October 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: kirkwood: Fix compile error due to patch 'add S/PDIF support'
by Jean-Francois Moine 25 Oct '13

25 Oct '13

This patch fixes the compilation error of kirkwood-i2s.c introduced by the commit 75b9b65ee5a80e99e 'ASoC: kirkwood: add S/PDIF support'. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- Mark, sorry for this error. This patch applies on your repository broonie/sound.git/ topic/kirkwood. I may give you an other patch to apply with the commit removed. --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 9ec38d1..d34d917 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -568,7 +568,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } else { dev_info(&pdev->dev, "found external clock\n"); clk_prepare_enable(priv->extclk); - soc_dai = &kirkwood_i2s_dai_extclk; + soc_dai = kirkwood_i2s_dai_extclk; } } -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] Variable Delay through ALSA driver
by Brandon Yates 25 Oct '13

25 Oct '13

I have an application using the ALSA API that writes to playback device and simultaneously reads from capture device. My processing loop is : 1)Read from microphone (blocking) 2)Process Data 3)Write to speaker I am measuring the delay from the time I play a series of frames to the speaker until I detect the echo of this coming back in through the microphone. This delay seems to be changing every time I initialize the devices with ALSA. I have measured in the range of 60-160ms delay. I know things like period size effect delay, but using the exact same software and hardware configuration I would expect this to be constant across runs. What are some parameters that can help me make the delay deterministic? Thanks, Brandon

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[alsa-devel] [PATCH] ALSA: hda - hdmi: Re-setup pin and infoframe on plug-in on all codecs
by Anssi Hannula 25 Oct '13

25 Oct '13

hdmi_setup_audio_infoframe() does not set up pin and infoframe if there is no connected sink. If a sink is connected while audio playback is already in progress, the pin and infoframe will not be properly set up, causing no audio or wrongly mapped audio. On Intel Haswell codecs the hdmi_setup_audio_infoframe() is already called again from hdmi_present_sense() when an ELD appears because transcoder:port mapping may have changed. Make the call non-Haswell-specific so that audio will be properly set up if the playback was started before a sink was connected. Tested on non-Haswell Intel HDMI codec by plugging sink in during playback. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- sound/pci/hda/patch_hdmi.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 804adb8..a1fe4bb 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1462,11 +1462,12 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) pin_eld->eld_size = eld->eld_size; pin_eld->info = eld->info; - /* Haswell-specific workaround: re-setup when the transcoder is - * changed during the stream playback + /* + * Re-setup pin and infoframe. This is needed e.g. when + * - sink is first plugged-in (infoframe is not set up if !monitor_present) + * - transcoder can change during stream playback on Haswell */ - if (is_haswell(codec) && - eld->eld_valid && !old_eld_valid && per_pin->setup) + if (eld->eld_valid && !old_eld_valid && per_pin->setup) hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); } -- 1.8.1.5

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[alsa-devel] Bug report Realtek ALC887-VD HDA Intel tone://15000
by Mihai Moise 24 Oct '13

