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

[alsa-devel] [PATCH] ES938 support for ES18xx driver
by Ondrej Zary 26 Sep '14

26 Sep '14

Add support for ES938 3-D Audio Effects Processor found on some ES18xx (and possibly other) sound cards, doing bass/treble and 3D control. ES938 is controlled by MIDI SysEx commands sent through card's MPU401 port. The code opens/closes the MIDI device everytime a mixer control value is changed so userspace apps can still use the MIDI port. Changing the mixer controls will fail when the MIDI port is open by an application. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/isa/Kconfig | 4 ++ sound/isa/Makefile | 2 + sound/isa/es18xx.c | 11 ++- sound/isa/es938.c | 203 ++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/es938.h | 29 ++++++++ 5 files changed, 248 insertions(+), 1 deletion(-) create mode 100644 sound/isa/es938.c create mode 100644 sound/isa/es938.h diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 0216475..db0b678 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -17,6 +17,9 @@ config SND_SB16_DSP select SND_PCM select SND_SB_COMMON +config SND_ES938 + tristate + menuconfig SND_ISA bool "ISA sound devices" depends on ISA && ISA_DMA_API @@ -183,6 +186,7 @@ config SND_ES18XX select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM + select SND_ES938 help Say Y here to include support for ESS AudioDrive ES18xx chips. diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 9a15f14..d59e0bf 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -8,6 +8,7 @@ snd-als100-objs := als100.o snd-azt2320-objs := azt2320.o snd-cmi8328-objs := cmi8328.o snd-cmi8330-objs := cmi8330.o +snd-es938-objs := es938.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o @@ -19,6 +20,7 @@ obj-$(CONFIG_SND_ALS100) += snd-als100.o obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o obj-$(CONFIG_SND_CMI8328) += snd-cmi8328.o obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o +obj-$(CONFIG_SND_ES938) += snd-es938.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 6faaac6..139cd3b 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -96,6 +96,7 @@ #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> +#include "es938.h" #define PFX "es18xx: " @@ -122,6 +123,7 @@ struct snd_es18xx { struct snd_pcm_substream *playback_b_substream; struct snd_rawmidi *rmidi; + struct snd_es938 es938; struct snd_kcontrol *hw_volume; struct snd_kcontrol *hw_switch; @@ -2167,7 +2169,14 @@ static int snd_audiodrive_probe(struct snd_card *card, int dev) return err; } - return snd_card_register(card); + err = snd_card_register(card); + if (err < 0) + return err; + + if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) + snd_es938_init(&chip->es938, card, 0, 0); + + return 0; } static int snd_es18xx_isa_match(struct device *pdev, unsigned int dev) diff --git a/sound/isa/es938.c b/sound/isa/es938.c new file mode 100644 index 0000000..ea1b25c --- /dev/null +++ b/sound/isa/es938.c @@ -0,0 +1,203 @@ +/* + * Driver for ESS ES938 3-D Audio Effects Processor + * Copyright (c) 2014 Ondrej Zary + * + * + * 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/module.h> +#include <sound/asoundef.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/rawmidi.h> +#include "es938.h" + +MODULE_AUTHOR("Ondrej Zary"); +MODULE_DESCRIPTION("ESS ES938"); +MODULE_LICENSE("GPL"); + +static int snd_es938_read_reg(struct snd_es938 *chip, u8 reg, u8 *out) +{ + u8 buf[8]; + int i = 0, res; + u8 sysex[] = { MIDI_CMD_COMMON_SYSEX, 0x00, 0x00, ES938_ID, + ES938_CMD_REG_R, reg, MIDI_CMD_COMMON_SYSEX_END }; + unsigned long end_time; + + snd_rawmidi_kernel_write(chip->rfile.output, sysex, sizeof(sysex)); + + memset(buf, 0, sizeof(buf)); + end_time = jiffies + msecs_to_jiffies(100); + while (i < sizeof(buf)) { + res = snd_rawmidi_kernel_read(chip->rfile.input, buf + i, + sizeof(buf) - i); + if (res > 0) + i += res; + if (time_after(jiffies, end_time)) + return -1; + } + + /* check reply */ + if (memcmp(buf, sysex, 6) || buf[7] != MIDI_CMD_COMMON_SYSEX_END) + return -1; + + if (out) + *out = buf[6]; + + return 0; +} + +static void snd_es938_write_reg(struct snd_es938 *chip, u8 reg, u8 val) +{ + u8 sysex[] = { MIDI_CMD_COMMON_SYSEX, 0x00, 0x00, ES938_ID, + ES938_CMD_REG_W, reg, val, MIDI_CMD_COMMON_SYSEX_END }; + + snd_rawmidi_kernel_write(chip->rfile.output, sysex, sizeof(sysex)); + chip->regs[reg] = val; +} + +#define ES938_MIXER(xname, xindex, reg, shift, mask) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ +.info = snd_es938_info_mixer, \ +.get = snd_es938_get_mixer, .put = snd_es938_put_mixer, \ +.private_value = reg | (shift << 8) | (mask << 16) } + +#define ES938_MIXER_TLV(xname, xindex, reg, shift, mask, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ +.