August 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC multi-component branch is no more.
by Liam Girdwood 20 Aug '10

20 Aug '10

Quick note, subject says it all. Since ASoC multi-component has now been merged (in -next and both Mark and my for-2.6.37 branches) I'm going to archive the multi-component branch in order to stop any future confusion over where patches should be against for submission. Also please note that 'multi-component' is no longer require in subject or commit message ;) Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

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[alsa-devel] [PATCH 1/1] ASoC: clean dead file pxa2xx-pcm.h
by Haojian Zhuang 20 Aug '10

20 Aug '10

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[PATCH 1/1] ASoC: clean dead file pxa2xx-pcm.h
by Haojian Zhuang 20 Aug '10

20 Aug '10

Since pxa2xx-pcm.h is removed from sound/soc/pxa directory, clean it for passing compilation. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/pxa2xx-ac97.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9c2bafa..ac51c6d 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -24,7 +24,6 @@ #include <mach/dma.h> #include <mach/audio.h> -#include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) -- 1.5.6.5 --0016367fa53455dab4048e3ee29a Content-Type: text/x-patch; charset=US-ASCII; name="0001-ASoC-clean-dead-file-pxa2xx-pcm.h.patch" Content-Disposition: attachment; filename="0001-ASoC-clean-dead-file-pxa2xx-pcm.h.patch" Content-Transfer-Encoding: base64 X-Attachment-Id: f_gd2wgfov0 RnJvbSBhM2QyMzA5YzI4ZDUwMmU4OTk5N2UwY2MxYWI5OGYzY2FjYzY4NGRmIE1vbiBTZXAgMTcg MDA6MDA6MDAgMjAwMQpGcm9tOiBIYW9qaWFuIFpodWFuZyA8aGFvamlhbi56aHVhbmdAbWFydmVs bC5jb20+CkRhdGU6IEZyaSwgMjAgQXVnIDIwMTAgMTg6Mjc6NDEgKzA4MDAKU3ViamVjdDogW1BB VENIIDEvMV0gQVNvQzogY2xlYW4gZGVhZCBmaWxlIHB4YTJ4eC1wY20uaAoKU2luY2UgcHhhMnh4 LXBjbS5oIGlzIHJlbW92ZWQgZnJvbSBzb3VuZC9zb2MvcHhhIGRpcmVjdG9yeSwgY2xlYW4gaXQK Zm9yIHBhc3NpbmcgY29tcGlsYXRpb24uCgpTaWduZWQtb2ZmLWJ5OiBIYW9qaWFuIFpodWFuZyA8 aGFvamlhbi56aHVhbmdAbWFydmVsbC5jb20+Ci0tLQogc291bmQvc29jL3B4YS9weGEyeHgtYWM5 Ny5jIHwgICAgMSAtCiAxIGZpbGVzIGNoYW5nZWQsIDAgaW5zZXJ0aW9ucygrKSwgMSBkZWxldGlv bnMoLSkKCmRpZmYgLS1naXQgYS9zb3VuZC9zb2MvcHhhL3B4YTJ4eC1hYzk3LmMgYi9zb3VuZC9z b2MvcHhhL3B4YTJ4eC1hYzk3LmMKaW5kZXggOWMyYmFmYS4uYWM1MWM2ZCAxMDA2NDQKLS0tIGEv c291bmQvc29jL3B4YS9weGEyeHgtYWM5Ny5jCisrKyBiL3NvdW5kL3NvYy9weGEvcHhhMnh4LWFj OTcuYwpAQCAtMjQsNyArMjQsNiBAQAogI2luY2x1ZGUgPG1hY2gvZG1hLmg+CiAjaW5jbHVkZSA8 bWFjaC9hdWRpby5oPgogCi0jaW5jbHVkZSAicHhhMnh4LXBjbS5oIgogI2luY2x1ZGUgInB4YTJ4 eC1hYzk3LmgiCiAKIHN0YXRpYyB2b2lkIHB4YTJ4eF9hYzk3X3dhcm1fcmVzZXQoc3RydWN0IHNu ZF9hYzk3ICphYzk3KQotLSAKMS41LjYuNQoK --0016367fa53455dab4048e3ee29a--

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[alsa-devel] [PATCH 3/3] asoc/multi-component: add support for separate codec DAIs to the fsl_dma driver
by Timur Tabi 20 Aug '10

20 Aug '10

Some codecs have separate DAIs for playback and capture, so the DMA driver should allocate a DMA buffer only for the streams that are valid when the driver is opened. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 34 +++++++++++++++++++++------------- 1 files changed, 21 insertions(+), 13 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index cb78698..4cf98c0 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -306,21 +306,29 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = fsl_dma_dmamask; - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, - fsl_dma_hardware.buffer_bytes_max, - &pcm->streams[0].substream->dma_buffer); - if (ret) { - dev_err(card->dev, "can't allocate playback dma buffer\n"); - return ret; + /* Some codecs have separate DAIs for playback and capture, so we + * should allocate a DMA buffer only for the streams that are valid. + */ + + if (dai->driver->playback.channels_min) { + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, + fsl_dma_hardware.buffer_bytes_max, + &pcm->streams[0].substream->dma_buffer); + if (ret) { + dev_err(card->dev, "can't alloc playback dma buffer\n"); + return ret; + } } - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, - fsl_dma_hardware.buffer_bytes_max, - &pcm->streams[1].substream->dma_buffer); - if (ret) { - snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); - dev_err(card->dev, "can't allocate capture dma buffer\n"); - return ret; + if (dai->driver->capture.channels_min) { + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, + fsl_dma_hardware.buffer_bytes_max, + &pcm->streams[1].substream->dma_buffer); + if (ret) { + snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); + dev_err(card->dev, "can't alloc capture dma buffer\n"); + return ret; + } } return 0; -- 1.7.0.1

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[alsa-devel] purify_inactive_streams and Azalia controllers with >1 codec
by Stephen Warren 20 Aug '10

20 Aug '10

NVIDIA GPUs that contain Azalia controllers currently all have 1 controller, each having 4 codecs, each having 1 converter. I think there's a logic bug in hda_codec.c:purify_inactive_streams(); streams are controller-wide, but purify_inactive_streams() only cleans up the converters a single codec as far as I can tell. I think the solution is to rewrite purify_inactive_streams() as follows: static void purify_inactive_streams(struct hda_codec *codec) { struct hda_codec *itercodec; int i; list_for_each_entry(itercodec, &codec->bus->codec_list, list) { for (i = 0; i < itercodec->cvt_setups.used; i++) { struct hda_cvt_setup *p = snd_array_elem(&itercodec->cvt_setups, i); if (p->dirty) really_cleanup_stream(itercodec, p); } } } This alone doesn't work; had_codec.c:snd_hda_codec_setup_stream() also needs a similar loop when setting the dirty flag. However, this still doesn't work; as best I can tell, patch_nvhdmi.c:nvhdmi_dig_playback_pcm_prepare_8ch_89() never calls snd_hda_codec_setup_stream(). This is where I need help. patch_nvhdmi.c:nvhdmi_dig_playback_pcm_prepare_8ch_89() and patch_intelhdmi.c:nvhdmi_dig_playback_pcm_prepare_8ch_89() are identical; both call patch_hdmi.c:hdmi_setup_stream() to do the bulk of the work, which doesn't ever call snd_hda_codec_setup_stream(). However, patch_atihdmi.c:atihdmi_dig_playback_pcm_prepare() does call snd_hda_multi_out_dig_prepare which eventually calls snd_hda_codec_setup_stream(). I'm not sure what the correct fix is here; shouldn't all 3 patch_*hdmi.c work in the same way, and also call snd_hda_codec_setup_stream()? The end-result of this is that once an audio stream has been played on the correct converter, any subsequent streams played on any converter will also be sent out the original converter. This is somewhat confusing! Thanks for any pointers. -- nvpublic

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Re: [alsa-devel] [PATCH] of/device: Replace struct of_device with struct of_platform
by Timur Tabi 20 Aug '10

20 Aug '10

On Sun, Aug 1, 2010 at 2:57 AM, Grant Likely <grant.likely(a)secretlab.ca> wrote: > sound/soc/fsl/mpc8610_hpcd.c | 4 +- This file is/was significantly rewritten for 2.6.37. I posted a fix to the alsa mailing list, but it will conflict with your patch. You might want to remove all changes to sound/soc/fsl from this commit. -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] [PATCH v4] ASoC: multi-component: fix au1x platform
by Manuel Lauss 19 Aug '10

