[alsa-devel] [PATCH V2 0/3] ASoC: au1x: update PSC AC97/I2S code
The following two patches depend on patchset "ASoC for Alchemy Au1000/1500/1100" to apply and work. Patch #1 moves the DBDMA (PCM) device registration from the AC97/I2S drivers into the board code. Patch #2 changes use of "soc-audio" platform device int db1200 machine code to the new way. Patch #3 removes the use of custom PCM_TX/RX constants in favour of the established SNDRV_PCM_SUBTREAM_PLAYBACK/CAPTURE.
Changes since V1: - added patch #3 - tidied patch #1 a bit more.
Run-tested on DB1200, DB1300 and DB1550 boards.
Manuel Lauss (3): ASoC: au1x: remove automatic DMA device registration from PSC drivers ASoC: au1x: update db1200 machine to the new way of things ASoC: au1x: use substream stream info directly
arch/mips/alchemy/devboards/db1200/platform.c | 16 +++++ sound/soc/au1x/db1200.c | 64 ++++++++++++------ sound/soc/au1x/dbdma2.c | 91 +++++-------------------- sound/soc/au1x/psc-ac97.c | 48 +++++++++----- sound/soc/au1x/psc-i2s.c | 42 +++++++---- sound/soc/au1x/psc.h | 11 --- 6 files changed, 133 insertions(+), 139 deletions(-)
The PSC audio drivers (psc-ac97/psc-i2s) register the DMA platform_device on their own. This is frowned upon, from now on board code must register a simple pcm dma platform device for each PSC with sound duties.
Signed-off-by: Manuel Lauss manuel.lauss@googlemail.com --- V2: remove now unnecessary snd_soc_unregister_dai() calls in probe() callback error path of the ac97/i2s glues.
arch/mips/alchemy/devboards/db1200/platform.c | 6 ++ sound/soc/au1x/dbdma2.c | 83 ++++--------------------- sound/soc/au1x/psc-ac97.c | 34 +++++++--- sound/soc/au1x/psc-i2s.c | 32 +++++++--- sound/soc/au1x/psc.h | 5 -- 5 files changed, 64 insertions(+), 96 deletions(-)
diff --git a/arch/mips/alchemy/devboards/db1200/platform.c b/arch/mips/alchemy/devboards/db1200/platform.c index fbb5593..cfb71ae 100644 --- a/arch/mips/alchemy/devboards/db1200/platform.c +++ b/arch/mips/alchemy/devboards/db1200/platform.c @@ -434,12 +434,18 @@ static struct platform_device db1200_stac_dev = { .id = 1, /* on PSC1 */ };
+static struct platform_device db1200_audiodma_dev = { + .name = "au1xpsc-pcm", + .id = 1, /* PSC ID */ +}; + static struct platform_device *db1200_devs[] __initdata = { NULL, /* PSC0, selected by S6.8 */ &db1200_ide_dev, &db1200_eth_dev, &db1200_rtc_dev, &db1200_nand_dev, + &db1200_audiodma_dev, &db1200_audio_dev, &db1200_stac_dev, }; diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 20bb53a..fd5378f 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -293,6 +293,16 @@ au1xpsc_pcm_pointer(struct snd_pcm_substream *substream)
static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) { + struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int stype = SUBSTREAM_TYPE(substream), *dmaids; + + dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dmaids) + return -ENODEV; /* whoa, has ordering changed? */ + + pcd->ddma_id = dmaids[stype]; + snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware); return 0; } @@ -340,36 +350,18 @@ struct snd_soc_platform_driver au1xpsc_soc_platform = { static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev) { struct au1xpsc_audio_dmadata *dmadata; - struct resource *r; int ret;
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; - } - dmadata[PCM_TX].ddma_id = r->start; - - /* RX DMA */ - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) { - ret = -ENODEV; - goto out1; - } - 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; + if (ret) + kfree(dmadata);
-out1: - kfree(dmadata); return ret; }
@@ -405,57 +397,6 @@ static void __exit au1xpsc_audio_dbdma_unload(void) module_init(au1xpsc_audio_dbdma_load); module_exit(au1xpsc_audio_dbdma_unload);