24 Oct '13

I am writing to report a bug with ALSA that affects XMMS and Audacious. I have a Realtek ALC887-VD HDA Intel. I tested with kernel 3.12-rc4 and alsa-libs 1.0.27.2.10.gc1fbd and alsa-oss-1.0.25.2.g39df1 library. I also compiled and installed alsa-utils-1.0.27.2.6.gf1e99 which made me realize I had to move the libraries from /usr/lib to /usr/lib64. Playing tone://15000 in Audacious or XMMS with the ALSA driver without resampling sounds bad (in XMMS there is no resampling option). How to reproduce the bug: Play in XMMS tone://10 In Audacious you would have to lower the volume by 6 dB to get the same sound volume as in XMMS because it pre-amplifies sounds. Lower the PC-volume until you stop hearing clicks. To hear the clicks you need headphones and to raise the real-life amplifier volume all the way up. For me it is, Master: 91% PCM: 100% Front: 60% Select the ALSA driver and no resampling in Audacious (I think Audacious comes with no resampling as the default setting which is why this is so important). Play tone://15000 It is supposed to sound like this: perl -e '$|++;binmode(STDOUT);$_ = 0;while(1){print pack("v",32767.0*sin(($_&(~1))*3.14159265*15000.0/48000.0));++$_;}' | aplay -f dat but it sounds like this: perl -e '$|++;binmode(STDOUT);$_ = 0;while(1){print pack("v",32767.0*sin(($_&(~1))*3.14159265*15000.0/44100.0));++$_;}' | aplay -f cd This is how it sounds like if the OSS driver is invoked: perl -e '$|++;binmode(STDOUT);$_ = 0;while(1){print pack("v",32767.0*sin(($_&(~1))*3.14159265*15000.0/44100.0));++$_;}' | sox -V4 -b 16 -c 2 -r 44100 -e signed-integer -t raw - -r 44100 -t oss -d On my computer, the first and third command lines sound good, the second command line sounds bad. When using the OSS driver in XMMS it sounds like the first and third command lines. To go into more details: when using the ALSA driver in XMMS, for frequencies higher than tone://10000 the main frequency gets swamped by lower harmonics, so the frequency heard seems to drop the higher it increases. At tone://15000 what is heard is a louder lower harmonic. My suggestion is: could there be a way to force ALSA to resample the sound when using an ALC887-VD and/or a Realtek chipset and/or HDA Intel...

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[alsa-devel] [PATCH] ALSA: hda - hdmi: Fix channel map switch not taking effect
by Anssi Hannula 24 Oct '13

24 Oct '13

Currently hdmi_setup_audio_infoframe() reprograms the HDA channel mapping only when the infoframe is not up-to-date or the non-PCM flag has changed. However, when just the channel map has been changed, the infoframe may still be up-to-date and non-PCM flag may not have changed, so the new channel map is not actually programmed into the HDA codec. Notably, this failing case is also always triggered when the device is already in a prepared state and a new channel map is configured while changing only the channel positions (for example, plain "speaker-test -c2 -m FR,FL"). Fix that by keeping track of the last programmed channel map and making hdmi_setup_audio_infoframe() always update the hardware mapping if the channel map has changed. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: <stable(a)vger.kernel.org> --- BTW, is there some mechanism preventing hdmi_chmap_ctl_put() to be called at the same time with generic_hdmi_playback_pcm_prepare() that I didn't see with a quick look? Both of them call hdmi_setup_audio_infoframe() which is not thread-safe. sound/pci/hda/patch_hdmi.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7ea0245..09f38a1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -74,6 +74,8 @@ struct hdmi_spec_per_pin { bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ + bool last_chmap_set; /* channel-map override programmed? */ + unsigned char last_chmap[8]; /* last programmed channel-map */ char pcm_name[8]; /* filled in build_pcm callbacks */ }; @@ -955,15 +957,21 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, ai.bytes, sizeof(ai)); hdmi_start_infoframe_trans(codec, pin_nid); } else { - /* For non-pcm audio switch, setup new channel mapping - * accordingly */ - if (per_pin->non_pcm != non_pcm) + /* For non-pcm audio switch or channel-map switch, + * setup new channel mapping accordingly */ + if (per_pin->non_pcm != non_pcm + || per_pin->chmap_set != per_pin->last_chmap_set + || (per_pin->chmap_set && memcmp(per_pin->chmap, per_pin->last_chmap, + ARRAY_SIZE(per_pin->chmap)) != 0)) hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, channels, per_pin->chmap, per_pin->chmap_set); } per_pin->non_pcm = non_pcm; + per_pin->last_chmap_set = per_pin->chmap_set; + if (per_pin->chmap_set) + memcpy(per_pin->last_chmap, per_pin->chmap, ARRAY_SIZE(per_pin->chmap)); } -- 1.8.1.5

