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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 23 Feb '10

23 Feb '10

The following changes since commit 6c5f1fed49f96a0600aa9a97ac3faf972c33a341: Mark Brown (1): ASoC: Make pmdown_time a long 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 Candelaria Villareal, Jorge (1): ASoC: OMAP4: Add support for McPDM Kuninori Morimoto (1): ASoC: fsi: Modify over/under run error settlement Misael Lopez Cruz (2): ASoC: OMAP: data_type and sync_mode configurable in audio dma ASoC: OMAP4: Add McPDM platform driver Peter Ujfalusi (1): ASoC: core: On resume also check the soc device state sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/mcpdm.c | 484 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 151 ++++++++++++++ sound/soc/omap/omap-mcbsp.c | 2 + sound/soc/omap/omap-mcpdm.c | 251 ++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 +++ sound/soc/omap/omap-pcm.c | 15 +- sound/soc/omap/omap-pcm.h | 4 +- sound/soc/sh/fsi.c | 46 +++-- sound/soc/soc-core.c | 6 + 11 files changed, 964 insertions(+), 29 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 v4 0/3] McPDM driver patch series
by Candelaria Villareal, Jorge 23 Feb '10

23 Feb '10

McPDM module is the interface between the OMAP and TWL6030 audio codec. 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. Changes from v3: - Implemented changes based on comments from Liam Girdwood. -- Jorge Eduardo Candelaria (1): ASoC: OMAP4: Add support for McPDM Misael Lopez Cruz (2): ASoC: OMAP: data_type and sync_mode configurable in audio dma ASoC: OMAP4: Add McPDM platform driver sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/mcpdm.c | 486 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 151 +++++++++++++ sound/soc/omap/omap-mcbsp.c | 2 + sound/soc/omap/omap-mcpdm.c | 251 ++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 +++ sound/soc/omap/omap-pcm.c | 15 +- sound/soc/omap/omap-pcm.h | 4 +- 9 files changed, 935 insertions(+), 8 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] ALSA: hda - Refer to macmini3 channel mode functions
by Luke Yelavich 23 Feb '10

23 Feb '10

From: Luke Yelavich <luke.yelavich(a)canonical.com> Refer to alc885_macmini3_{2,6}ch_mode_init functions instead of those for the mb5 model quirk. While they are the functino contents are the same now, things may change later. This also gets rid of compiler warnings about initialized, but unused functions. Signed-off-by: Luke Yelavich <luke.yelavich(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f8fb260..9e5ea9e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7033,8 +7033,8 @@ static struct hda_verb alc885_macmini3_ch6_init[] = { }; static struct hda_channel_mode alc885_macmini3_6ch_modes[2] = { - { 2, alc885_mb5_ch2_init }, - { 6, alc885_mb5_ch6_init }, + { 2, alc885_macmini3_ch2_init }, + { 6, alc885_macmini3_ch6_init }, }; -- 1.7.0

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[alsa-devel] [PATCH] ASoC: fsi: Modify over/under run error settlement
by Kuninori Morimoto 23 Feb '10