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ +.info = snd_es938_info_mixer, \ +.get = snd_es938_get_mixer, .put = snd_es938_put_mixer, \ +.private_value = reg | (shift << 8) | (mask << 16), \ +.tlv = { .p = (xtlv) } } + +static const DECLARE_TLV_DB_SCALE(db_scale_tone, -900, 300, 0); + +static struct snd_kcontrol_new snd_es938_controls[] = { +ES938_MIXER_TLV("Tone Control - Bass", 0, ES938_REG_TONE, 0, 7, db_scale_tone), +ES938_MIXER_TLV("Tone Control - Treble", 0, ES938_REG_TONE, 4, 7, db_scale_tone), +ES938_MIXER("3D Control - Level", 0, ES938_REG_SPATIAL, 1, 63), +ES938_MIXER("3D Control - Switch", 0, ES938_REG_SPATIAL_EN, 0, 1), +}; + +int snd_es938_init(struct snd_es938 *chip, struct snd_card *card, int device, + int subdevice) +{ + int ret, i; + + ret = snd_rawmidi_kernel_open(card, device, subdevice, + SNDRV_RAWMIDI_LFLG_OPEN | SNDRV_RAWMIDI_LFLG_APPEND, + &chip->rfile); + if (ret < 0) { + snd_printk(KERN_WARNING "es938: unable to open MIDI device\n"); + return ret; + } + + /* try to read a register to detect chip presence */ + if (snd_es938_read_reg(chip, ES938_REG_MISC, NULL) < 0) { + ret = -ENODEV; + goto err; + } + + /* write default values (there's no reset) */ + snd_es938_write_reg(chip, ES938_REG_MISC, 0x49); + snd_es938_write_reg(chip, ES938_REG_TONE, 0x33); + /* reserved bit 0 must be set for 3D to work */ + snd_es938_write_reg(chip, ES938_REG_SPATIAL, 0x01); + /* datasheet specifies invalid value 0x00 as default */ + snd_es938_write_reg(chip, ES938_REG_SPATIAL_EN, 0x02); + snd_es938_write_reg(chip, ES938_REG_POWER, 0x0e); + + strlcat(card->mixername, " + ES938", sizeof(card->mixername)); + for (i = 0; i < ARRAY_SIZE(snd_es938_controls); i++) { + ret = snd_ctl_add(card, + snd_ctl_new1(&snd_es938_controls[i], chip)); + if (ret < 0) + goto err; + } + + chip->card = card; + chip->device = device; + chip->subdevice = subdevice; +err: + snd_rawmidi_kernel_release(&chip->rfile); + + return ret; +} +EXPORT_SYMBOL(snd_es938_init); + +int snd_es938_info_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : + SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + return 0; +} +EXPORT_SYMBOL(snd_es938_info_mixer); + +int snd_es938_get_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_es938 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u8 val = chip->regs[reg]; + + ucontrol->value.integer.value[0] = (val >> shift) & mask; + return 0; +} +EXPORT_SYMBOL(snd_es938_get_mixer); + +int snd_es938_put_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_es938 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + u8 val = ucontrol->value.integer.value[0] & mask; + u8 oldval = chip->regs[reg]; + u8 newval; + int err; + + mask <<= shift; + val <<= shift; + + newval = (oldval & ~mask) | val; + if (newval == oldval) + return 0; + + /* this will fail if the MIDI port is used by an application */ + err = snd_rawmidi_kernel_open(chip->card, chip->device, + chip->subdevice, + SNDRV_RAWMIDI_LFLG_OPEN | SNDRV_RAWMIDI_LFLG_APPEND, + &chip->rfile); + if (err < 0) + return err; + + snd_es938_write_reg(chip, reg, newval); + + snd_rawmidi_kernel_release(&chip->rfile); + + return 1; +} +EXPORT_SYMBOL(snd_es938_put_mixer); diff --git a/sound/isa/es938.h b/sound/isa/es938.h new file mode 100644 index 0000000..b67dfc7 --- /dev/null +++ b/sound/isa/es938.h @@ -0,0 +1,29 @@ +#include <sound/tlv.h> + +#define ES938_ID 0x7b + +#define ES938_CMD_REG_R 0x7e +#define ES938_CMD_REG_W 0x7f + +#define ES938_REG_MISC 0 +#define ES938_REG_TONE 1 +#define ES938_REG_SPATIAL 5 +#define ES938_REG_SPATIAL_EN 6 +#define ES938_REG_POWER 7 + +struct snd_es938 { + u8 regs[8]; + struct snd_card *card; + int device; + int subdevice; + struct snd_rawmidi_file rfile; +}; + +int snd_es938_init(struct snd_es938 *chip, struct snd_card *card, int device, + int subdevice); +int snd_es938_info_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_es938_get_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_es938_put_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); -- Ondrej Zary