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[alsa-devel] [PATCH 1/2] ASoC: cs42l52: Add support for platform data reset GPIO
by Brian Austin 24 Oct '13

24 Oct '13

This patch adds support for grabbing a reset GPIO from platform data and resetting the codec Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- include/sound/cs42l52.h | 3 +++ sound/soc/codecs/cs42l52.c | 25 ++++++++++++++++++++----- sound/soc/codecs/cs42l52.h | 2 +- 3 files changed, 24 insertions(+), 6 deletions(-) diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h index a902e51..853df80 100644 --- a/include/sound/cs42l52.h +++ b/include/sound/cs42l52.h @@ -33,6 +33,9 @@ struct cs42l52_platform_data { /* MCLK Freq */ unsigned int mclk_freq; + + /* !RST GPIO */ + unsigned int reset_gpio; }; #endif /* __CS42L52_H */ diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index be2ba1b..6782d4875 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -17,6 +17,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/gpio.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/input.h> @@ -1205,6 +1206,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l52_private *cs42l52; + struct cs42l52_platform_data *pdata = dev_get_platdata(&i2c_client->dev); int ret; unsigned int devid = 0; unsigned int reg; @@ -1222,12 +1224,23 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, return ret; } + if (pdata) + cs42l52->pdata = *pdata; + + if ( cs42l52->pdata.reset_gpio) { + ret = gpio_request_one(cs42l52->pdata.reset_gpio, + GPIOF_OUT_INIT_HIGH, "CS42L52 /RST"); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n", + cs42l52->pdata.reset_gpio, ret); + return ret; + } + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 0); + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 1); + } + i2c_set_clientdata(i2c_client, cs42l52); - - if (dev_get_platdata(&i2c_client->dev)) - memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev), - sizeof(cs42l52->pdata)); - + ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch, ARRAY_SIZE(cs42l52_threshold_patch)); if (ret != 0) @@ -1243,6 +1256,8 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, devid, CS42L52_CHIP_ID); return ret; } + dev_info(&i2c_client->dev, + "Cirrus Logic CS42L52, Revision: %02X\n", reg & 0xFF); regcache_cache_only(cs42l52->regmap, true); diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 4277012..1a9412d 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -269,6 +269,6 @@ #define CS42L52_FIX_BITS1 0x3E #define CS42L52_FIX_BITS2 0x47 -#define CS42L52_MAX_REGISTER 0x34 +#define CS42L52_MAX_REGISTER 0x47 #endif -- 1.8.4.rc3

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[alsa-devel] [PATCH v7] ALSA: Add SoC on-chip internal ram support for DMA buffer allocation
by Nicolin Chen 24 Oct '13