19 Aug '10

This patch fixes up the au1x audio platform to compile, and prepares the I2S and PCM modules for multiple card operation (no more globals for dma and i2s). The AC97 part is limited to one instance only for now since I couldn't find a way to reliably get at the PSC's platform information in the ac97 bus ops. Signed-off-by: Manuel Lauss <manuel.lauss(a)googlemail.com> --- v4: fixed a bug in the previous bugfix, and added DAI drvdata accessors. v3: fixed a bug which caused cat /proc/iomem to loop endlessly. v2: prepare PCM,I2S for multiple card operation, use dev_name() for DAI name. Run-tested on DB1200, both I2S and AC97 playback and recording work. AC97 spits out this kobject warning, which is caused by the "device_register()" in soc-core.c::soc_ac97_dev_register(): asoc: ac97-hifi <-> au1xpsc-ac97.1 mapping ok kobject (8fdc59b0): tried to init an initialized object, something is seriously wrong. Call Trace: [<804959c4>] dump_stack+0x8/0x34 [<802a30e4>] kobject_init+0x50/0xcc [<802ec27c>] device_initialize+0x2c/0x70 [<802ecb64>] device_register+0x14/0x28 [<8039e058>] snd_soc_instantiate_cards+0xa00/0xb10 [<8039e27c>] soc_probe+0x114/0x154 [<802ef0fc>] driver_probe_device+0xe4/0x1a0 [<802ee404>] bus_for_each_drv+0x60/0xb0 [<802ef370>] device_attach+0x74/0xa8 [<802ee1e8>] bus_probe_device+0x30/0x54 [<802ec9a0>] device_add+0x384/0x534 [<802f09a4>] platform_device_add+0x15c/0x1c8 [<805ccb14>] db1200_audio_load+0x70/0x9c [<801004fc>] do_one_initcall+0xfc/0x1e0 [<805ba32c>] kernel_init+0xc8/0x168 [<80105904>] kernel_thread_helper+0x10/0x18 ALSA device list: #0: DB1200_AC97 arch/mips/alchemy/devboards/db1200/platform.c | 10 ++- sound/soc/au1x/db1200.c | 18 ++-- sound/soc/au1x/dbdma2.c | 104 ++++++++++--------------- sound/soc/au1x/psc-ac97.c | 56 ++++++++------ sound/soc/au1x/psc-i2s.c | 42 ++++------- sound/soc/au1x/psc.h | 7 +- 6 files changed, 106 insertions(+), 131 deletions(-) diff --git a/arch/mips/alchemy/devboards/db1200/platform.c b/arch/mips/alchemy/devboards/db1200/platform.c index 3fa34c3..f985373 100644 --- a/arch/mips/alchemy/devboards/db1200/platform.c +++ b/arch/mips/alchemy/devboards/db1200/platform.c @@ -429,8 +429,14 @@ static struct platform_device db1200_audio_dev = { .resource = au1200_psc1_res, }; +static struct platform_device db1200_ac97_codec = { + .name = "ac97-codec", + .id = 1, /* on PSC1 */ +}; + static struct platform_device *db1200_devs[] __initdata = { NULL, /* PSC0, selected by S6.8 */ + &db1200_ac97_codec, &db1200_ide_dev, &db1200_eth_dev, &db1200_rtc_dev, @@ -494,11 +500,11 @@ static int __init db1200_dev_init(void) sw &= BCSR_SWITCHES_DIP_8 | BCSR_SWITCHES_DIP_7; if (sw == BCSR_SWITCHES_DIP_8) { bcsr_mod(BCSR_RESETS, 0, BCSR_RESETS_PSC1MUX); - db1200_audio_dev.name = "au1xpsc_i2s"; + db1200_audio_dev.name = "au1xpsc-i2s"; printk(KERN_INFO " S6.7 ON : PSC1 mode I2S\n"); } else { bcsr_mod(BCSR_RESETS, BCSR_RESETS_PSC1MUX, 0); - db1200_audio_dev.name = "au1xpsc_ac97"; + db1200_audio_dev.name = "au1xpsc-ac97"; printk(KERN_INFO " S6.7 OFF: PSC1 mode AC97\n"); } diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 8780c90..2065d03 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -26,11 +26,11 @@ static struct snd_soc_dai_link db1200_ac97_dai = { .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai_name = "au1xpsc-ac97", + .stream_name = "AC97 Playback", + .cpu_dai_name = "au1xpsc-ac97.1", .codec_dai_name = "ac97-hifi", - .platform_name = "au1xpsc-pcm-audio", - .codec_name = "ac97-codec", + .platform_name = "au1xpsc-pcm-audio.1", + .codec_name = "ac97-codec.1", }; static struct snd_soc_card db1200_ac97_machine = { @@ -74,11 +74,11 @@ static struct snd_soc_ops db1200_i2s_wm8731_ops = { static struct snd_soc_dai_link db1200_i2s_dai = { .name = "WM8731", - .stream_name = "WM8731 PCM", - .cpu_dai_name = "au1xpsc", - .codec_dai_name = "wm8731-hifi" - .platform_name = "au1xpsc-pcm-audio", - .codec_name = "wm8731-codec.0-001a", + .stream_name = "WM8731", + .cpu_dai_name = "au1xpsc-i2s.1", + .codec_dai_name = "wm8731-hifi", + .platform_name = "au1xpsc-pcm-audio.1", + .codec_name = "wm8731-codec.0-001b", .ops = &db1200_i2s_wm8731_ops, }; diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 00fdb9c..9a58831 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -10,9 +10,6 @@ * * DMA glue for Au1x-PSC audio. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ @@ -61,9 +58,6 @@ struct au1xpsc_audio_dmadata { int msbits; }; -/* instance data. There can be only one, MacLeod!!!! */ -static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2]; - /* * These settings are somewhat okay, at least on my machine audio plays * almost skip-free. Especially the 64kB buffer seems to help a LOT. @@ -199,19 +193,26 @@ out: return 0; } -static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, +static inline struct au1xpsc_audio_dmadata *to_dmadata(struct snd_pcm_substream *ss) +{ + struct snd_soc_pcm_runtime *rtd = ss->private_data; + struct au1xpsc_audio_dmadata *pcd = + snd_soc_platform_get_drvdata(rtd->platform); + return &pcd[SUBSTREAM_TYPE(ss)]; +} + +static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *ss, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct au1xpsc_audio_dmadata *pcd; + struct snd_pcm_runtime *runtime = ss->runtime; + struct au1xpsc_audio_dmadata *pcd = to_dmadata(ss); int stype, ret; - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + ret = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(params)); if (ret < 0) goto out; - stype = SUBSTREAM_TYPE(substream); - pcd = au1xpsc_audio_pcmdma[stype]; + stype = SUBSTREAM_TYPE(ss); DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " "runtime->min_align %d\n", @@ -222,13 +223,13 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, DBG("bits %d frags %d frag_bytes %d is_rx %d\n", params->msbits, params_periods(params), params_period_bytes(params), stype); - ret = au1x_pcm_dbdma_realloc(pcd, stype, params->msbits); + ret = au1x_pcm_dbdma_realloc(pcd, SUBSTREAM_TYPE(ss), params->msbits); if (ret) { MSG("DDMA channel (re)alloc failed!\n"); goto out; } - pcd->substream = substream; + pcd->substream = ss; pcd->period_bytes = params_period_bytes(params); pcd->periods = params_periods(params); pcd->dma_area_s = pcd->dma_area = runtime->dma_addr; @@ -249,8 +250,7 @@ static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream) static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) { - struct au1xpsc_audio_dmadata *pcd = - au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]; + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); au1xxx_dbdma_reset(pcd->ddma_chan); @@ -267,16 +267,16 @@ static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - u32 c = au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->ddma_chan; + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - au1xxx_dbdma_start(c); + au1xxx_dbdma_start(pcd->ddma_chan); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - au1xxx_dbdma_stop(c); + au1xxx_dbdma_stop(pcd->ddma_chan); break; default: return -EINVAL; @@ -287,8 +287,9 @@ static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static snd_pcm_uframes_t au1xpsc_pcm_pointer(struct snd_pcm_substream *substream) { - return bytes_to_frames(substream->runtime, - au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->pos); + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); + + return bytes_to_frames(substream->runtime, pcd->pos); } static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) @@ -299,7 +300,9 @@ static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) static int au1xpsc_pcm_close(struct snd_pcm_substream *substream) { - au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]); + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); + + au1x_pcm_dbdma_free(pcd); return 0; } @@ -329,83 +332,55 @@ static int au1xpsc_pcm_new(struct snd_card *card, return 0; } -static int au1xpsc_pcm_probe(struct snd_soc_platform *platform) -{ - if (!au1xpsc_audio_pcmdma[PCM_TX] || !au1xpsc_audio_pcmdma[PCM_RX]) - return -ENODEV; - - return 0; -} - /* au1xpsc audio platform */ struct snd_soc_platform_driver au1xpsc_soc_platform = { - .probe = au1xpsc_pcm_probe, .ops = &au1xpsc_pcm_ops, .pcm_new = au1xpsc_pcm_new, .pcm_free = au1xpsc_pcm_free_dma_buffers, }; -EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev) { + struct au1xpsc_audio_dmadata *dmadata; struct resource *r; int ret; - if (au1xpsc_audio_pcmdma[PCM_TX] || au1xpsc_audio_pcmdma[PCM_RX]) - return -EBUSY; - - /* TX DMA */ - au1xpsc_audio_pcmdma[PCM_TX] - = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); - if (!au1xpsc_audio_pcmdma[PCM_TX]) + dmadata = kzalloc(2 * sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); + if (!dmadata) return -ENOMEM; r = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!r) { ret = -ENODEV; - goto