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[alsa-devel] codec->card removed
by jonsmirl＠gmail.com 26 Sep '14

26 Sep '14

How should I rewrite this to reflect that codec->card has been removed? This is codec is on the SOC chip, not an external one. static int sunxi_codec_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_card *card = codec->card; struct sunxi_priv *priv = snd_soc_card_get_drvdata(card); ... Complete file... https://bitbucket.org/emiliolopez/linux/src/fbfc3e9c092c90574aee454062fc465… -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] Question about SNDRV_PCM_STATE_DRAINING and DMA transfer
by Kuninori Morimoto 26 Sep '14

26 Sep '14

Hi Mark, ALSA ML We noticed that DMA seems transfered +1 time when Ctrl-C happen. But, is this correct ? is this our driver bug ? our buffer_size is 8192, and DMA transfers 2048 in 1 time. 1. Ctrl-C happen wait_for_avail() of snd_pcm_lib_write1() returns -ERESTARTSYS immediately no transfer() are called any more. 2. DMA transfer interrupt happen This is normal from driver point of view It calls snd_pcm_period_elapsed() and try to transfer next 2048 byte 3. snd_pcm_common_ioctl1() :: SNDRV_PCM_IOCTL_DRAIN is called it goes to SNDRV_PCM_STATE_DRAINING status in snd_pcm_drain() and it wait until all drained. 4. DMA transfer interrupt happen It calls snd_pcm_period_elapsed() and try to transfer next 2048 byte snd_pcm_playback_avail() in snd_pcm_update_state() return 2048 this time. 5. DMA transfer interrupt happen It calls snd_pcm_period_elapsed() and try to transfer next 2048 byte snd_pcm_playback_avail() in snd_pcm_update_state() return 4096 this time. 6. DMA transfer interrupt happen It calls snd_pcm_period_elapsed() and try to transfer next 2048 byte snd_pcm_playback_avail() in snd_pcm_update_state() return 6144 this time. 7. DMA transfer interrupt happen It calls snd_pcm_period_elapsed() and try to transfer next 2048 byte snd_pcm_playback_avail() in snd_pcm_update_state() return 8192 this time. then, it calls snd_pcm_drain_done() 9. snd_soc_dai_ops :: trigger called with SNDRV_PCM_TRIGGER_STOP driver stops DMA transfer 2) / 4) / 5) / 6) / 7) continue DMA transfer. but, 1) already quited from snd_pcm_lib_write1() I guess. So, 2) and 7) uses same buffer ? What/Where is wrong ? Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: Intel: byt-max98090: Set card as fully routed
by Jarkko Nikula 26 Sep '14