- -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", pdev->id); - if (!pd) - goto out; - - pd->resource = res; - pd->num_resources = 2; - - ret = platform_device_add(pd); - if (!ret) - return pd; - - platform_device_put(pd); -out: - kfree(res); - return NULL; -} -EXPORT_SYMBOL_GPL(au1xpsc_pcm_add); - -void au1xpsc_pcm_destroy(struct platform_device *dmapd) -{ - if (dmapd) - platform_device_unregister(dmapd); -} -EXPORT_SYMBOL_GPL(au1xpsc_pcm_destroy); - MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index d0db66f..44296ab 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -324,12 +324,21 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, return ret; }
+static int au1xpsc_ac97_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]); + return 0; +} + static int au1xpsc_ac97_probe(struct snd_soc_dai *dai) { return au1xpsc_ac97_workdata ? 0 : -ENODEV; }
static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .startup = au1xpsc_ac97_startup, .trigger = au1xpsc_ac97_trigger, .hw_params = au1xpsc_ac97_hw_params, }; @@ -379,6 +388,16 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) if (!wd->mmio) goto out1;
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) + goto out2; + wd->dmaids[PCM_TX] = r->start; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) + goto out2; + wd->dmaids[PCM_RX] = r->start; + /* configuration: max dma trigger threshold, enable ac97 */ wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | PSC_AC97CFG_DE_ENABLE; @@ -401,15 +420,13 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv); if (ret) - goto out1; + goto out2;
- wd->dmapd = au1xpsc_pcm_add(pdev); - if (wd->dmapd) { - au1xpsc_ac97_workdata = wd; - return 0; - } + au1xpsc_ac97_workdata = wd; + return 0;
- snd_soc_unregister_dai(&pdev->dev); +out2: + iounmap(wd->mmio); out1: release_mem_region(r->start, resource_size(r)); out0: @@ -422,9 +439,6 @@ static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev) struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (wd->dmapd) - au1xpsc_pcm_destroy(wd->dmapd); - snd_soc_unregister_dai(&pdev->dev);
/* disable PSC completely */ diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index fca0912..1b7ab5d 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -257,7 +257,16 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; }
+static int au1xpsc_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]); + return 0; +} + static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { + .startup = au1xpsc_i2s_startup, .trigger = au1xpsc_i2s_trigger, .hw_params = au1xpsc_i2s_hw_params, .set_fmt = au1xpsc_i2s_set_fmt, @@ -304,6 +313,16 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) if (!wd->mmio) goto out1;
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) + goto out2; + wd->dmaids[PCM_TX] = r->start; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) + goto out2; + wd->dmaids[PCM_RX] = r->start; + /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) */ @@ -330,15 +349,11 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) 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) + if (!ret) return 0;
- snd_soc_unregister_dai(&pdev->dev); +out2: + iounmap(wd->mmio); out1: release_mem_region(r->start, resource_size(r)); out0: @@ -351,9 +366,6 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- if (wd->dmapd) - au1xpsc_pcm_destroy(wd->dmapd); - snd_soc_unregister_dai(&pdev->dev);
au_writel(0, I2S_CFG(wd)); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index c59b9e5..1b21c4f 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -19,10 +19,6 @@ #define SUBSTREAM_TYPE(substream) \ ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX)
-/* PSC/DBDMA helpers */ -extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev); -extern void au1xpsc_pcm_destroy(struct platform_device *dmapd); - struct au1xpsc_audio_data { void __iomem *mmio;
@@ -34,7 +30,6 @@ struct au1xpsc_audio_data { unsigned long pm[2]; struct mutex lock; int dmaids[2]; - struct platform_device *dmapd; };
/* easy access macros */
The use of the "soc-audio" platform device is no longer en vogue, update the code to the newer, simpler way of doing things.
