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[alsa-devel] [PATCH 0/3] McPDM driver patch series
by Candelaria Villareal, Jorge 19 Dec '09

19 Dec '09

McPDM module is the interface between the OMAP and TWL6030 audio codec. It transfers data between the two chips using a Pulse Density Modulation protocol, which allows sound to be played or recorded at speeds of up to 96000 Hz. Besides data, McPDM can transfer commands and read status to/from the codec using the frame synchronization line. The interface supports the following: * Up to 5 downlink channels and 3 uplink channels * 88.2 and 96 KHz * 32 bit sample size Currently, driver is set to support up to 4 downlink channels. If 2 channels are selected, output will go to the headset, and if 4 channels are selected, output will go to the headset and speakers. Functionality to select the output at runtime will be added afterwards. Command and status mode is not supported in current patches. -- Jorge Eduardo Candelaria (2): OMAP4: IRQ: Add McPDM IRQ definition. ASoC: OMAP4: Add support for McPDM Misael Lopez Cruz (1): ASoC: OMAP4: Add McPDM platform driver arch/arm/plat-omap/include/plat/irqs.h | 2 +- sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/mcpdm.c | 505 ++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 156 ++++++++++ sound/soc/omap/omap-mcpdm.c | 248 ++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 ++ 7 files changed, 944 insertions(+), 1 deletions(-) create mode 100644 sound/soc/omap/mcpdm.c create mode 100644 sound/soc/omap/mcpdm.h create mode 100644 sound/soc/omap/omap-mcpdm.c create mode 100644 sound/soc/omap/omap-mcpdm.h

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[alsa-devel] [PATCH 7/7] sound: Use kzalloc for allocating only one thing
by Julia Lawall 19 Dec '09

19 Dec '09

From: Julia Lawall <julia(a)diku.dk> Use kzalloc rather than kcalloc(1,...) The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ @@ - kcalloc(1, + kzalloc( ...) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/isa/msnd/msnd_midi.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff -u -p a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c --- a/sound/isa/msnd/msnd_midi.c 2009-03-28 23:34:55.000000000 +0100 +++ b/sound/isa/msnd/msnd_midi.c 2009-12-19 07:52:27.000000000 +0100 @@ -162,7 +162,7 @@ int snd_msndmidi_new(struct snd_card *ca err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi); if (err < 0) return err; - mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL); + mpu = kzalloc(sizeof(*mpu), GFP_KERNEL); if (mpu == NULL) { snd_device_free(card, rmidi); return -ENOMEM; diff -u -p a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c --- a/sound/pci/hda/patch_realtek.c 2009-12-17 08:30:15.000000000 +0100 +++ b/sound/pci/hda/patch_realtek.c 2009-12-19 07:52:50.000000000 +0100 @@ -12857,7 +12857,7 @@ static int patch_alc268(struct hda_codec int board_config; int i, has_beep, err; - spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM;

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[alsa-devel] [PATCH 0/5] AACI: cleanups
by Russell King - ARM Linux 18 Dec '09

18 Dec '09

This patch series is in preparation of adding DMA support to the AACI driver. I'd appreciate it if ALSA people could do me the curtesy of copying me on patches they dream up on ALSA drivers maintained by me, rather than running rough-shod over work that they may have queued - especially when they say "you can also do this" "yes I know, I have a patch" and then they do it anyway. If this is how they want to operate, I'll avoid working on any ALSA stuff in the future. Its just not worth me wasting my time.