out1; + goto out2; } - (au1xpsc_audio_pcmdma[PCM_TX])->ddma_id = r->start; + dmadata[PCM_TX].ddma_id = r->start; /* RX DMA */ - au1xpsc_audio_pcmdma[PCM_RX] - = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); - if (!au1xpsc_audio_pcmdma[PCM_RX]) - return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!r) { ret = -ENODEV; goto out2; } - (au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start; + dmadata[PCM_RX].ddma_id = r->start; + + platform_set_drvdata(pdev, dmadata); ret = snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform); if (!ret) return ret; out2: - kfree(au1xpsc_audio_pcmdma[PCM_RX]); - au1xpsc_audio_pcmdma[PCM_RX] = NULL; -out1: - kfree(au1xpsc_audio_pcmdma[PCM_TX]); - au1xpsc_audio_pcmdma[PCM_TX] = NULL; + kfree(dmadata); return ret; } static int __devexit au1xpsc_pcm_drvremove(struct platform_device *pdev) { - int i; + struct au1xpsc_audio_dmadata *dmadata = platform_get_drvdata(pdev); snd_soc_unregister_platform(&pdev->dev); - - for (i = 0; i < 2; i++) { - if (au1xpsc_audio_pcmdma[i]) { - au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]); - kfree(au1xpsc_audio_pcmdma[i]); - au1xpsc_audio_pcmdma[i] = NULL; - } - } + kfree(dmadata); return 0; } @@ -421,8 +396,6 @@ static struct platform_driver au1xpsc_pcm_driver = { static int __init au1xpsc_audio_dbdma_load(void) { - au1xpsc_audio_pcmdma[PCM_TX] = NULL; - au1xpsc_audio_pcmdma[PCM_RX] = NULL; return platform_driver_register(&au1xpsc_pcm_driver); } @@ -435,6 +408,9 @@ module_init(au1xpsc_audio_dbdma_load); module_exit(au1xpsc_audio_dbdma_unload); +/* psc-ac97/psc-i2s call this function to create a pcm platform device with + * their DMA channel information. + */ struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev) { struct resource *res, *r; @@ -460,7 +436,7 @@ struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev) res[1].start = res[1].end = id[1]; res[0].flags = res[1].flags = IORESOURCE_DMA; - pd = platform_device_alloc("au1xpsc-pcm", -1); + pd = platform_device_alloc("au1xpsc-pcm-audio", pdev->id); if (!pd) goto out; diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 6a9516c..415b5aa 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -10,9 +10,6 @@ * * Au1xxx-PSC AC97 glue. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ #include <linux/init.h> @@ -56,12 +53,28 @@ /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; +#if 0 + +/* this could theoretically work, but ac97->bus->card can be NULL when + * snd_ac97_mixer() is called; I don't know if the rest further down the chain + * are always valid either. + */ +static inline struct au1xpsc_audio_data *ac97_to_pscdata(struct snd_ac97 *x) +{ + struct snd_soc_card *c = x->bus->card->private_data; + return snd_soc_dai_get_drvdata(c->rtd->cpu_dai); +} + +#else + +#define ac97_to_pscdata(x) au1xpsc_ac97_workdata +#endif + /* AC97 controller reads codec register */ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); unsigned short retry, tmo; unsigned long data; @@ -102,8 +115,7 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); unsigned int tmo, retry; au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); @@ -134,8 +146,7 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, /* AC97 controller asserts a warm reset */ static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); au_writel(PSC_AC97RST_SNC, AC97_RST(pscdata)); au_sync(); @@ -146,8 +157,7 @@ static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); int i; /* disable PSC during cold reset */ @@ -202,8 +212,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = dev_get_drvdata(dai->dev); unsigned long r, ro, stat; int chans, t, stype = SUBSTREAM_TYPE(substream); @@ -283,8 +292,7 @@ out: static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = dev_get_drvdata(dai->dev); int ret, stype = SUBSTREAM_TYPE(substream); ret = 0; @@ -325,7 +333,7 @@ static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { .hw_params = au1xpsc_ac97_hw_params, }; -struct snd_soc_dai_driver au1xpsc_ac97_dai = { +static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { .ac97_control = 1, .probe = au1xpsc_ac97_probe, .playback = { @@ -342,7 +350,6 @@ struct snd_soc_dai_driver au1xpsc_ac97_dai = { }, .ops = &au1xpsc_ac97_dai_ops, }; -EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) { @@ -351,9 +358,6 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) unsigned long sel; struct au1xpsc_audio_data *wd; - if (au1xpsc_ac97_workdata) - return -EBUSY; - wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); if (!wd) return -ENOMEM; @@ -387,14 +391,20 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(wd)); au_sync(); - ret = snd_soc_register_dai(&pdev->dev, &au1xpsc_ac97_dai); + /* name the DAI like this device instance ("au1xpsc-ac97.PSCINDEX") */ + memcpy(&wd->dai_drv, &au1xpsc_ac97_dai_template, + sizeof(struct snd_soc_dai_driver)); + wd->dai_drv.name = dev_name(&pdev->dev); + + platform_set_drvdata(pdev, wd); + + ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); if (ret) goto out1; wd->dmapd = au1xpsc_pcm_add(pdev); if (wd->dmapd) { - platform_set_drvdata(pdev, wd); - au1xpsc_ac97_workdata = wd; /* MDEV */ + au1xpsc_ac97_workdata = wd; return 0; } diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 94e560a..a8538a0 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -10,9 +10,6 @@ * * Au1xxx-PSC I2S glue. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. * NOTE: so far only PSC slave mode (bit- and frameclock) is supported. */ @@ -54,13 +51,10 @@ ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) -/* instance data. There can be only one, MacLeod!!!! */ -static struct au1xpsc_audio_data *au1xpsc_i2s_workdata; - static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(cpu_dai); unsigned long ct; int ret; @@ -120,7 +114,7 @@ static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); int cfgbits; unsigned long stat; @@ -245,7 +239,7 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); int ret, stype = SUBSTREAM_TYPE(substream); switch (cmd) { @@ -263,19 +257,13 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int au1xpsc_i2s_probe(struct snd_soc_dai *dai) -{ - return au1xpsc_i2s_workdata ? 0 : -ENODEV; -} - static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { .trigger = au1xpsc_i2s_trigger, .hw_params = au1xpsc_i2s_hw_params, .set_fmt = au1xpsc_i2s_set_fmt, }; -static struct snd_soc_dai_driver au1xpsc_i2s_dai = { - .probe = au1xpsc_i2s_probe, +static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { .playback = { .rates = AU1XPSC_I2S_RATES, .formats = AU1XPSC_I2S_FMTS, @@ -298,9 +286,6 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) int ret; struct au1xpsc_audio_data *wd; - if (au1xpsc_i2s_workdata) - return -EBUSY; - wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); if (!wd) return -ENOMEM; @@ -337,17 +322,21 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) * time out. */ - ret = snd_soc_register_dai(&pdev->dev, &au1xpsc_i2s_dai); + /* name the DAI like this device instance ("au1xpsc-i2s.PSCINDEX") */ + memcpy(&wd->dai_drv, &au1xpsc_i2s_dai_template, + sizeof(struct snd_soc_dai_driver)); + wd->dai_drv.name = dev_name(&pdev->dev); + + platform_set_drvdata(pdev, wd); + + ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); if (ret) goto out1; /* finally add the DMA device for this PSC */ wd->dmapd = au1xpsc_pcm_add(pdev); - if (wd->dmapd) { - platform_set_drvdata(pdev, wd); - au1xpsc_i2s_workdata = wd; + if (wd->dmapd) return 0; - } snd_soc_unregister_dai(&pdev->dev); out1: @@ -376,8 +365,6 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) release_mem_region(r->start, resource_size(r)); kfree(wd); - au1xpsc_i2s_workdata = NULL; /* MDEV */ - return 0; } @@ -427,7 +414,7 @@ static struct dev_pm_ops au1xpsci2s_pmops = { static struct platform_driver au1xpsc_i2s_driver = { .driver = { - .name = "au1xpsc", + .name = "au1xpsc-i2s", .owner = THIS_MODULE, .pm = AU1XPSCI2S_PMOPS, }, @@ -437,7 +424,6 @@ static struct platform_driver au1xpsc_i2s_driver = { static int __init au1xpsc_i2s_load(void) { - au1xpsc_i2s_workdata = NULL; return platform_driver_register(&au1xpsc_i2s_driver); } diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index f281443..b30eadd 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -8,16 +8,11 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ #ifndef _AU1X_PCM_H #define _AU1X_PCM_H -extern struct snd_ac97_bus_ops soc_ac97_ops; - /* DBDMA helpers */ extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev); extern void au1xpsc_pcm_destroy(struct platform_device *dmapd); @@ -28,6 +23,8 @@ struct au1xpsc_audio_data { unsigned long cfg; unsigned long rate; + struct snd_soc_dai_driver dai_drv; + unsigned long pm[2]; struct mutex lock; struct platform_device *dmapd; -- 1.7.2 -- 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