26 Sep '14

All byt-max98090 audio connections are known and described in DAPM routing table. Set the fully_routed flag in order to disable unused codec pins. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/byt-max98090.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index b8b8af571ef1..d52681e7225e 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -139,6 +139,7 @@ static struct snd_soc_card byt_max98090_card = { .num_dapm_routes = ARRAY_SIZE(byt_max98090_audio_map), .controls = byt_max98090_controls, .num_controls = ARRAY_SIZE(byt_max98090_controls), + .fully_routed = true, }; static int byt_max98090_probe(struct platform_device *pdev) -- 2.1.0

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[alsa-devel] [PATCH 1/2] ASoC: rt286: Correct default value
by Bard Liao 26 Sep '14

26 Sep '14

This patch corrects some incorrect default value in the cache. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index e4f6102..7a66084 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -51,7 +51,7 @@ static struct reg_default rt286_index_def[] = { { 0x04, 0xaf01 }, { 0x08, 0x000d }, { 0x09, 0xd810 }, - { 0x0a, 0x0060 }, + { 0x0a, 0x0120 }, { 0x0b, 0x0000 }, { 0x0d, 0x2800 }, { 0x0f, 0x0000 }, @@ -60,7 +60,7 @@ static struct reg_default rt286_index_def[] = { { 0x33, 0x0208 }, { 0x49, 0x0004 }, { 0x4f, 0x50e9 }, - { 0x50, 0x2c00 }, + { 0x50, 0x2000 }, { 0x63, 0x2902 }, { 0x67, 0x1111 }, { 0x68, 0x1016 }, @@ -104,7 +104,6 @@ static const struct reg_default rt286_reg[] = { { 0x02170700, 0x00000000 }, { 0x02270100, 0x00000000 }, { 0x02370100, 0x00000000 }, - { 0x02040000, 0x00004002 }, { 0x01870700, 0x00000020 }, { 0x00830000, 0x000000c3 }, { 0x00930000, 0x000000c3 }, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] ASoC DPCM has issues when multiple FEs links to same type of codec.
by Tan, Joshua Chu Yeaw 26 Sep '14