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[alsa-devel] Help needed to develop quirks for USB Sound device for Roland Edirol PC-80 MIDI controller
by Alain Lauzon 18 Dec '09

18 Dec '09

Heloo I just bought a Roland Edirol PC-80 MIDI controller and found out that it was not supported on LINUX. I looked on the web and found the /usr/src/alsa/alsa-driver-xxx/usbquirks.h file to be the place to change some code to support it. I have tried many things but I don't enough info to get it right once and for all. Here is what I did : { /* PC-80 Wave */ USB_DEVICE(0x0582, 0x0092), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "EDIROL", */ /* .product_name = "PC-80", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 0, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x81, .ep_attr = 0x01, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 48000, .rate_max = 48000, //(96?) } }, { .ifnum = 2, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { .out_cables = 0x0007, .in_cables = 0x0007 } }, { .ifnum = -1 } } } What I got working is the audio out but cannot have the audio working. Is there some resources somewhere that I could read to understand what I need to do? Thank you, Alain Lauzon -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 18 Dec '09

18 Dec '09

The following changes since commit bc2580061e42c323d7777029f01318f395edac0d: Julia Lawall (1): ASoC: Correct code taking the size of a pointer are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Nothing terribly exciting here, almost all of it is a bunch of new drivers. Chaithrika U S (1): ASoC: DaVinci: Update suspend/resume support for McASP driver Daniel Mack (1): ASoC: Add regulator support to CS4270 codec driver Guennadi Liakhovetski (1): ASoC: wm8974: fix a wrong bit definition Ilkka Koskinen (2): ASoC: tlv320dac33: Add support for regulator framework ASoC: tpa6130a2: Add support for regulator framework Jassi Brar (2): ASoC: S3C64XX: Remove unnecessary header includes ASoC: S3C64XX: Compress and generalize the CPU driver Jon Smirl (1): ASoC: Fix disable of SPDIF on STAC9766 codec Kuninori Morimoto (4): ASoC: sh_fsi: avoid using global variable ASoC: ak4642: Add default return value in ak4642_modinit ASoC: Add DA7210 codec device support for ALSA ASoC: Add FSI-DA7210 sound support for SuperH Mark Brown (12): Merge branch 'for-2.6.33' into for-2.6.34 ASoC: Export snd_soc_update_bits_unlocked() ASoC: Display the power register in DAPM widget debugfs ASoC: Initial WM8904 CODEC driver ASoC: Push registers out of mixer power decision ASoC: Sort DAPM sequences by CODEC as well ASoC: Register the CODEC in WM8727 ASoC: Fix sorting of codecs Makefile entries ASoC: Fix build of DA7210 ASoC: Add initial WM8955 CODEC driver ASoC: Set AIF word length for WM8904 Merge branch 'for-2.6.33' into for-2.6.34 Peter Meerwald (1): ASoC: PLL computation in TLV320AIC3x SoC driver include/sound/soc.h | 3 + include/sound/wm8904.h | 57 + include/sound/wm8955.h | 26 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 10 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/cs4270.c | 43 +- sound/soc/codecs/da7210.c | 589 +++++++++ sound/soc/codecs/da7210.h | 24 + sound/soc/codecs/stac9766.c | 18 +- sound/soc/codecs/tlv320aic3x.c | 75 +- sound/soc/codecs/tlv320dac33.c | 92 ++- sound/soc/codecs/tpa6130a2.c | 87 +- sound/soc/codecs/wm8727.c | 66 +- sound/soc/codecs/wm8904.c | 2555 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8904.h | 1681 ++++++++++++++++++++++++ sound/soc/codecs/wm8955.c | 1151 +++++++++++++++++ sound/soc/codecs/wm8955.h | 489 +++++++ sound/soc/codecs/wm8974.c | 2 +- sound/soc/davinci/davinci-mcasp.c | 18 +- sound/soc/davinci/davinci-mcasp.h | 1 + sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 120 +-- sound/soc/sh/Kconfig | 8 + sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-da7210.c | 83 ++ sound/soc/sh/fsi.c | 115 +- sound/soc/soc-core.c | 7 +- sound/soc/soc-dapm.c | 37 +- 29 files changed, 7127 insertions(+), 248 deletions(-) create mode 100644 include/sound/wm8904.h create mode 100644 include/sound/wm8955.h create mode 100644 sound/soc/codecs/da7210.c create mode 100644 sound/soc/codecs/da7210.h create mode 100644 sound/soc/codecs/wm8904.c create mode 100644 sound/soc/codecs/wm8904.h create mode 100644 sound/soc/codecs/wm8955.c create mode 100644 sound/soc/codecs/wm8955.h create mode 100644 sound/soc/sh/fsi-da7210.c