Signed-off-by: Manuel Lauss manuel.lauss@googlemail.com --- V2: no changes.
arch/mips/alchemy/devboards/db1200/platform.c | 10 ++++ sound/soc/au1x/db1200.c | 64 +++++++++++++++++-------- 2 files changed, 53 insertions(+), 21 deletions(-)
diff --git a/arch/mips/alchemy/devboards/db1200/platform.c b/arch/mips/alchemy/devboards/db1200/platform.c index cfb71ae..dda090b 100644 --- a/arch/mips/alchemy/devboards/db1200/platform.c +++ b/arch/mips/alchemy/devboards/db1200/platform.c @@ -422,6 +422,7 @@ static struct resource au1200_psc1_res[] = { }, };
+/* AC97 or I2S device */ static struct platform_device db1200_audio_dev = { /* name assigned later based on switch setting */ .id = 1, /* PSC ID */ @@ -429,6 +430,12 @@ static struct platform_device db1200_audio_dev = { .resource = au1200_psc1_res, };
+/* DB1200 ASoC card device */ +static struct platform_device db1200_sound_dev = { + /* name assigned later based on switch setting */ + .id = 1, /* PSC ID */ +}; + static struct platform_device db1200_stac_dev = { .name = "ac97-codec", .id = 1, /* on PSC1 */ @@ -448,6 +455,7 @@ static struct platform_device *db1200_devs[] __initdata = { &db1200_audiodma_dev, &db1200_audio_dev, &db1200_stac_dev, + &db1200_sound_dev, };
static int __init db1200_dev_init(void) @@ -507,10 +515,12 @@ static int __init db1200_dev_init(void) if (sw == BCSR_SWITCHES_DIP_8) { bcsr_mod(BCSR_RESETS, 0, BCSR_RESETS_PSC1MUX); db1200_audio_dev.name = "au1xpsc_i2s"; + db1200_sound_dev.name = "db1200-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_sound_dev.name = "db1200-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 1d3e258..289312c 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -1,7 +1,7 @@ /* * DB1200 ASoC audio fabric support code. * - * (c) 2008-9 Manuel Lauss manuel.lauss@gmail.com + * (c) 2008-2011 Manuel Lauss manuel.lauss@googlemail.com * */
@@ -21,6 +21,17 @@ #include "../codecs/wm8731.h" #include "psc.h"
+static struct platform_device_id db1200_pids[] = { + { + .name = "db1200-ac97", + .driver_data = 0, + }, { + .name = "db1200-i2s", + .driver_data = 1, + }, + {}, +}; + /*------------------------- AC97 PART ---------------------------*/
static struct snd_soc_dai_link db1200_ac97_dai = { @@ -89,36 +100,47 @@ static struct snd_soc_card db1200_i2s_machine = {
/*------------------------- COMMON PART ---------------------------*/
-static struct platform_device *db1200_asoc_dev; +static struct snd_soc_card *db1200_cards[] __devinitdata = { + &db1200_ac97_machine, + &db1200_i2s_machine, +};
-static int __init db1200_audio_load(void) +static int __devinit db1200_audio_probe(struct platform_device *pdev) { - int ret; + const struct platform_device_id *pid = platform_get_device_id(pdev); + struct snd_soc_card *card;
- ret = -ENOMEM; - db1200_asoc_dev = platform_device_alloc("soc-audio", 1); /* PSC1 */ - if (!db1200_asoc_dev) - goto out; + card = db1200_cards[pid->driver_data]; + card->dev = &pdev->dev; + return snd_soc_register_card(card); +}
- /* DB1200 board setup set PSC1MUX to preferred audio device */ - if (bcsr_read(BCSR_RESETS) & BCSR_RESETS_PSC1MUX) - platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_machine); - else - platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_machine); +static int __devexit db1200_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + snd_soc_unregister_card(card); + return 0; +}
- ret = platform_device_add(db1200_asoc_dev); +static struct platform_driver db1200_audio_driver = { + .driver = { + .name = "db1200-ac97", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .id_table = db1200_pids, + .probe = db1200_audio_probe, + .remove = __devexit_p(db1200_audio_remove), +};
- if (ret) { - platform_device_put(db1200_asoc_dev); - db1200_asoc_dev = NULL; - } -out: - return ret; +static int __init db1200_audio_load(void) +{ + return platform_driver_register(&db1200_audio_driver); }
static void __exit db1200_audio_unload(void) { - platform_device_unregister(db1200_asoc_dev); + platform_driver_unregister(&db1200_audio_driver); }
module_init(db1200_audio_load);
PCM_TX/RX are the same as SNDRV_PCM_STREAM_PLAYBACK/CAPTURE. Use them directly.