23 Feb '10

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/sh/fsi.c | 46 +++++++++++++++++++++++++--------------------- 1 files changed, 25 insertions(+), 21 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3c36d24..993abb7 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -388,7 +388,7 @@ static void fsi_soft_all_reset(struct fsi_master *master) } /* playback interrupt */ -static int fsi_data_push(struct fsi_priv *fsi) +static int fsi_data_push(struct fsi_priv *fsi, int startup) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; @@ -397,7 +397,7 @@ static int fsi_data_push(struct fsi_priv *fsi) int fifo_free; int width; u8 *start; - int i, ret, over_period; + int i, over_period; if (!fsi || !fsi->substream || @@ -453,24 +453,26 @@ static int fsi_data_push(struct fsi_priv *fsi) fsi->byte_offset += send * width; - ret = 0; status = fsi_reg_read(fsi, DOFF_ST); - if (status & ERR_OVER) { + if (!startup) { struct snd_soc_dai *dai = fsi_get_dai(substream); - dev_err(dai->dev, "over run error\n"); - fsi_reg_write(fsi, DOFF_ST, status & ~ST_ERR); - ret = -EIO; + + if (status & ERR_OVER) + dev_err(dai->dev, "over run\n"); + if (status & ERR_UNDER) + dev_err(dai->dev, "under run\n"); } + fsi_reg_write(fsi, DOFF_ST, 0); fsi_irq_enable(fsi, 1); if (over_period) snd_pcm_period_elapsed(substream); - return ret; + return 0; } -static int fsi_data_pop(struct fsi_priv *fsi) +static int fsi_data_pop(struct fsi_priv *fsi, int startup) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; @@ -479,7 +481,7 @@ static int fsi_data_pop(struct fsi_priv *fsi) int fifo_fill; int width; u8 *start; - int i, ret, over_period; + int i, over_period; if (!fsi || !fsi->substream || @@ -534,21 +536,23 @@ static int fsi_data_pop(struct fsi_priv *fsi) fsi->byte_offset += fifo_fill * width; - ret = 0; status = fsi_reg_read(fsi, DIFF_ST); - if (status & ERR_UNDER) { + if (!startup) { struct snd_soc_dai *dai = fsi_get_dai(substream); - dev_err(dai->dev, "under run error\n"); - fsi_reg_write(fsi, DIFF_ST, status & ~ST_ERR); - ret = -EIO; + + if (status & ERR_OVER) + dev_err(dai->dev, "over run\n"); + if (status & ERR_UNDER) + dev_err(dai->dev, "under run\n"); } + fsi_reg_write(fsi, DIFF_ST, 0); fsi_irq_enable(fsi, 0); if (over_period) snd_pcm_period_elapsed(substream); - return ret; + return 0; } static irqreturn_t fsi_interrupt(int irq, void *data) @@ -562,13 +566,13 @@ static irqreturn_t fsi_interrupt(int irq, void *data) fsi_master_write(master, SOFT_RST, status | 0x00000010); if (int_st & INT_A_OUT) - fsi_data_push(&master->fsia); + fsi_data_push(&master->fsia, 0); if (int_st & INT_B_OUT) - fsi_data_push(&master->fsib); + fsi_data_push(&master->fsib, 0); if (int_st & INT_A_IN) - fsi_data_pop(&master->fsia); + fsi_data_pop(&master->fsia, 0); if (int_st & INT_B_IN) - fsi_data_pop(&master->fsib); + fsi_data_pop(&master->fsib, 0); fsi_master_write(master, INT_ST, 0x0000000); @@ -726,7 +730,7 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, fsi_stream_push(fsi, substream, frames_to_bytes(runtime, runtime->buffer_size), frames_to_bytes(runtime, runtime->period_size)); - ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi); + ret = is_play ? fsi_data_push(fsi, 1) : fsi_data_pop(fsi, 1); break; case SNDRV_PCM_TRIGGER_STOP: fsi_irq_disable(fsi, is_play); -- 1.6.3.3

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[alsa-devel] Separate dma driver for cpu_dais
by jassi brar 23 Feb '10

23 Feb '10

Hi, The query concerns ASOC. Currently we have one platform per card. What if my card has two dai_links with each cpu_dai fed data with different mechanism(one with system dma controller and the other with a dma dedicated to the Audio block). The cpu_dai's are tightly coupled and I don't wanna have separate socdev's for them. Presently, I define a dma_id flag in snd_soc_dai, make the snd_soc_card point to a 'wrapper/muxed' platform driver that checks the dma_id flag and calls the relevant platform's callbacks. Of course, each cpu_dai initializes the dma_id to a unique value that corresponds to it's real dma mechanism. What approach/changes would you suggest I take, so that someday I could submit the implementation for mainline inclusion? (For those curious to have a reference - I am talking about Samsung's SoCs <http://dev.odroid.com/wiki/odroid/pds/HardwareInformation/S5PC100_UM_REV101… refer to figures on page-1635 & 1637> that have I2S controllers with h/w mixing capability and where 'secondary fifo' can be fed with I2S internal DMA controller with dedicated h/w ring buffer) Thanks.