1 0

[alsa-devel] [PATCH v3] ASoC: multi-component: fix au1x platform
by Manuel Lauss 19 Aug '10

19 Aug '10

This patch fixes up the au1x audio platform to compile, and prepares the I2S and PCM modules for multiple card operation (no more globals for dma and i2s). The AC97 part is limited to one instance only for now due to other issues (see code). Signed-off-by: Manuel Lauss <manuel.lauss(a)googlemail.com> --- v3: fixed a bug which caused cat /proc/iomem to loop endlessly. v2: prepare PCM,I2S for multiple card operation, use dev_name() for DAI name. Run-tested on DB1200, both I2S and AC97 playback and recording work. AC97 spits out this kobject warning, which is caused by the "device_register()" in soc-core.c::soc_ac97_dev_register(): asoc: ac97-hifi <-> au1xpsc-ac97.1 mapping ok kobject (8fdc59b0): tried to init an initialized object, something is seriously wrong. Call Trace: [<804959c4>] dump_stack+0x8/0x34 [<802a30e4>] kobject_init+0x50/0xcc [<802ec27c>] device_initialize+0x2c/0x70 [<802ecb64>] device_register+0x14/0x28 [<8039e058>] snd_soc_instantiate_cards+0xa00/0xb10 [<8039e27c>] soc_probe+0x114/0x154 [<802ef0fc>] driver_probe_device+0xe4/0x1a0 [<802ee404>] bus_for_each_drv+0x60/0xb0 [<802ef370>] device_attach+0x74/0xa8 [<802ee1e8>] bus_probe_device+0x30/0x54 [<802ec9a0>] device_add+0x384/0x534 [<802f09a4>] platform_device_add+0x15c/0x1c8 [<805ccb14>] db1200_audio_load+0x70/0x9c [<801004fc>] do_one_initcall+0xfc/0x1e0 [<805ba32c>] kernel_init+0xc8/0x168 [<80105904>] kernel_thread_helper+0x10/0x18 ALSA device list: #0: DB1200_AC97 arch/mips/alchemy/devboards/db1200/platform.c | 10 ++- sound/soc/au1x/db1200.c | 18 ++-- sound/soc/au1x/dbdma2.c | 102 ++++++++++--------------- sound/soc/au1x/psc-ac97.c | 56 ++++++++------ sound/soc/au1x/psc-i2s.c | 42 ++++------- sound/soc/au1x/psc.h | 7 +- 6 files changed, 105 insertions(+), 130 deletions(-) diff --git a/arch/mips/alchemy/devboards/db1200/platform.c b/arch/mips/alchemy/devboards/db1200/platform.c index 3fa34c3..f985373 100644 --- a/arch/mips/alchemy/devboards/db1200/platform.c +++ b/arch/mips/alchemy/devboards/db1200/platform.c @@ -429,8 +429,14 @@ static struct platform_device db1200_audio_dev = { .resource = au1200_psc1_res, }; +static struct platform_device db1200_ac97_codec = { + .name = "ac97-codec", + .id = 1, /* on PSC1 */ +}; + static struct platform_device *db1200_devs[] __initdata = { NULL, /* PSC0, selected by S6.8 */ + &db1200_ac97_codec, &db1200_ide_dev, &db1200_eth_dev, &db1200_rtc_dev, @@ -494,11 +500,11 @@ static int __init db1200_dev_init(void) sw &= BCSR_SWITCHES_DIP_8 | BCSR_SWITCHES_DIP_7; if (sw == BCSR_SWITCHES_DIP_8) { bcsr_mod(BCSR_RESETS, 0, BCSR_RESETS_PSC1MUX); - db1200_audio_dev.name = "au1xpsc_i2s"; + db1200_audio_dev.name = "au1xpsc-i2s"; printk(KERN_INFO " S6.7 ON : PSC1 mode I2S\n"); } else { bcsr_mod(BCSR_RESETS, BCSR_RESETS_PSC1MUX, 0); - db1200_audio_dev.name = "au1xpsc_ac97"; + db1200_audio_dev.name = "au1xpsc-ac97"; printk(KERN_INFO " S6.7 OFF: PSC1 mode AC97\n"); } diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 8780c90..2065d03 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -26,11 +26,11 @@ static struct snd_soc_dai_link db1200_ac97_dai = { .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai_name = "au1xpsc-ac97", + .stream_name = "AC97 Playback", + .cpu_dai_name = "au1xpsc-ac97.1", .codec_dai_name = "ac97-hifi", - .platform_name = "au1xpsc-pcm-audio", - .codec_name = "ac97-codec", + .platform_name = "au1xpsc-pcm-audio.1", + .codec_name = "ac97-codec.1", }; static struct snd_soc_card db1200_ac97_machine = { @@ -74,11 +74,11 @@ static struct snd_soc_ops db1200_i2s_wm8731_ops = { static struct snd_soc_dai_link db1200_i2s_dai = { .name = "WM8731", - .stream_name = "WM8731 PCM", - .cpu_dai_name = "au1xpsc", - .codec_dai_name = "wm8731-hifi" - .platform_name = "au1xpsc-pcm-audio", - .codec_name = "wm8731-codec.0-001a", + .stream_name = "WM8731", + .cpu_dai_name = "au1xpsc-i2s.1", + .codec_dai_name = "wm8731-hifi", + .platform_name = "au1xpsc-pcm-audio.1", + .codec_name = "wm8731-codec.0-001b", .ops = &db1200_i2s_wm8731_ops, }; diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 00fdb9c..fd453c5 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -10,9 +10,6 @@ * * DMA glue for Au1x-PSC audio. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ @@ -61,9 +58,6 @@ struct au1xpsc_audio_dmadata { int msbits; }; -/* instance data. There can be only one, MacLeod!!!! */ -static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2]; - /* * These settings are somewhat okay, at least on my machine audio plays * almost skip-free. Especially the 64kB buffer seems to help a LOT. @@ -199,19 +193,26 @@ out: return 0; } -static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, +static inline struct au1xpsc_audio_dmadata *to_dmadata(struct snd_pcm_substream *ss) +{ + struct snd_soc_pcm_runtime *rtd = ss->private_data; + struct au1xpsc_audio_dmadata *pcd = + snd_soc_platform_get_drvdata(rtd->platform); + return &pcd[SUBSTREAM_TYPE(ss)]; +} + +static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *ss, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct au1xpsc_audio_dmadata *pcd; + struct snd_pcm_runtime *runtime = ss->runtime; + struct au1xpsc_audio_dmadata *pcd = to_dmadata(ss); int stype, ret; - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + ret = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(params)); if (ret < 0) goto out; - stype = SUBSTREAM_TYPE(substream); - pcd = au1xpsc_audio_pcmdma[stype]; + stype = SUBSTREAM_TYPE(ss); DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " "runtime->min_align %d\n", @@ -222,13 +223,13 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, DBG("bits %d frags %d frag_bytes %d is_rx %d\n", params->msbits, params_periods(params), params_period_bytes(params), stype); - ret = au1x_pcm_dbdma_realloc(pcd, stype, params->msbits); + ret = au1x_pcm_dbdma_realloc(pcd, SUBSTREAM_TYPE(ss), params->msbits); if (ret) { MSG("DDMA channel (re)alloc failed!\n"); goto out; } - pcd->substream = substream; + pcd->substream = ss; pcd->period_bytes = params_period_bytes(params); pcd->periods = params_periods(params); pcd->dma_area_s = pcd->dma_area = runtime->dma_addr; @@ -249,8 +250,7 @@ static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream) static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) { - struct au1xpsc_audio_dmadata *pcd = - au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]; + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); au1xxx_dbdma_reset(pcd->ddma_chan); @@ -267,16 +267,16 @@ static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - u32 c = au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->ddma_chan; + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - au1xxx_dbdma_start(c); + au1xxx_dbdma_start(pcd->ddma_chan); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - au1xxx_dbdma_stop(c); + au1xxx_dbdma_stop(pcd->ddma_chan); break; default: return -EINVAL; @@ -287,8 +287,9 @@ static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static snd_pcm_uframes_t au1xpsc_pcm_pointer(struct snd_pcm_substream *substream) { - return bytes_to_frames(substream->runtime, - au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->pos); + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); + + return bytes_to_frames(substream->runtime, pcd->pos); } static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) @@ -299,7 +300,9 @@ static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) static int au1xpsc_pcm_close(struct snd_pcm_substream *substream) { - au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]); + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); + + au1x_pcm_dbdma_free(pcd); return 0; } @@ -329,83 +332,55 @@ static int au1xpsc_pcm_new(struct snd_card *card, return 0; } -static int au1xpsc_pcm_probe(struct snd_soc_platform *platform) -{ - if (!au1xpsc_audio_pcmdma[PCM_TX] || !au1xpsc_audio_pcmdma[PCM_RX]) - return -ENODEV; - - return 0; -} - /* au1xpsc audio platform */ struct snd_soc_platform_driver au1xpsc_soc_platform = { - .probe = au1xpsc_pcm_probe, .ops = &au1xpsc_pcm_ops, .pcm_new = au1xpsc_pcm_new, .pcm_free = au1xpsc_pcm_free_dma_buffers, }; -EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev) { + struct au1xpsc_audio_dmadata *dmadata; struct resource *r; int ret; - if (au1xpsc_audio_pcmdma[PCM_TX] || au1xpsc_audio_pcmdma[PCM_RX]) - return -EBUSY; - - /* TX DMA */ - au1xpsc_audio_pcmdma[PCM_TX] - = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); - if (!au1xpsc_audio_pcmdma[PCM_TX]) + dmadata = kzalloc(2 * sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); + if (!dmadata) return -ENOMEM; r = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!r) { ret = -ENODEV; - goto out1; + goto out2; } - (au1xpsc_audio_pcmdma[PCM_TX])->ddma_id = r->start; + dmadata[PCM_TX].ddma_id = r->start; /* RX DMA */ - au1xpsc_audio_pcmdma[PCM_RX] - = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); - if (!au1xpsc_audio_pcmdma[PCM_RX]) - return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!r) { ret = -ENODEV; goto out2; } - (au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start; + dmadata[PCM_RX].ddma_id = r->start; + + platform_set_drvdata(pdev, dmadata); ret = snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform); if (!ret) return ret; out2: - kfree(au1xpsc_audio_pcmdma[PCM_RX]); - au1xpsc_audio_pcmdma[PCM_RX] = NULL; -out1: - kfree(au1xpsc_audio_pcmdma[PCM_TX]); - au1xpsc_audio_pcmdma[PCM_TX] = NULL; + kfree(dmadata); return ret; } static int __devexit au1xpsc_pcm_drvremove(struct platform_device *pdev) { - int i; + struct au1xpsc_audio_dmadata *dmadata = platform_get_drvdata(pdev); snd_soc_unregister_platform(&pdev->dev); - - for (i = 0; i < 2; i++) { - if (au1xpsc_audio_pcmdma[i]) { - au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]); - kfree(au1xpsc_audio_pcmdma[i]); - au1xpsc_audio_pcmdma[i] = NULL; - } - } + kfree(dmadata); return 0; } @@ -421,8 +396,6 @@ static struct platform_driver au1xpsc_pcm_driver = { static int __init au1xpsc_audio_dbdma_load(void) { - au1xpsc_audio_pcmdma[PCM_TX] = NULL; - au1xpsc_audio_pcmdma[PCM_RX] = NULL; return platform_driver_register(&au1xpsc_pcm_driver); } @@ -435,6 +408,9 @@ module_init(au1xpsc_audio_dbdma_load); module_exit(au1xpsc_audio_dbdma_unload); +/* psc-ac97/psc-i2s call this function to create a pcm platform device with + * their DMA channel information. + */ struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev) { struct resource *res, *r; diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 6a9516c..415b5aa 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -10,9 +10,6 @@ * * Au1xxx-PSC AC97 glue. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ #include <linux/init.h> @@ -56,12 +53,28 @@ /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; +#if 0 + +/* this could theoretically work, but ac97->bus->card can be NULL when + * snd_ac97_mixer() is called; I don't know if the rest further down the chain + * are always valid either. + */ +static inline struct au1xpsc_audio_data *ac97_to_pscdata(struct snd_ac97 *x) +{ + struct snd_soc_card *c = x->bus->card->private_data; + return snd_soc_dai_get_drvdata(c->rtd->cpu_dai); +} + +#else + +#define ac97_to_pscdata(x) au1xpsc_ac97_workdata +#endif + /* AC97 controller reads codec register */ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); unsigned short retry, tmo; unsigned long data; @@ -102,8 +115,7 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); unsigned int tmo, retry; au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); @@ -134,8 +146,7 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, /* AC97 controller asserts a warm reset */ static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); au_writel(PSC_AC97RST_SNC, AC97_RST(pscdata)); au_sync(); @@ -146,8 +157,7 @@ static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); int i; /* disable PSC during cold reset */ @@ -202,8 +212,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = dev_get_drvdata(dai->dev); unsigned long r, ro, stat; int chans, t, stype = SUBSTREAM_TYPE(substream); @@ -283,8 +292,7 @@ out: static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = dev_get_drvdata(dai->dev); int ret, stype = SUBSTREAM_TYPE(substream); ret = 0; @@ -325,7 +333,7 @@ static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { .hw_params = au1xpsc_ac97_hw_params, }; -struct snd_soc_dai_driver au1xpsc_ac97_dai = { +static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { .ac97_control = 1, .probe = au1xpsc_ac97_probe, .playback = { @@ -342,7 +350,6 @@ struct snd_soc_dai_driver au1xpsc_ac97_dai = { }, .ops = &au1xpsc_ac97_dai_ops, }; -EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) { @@ -351,9 +358,6 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) unsigned long sel; struct au1xpsc_audio_data *wd; - if (au1xpsc_ac97_workdata) - return -EBUSY; - wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); if (!wd) return -ENOMEM; @@ -387,14 +391,20 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(wd)); au_sync(); - ret = snd_soc_register_dai(&pdev->dev, &au1xpsc_ac97_dai); + /* name the DAI like this device instance ("au1xpsc-ac97.PSCINDEX") */ + memcpy(&wd->dai_drv, &au1xpsc_ac97_dai_template, + sizeof(struct snd_soc_dai_driver)); + wd->dai_drv.name = dev_name(&pdev->dev); + + platform_set_drvdata(pdev, wd); + + ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); if (ret) goto out1; wd->dmapd = au1xpsc_pcm_add(pdev); if (wd->dmapd) { - platform_set_drvdata(pdev, wd); - au1xpsc_ac97_workdata = wd; /* MDEV */ + au1xpsc_ac97_workdata = wd; return 0; } diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 94e560a..28c0ec4 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -10,9 +10,6 @@ * * Au1xxx-PSC I2S glue. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. * NOTE: so far only PSC slave mode (bit- and frameclock) is supported. */ @@ -54,13 +51,10 @@ ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) -/* instance data. There can be only one, MacLeod!!!! */ -static struct au1xpsc_audio_data *au1xpsc_i2s_workdata; - static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = dev_get_drvdata(cpu_dai->dev); unsigned long ct; int ret; @@ -120,7 +114,7 @@ static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = dev_get_drvdata(dai->dev); int cfgbits; unsigned long stat; @@ -245,7 +239,7 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = dev_get_drvdata(dai->dev); int ret, stype = SUBSTREAM_TYPE(substream); switch (cmd) { @@ -263,19 +257,13 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int au1xpsc_i2s_probe(struct snd_soc_dai *dai) -{ - return au1xpsc_i2s_workdata ? 0 : -ENODEV; -} - static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { .trigger = au1xpsc_i2s_trigger, .hw_params = au1xpsc_i2s_hw_params, .set_fmt = au1xpsc_i2s_set_fmt, }; -static struct snd_soc_dai_driver au1xpsc_i2s_dai = { - .probe = au1xpsc_i2s_probe, +static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { .playback = { .rates = AU1XPSC_I2S_RATES, .formats = AU1XPSC_I2S_FMTS, @@ -298,9 +286,6 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) int ret; struct au1xpsc_audio_data *wd; - if (au1xpsc_i2s_workdata) - return -EBUSY; - wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); if (!wd) return -ENOMEM; @@ -337,17 +322,21 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) * time out. */ - ret = snd_soc_register_dai(&pdev->dev, &au1xpsc_i2s_dai); + /* name the DAI like this device instance ("au1xpsc-i2s.PSCINDEX") */ + memcpy(&wd->dai_drv, &au1xpsc_i2s_dai_template, + sizeof(struct snd_soc_dai_driver)); + wd->dai_drv.name = dev_name(&pdev->dev); + + platform_set_drvdata(pdev, wd); + + ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); if (ret) goto out1; /* finally add the DMA device for this PSC */ wd->dmapd = au1xpsc_pcm_add(pdev); - if (wd->dmapd) { - platform_set_drvdata(pdev, wd); - au1xpsc_i2s_workdata = wd; + if (wd->dmapd) return 0; - } snd_soc_unregister_dai(&pdev->dev); out1: @@ -376,8 +365,6 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) release_mem_region(r->start, resource_size(r)); kfree(wd); - au1xpsc_i2s_workdata = NULL; /* MDEV */ - return 0; } @@ -427,7 +414,7 @@ static struct dev_pm_ops au1xpsci2s_pmops = { static struct platform_driver au1xpsc_i2s_driver = { .driver = { - .name = "au1xpsc", + .name = "au1xpsc-i2s", .owner = THIS_MODULE, .pm = AU1XPSCI2S_PMOPS, }, @@ -437,7 +424,6 @@ static struct platform_driver au1xpsc_i2s_driver = { static int __init au1xpsc_i2s_load(void) { - au1xpsc_i2s_workdata = NULL; return platform_driver_register(&au1xpsc_i2s_driver); } diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index f281443..b30eadd 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -8,16 +8,11 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ #ifndef _AU1X_PCM_H #define _AU1X_PCM_H -extern struct snd_ac97_bus_ops soc_ac97_ops; - /* DBDMA helpers */ extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev); extern void au1xpsc_pcm_destroy(struct platform_device *dmapd); @@ -28,6 +23,8 @@ struct au1xpsc_audio_data { unsigned long cfg; unsigned long rate; + struct snd_soc_dai_driver dai_drv; + unsigned long pm[2]; struct mutex lock; struct platform_device *dmapd; -- 1.7.2 -- 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