Signed-off-by: Manuel Lauss manuel.lauss@googlemail.com --- V2: new patch based on feedback from Liam Girdwood.
sound/soc/au1x/dbdma2.c | 10 +++++----- sound/soc/au1x/psc-ac97.c | 18 +++++++++--------- sound/soc/au1x/psc-i2s.c | 14 +++++++------- sound/soc/au1x/psc.h | 6 ------ 4 files changed, 21 insertions(+), 27 deletions(-)
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index fd5378f..d7d04e2 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -169,7 +169,7 @@ static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd,
au1x_pcm_dbdma_free(pcd);
- if (stype == PCM_RX) + if (stype == SNDRV_PCM_STREAM_CAPTURE) pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id, DSCR_CMD0_ALWAYS, au1x_pcm_dmarx_cb, (void *)pcd); @@ -198,7 +198,7 @@ static inline struct au1xpsc_audio_dmadata *to_dmadata(struct snd_pcm_substream 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)]; + return &pcd[ss->stream]; }
static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, @@ -212,7 +212,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, if (ret < 0) goto out;
- stype = SUBSTREAM_TYPE(substream); + stype = substream->stream; pcd = to_dmadata(substream);
DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " @@ -255,7 +255,7 @@ static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream)
au1xxx_dbdma_reset(pcd->ddma_chan);
- if (SUBSTREAM_TYPE(substream) == PCM_RX) { + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { au1x_pcm_queue_rx(pcd); au1x_pcm_queue_rx(pcd); } else { @@ -295,7 +295,7 @@ static int au1xpsc_pcm_open(struct snd_pcm_substream *substream) { struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream); struct snd_soc_pcm_runtime *rtd = substream->private_data; - int stype = SUBSTREAM_TYPE(substream), *dmaids; + int stype = substream->stream, *dmaids;
dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if (!dmaids) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 44296ab..172eefd 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -41,14 +41,14 @@ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE)
#define AC97PCR_START(stype) \ - ((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TS : PSC_AC97PCR_RS) #define AC97PCR_STOP(stype) \ - ((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TP : PSC_AC97PCR_RP) #define AC97PCR_CLRFIFO(stype) \ - ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
#define AC97STAT_BUSY(stype) \ - ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
/* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -215,7 +215,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, { struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); unsigned long r, ro, stat; - int chans, t, stype = SUBSTREAM_TYPE(substream); + int chans, t, stype = substream->stream;
chans = params_channels(params);
@@ -235,7 +235,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_SET_LEN(params->msbits);
/* channels: enable slots for front L/R channel */ - if (stype == PCM_TX) { + if (stype == SNDRV_PCM_STREAM_PLAYBACK) { r &= ~PSC_AC97CFG_TXSLOT_MASK; r |= PSC_AC97CFG_TXSLOT_ENA(3); r |= PSC_AC97CFG_TXSLOT_ENA(4); @@ -294,7 +294,7 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); - int ret, stype = SUBSTREAM_TYPE(substream); + int ret, stype = substream->stream;
ret = 0;
@@ -391,12 +391,12 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) r = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!r) goto out2; - wd->dmaids[PCM_TX] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
r = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!r) goto out2; - wd->dmaids[PCM_RX] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
/* configuration: max dma trigger threshold, enable ac97 */ wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 1b7ab5d..7c5ae92 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -42,13 +42,13 @@ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