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[alsa-devel] [PATCH] ALSA: add support for Macbook Air 2, 1 internal speaker.
by Reimundo Heluani 23 Feb '10

23 Feb '10

From: Reimundo Heluani <rheluani(a)gmail.com> Add support for Macbook Air 2,1 (late 2008) internal speaker and headphones. Create a "mba21" model for snd-hda-intel. Signed-off-by: Reimundo Heluani <rheluani(a)gmail.com> --- Still missing internal microphone support --- linux-2.6/sound/pci/hda/patch_realtek.c.orig 2010-02-23 00:36:16.000000000 -0800 +++ linux-2.6/sound/pci/hda/patch_realtek.c 2010-02-23 00:44:35.000000000 -0800 @@ -207,6 +207,7 @@ enum { ALC882_ASUS_A7J, ALC882_ASUS_A7M, ALC885_MACPRO, + ALC885_MBA21, ALC885_MBP3, ALC885_MB5, ALC885_IMAC24, @@ -6852,6 +6853,13 @@ static struct hda_channel_mode alc882_si { 8, alc882_sixstack_ch8_init }, }; + +/* Macbook Air 2,1 */ + +static struct hda_channel_mode alc885_mba21_ch_modes[1] = { + { 2, NULL }, +}; + /* * macbook pro ALC885 can switch LineIn to LineOut without losing Mic */ @@ -7123,6 +7131,15 @@ static struct snd_kcontrol_new alc882_ba { } /* end */ }; +/* Macbook Air 2,1 same control for HP and internal Speaker */ + +static struct snd_kcontrol_new alc885_mba21_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT), + { } +}; + + static struct snd_kcontrol_new alc885_mbp3_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), @@ -7562,6 +7579,29 @@ static struct hda_verb alc885_mb5_init_v { } }; + +static struct hda_verb alc885_mba21_init_verbs[] = { + /*Internal and HP Speaker Mixer*/ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /*Internal Speaker Pin (0x0c)*/ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP Pin: output 0 (0x0e) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC880_HP_EVENT | AC_USRSP_EN)}, + /* Line in (is hp when jack connected)*/ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + { } + }; + + /* Macbook Pro rev3 */ static struct hda_verb alc885_mbp3_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -7724,6 +7764,17 @@ static void alc885_imac24_setup(struct h spec->autocfg.speaker_pins[1] = 0x1a; } +/* Macbook Air 2,1 */ +static void alc885_mba21_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x18; +} + + + static void alc885_mbp3_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -8938,6 +8989,7 @@ static const char *alc882_models[ALC882_ [ALC882_ASUS_A7M] = "asus-a7m", [ALC885_MACPRO] = "macpro", [ALC885_MB5] = "mb5", + [ALC885_MBA21] = "mba21", [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC885_IMAC91] = "imac91", @@ -9172,6 +9224,18 @@ static struct alc_config_preset alc882_p .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, }, + [ALC885_MBA21] = { + .mixers = { alc885_mba21_mixer }, + .init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs }, + .num_dacs = 2, + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_mba21_ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), + .input_mux = &alc882_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc885_mba21_setup, + .init_hook = alc_automute_amp, + }, [ALC885_MBP3] = { .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_mbp3_init_verbs,

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[alsa-devel] Conexant CX20561 alsa driver not muting headphone jack on Thinkpads when mute button pressed .. any ideas?
by Jerone Young 23 Feb '10

23 Feb '10

There is an interesting problem with Lenovo Thinkpads X200, X200s, X301, & .. looks like W500 using the Conexant CX20561 codec under Linux with alsa. But if you have your headphone plugged in. You get audio .. but if you press the hardware mute button .. it will not mute the headphones. By pulling out the headphones you will notice though that the internal speakers are muted. At first I blamed the bios. But then I did a test under FreeDOS where I used the beep command for sound. Sure enough I press the mute button and the headphone jack is muted. I've been looking for any documentation I can find on the CX20561 but there is non out there I can find. I could be wrong, but it seems some bit isn't being flipped for this codec to work correctly with this functionality with the alsa driver. Any ideas? I've created a bug here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4928 Thanks, Jerone