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[alsa-devel] Play multiple sounds concurrently
by Markus Luttenberger 18 Dec '09

18 Dec '09

Hey, I want to play multiple sounds concurrently with ALSA. The audio data is mono and G.711 a-law encoded. I can successfully play a single source of audio data but the sound is distorted when I try to play two sources. Basically, this is what I'm doing: -----8<----------------- char buf1[1024]; // filled with audio data from source 1 char buf2[1024]; // filled with audio data from source 2 char wr_buf[1024]; for (int count = 0; count < 1024; count++) wr_buf[count] = (char) (0.5 * buf1[count]) + (char) (0.5 * buf2[count]); snd_pcm_writei(dev_handle, wr_buf, 1024); -----8<----------------- I think I have to use a scale value to correct the values or else clipping occures. However, this is the problem. How do I have to modify the audio bytes in order to add them together and write them into one buffer? greetings.. ..Markus

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[alsa-devel] alsa device drain power
by Michael Trimarchi 18 Dec '09

18 Dec '09

Hi all, I have a simple question is the general behavior that an alsa device start to drain power when it starts to play sound or start to recording. What happen if there is an external amplifier connect to sound device? Michael

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[alsa-devel] [PATCH] wm8974: fix a wrong bit definition
by Guennadi Liakhovetski 18 Dec '09

18 Dec '09

The wm8974 datasheet defines BUFIOEN as bit 2. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- no idea if this would break anything, so, please test. Maybe the correct fix would be to actually use a correct name for bit 4, but who wants to enable Mic bias when putting bias into standby?... diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index eff2933..fb99433 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -47,7 +47,7 @@ static const u16 wm8974_reg[WM8974_CACHEREGNUM] = { }; #define WM8974_POWER1_BIASEN 0x08 -#define WM8974_POWER1_BUFIOEN 0x10 +#define WM8974_POWER1_BUFIOEN 0x04 struct wm8974_priv { struct snd_soc_codec codec;

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[alsa-devel] 2.6.32-gitX: oops in ac97 codec registration
by Manuel Lauss 18 Dec '09