1 0

[alsa-devel] [PATCH v2] ASoC: multi-component: fix au1x platform
by Manuel Lauss 19 Aug '10

19 Aug '10

This patch fixes up the au1x audio platform to compile, and prepares the I2S and PCM modules for multiple card operation (no more globals for dma and i2s). The AC97 part is limited to one instance only for now due to other issues (see code). Signed-off-by: Manuel Lauss <manuel.lauss(a)googlemail.com> --- Run-tested on DB1200, both I2S and AC97 playback and recording work. AC97 spits out this kobject warning, which is caused by the "device_register()" in soc-core.c::soc_ac97_dev_register(): asoc: ac97-hifi <-> au1xpsc-ac97.1 mapping ok kobject (8fdc59b0): tried to init an initialized object, something is seriously wrong. Call Trace: [<804959c4>] dump_stack+0x8/0x34 [<802a30e4>] kobject_init+0x50/0xcc [<802ec27c>] device_initialize+0x2c/0x70 [<802ecb64>] device_register+0x14/0x28 [<8039e058>] snd_soc_instantiate_cards+0xa00/0xb10 [<8039e27c>] soc_probe+0x114/0x154 [<802ef0fc>] driver_probe_device+0xe4/0x1a0 [<802ee404>] bus_for_each_drv+0x60/0xb0 [<802ef370>] device_attach+0x74/0xa8 [<802ee1e8>] bus_probe_device+0x30/0x54 [<802ec9a0>] device_add+0x384/0x534 [<802f09a4>] platform_device_add+0x15c/0x1c8 [<805ccb14>] db1200_audio_load+0x70/0x9c [<801004fc>] do_one_initcall+0xfc/0x1e0 [<805ba32c>] kernel_init+0xc8/0x168 [<80105904>] kernel_thread_helper+0x10/0x18 ALSA device list: #0: DB1200_AC97 arch/mips/alchemy/devboards/db1200/platform.c | 10 ++- sound/soc/au1x/db1200.c | 18 ++-- sound/soc/au1x/dbdma2.c | 130 ++++++++----------------- sound/soc/au1x/psc-ac97.c | 56 ++++++----- sound/soc/au1x/psc-i2s.c | 42 +++------ sound/soc/au1x/psc.h | 7 +- 6 files changed, 109 insertions(+), 154 deletions(-) diff --git a/arch/mips/alchemy/devboards/db1200/platform.c b/arch/mips/alchemy/devboards/db1200/platform.c index 3fa34c3..f985373 100644 --- a/arch/mips/alchemy/devboards/db1200/platform.c +++ b/arch/mips/alchemy/devboards/db1200/platform.c @@ -429,8 +429,14 @@ static struct platform_device db1200_audio_dev = { .resource = au1200_psc1_res, }; +static struct platform_device db1200_ac97_codec = { + .name = "ac97-codec", + .id = 1, /* on PSC1 */ +}; + static struct platform_device *db1200_devs[] __initdata = { NULL, /* PSC0, selected by S6.8 */ + &db1200_ac97_codec, &db1200_ide_dev, &db1200_eth_dev, &db1200_rtc_dev, @@ -494,11 +500,11 @@ static int __init db1200_dev_init(void) sw &= BCSR_SWITCHES_DIP_8 | BCSR_SWITCHES_DIP_7; if (sw == BCSR_SWITCHES_DIP_8) { bcsr_mod(BCSR_RESETS, 0, BCSR_RESETS_PSC1MUX); - db1200_audio_dev.name = "au1xpsc_i2s"; + db1200_audio_dev.name = "au1xpsc-i2s"; printk(KERN_INFO " S6.7 ON : PSC1 mode I2S\n"); } else { bcsr_mod(BCSR_RESETS, BCSR_RESETS_PSC1MUX, 0); - db1200_audio_dev.name = "au1xpsc_ac97"; + db1200_audio_dev.name = "au1xpsc-ac97"; printk(KERN_INFO " S6.7 OFF: PSC1 mode AC97\n"); } diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 8780c90..2065d03 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -26,11 +26,11 @@ static struct snd_soc_dai_link db1200_ac97_dai = { .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai_name = "au1xpsc-ac97", + .stream_name = "AC97 Playback", + .cpu_dai_name = "au1xpsc-ac97.1", .codec_dai_name = "ac97-hifi", - .platform_name = "au1xpsc-pcm-audio", - .codec_name = "ac97-codec", + .platform_name = "au1xpsc-pcm-audio.1", + .codec_name = "ac97-codec.1", }; static struct snd_soc_card db1200_ac97_machine = { @@ -74,11 +74,11 @@ static struct snd_soc_ops db1200_i2s_wm8731_ops = { static struct snd_soc_dai_link db1200_i2s_dai = { .name = "WM8731", - .stream_name = "WM8731 PCM", - .cpu_dai_name = "au1xpsc", - .codec_dai_name = "wm8731-hifi" - .platform_name = "au1xpsc-pcm-audio", - .codec_name = "wm8731-codec.0-001a", + .stream_name = "WM8731", + .cpu_dai_name = "au1xpsc-i2s.1", + .codec_dai_name = "wm8731-hifi", + .platform_name = "au1xpsc-pcm-audio.1", + .codec_name = "wm8731-codec.0-001b", .ops = &db1200_i2s_wm8731_ops, }; diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 00fdb9c..383c241 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -10,9 +10,6 @@ * * DMA glue for Au1x-PSC audio. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ @@ -61,9 +58,6 @@ struct au1xpsc_audio_dmadata { int msbits; }; -/* instance data. There can be only one, MacLeod!!!! */ -static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2]; - /* * These settings are somewhat okay, at least on my machine audio plays * almost skip-free. Especially the 64kB buffer seems to help a LOT. @@ -199,19 +193,26 @@ out: return 0; } -static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, +static inline struct au1xpsc_audio_dmadata *to_dmadata(struct snd_pcm_substream *ss) +{ + struct snd_soc_pcm_runtime *rtd = ss->private_data; + struct au1xpsc_audio_dmadata *pcd = + snd_soc_platform_get_drvdata(rtd->platform); + return &pcd[SUBSTREAM_TYPE(ss)]; +} + +static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *ss, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct au1xpsc_audio_dmadata *pcd; + struct snd_pcm_runtime *runtime = ss->runtime; + struct au1xpsc_audio_dmadata *pcd = to_dmadata(ss); int stype, ret; - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + ret = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(params)); if (ret < 0) goto out; - stype = SUBSTREAM_TYPE(substream); - pcd = au1xpsc_audio_pcmdma[stype]; + stype = SUBSTREAM_TYPE(ss); DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " "runtime->min_align %d\n", @@ -222,13 +223,13 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, DBG("bits %d frags %d frag_bytes %d is_rx %d\n", params->msbits, params_periods(params), params_period_bytes(params), stype); - ret = au1x_pcm_dbdma_realloc(pcd, stype, params->msbits); + ret = au1x_pcm_dbdma_realloc(pcd, SUBSTREAM_TYPE(ss), params->msbits); if (ret) { MSG("DDMA channel (re)alloc failed!