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[alsa-devel] [PATCHES_v1] ASoC: Towards dropping runtime field from snd_soc_dai
by jassi brar 23 Feb '10

23 Feb '10

Hi, <Due to some problem I am unable to send inline patches, please accept the attached patches for this time> The following three patches are aimed at discarding runtime field from struct snd_soc_dai, which comes in the way of working towards making it possible to share cpu_dai/codec_dai among two or more dai_links. [PATCH 1/3] ASoC: PLATFORM: Pass dai_link as argument to suspend, resume Also a good idea to provide pointer to relevant dai_link to platform suspend/resume. It also makes easy traversing the ASOC links to reach the substream(when snd_soc_dai.runtime is dropped). [PATCH 2/3] ASOC: ATMEL: Getting cpu_dai method change [PATCH 3/3] ASOC: SND_SOC_DAI: Get rid of 'runtime' field Regards.

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[alsa-devel] Macbook Air 2,1 - no mic nor main speakers
by Reimundo Heluani 23 Feb '10

23 Feb '10

Hello list, this came up a year ago and Takashi's answer was "a broken bios". I was hoping someone here could point me to documentation to read in order to diagnose the problem if indeed is a broken bios. I don't want to start randombly reading the driver's code. Macbook Air 2,1 2.6.32-gentoo-r3 SMP x86_64 Alsa Driver 1.0.21 libs 1.0.22 nVidia Corporation MCP79 High Definition Audio (rev b1) HDA NVidia at 0x93380000 irq 19 00:08.0 0403: 10de:0ac0 (rev b1) Subsystem: 10de:cb79 Codec: Realtek ALC889A Booting with snd_hda_intel.model=mbp3 get sound on headphones, but not on mainspeakers. The mic never worked.

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[alsa-devel] [PATCH v4 3/3] ASoC: OMAP4: Add McPDM platform driver
by Candelaria Villareal, Jorge 22 Feb '10

22 Feb '10

From: Misael Lopez Cruz <x0052729(a)ti.com> 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 <jorge.candelaria(a)ti.com> --- sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-mcpdm.c | 251 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 +++++ 4 files changed, 285 insertions(+), 0 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/Kconfig b/sound/soc/omap/Kconfig index 61952aa..b96e9d8 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -6,6 +6,9 @@ config SND_OMAP_SOC_MCBSP tristate select OMAP_MCBSP +config SND_OMAP_SOC_MCPDM + tristate + config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3db8a6c..41ae838 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,9 +1,11 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o +snd-soc-omap-mcpdm-objs := omap-mcpdm.o mcpdm.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o +obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o # OMAP Machine Support snd-soc-n810-objs := n810.o diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c new file mode 100644 index 0000000..25f19e4 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.c @@ -0,0 +1,251 @@ +/* + * 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 <plat/control.h> +#include <plat/dma.h> +#include <plat/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 = MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, + /* uplink */ + { + .irq_mask = MCPDM_UP_IRQ_EMPTY | MCPDM_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 playback", + .dma_req = OMAP44XX_DMA_MCPDM_DL, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .packet_size = 16, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA, + }, + { + .name = "Audio capture", + .dma_req = OMAP44XX_DMA_MCPDM_UP, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .packet_size = 16, + .port_addr = OMAP44XX_MCPDM_L3_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 == SNDRV_PCM_STREAM_CAPTURE) + /* up to 2 channels for capture */ + return -EINVAL; + link_mask |= 1 << 3; + case 3: + if (stream == SNDRV_PCM_STREAM_CAPTURE) + /* up to 2 channels for capture */ + 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 == SNDRV_PCM_STREAM_PLAYBACK) { + mcpdm_links[stream].channels = link_mask << 3; + err = omap_mcpdm_playback_open(&mcpdm_links[stream]); + } else { + mcpdm_links[stream].channels = link_mask << 0; + err = omap_mcpdm_capture_open(&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 stream = substream->stream; + int err; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + err = omap_mcpdm_playback_close(&mcpdm_links[stream]); + else + err = omap_mcpdm_capture_close(&mcpdm_links[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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