24 Oct '13

Now it's quite common that an SoC contains its on-chip internal RAM. By using this RAM space for DMA buffer during audio playback/record, we can shutdown the voltage for external RAM to save power. So add new DEV type with iram malloc()/free() and accordingly modify current default mmap() for the iram circumstance. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- Changelog v6->v7: * Assigned pool to private_data field and accordingly refined helper functions. v5->v6: * Dropped remaining OF dependency bacause no need to add them: of_node doesn't care about OF; of_get_named_gen_pool() returns NULL if !OF. v4->v5: * Minimized the OF dependency. v3->v4: * Appropriately placed '#ifdef CONFIG_OF' * Tested by adding '#undef CONFIG_OF' to modified files, and passed compiling. v2->v3: * Moved iram specific mmap procedure out of ifdef ARCH_HAS_DMA_MMAP_COHERENT v1->v2: * Added of_node check * Dropped noops and unused physical addr in snd_free_dev_iram() --- include/sound/memalloc.h | 1 + sound/core/memalloc.c | 52 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/core/pcm_native.c | 6 ++++++ 3 files changed, 59 insertions(+) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index cf15b82..510aec4 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -52,6 +52,7 @@ struct snd_dma_device { #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #endif +#define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index bdf826f..18c1d47 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -30,6 +30,7 @@ #include <linux/seq_file.h> #include <asm/uaccess.h> #include <linux/dma-mapping.h> +#include <linux/genalloc.h> #include <linux/moduleparam.h> #include <linux/mutex.h> #include <sound/memalloc.h> @@ -157,6 +158,46 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, dec_snd_pages(pg); dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } + +/** + * snd_malloc_dev_iram - allocate memory from on-chip internal ram + * @dmab: buffer allocation record to store the allocated data + * @size: number of bytes to allocate from the iram + * + * This function requires iram phandle provided via of_node + */ +void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) +{ + struct device *dev = dmab->dev.dev; + struct gen_pool *pool = NULL; + + if (dev->of_node) + pool = of_get_named_gen_pool(dev->of_node, "iram", 0); + + if (!pool) + return; + + /* Assign the pool into private_data field */ + dmab->private_data = pool; + + dmab->area = (void *)gen_pool_alloc(pool, size); + if (!dmab->area) + return; + + dmab->addr = gen_pool_virt_to_phys(pool, (unsigned long)dmab->area); +} + +/** + * snd_free_dev_iram - free allocated specific memory from on-chip internal ram + * @dmab: buffer allocation record to store the allocated data + */ +void snd_free_dev_iram(struct snd_dma_buffer *dmab) +{ + struct gen_pool *pool = dmab->private_data; + + if (pool && dmab->area) + gen_pool_free(pool, (unsigned long)dmab->area, dmab->bytes); +} #endif /* CONFIG_HAS_DMA */ /* @@ -197,6 +238,14 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: + snd_malloc_dev_iram(dmab, size); + if (dmab->area) + break; + /* Internal memory might have limited size and no enough space, + * so if we fail to malloc, try to fetch memory traditionally. + */ + dmab->dev.type = SNDRV_DMA_TYPE_DEV; case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; @@ -269,6 +318,9 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) snd_free_pages(dmab->area, dmab->bytes); break; #ifdef CONFIG_HAS_DMA + case SNDRV_DMA_TYPE_DEV_IRAM: + snd_free_dev_iram(dmab); + break; case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a68d4c6..513f095 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3199,6 +3199,12 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_IRAM) { + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return remap_pfn_range(area, area->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + area->vm_end - area->vm_start, area->vm_page_prot); + } #ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) -- 1.8.4

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[alsa-devel] [PATCH v4] ASoC: kirkwood: add S/PDIF support
by Jean-Francois Moine 24 Oct '13