18 Dec '09

Hi Mark, current 2.6.32-git spits out the following oops when I load my machine driver with the generic AC97 codec: No device for DAI au1xpsc_ac97 AC97 SoC Audio Codec 0.6 asoc: AC97 HiFi <-> au1xpsc_ac97 mapping ok CPU 0 Unable to handle kernel paging request at virtual address 000001f0, epc == 80366320, ra == 803662c0 Oops[#1]: Cpu 0 $ 0 : 00000000 10003c00 00000000 00000000 $ 4 : 00000000 00004b39 00000001 00004b39 $ 8 : 00000000 00000002 8041a7a0 8d140af0 $12 : 00000004 ad55ad55 00000003 00000001 $16 : 00000000 c1179a94 8d1f2520 c007b420 $20 : 8d49cfe0 8d1f2520 804fc03c 804fc130 $24 : 00000000 0000ad55 $28 : 8d590000 8d591af0 00000000 803662c0 Hi : 00000000 Lo : 00000004 epc : 80366320 snd_soc_new_pcms+0x300/0x358 Not tainted ra : 803662c0 snd_soc_new_pcms+0x2a0/0x358 Status: 10003c03 KERNEL EXL IE Cause : 0080800c BadVA : 000001f0 PrId : 04030202 (Au1250) Modules linked in: snd_soc_exm32cexx(+) snd_soc_cs4251x snd_soc_ac97 snd_ac97_codec snd_soc_au1xpsc_ac97 snd_soc_au1xpsc_i2s Process modprobe (pid: 673, threadinfo=8d590000, task=8d430938, tls=2aacf2d0) Stack : 8d591b30 c007b3c4 c0074afc 00010000 00000001 8d591b08 8d17d1c0 37394341 46694820 43412069 48203739 2d694669 8d170030 804fc16c 8d17d2c8 0000001c 10003c01 8d17d2c8 804fc16c 8d17d2c8 10003c01 8d17d2d0 00000024 80500000 c0074b58 00000001 c11799f4 00000000 8d1f2528 00000001 fffffff4 8d1f2520 c11798f0 00000001 8d17d0c0 00000001 804fc130 00000000 804f4980 c007b288 ... Call Trace: [<80366320>] snd_soc_new_pcms+0x300/0x358 [<c007b288>] ac97_soc_probe+0x108/0x1d0 [snd_soc_ac97] [<80365108>] snd_soc_instantiate_cards+0x22c/0x694 [<803655ec>] soc_probe+0x7c/0xc4 [<802c73b8>] driver_probe_device+0xbc/0x194 [<802c6748>] bus_for_each_drv+0x60/0xb0 [<802c7638>] device_attach+0x60/0x88 [<802c6524>] bus_probe_device+0x30/0x54 [<802c4d74>] device_add+0x3a4/0x53c [<802c8e48>] platform_device_add+0x14c/0x1b8 [<c11794ec>] cexx_audio_probe+0x310/0x374 [snd_soc_exm32cexx] [<802c73b8>] driver_probe_device+0xbc/0x194 [<802c74fc>] __driver_attach+0x6c/0xa4 [<802c6afc>] bus_for_each_dev+0x60/0xb0 [<802c6264>] bus_add_driver+0xc8/0x268 [<802c78ac>] driver_register+0xdc/0x194 [<8010a010>] do_one_initcall+0x68/0x1f4 [<80154268>] sys_init_module+0xec/0x23c [<801022a0>] stack_done+0x20/0x3c Oops is inside snd_ac97_dev_add_pdata() call: seems ac97->dev is NULL. I'm not entirely sure that this isn't somehow the fault of my machine driver; has the procedure to register the asoc device changed since 2.6.32? Thanks! Manuel Lauss

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[alsa-devel] [PATCH 3/3] ASoC: OMAP4: Add McPDM platform driver
by Candelaria Villareal, Jorge 17 Dec '09