26 Sep '14

Hi, ALSA ASoC DPCM FE - BE connection doesn't handle the scenario when a machine driver has multiple FEs linking to same codec type. Let me use dummy codec as example codec: FE1 - BE1 (connected to dummy codec instance #1) FE2 - BE2 (connected to dummy codec instance #2) FE3 - BE3 (connected to dummy codec instance #3) DPCM will not be able to correctly identify the Back End from the 2nd pair onwards. This happens as the codec .stream_name is used 3 times to connect to different Back-Ends. Say I am using the dummy codec that has a playback .stream_name "Dummy Playback" with 3 I2S AIF. E.g. I have 3 BE dai links: static struct snd_soc_dai_link xxx_dailink[] = { ... { .name = "I2S0-Codec", .be_id = 1, .cpu_dai_name = "i2s0-port", .platform_name = "xxx-platform", .no_pcm = 1, .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", } { .name = "I2S1-Codec", .be_id = 2, .cpu_dai_name = "i2s1-port", .platform_name = "xxx-platform", .no_pcm = 1, .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", } { .name = "I2S2-Codec", .be_id = 3, .cpu_dai_name = "i2s2-port", .platform_name = "xxx-platform", .no_pcm = 1, .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", } /*DAPM routing*/ static const struct snd_soc_dapm_route map[] = { { "Dummy Playback", NULL, "I2S0 Tx"}, { "Dummy Playback", NULL, "I2S1 Tx"}, { "Dummy Playback", NULL, "I2S2 Tx"}, ... } The order of the DAI Link affects the connection of the codec stream name widget on a first come first serve basis, in this case, first declared DAI link codec is used. In soc-pcm.c, when dpcm_fe_dai_open tries to add paths for the 2nd(and later) FE, dpcm_add_paths function will always discover the "Dummy Playback" widget belonging to the back-end of the first DAI link and link it incorrectly. 1. dpcm_add_paths() finds "Dummy Playback" is a snd_soc_dapm_dai_in type, and calls dpcm_get_be() 1101 static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, 1102 struct snd_soc_dapm_widget_list **list_) ... 1120 /* is there a valid BE rtd for this widget */ 1121 be = dpcm_get_be(card, list->widgets[i], stream); 2. The dpcm_get_be will traverse the list for "Dummy Playback" and return the first back end found. Line 994 below always finds that I2S0-Codec link matches this condition and will not proceed further to match I2S1-Codec 990 if (stream == SNDRV_PCM_STREAM_PLAYBACK) { 991 for (i = 0; i < card->num_links; i++) { 992 be = &card->rtd[i]; 993 994 if (!be->dai_link->no_pcm) 995 continue; 996 997 if (be->cpu_dai->playback_widget == widget || 998 be->codec_dai->playback_widget == widget) 999 return be; 1000 } 1001 } else { --> Same effect also happens in the capture path. E.g. When I try to playback on the i2s1 (2nd playback device), the following fe-be link happens : Widget = "I2S1 Tx", Back End = "I2S1-Codec" --> This is the intended link. Widget = "Dummy Playback", Back End = "I2S0-Codec" --> This is additional problematic link because the first discovered "Dummy Playback" is from the codec in the first DAI link. This results in soc_pcm_open called twice in dpcm_be_dai_startup function; first for the first Back End (undesired), and then only the correct Back End Is it possible to add some instance id to the codec name, or some check on the intended FE-BE pairing? Regards, Joshua

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[alsa-devel] [PATCH] ASoC: wm8741: Remove unused wm8741_suspend define
by Lars-Peter Clausen 25 Sep '14

25 Sep '14

This driver has no suspend callback. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8741.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index a237f16..31bb480 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -413,7 +413,6 @@ static int wm8741_resume(struct snd_soc_codec *codec) return 0; } #else -#define wm8741_suspend NULL #define wm8741_resume NULL #endif -- 1.8.0

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[alsa-devel] [PATCH] ASoC: blackfin: add bf6xx audio dma driver
by Scott Jiang 25 Sep '14