24 Oct '13

This patch adds S/PDIF input/output for mvebu DT boards. Three output DAI's are created: I2S only, S/PDIF only, both I2S and S/PDIF. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v4: declare 3 DAI's v3: (Mark Brown) - giving a name to the DAI's impacts the Openrd and HP t5325 clients - test i2s or spdif from the DAI's ID instead of its name v2: declare 2 DAI's (Mark Brown) --- sound/soc/kirkwood/kirkwood-i2s.c | 119 ++++++++++++--------- sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- 3 file changed, 72 insertions(+), 51 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d0504a2..c3a5a3e 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -160,9 +160,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16; ctl_play = KIRKWOOD_PLAYCTL_SIZE_16_C | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_16_C | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; /* * doesn't work... S20_3LE != kirkwood 20bit format ? @@ -178,9 +180,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S24_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24; ctl_play = KIRKWOOD_PLAYCTL_SIZE_24 | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_24 | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; case SNDRV_PCM_FORMAT_S32_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32; @@ -240,6 +244,11 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl); } + if (dai->id == 0) + ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s only */ + else if (dai->id == 1) + ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif only */ + switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* configure */ @@ -258,7 +267,8 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: /* stop audio, disable interrupts */ - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); @@ -272,13 +282,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -301,7 +313,13 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_rec; - value = ctl & ~KIRKWOOD_RECCTL_I2S_EN; + if (dai->id == 0) + ctl &= ~KIRKWOOD_RECCTL_SPDIF_EN; /* i2s only */ + else if (dai->id == 1) + ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif only */ + + value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN); writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -361,9 +379,8 @@ static int kirkwood_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int kirkwood_i2s_probe(struct snd_soc_dai *dai) +static int kirkwood_i2s_init(struct kirkwood_dma_data *priv) { - struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); unsigned long value; unsigned int reg_data; @@ -404,45 +421,46 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .set_fmt = kirkwood_i2s_set_fmt, }; +#define DAI_DEF(name_p, id_p, rates_p) {\ + .name = name_p,\ + .id = id_p,\ + .playback = {\ + .channels_min = 1,\ + .channels_max = 2,\ + .rates = rates_p,\ + .formats = KIRKWOOD_I2S_FORMATS,\ + },\ + .capture = {\ + .channels_min = 1,\ + .channels_max = 2,\ + .rates = rates_p,\ + .formats = KIRKWOOD_I2S_FORMATS,\ + },\ + .ops = &kirkwood_i2s_dai_ops,\ +} -static struct snd_soc_dai_driver kirkwood_i2s_dai = { - .probe = kirkwood_i2s_probe, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .ops = &kirkwood_i2s_dai_ops, +static struct snd_soc_dai_driver kirkwood_i2s_dai[3] = { + DAI_DEF("i2s", 0, + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000), + DAI_DEF("spdif", 1, + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000), + DAI_DEF("i2s/spdif", 2, + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000), }; -static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { - .probe = kirkwood_i2s_probe, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, - }, - .ops = &kirkwood_i2s_dai_ops, +static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[3] = { + DAI_DEF("i2s", 0, + SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT), + DAI_DEF("spdif", 1, + SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT), + DAI_DEF("i2s/spdif", 2, + SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT), }; static const struct snd_soc_component_driver kirkwood_i2s_component = { @@ -452,7 +470,7 @@ static const struct snd_soc_component_driver kirkwood_i2s_component = { static int kirkwood_i2s_dev_probe(struct platform_device *pdev) { struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data; - struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; + struct snd_soc_dai_driver *soc_dai = kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; struct device_node *np = pdev->dev.of_node; @@ -506,7 +524,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } else { dev_info(&pdev->dev, "found external clock\n"); clk_prepare_enable(priv->extclk); - soc_dai = &kirkwood_i2s_dai_extclk; + soc_dai = kirkwood_i2s_dai_extclk; } } @@ -524,7 +542,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, - soc_dai, 1); + soc_dai, 3); if (err) { dev_err(&pdev->dev, "snd_soc_register_component failed\n"); goto err_component; @@ -535,6 +553,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); goto err_platform; } + + kirkwood_i2s_init(priv); + return 0; err_platform: snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index 025be0e..65f2a5b 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -52,7 +52,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { { .name = "CS42L51", .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 27545b0..d213832 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -68,7 +68,7 @@ static struct snd_soc_dai_link t5325_dai[] = { { .name = "ALC5621", .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH v2] ASoC: kirkwood: add S/PDIF support
by Jean-Francois Moine 24 Oct '13