17 Dec '09

McPDM platform driver is configured to use sDMA in order to transfer to/from memory. Support for interfacing with ABE will be added later. McPDM dai currently supports up to 4 downlink channels and 2 uplink channels simultaneously, as well as 88.2 and 96 KHz, and a sample size of 32 bits. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Margarita Olaya <x0080101(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/Makefile | 2 +- sound/soc/omap/omap-mcpdm.c | 248 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 +++++ 3 files changed, 278 insertions(+), 1 deletions(-) create mode 100644 sound/soc/omap/omap-mcpdm.c create mode 100644 sound/soc/omap/omap-mcpdm.h diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index bf8b388..d6382fa 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,7 +1,7 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o -snd-soc-omap-mcpdm-objs := mcpdm.o +snd-soc-omap-mcpdm-objs := mcpdm.o omap-mcpdm.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c new file mode 100644 index 0000000..9dc0ff5 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.c @@ -0,0 +1,248 @@ +/* + * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port + * + * Copyright (C) 2009 Texas Instruments + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * Contact: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * Margarita Olaya <magi.olaya(a)ti.com> + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/control.h> +#include <mach/dma.h> +#include <mach/mcbsp.h> +#include "mcpdm.h" +#include "omap-mcpdm.h" +#include "omap-pcm.h" + +struct omap_mcpdm_data { + struct omap_mcpdm_link *links; + int active; +}; + +static struct omap_mcpdm_link omap_mcpdm_links[] = { + /* downlink */ + { + .irq_mask = DN_IRQ_EMTPY | DN_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, + /* uplink */ + { + .irq_mask = UP_IRQ_EMPTY | UP_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, +}; + +static struct omap_mcpdm_data mcpdm_data = { + .links = omap_mcpdm_links, + .active = 0, +}; + +/* + * Stream DMA parameters + */ +static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { + { + .name = "Audio downlink", + .dma_req = OMAP44XX_DMA_MCPDM_DL, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_ELEMENT, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA, + }, + { + .name = "Audio uplink", + .dma_req = OMAP44XX_DMA_MCPDM_UP, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_ELEMENT, + .port_addr = OMAP44XX_MCPDM_BASE + MCPDM_UP_DATA, + }, +}; + +static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err = 0; + + if (!cpu_dai->active) + err = omap_mcpdm_request(); + + return err; +} + +static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (!cpu_dai->active) + omap_mcpdm_free(); +} + +static int omap_mcpdm_dai_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 *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + int stream = substream->stream; + int err = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!mcpdm_priv->active++) + omap_mcpdm_start(stream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!--mcpdm_priv->active) + omap_mcpdm_stop(stream); + break; + default: + err = -EINVAL; + } + + return err; +} + +static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int stream = substream->stream; + int channels, err, link_mask = 0; + + cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream]; + + channels = params_channels(params); + switch (channels) { + case 4: + if (stream) + /* up to 2 channels for uplink */ + return -EINVAL; + link_mask |= 1 << 3; + case 3: + if (stream) + /* up to 2 channels for uplink */ + return -EINVAL; + link_mask |= 1 << 2; + case 2: + link_mask |= 1 << 1; + case 1: + link_mask |= 1 << 0; + break; + default: + /* unsupported number of channels */ + return -EINVAL; + } + + if (stream) { + mcpdm_links[stream].channels = link_mask << 0; + err = omap_mcpdm_set_uplink(&mcpdm_links[stream]); + } else { + mcpdm_links[stream].channels = link_mask << 3; + err = omap_mcpdm_set_downlink(&mcpdm_links[stream]); + } + + return err; +} + +static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int err; + + if (substream->stream) + err = omap_mcpdm_clr_uplink(&mcpdm_links[substream->stream]); + else + err = omap_mcpdm_clr_downlink(&mcpdm_links[substream->stream]); + + return err; +} + +static struct snd_soc_dai_ops omap_mcpdm_dai_ops = { + .startup = omap_mcpdm_dai_startup, + .shutdown = omap_mcpdm_dai_shutdown, + .trigger = omap_mcpdm_dai_trigger, + .hw_params = omap_mcpdm_dai_hw_params, + .hw_free = omap_mcpdm_dai_hw_free, +}; + +#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define OMAP_MCPDM_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai omap_mcpdm_dai = { + .name = "omap-mcpdm", + .id = -1, + .playback = { + .channels_min = 1, + .channels_max = 4, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .ops = &omap_mcpdm_dai_ops, + .private_data = &mcpdm_data, +}; +EXPORT_SYMBOL_GPL(omap_mcpdm_dai); + +static int __init snd_omap_mcpdm_init(void) +{ + return snd_soc_register_dai(&omap_mcpdm_dai); +} +module_init(snd_omap_mcpdm_init); + +static void __exit snd_omap_mcpdm_exit(void) +{ + snd_soc_unregister_dai(&omap_mcpdm_dai); +} +module_exit(snd_omap_mcpdm_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("OMAP PDM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h new file mode 100644 index 0000000..73b80d5 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.h @@ -0,0 +1,29 @@ +/* + * omap-mcpdm.h + * + * Copyright (C) 2009 Texas Instruments + * + * Contact: Misael Lopez Cruz <x0052729(a)ti.com> + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_MCPDM_H__ +#define __OMAP_MCPDM_H__ + +extern struct snd_soc_dai omap_mcpdm_dai; + +#endif /* End of __OMAP_MCPDM_H__ */ -- 1.6.0.4

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