#define I2SSTAT_BUSY(stype) \ - ((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB) #define I2SPCR_START(stype) \ - ((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TS : PSC_I2SPCR_RS) #define I2SPCR_STOP(stype) \ - ((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TP : PSC_I2SPCR_RP) #define I2SPCR_CLRFIFO(stype) \ - ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) + ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, @@ -240,7 +240,7 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai); - int ret, stype = SUBSTREAM_TYPE(substream); + int ret, stype = substream->stream;
switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -316,12 +316,12 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) r = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!r) goto out2; - wd->dmaids[PCM_TX] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
r = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!r) goto out2; - wd->dmaids[PCM_RX] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
/* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 1b21c4f..b16b2e0 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -13,12 +13,6 @@ #ifndef _AU1X_PCM_H #define _AU1X_PCM_H
-#define PCM_TX 0 -#define PCM_RX 1 - -#define SUBSTREAM_TYPE(substream) \ - ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX) - struct au1xpsc_audio_data { void __iomem *mmio;
On 25/07/11 12:45, Manuel Lauss wrote:
The following two patches depend on patchset "ASoC for Alchemy Au1000/1500/1100" to apply and work. Patch #1 moves the DBDMA (PCM) device registration from the AC97/I2S drivers into the board code. Patch #2 changes use of "soc-audio" platform device int db1200 machine code to the new way. Patch #3 removes the use of custom PCM_TX/RX constants in favour of the established SNDRV_PCM_SUBTREAM_PLAYBACK/CAPTURE.
Changes since V1:
- added patch #3
- tidied patch #1 a bit more.
Run-tested on DB1200, DB1300 and DB1550 boards.
Manuel Lauss (3): ASoC: au1x: remove automatic DMA device registration from PSC drivers ASoC: au1x: update db1200 machine to the new way of things ASoC: au1x: use substream stream info directly
arch/mips/alchemy/devboards/db1200/platform.c | 16 +++++ sound/soc/au1x/db1200.c | 64 ++++++++++++------ sound/soc/au1x/dbdma2.c | 91 +++++-------------------- sound/soc/au1x/psc-ac97.c | 48 +++++++++----- sound/soc/au1x/psc-i2s.c | 42 +++++++---- sound/soc/au1x/psc.h | 11 --- 6 files changed, 133 insertions(+), 139 deletions(-)
Acked-by: Liam Girdwood lrg@ti.com -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo@vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html
On Mon, Jul 25, 2011 at 01:45:01PM +0200, Manuel Lauss wrote:
The following two patches depend on patchset "ASoC for Alchemy Au1000/1500/1100" to apply and work.
This all looks good but again I need the MIPS guys to be happy with it before applying. -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo@vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html
On Tue, Jul 26, 2011 at 03:43:42PM +0100, Mark Brown wrote:
On Mon, Jul 25, 2011 at 01:45:01PM +0200, Manuel Lauss wrote:
The following two patches depend on patchset "ASoC for Alchemy Au1000/1500/1100" to apply and work.
This all looks good but again I need the MIPS guys to be happy with it before applying.
Acked-by: Ralf Baechle ralf@linux-mips.org
Ralf -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo@vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html
On Mon, Jul 25, 2011 at 01:45:01PM +0200, Manuel Lauss wrote:
The following two patches depend on patchset "ASoC for Alchemy Au1000/1500/1100" to apply and work.
All applied, thanks. -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo@vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html
participants (4)
-
Liam Girdwood -
Manuel Lauss -
Mark Brown -
Ralf Baechle