24 Oct '13

This patch adds S/PDIF input/output for mvebu DT boards. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v3: (Mark Brown) - giving a name to the DAI's impacts the Openrd and HP t5325 clients - test i2s or spdif from the DAI's ID instead of its name v2: declare 2 DAI's (Mark Brown) --- sound/soc/kirkwood/kirkwood-i2s.c | 99 ++++++++++++++++++---- sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/kirkwood/kirkwood-t5325.c | 2 +- 3 file changed, 84 insertions(+), 19 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index c2433ea..f3e1547 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -160,9 +160,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16; ctl_play = KIRKWOOD_PLAYCTL_SIZE_16_C | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_16_C | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; /* * doesn't work... S20_3LE != kirkwood 20bit format ? @@ -178,9 +180,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S24_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24; ctl_play = KIRKWOOD_PLAYCTL_SIZE_24 | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_24 | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; case SNDRV_PCM_FORMAT_S32_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32; @@ -240,6 +244,11 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl); } + if (dai->id == 0) + ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ + switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* configure */ @@ -258,7 +267,8 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: /* stop audio, disable interrupts */ - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); @@ -272,13 +282,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -301,7 +313,13 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_rec; - value = ctl & ~KIRKWOOD_RECCTL_I2S_EN; + if (dai->id == 0) + ctl &= ~KIRKWOOD_RECCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif */ + + value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN); writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -361,9 +379,8 @@ static int kirkwood_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int kirkwood_i2s_probe(struct snd_soc_dai *dai) +static int kirkwood_i2s_init(struct kirkwood_dma_data *priv) { - struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); unsigned long value; unsigned int reg_data; @@ -404,9 +421,29 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .set_fmt = kirkwood_i2s_set_fmt, }; - -static struct snd_soc_dai_driver kirkwood_i2s_dai = { - .probe = kirkwood_i2s_probe, +static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = { + { + .name = "i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .ops = &kirkwood_i2s_dai_ops, + }, + { + .name = "spdif", + .id = 1, .playback = { .channels_min = 1, .channels_max = 2, @@ -422,10 +459,34 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai = { .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, + }, }; -static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { - .probe = kirkwood_i2s_probe, +static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { + { + .name = "i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .ops = &kirkwood_i2s_dai_ops, + }, + { + .name = "spdif", + .id = 1, .playback = { .channels_min = 1, .channels_max = 2, @@ -443,6 +504,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, + }, }; static const struct snd_soc_component_driver kirkwood_i2s_component = { @@ -452,7 +514,7 @@ static const struct snd_soc_component_driver kirkwood_i2s_component = { static int kirkwood_i2s_dev_probe(struct platform_device *pdev) { struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data; - struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; + struct snd_soc_dai_driver *soc_dai = kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; struct device_node *np = pdev->dev.of_node; @@ -521,7 +583,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, - soc_dai, 1); + soc_dai, 2); if (err) { dev_err(&pdev->dev, "snd_soc_register_component failed\n"); goto err_component; @@ -532,6 +594,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); goto err_platform; } + + kirkwood_i2s_init(priv); + return 0; err_platform: snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index 025be0e..65f2a5b 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -52,7 +52,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { { .name = "CS42L51", .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 27545b0..d213832 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -68,7 +68,7 @@ static struct snd_soc_dai_link t5325_dai[] = { { .name = "ALC5621", .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH] ASoC: fsl: Add missing pm to current machine drivers
by Nicolin Chen 24 Oct '13

24 Oct '13

Add missing pm to current machine drivers so that all of them would correctly do suspend/resume. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/imx-mc13783.c | 1 + sound/soc/fsl/imx-sgtl5000.c | 1 + sound/soc/fsl/imx-wm8962.c | 1 + 3 files changed, 3 insertions(+) diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index a3d60d4..8ded5ad 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -160,6 +160,7 @@ static struct platform_driver imx_mc13783_audio_driver = { .driver = { .name = "imx_mc13783", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = imx_mc13783_probe, .remove = imx_mc13783_remove diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 6f4bdc8..4b65c04 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -201,6 +201,7 @@ static struct platform_driver imx_sgtl5000_driver = { .driver = { .name = "imx-sgtl5000", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 6c60666..72064e9 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -310,6 +310,7 @@ static struct platform_driver imx_wm8962_driver = { .driver = { .name = "imx-wm8962", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = imx_wm8962_dt_ids, }, .probe = imx_wm8962_probe, -- 1.8.4

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