367 if (ids->parent_wname != NULL) 1368 sst_fill_linked_widgets(platform, ids); 1369 } 1370 } 1371 return 0; 1372 } regards, dan carpenter

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[alsa-devel] [PATCH 0/4] ASoC: Intel: add compressed support
by Vinod Koul 03 Nov '14

03 Nov '14

This add compressed support in sst ipc driver. Along with it two minor fixes on driver as well Fang, Yang A (1): ASoC: Intel: use correct firmware name Subhransu S. Prusty (1): ASoC: mfld-compress: implement .power callback Vinod Koul (2): ASoC: Intel: sst: load firmware using async callback ASoC: Intel: sst - add compressed ops handling sound/soc/intel/sst-mfld-platform-compress.c | 8 +- sound/soc/intel/sst-mfld-platform.h | 1 + sound/soc/intel/sst/sst.c | 15 ++ sound/soc/intel/sst/sst_drv_interface.c | 285 ++++++++++++++++++++++++++ sound/soc/intel/sst/sst_loader.c | 5 +- 5 files changed, 309 insertions(+), 5 deletions(-)

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