\n"); goto out; } - pcd->substream = substream; + pcd->substream = ss; pcd->period_bytes = params_period_bytes(params); pcd->periods = params_periods(params); pcd->dma_area_s = pcd->dma_area = runtime->dma_addr; @@ -249,8 +250,7 @@ static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream) static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) { - struct au1xpsc_audio_dmadata *pcd = - au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]; + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); au1xxx_dbdma_reset(pcd->ddma_chan); @@ -267,16 +267,16 @@ static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream) static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - u32 c = au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->ddma_chan; + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - au1xxx_dbdma_start(c); + au1xxx_dbdma_start(pcd->ddma_chan); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - au1xxx_dbdma_stop(c); + au1xxx_dbdma_stop(pcd->ddma_chan); break; default: return -EINVAL; @@ -287,8 +287,9 @@ static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static snd_pcm_uframes_t au1xpsc_pcm_pointer(struct snd_pcm_substream *substream) { - return bytes_to_frames(substream->runtime, - au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->pos); + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); + + return bytes_to_frames(substream->runtime, pcd->pos); } static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) @@ -299,7 +300,9 @@ static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) static int au1xpsc_pcm_close(struct snd_pcm_substream *substream) { - au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]); + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); + + au1x_pcm_dbdma_free(pcd); return 0; } @@ -329,83 +332,55 @@ static int au1xpsc_pcm_new(struct snd_card *card, return 0; } -static int au1xpsc_pcm_probe(struct snd_soc_platform *platform) -{ - if (!au1xpsc_audio_pcmdma[PCM_TX] || !au1xpsc_audio_pcmdma[PCM_RX]) - return -ENODEV; - - return 0; -} - /* au1xpsc audio platform */ struct snd_soc_platform_driver au1xpsc_soc_platform = { - .probe = au1xpsc_pcm_probe, .ops = &au1xpsc_pcm_ops, .pcm_new = au1xpsc_pcm_new, .pcm_free = au1xpsc_pcm_free_dma_buffers, }; -EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev) { + struct au1xpsc_audio_dmadata *dmadata; struct resource *r; int ret; - if (au1xpsc_audio_pcmdma[PCM_TX] || au1xpsc_audio_pcmdma[PCM_RX]) - return -EBUSY; - - /* TX DMA */ - au1xpsc_audio_pcmdma[PCM_TX] - = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); - if (!au1xpsc_audio_pcmdma[PCM_TX]) + dmadata = kzalloc(2 * sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); + if (!dmadata) return -ENOMEM; r = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!r) { ret = -ENODEV; - goto out1; + goto out2; } - (au1xpsc_audio_pcmdma[PCM_TX])->ddma_id = r->start; + dmadata[PCM_TX].ddma_id = r->start; /* RX DMA */ - au1xpsc_audio_pcmdma[PCM_RX] - = kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL); - if (!au1xpsc_audio_pcmdma[PCM_RX]) - return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!r) { ret = -ENODEV; goto out2; } - (au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start; + dmadata[PCM_RX].ddma_id = r->start; + + platform_set_drvdata(pdev, dmadata); ret = snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform); if (!ret) return ret; out2: - kfree(au1xpsc_audio_pcmdma[PCM_RX]); - au1xpsc_audio_pcmdma[PCM_RX] = NULL; -out1: - kfree(au1xpsc_audio_pcmdma[PCM_TX]); - au1xpsc_audio_pcmdma[PCM_TX] = NULL; + kfree(dmadata); return ret; } static int __devexit au1xpsc_pcm_drvremove(struct platform_device *pdev) { - int i; + struct au1xpsc_audio_dmadata *dmadata = platform_get_drvdata(pdev); snd_soc_unregister_platform(&pdev->dev); - - for (i = 0; i < 2; i++) { - if (au1xpsc_audio_pcmdma[i]) { - au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]); - kfree(au1xpsc_audio_pcmdma[i]); - au1xpsc_audio_pcmdma[i] = NULL; - } - } + kfree(dmadata); return 0; } @@ -421,8 +396,6 @@ static struct platform_driver au1xpsc_pcm_driver = { static int __init au1xpsc_audio_dbdma_load(void) { - au1xpsc_audio_pcmdma[PCM_TX] = NULL; - au1xpsc_audio_pcmdma[PCM_RX] = NULL; return platform_driver_register(&au1xpsc_pcm_driver); } @@ -435,37 +408,21 @@ module_init(au1xpsc_audio_dbdma_load); module_exit(au1xpsc_audio_dbdma_unload); +/* psc-ac97/psc-i2s call this function to create a pcm platform device with + * their DMA channel information. + */ struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev) { - struct resource *res, *r; struct platform_device *pd; - int id[2]; int ret; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) - return NULL; - id[0] = r->start; - - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) - return NULL; - id[1] = r->start; - - res = kzalloc(sizeof(struct resource) * 2, GFP_KERNEL); - if (!res) - return NULL; - - res[0].start = res[0].end = id[0]; - res[1].start = res[1].end = id[1]; - res[0].flags = res[1].flags = IORESOURCE_DMA; - - pd = platform_device_alloc("au1xpsc-pcm", -1); + /* pass on the resources from the PSC, and match its ID */ + pd = platform_device_alloc("au1xpsc-pcm-audio", pdev->id); if (!pd) goto out; - pd->resource = res; - pd->num_resources = 2; + pd->resource = pdev->resource; + pd->num_resources = pdev->num_resources; ret = platform_device_add(pd); if (!ret) @@ -473,7 +430,6 @@ struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev) platform_device_put(pd); out: - kfree(res); return NULL; } EXPORT_SYMBOL_GPL(au1xpsc_pcm_add); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 6a9516c..415b5aa 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -10,9 +10,6 @@ * * Au1xxx-PSC AC97 glue. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ #include <linux/init.h> @@ -56,12 +53,28 @@ /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; +#if 0 + +/* this could theoretically work, but ac97->bus->card can be NULL when + * snd_ac97_mixer() is called; I don't know if the rest further down the chain + * are always valid either. + */ +static inline struct au1xpsc_audio_data *ac97_to_pscdata(struct snd_ac97 *x) +{ + struct snd_soc_card *c = x->bus->card->private_data; + return snd_soc_dai_get_drvdata(c->rtd->cpu_dai); +} + +#else + +#define ac97_to_pscdata(x) au1xpsc_ac97_workdata +#endif + /* AC97 controller reads codec register */ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); unsigned short retry, tmo; unsigned long data; @@ -102,8 +115,7 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); unsigned int tmo, retry; au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); @@ -134,8 +146,7 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, /* AC97 controller asserts a warm reset */ static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); au_writel(PSC_AC97RST_SNC, AC97_RST(pscdata)); au_sync(); @@ -146,8 +157,7 @@ static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97) static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97); int i; /* disable PSC during cold reset */ @@ -202,8 +212,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = dev_get_drvdata(dai->dev); unsigned long r, ro, stat; int chans, t, stype = SUBSTREAM_TYPE(substream); @@ -283,8 +292,7 @@ out: static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - /* FIXME */ - struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; + struct au1xpsc_audio_data *pscdata = dev_get_drvdata(dai->dev); int ret, stype = SUBSTREAM_TYPE(substream); ret = 0; @@ -325,7 +333,7 @@ static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { .hw_params = au1xpsc_ac97_hw_params, }; -struct snd_soc_dai_driver au1xpsc_ac97_dai = { +static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { .ac97_control = 1, .probe = au1xpsc_ac97_probe, .playback = { @@ -342,7 +350,6 @@ struct snd_soc_dai_driver au1xpsc_ac97_dai = { }, .ops = &au1xpsc_ac97_dai_ops, }; -EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) { @@ -351,9 +358,6 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) unsigned long sel; struct au1xpsc_audio_data *wd; - if (au1xpsc_ac97_workdata) - return -EBUSY; - wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); if (!wd) return -ENOMEM; @@ -387,14 +391,20 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(wd)); au_sync(); - ret = snd_soc_register_dai(&pdev->dev, &au1xpsc_ac97_dai); + /* name the DAI like this device instance ("au1xpsc-ac97.PSCINDEX") */ + memcpy(&wd->dai_drv, &au1xpsc_ac97_dai_template, + sizeof(struct snd_soc_dai_driver)); + wd->dai_drv.name = dev_name(&pdev->dev); + + platform_set_drvdata(pdev, wd); + + ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); if (ret) goto out1; wd->dmapd = au1xpsc_pcm_add(pdev); if (wd->dmapd) { - platform_set_drvdata(pdev, wd); - au1xpsc_ac97_workdata = wd; /* MDEV */ + au1xpsc_ac97_workdata = wd; return 0; } diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 94e560a..28c0ec4 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -10,9 +10,6 @@ * * Au1xxx-PSC I2S glue. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. * NOTE: so far only PSC slave mode (bit- and frameclock) is supported. */ @@ -54,13 +51,10 @@ ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) -/* instance data. There can be only one, MacLeod!!!! */ -static struct au1xpsc_audio_data *au1xpsc_i2s_workdata; - static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = dev_get_drvdata(cpu_dai->dev); unsigned long ct; int ret; @@ -120,7 +114,7 @@ static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = dev_get_drvdata(dai->dev); int cfgbits; unsigned long stat; @@ -245,7 +239,7 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; + struct au1xpsc_audio_data *pscdata = dev_get_drvdata(dai->dev); int ret, stype = SUBSTREAM_TYPE(substream); switch (cmd) { @@ -263,19 +257,13 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int au1xpsc_i2s_probe(struct snd_soc_dai *dai) -{ - return au1xpsc_i2s_workdata ? 0 : -ENODEV; -} - static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { .trigger = au1xpsc_i2s_trigger, .hw_params = au1xpsc_i2s_hw_params, .set_fmt = au1xpsc_i2s_set_fmt, }; -static struct snd_soc_dai_driver au1xpsc_i2s_dai = { - .probe = au1xpsc_i2s_probe, +static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { .playback = { .rates = AU1XPSC_I2S_RATES, .formats = AU1XPSC_I2S_FMTS, @@ -298,9 +286,6 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) int ret; struct au1xpsc_audio_data *wd; - if (au1xpsc_i2s_workdata) - return -EBUSY; - wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); if (!wd) return -ENOMEM; @@ -337,17 +322,21 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) * time out. */ - ret = snd_soc_register_dai(&pdev->dev, &au1xpsc_i2s_dai); + /* name the DAI like this device instance ("au1xpsc-i2s.PSCINDEX") */ + memcpy(&wd->dai_drv, &au1xpsc_i2s_dai_template, + sizeof(struct snd_soc_dai_driver)); + wd->dai_drv.name = dev_name(&pdev->dev); + + platform_set_drvdata(pdev, wd); + + ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); if (ret) goto out1; /* finally add the DMA device for this PSC */ wd->dmapd = au1xpsc_pcm_add(pdev); - if (wd->dmapd) { - platform_set_drvdata(pdev, wd); - au1xpsc_i2s_workdata = wd; + if (wd->dmapd) return 0; - } snd_soc_unregister_dai(&pdev->dev); out1: @@ -376,8 +365,6 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) release_mem_region(r->start, resource_size(r)); kfree(wd); - au1xpsc_i2s_workdata = NULL; /* MDEV */ - return 0; } @@ -427,7 +414,7 @@ static struct dev_pm_ops au1xpsci2s_pmops = { static struct platform_driver au1xpsc_i2s_driver = { .driver = { - .name = "au1xpsc", + .name = "au1xpsc-i2s", .owner = THIS_MODULE, .pm = AU1XPSCI2S_PMOPS, }, @@ -437,7 +424,6 @@ static struct platform_driver au1xpsc_i2s_driver = { static int __init au1xpsc_i2s_load(void) { - au1xpsc_i2s_workdata = NULL; return platform_driver_register(&au1xpsc_i2s_driver); } diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index f281443..b30eadd 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -8,16 +8,11 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. * - * NOTE: all of these drivers can only work with a SINGLE instance - * of a PSC. Multiple independent audio devices are impossible - * with ASoC v1. */ #ifndef _AU1X_PCM_H #define _AU1X_PCM_H -extern struct snd_ac97_bus_ops soc_ac97_ops; - /* DBDMA helpers */ extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev); extern void au1xpsc_pcm_destroy(struct platform_device *dmapd); @@ -28,6 +23,8 @@ struct au1xpsc_audio_data { unsigned long cfg; unsigned long rate; + struct snd_soc_dai_driver dai_drv; + unsigned long pm[2]; struct mutex lock; struct platform_device *dmapd; -- 1.7.2 -- 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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[PATCH 1/2] Show processing coefficients in codec proc file
by David Henningsson 19 Aug '10

19 Aug '10

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_proc.c | 16 ++++++++++++++-- 1 files changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index f025200..88e40f0 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -467,11 +467,23 @@ static void print_unsol_cap(struct snd_info_buffer *buffer, static void print_proc_caps(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { + int i, val; unsigned int proc_caps = snd_hda_param_read(codec, nid, AC_PAR_PROC_CAP); + unsigned int ncoeff = (proc_caps & AC_PCAP_NUM_COEF) >> + AC_PCAP_NUM_COEF_SHIFT; + snd_iprintf(buffer, " Processing caps: benign=%d, ncoeff=%d\n", - proc_caps & AC_PCAP_BENIGN, - (proc_caps & AC_PCAP_NUM_COEF) >> AC_PCAP_NUM_COEF_SHIFT); + proc_caps & AC_PCAP_BENIGN, ncoeff); + for (i = 0; i < ncoeff; i++) { + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, + i); + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PROC_COEF, 0); + snd_iprintf(buffer, " Processing coef %d: 0x%x\n", i, + val); + } + } static void print_conn_list(struct snd_info_buffer *buffer, -- 1.7.0.4 --------------010409060503090209020409--

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