25 Sep '14

Bf5xx and bf6xx reused bf5xx-i2s-pcm.c before. Recently tdm driver has been merged into bf5xx-i2s-pcm.c and it's not compatible with bf6xx audio. So create this dma driver and add tdm support later. Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/blackfin/Kconfig | 22 ++-- sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf6xx-pcm.c | 230 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 248 insertions(+), 6 deletions(-) create mode 100644 sound/soc/blackfin/bf6xx-pcm.c diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 6410aa2..9d6bd9d 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -1,8 +1,17 @@ config SND_BF5XX_I2S tristate "SoC I2S Audio for the ADI Blackfin chip" - depends on BLACKFIN - select SND_BF5XX_SOC_SPORT if !BF60x - select SND_BF6XX_SOC_SPORT if BF60x + depends on BLACKFIN && !BF60x + select SND_BF5XX_SOC_SPORT + help + Say Y or M if you want to add support for codecs attached to + the Blackfin SPORT (synchronous serial ports) interface in I2S + mode (supports single stereo In/Out). + You will also need to select the audio interfaces to support below. + +config SND_BF6XX_PCM + tristate "SoC Audio for the ADI Blackfin BF6xx chip" + depends on BF60x + select SND_BF6XX_SOC_SPORT help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in I2S @@ -11,7 +20,7 @@ config SND_BF5XX_I2S config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio Codec Add-On Card support" - depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI + depends on (SND_BF5XX_I2S || SND_BF6XX_PCM) && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S if !BF60x select SND_BF6XX_SOC_I2S if BF60x select SND_SOC_SSM2602_SPI if SPI_MASTER @@ -45,8 +54,9 @@ config SND_SOC_BFIN_EVAL_ADAU1373 config SND_SOC_BFIN_EVAL_ADAU1X61 tristate "Support for the EVAL-ADAU1X61 board on Blackfin eval boards" - depends on SND_BF5XX_I2S && I2C - select SND_BF5XX_SOC_I2S + depends on (SND_BF5XX_I2S || SND_BF6XX_PCM) && I2C + select SND_BF5XX_SOC_I2S if !BF60x + select SND_BF6XX_SOC_I2S if BF60x select SND_SOC_ADAU1761_I2C help Say Y if you want to add support for the Analog Devices EVAL-ADAU1X61 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index f21e948..cef39f6 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -1,6 +1,7 @@ # Blackfin Platform Support snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o +snd-bf6xx-pcm-objs := bf6xx-pcm.o snd-soc-bf5xx-sport-objs := bf5xx-sport.o snd-soc-bf6xx-sport-objs := bf6xx-sport.o snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o @@ -9,6 +10,7 @@ snd-soc-bf6xx-i2s-objs := bf6xx-i2s.o obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o +obj-$(CONFIG_SND_BF6XX_PCM) += snd-bf6xx-pcm.o obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o diff --git a/sound/soc/blackfin/bf6xx-pcm.c b/sound/soc/blackfin/bf6xx-pcm.c new file mode 100644 index 0000000..bad9fbf --- /dev/null +++ b/sound/soc/blackfin/bf6xx-pcm.c @@ -0,0 +1,230 @@ +/* + * bf6xx-pcm.c - Analog Devices BF6XX audio dma driver + * + * Copyright (c) 2014 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * 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. + */ + +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> + +#include "bf6xx-sport.h" + +static void bf6xx_dma_irq(void *data) +{ + struct snd_pcm_substream *pcm = data; + snd_pcm_period_elapsed(pcm); +} + +static const struct snd_pcm_hardware bf6xx_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .period_bytes_min = 32, + .period_bytes_max = 0x10000, + .periods_min = 1, + .periods_max = PAGE_SIZE/32, + .buffer_bytes_max = 0x20000, /* 128 kbytes */ + .fifo_size = 16, +}; + +static int bf6xx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + size_t size = params_buffer_bytes(params); + snd_pcm_lib_malloc_pages(substream, size); + + return 0; +} + +static int bf6xx_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int bf6xx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + int period_bytes = frames_to_bytes(runtime, runtime->period_size); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sport_set_tx_callback(sport, bf6xx_dma_irq, substream); + sport_config_tx_dma(sport, runtime->dma_area, + runtime->periods, period_bytes); + } else { + sport_set_rx_callback(sport, bf6xx_dma_irq, substream); + sport_config_rx_dma(sport, runtime->dma_area, + runtime->periods, period_bytes); + } + + return 0; +} + +static int bf6xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_tx_start(sport); + else + sport_rx_start(sport); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_tx_stop(sport); + else + sport_rx_stop(sport); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t bf6xx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + unsigned int diff; + snd_pcm_uframes_t frames; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + diff = sport_curr_offset_tx(sport); + else + diff = sport_curr_offset_rx(sport); + + /* + * TX at least can report one frame beyond the end of the + * buffer if we hit the wraparound case - clamp to within the + * buffer as the ALSA APIs require. + */ + if (diff == snd_pcm_lib_buffer_bytes(substream)) + diff = 0; + + frames = bytes_to_frames(substream->runtime, diff); + + return frames; +} + +static int bf6xx_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct sport_device *sport = snd_soc_dai_get_drvdata(cpu_dai); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = &substream->dma_buffer; + int ret; + + snd_soc_set_runtime_hwparams(substream, &bf6xx_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sport->tx_buf = buf->area; + else + sport->rx_buf = buf->area; + + runtime->private_data = sport; + return 0; +} + +static int bf6xx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + size_t size = vma->vm_end - vma->vm_start; + vma->vm_start = (unsigned long)runtime->dma_area; + vma->vm_end = vma->vm_start + size; + vma->vm_flags |= VM_SHARED; + + return 0 ; +} + +static struct snd_pcm_ops bf6xx_pcm_ops = { + .open = bf6xx_pcm_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = bf6xx_pcm_hw_params, + .hw_free = bf6xx_pcm_hw_free, + .prepare = bf6xx_pcm_prepare, + .trigger = bf6xx_pcm_trigger, + .pointer = bf6xx_pcm_pointer, + .mmap = bf6xx_pcm_mmap, +}; + +static int bf6xx_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = bf6xx_pcm_hardware.buffer_bytes_max; + int ret = 0; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, card->dev, size, size); +} + +static struct snd_soc_platform_driver bf6xx_soc_platform = { + .ops = &bf6xx_pcm_ops, + .pcm_new = bf6xx_pcm_new, +}; + +static int bfin_soc_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &bf6xx_soc_platform); +} + +static int bfin_soc_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver bfin_pcm_driver = { + .driver = { + .name = "bfin-i2s-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = bfin_soc_platform_probe, + .remove = bfin_soc_platform_remove, +}; + +module_platform_driver(bfin_pcm_driver); + +MODULE_DESCRIPTION("Analog Devices BF6XX audio dma driver"); +MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux(a)gmail.com>"); +MODULE_LICENSE("GPL v2"); -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: rt5677: Add gpio-controller DTS documentation
by Anatol Pomozov 25 Sep '14

25 Sep '14

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5677.txt | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index 98509fb..bd28df6 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -10,6 +10,11 @@ Required properties: - interrupts : The CODEC's interrupt output. +- gpio-controller : Indicates this device is a GPIO controller. + +- #gpio-cells : Should be two. The first cell is the pin number and the + second cell is used to specify optional parameters (currently unused). + Optional properties: - realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin. @@ -36,6 +41,10 @@ rt5677 { reg = <0x2c>; interrupt-parent = <&gpio>; interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + + gpio-controller; + #gpio-cells = <2>; + realtek,pow-ldo2-gpio = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; -- 2.1.0.rc2.206.gedb03e5

2 1

[alsa-devel] [PATCH] ASoC: trace: Remove trailing new-lines in trace messages
by Anatol Pomozov 25 Sep '14

25 Sep '14

These new-lines add empty lines to trace output Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- include/trace/events/asoc.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 0194a64..b04ee7e 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -175,7 +175,7 @@ TRACE_EVENT(snd_soc_dapm_output_path, __entry->path_sink = (long)path->sink; ), - TP_printk("%c%s -> %s -> %s\n", + TP_printk("%c%s -> %s -> %s", (int) __entry->path_sink && (int) __entry->path_connect ? '*' : ' ', __get_str(wname), __get_str(pname), __get_str(psname)) @@ -204,7 +204,7 @@ TRACE_EVENT(snd_soc_dapm_input_path, __entry->path_source = (long)path->source; ), - TP_printk("%c%s <- %s <- %s\n", + TP_printk("%c%s <- %s <- %s", (int) __entry->path_source && (int) __entry->path_connect ? '*' : ' ', __get_str(wname), __get_str(pname), __get_str(psname)) @@ -226,7 +226,7 @@ TRACE_EVENT(snd_soc_dapm_connected, __entry->stream = stream; ), - TP_printk("%s: found %d paths\n", + TP_printk("%s: found %d paths", __entry->stream ? "capture" : "playback", __entry->paths) ); -- 2.1.0.rc2.206.gedb03e5

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