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Diffstat (limited to 'drivers/devfreq/Makefile')

0 files changed, 0 insertions, 0 deletions

ted & GPIO_EXT_MIC_EN) gpio_free(pdata->gpio_ext_mic_en); if (machine->gpio_requested & GPIO_INT_MIC_EN) @@ -441,6 +443,11 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) gpio_free(pdata->gpio_hp_mute); if (machine->gpio_requested & GPIO_SPKR_EN) gpio_free(pdata->gpio_spkr_en); + machine->gpio_requested = 0; + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); kfree(machine); -- cgit v1.2.3-70-g09d2 From f99847a6909b95f857ee502ec98c372dcfd90b12 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 4 Aug 2011 16:44:44 -0600 Subject: ASoC: WM8903: Free IRQ on device removal Without this, request_irq on subsequent device initialization fails, and the codec cannot be used. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 43e3d760766..4ad8ebd290e 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2046,8 +2046,13 @@ static int wm8903_probe(struct snd_soc_codec *codec) /* power down chip */ static int wm8903_remove(struct snd_soc_codec *codec) { + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + wm8903_free_gpio(codec); wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (wm8903->irq) + free_irq(wm8903->irq, codec); + return 0; } -- cgit v1.2.3-70-g09d2 From 6678050442e90a4e9511a9ed14b9bdfc5e393323 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Aug 2011 17:36:48 +0900 Subject: ASoC: Fix binding of WM8750 on Jive The I2C address is misformatted and would never match. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/samsung/jive_wm8750.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 3b53ad54bc3..14eb6ea69e7 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -131,7 +131,7 @@ static struct snd_soc_dai_link jive_dai = { .cpu_dai_name = "s3c2412-i2s", .codec_dai_name = "wm8750-hifi", .platform_name = "samsung-audio", - .codec_name = "wm8750-codec.0-0x1a", + .codec_name = "wm8750-codec.0-001a", .init = jive_wm8750_init, .ops = &jive_ops, }; -- cgit v1.2.3-70-g09d2 From 40045a85df0ec4406fe611967ea9cf9fa668f493 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Aug 2011 18:32:09 +0900 Subject: ASoC: Fix SPI driver binding for WM8987 As we had no id_table only the driver name would be matched against meaning that WM8987 devices wouldn't be bound. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8750.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 38f38fddd19..65fe78aa375 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -778,11 +778,18 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id wm8750_spi_ids[] = { + { "wm8750", 0 }, + { "wm8987", 0 }, +}; +MODULE_DEVICE_TABLE(spi, wm8750_spi_id); + static struct spi_driver wm8750_spi_driver = { .driver = { .name = "wm8750-codec", .owner = THIS_MODULE, }, + .id_table = wm8750_spi_ids, .probe = wm8750_spi_probe, .remove = __devexit_p(wm8750_spi_remove), }; -- cgit v1.2.3-70-g09d2 From 371e7305c6c348d9e14a98fe337fadbd4106cfef Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 4 Aug 2011 10:54:17 +0900 Subject: ASoC: Fix warning in Speyside WM8962 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/samsung/speyside_wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 8ac42bf8209..0b9eb5f7ec4 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -37,7 +37,7 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, 44100 * 256, SND_SOC_CLOCK_IN); if (ret < 0) { - pr_err("Failed to set SYSCLK: %d\n"); + pr_err("Failed to set SYSCLK: %d\n", ret); return ret; } } -- cgit v1.2.3-70-g09d2 From 511d8cf0ab3d2e4ec3f3f672b06a83f17874b83b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Aug 2011 09:41:26 +0900 Subject: ASoC: Fix typo in wm8750 spi_ids Signed-off-by: Mark Brown Reported-by: Stephen Rothwell --- sound/soc/codecs/wm8750.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 65fe78aa375..e6f47f49357 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -782,7 +782,7 @@ static const struct spi_device_id wm8750_spi_ids[] = { { "wm8750", 0 }, { "wm8987", 0 }, }; -MODULE_DEVICE_TABLE(spi, wm8750_spi_id); +MODULE_DEVICE_TABLE(spi, wm8750_spi_ids); static struct spi_driver wm8750_spi_driver = { .driver = { -- cgit v1.2.3-70-g09d2 From c9c9e4e4252c9d554222906e4a843efd27c0ac96 Mon Sep 17 00:00:00 2001 From: Kazutomo Yoshii Date: Tue, 9 Aug 2011 23:39:13 -0500 Subject: ALSA: usb-audio - Add quirk for BOSS Micro BR-80 Signed-off-by: Kazutomo Yoshii Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 4d4f86552a2..a42e3ef3832 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1707,6 +1707,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x0130), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "MICRO BR-80", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { -- cgit v1.2.3-70-g09d2 From 96b635977984a88ecdb9cc76b8a54db7297f36e0 Mon Sep 17 00:00:00 2001 From: Wang Shaoyan Date: Wed, 10 Aug 2011 16:01:04 +0800 Subject: ALSA: hda - Add CONFIG_SND_HDA_POWER_SAVE to stac_vrefout_set() In commit 45eebda7, it add new function stac_vrefout_set, but it is only used in code between CONFIG_SND_HDA_POWER_SAVE macro, so add the macro to avoid such warning: sound/pci/hda/patch_sigmatel.c:676:12: warning: 'stac_vrefout_set' defined but not used Signed-off-by: Wang Shaoyan Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index aa376b59c00..5145b663ef6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -673,6 +673,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, return 0; } +#ifdef CONFIG_SND_HDA_POWER_SAVE static int stac_vrefout_set(struct hda_codec *codec, hda_nid_t nid, unsigned int new_vref) { @@ -696,6 +697,7 @@ static int stac_vrefout_set(struct hda_codec *codec, return 1; } +#endif static unsigned int stac92xx_vref_set(struct hda_codec *codec, hda_nid_t nid, unsigned int new_vref) -- cgit v1.2.3-70-g09d2 From a5a3973da8b52944bc5909852714e55771c31ce7 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Wed, 10 Aug 2011 11:49:04 +0200 Subject: ALSA: azt3328 - adjust error handling code to include debugging code snd_azf3328_dbgcallenter is called at the very beginning of the function, so it could be useful to call snd_azf3328_dbgcallleave at all exit points. Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/pci/azt3328.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index e4d76a270c9..579fc0dce12 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2625,16 +2625,19 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) int err; snd_azf3328_dbgcallenter(); - if (dev >= SNDRV_CARDS) - return -ENODEV; + if (dev >= SNDRV_CARDS) { + err = -ENODEV; + goto out; + } if (!enable[dev]) { dev++; - return -ENOENT; + err = -ENOENT; + goto out; } err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) - return err; + goto out; strcpy(card->driver, "AZF3328"); strcpy(card->shortname, "Aztech AZF3328 (PCI168)"); -- cgit v1.2.3-70-g09d2 From 15439bde3af7ff88459ea2b5520b77312e958df2 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 5 Aug 2011 13:49:52 +0200 Subject: ALSA: snd-usb-caiaq: Correct offset fields of outbound iso_frame_desc This fixes faulty outbount packets in case the inbound packets received from the hardware are fragmented and contain bogus input iso frames. The bug has been there for ages, but for some strange reasons, it was only triggered by newer machines in 64bit mode. Signed-off-by: Daniel Mack Reported-and-tested-by: William Light Reported-by: Pedro Ribeiro Cc: stable@kernel.org Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index d0d493ca28a..aa52b3e13bb 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -614,6 +614,7 @@ static void read_completed(struct urb *urb) struct snd_usb_caiaqdev *dev; struct urb *out; int frame, len, send_it = 0, outframe = 0; + size_t offset = 0; if (urb->status || !info) return; @@ -634,7 +635,8 @@ static void read_completed(struct urb *urb) len = urb->iso_frame_desc[outframe].actual_length; out->iso_frame_desc[outframe].length = len; out->iso_frame_desc[outframe].actual_length = 0; - out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame; + out->iso_frame_desc[outframe].offset = offset; + offset += len; if (len > 0) { spin_lock(&dev->spinlock); @@ -650,7 +652,7 @@ static void read_completed(struct urb *urb) } if (send_it) { - out->number_of_packets = FRAMES_PER_URB; + out->number_of_packets = outframe; out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); } -- cgit v1.2.3-70-g09d2 From 280ec8b718e8565333ace339d6bba91239440b20 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Aug 2011 22:19:19 +0900 Subject: ASoC: Add missing break in WM8994 probe This error would have no effect on current silicon revisions, the fall through case has the same behaviour. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 09e680ae88b..b393f9fac97 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2981,6 +2981,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.dcs_readback_mode = 1; break; } + break; case WM8958: wm8994->hubs.dcs_readback_mode = 1; -- cgit v1.2.3-70-g09d2 From feb00dceb5af57ce34514ce66096b32d133ded3d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 11 Aug 2011 12:23:22 +0900 Subject: ASoC: Terminate WM8750 SPI device ID table Signed-off-by: Mark Brown Reported-by: Stephen Rothwell --- sound/soc/codecs/wm8750.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index e6f47f49357..82ac5fcaa2b 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -781,6 +781,7 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi) static const struct spi_device_id wm8750_spi_ids[] = { { "wm8750", 0 }, { "wm8987", 0 }, + { 0, 0 }, }; MODULE_DEVICE_TABLE(spi, wm8750_spi_ids); -- cgit v1.2.3-70-g09d2 From f09aecd50f39d35372e551491d9f36ff0f51ee4d Mon Sep 17 00:00:00 2001 From: Sangbeom Kim Date: Wed, 20 Jul 2011 17:07:13 +0900 Subject: ASoC: SAMSUNG: Add I2S0 internal dma driver I2S in Exynos4 and S5PC110(S5PV210) has a internal dma. It can be used low power audio mode and 2nd channel transfer. This patch can support idma. [Reapplied after dependencies propagated through in 3.1-rc1. --broonie] Signed-off-by: Sangbeom Kim Acked-by: Jassi Brar Acked-by: Liam Girdwood Acked-by: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/samsung/Makefile | 2 + sound/soc/samsung/idma.c | 453 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/idma.h | 26 +++ 3 files changed, 481 insertions(+) create mode 100644 sound/soc/samsung/idma.c create mode 100644 sound/soc/samsung/idma.h (limited to 'sound') diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 9eb3b12eb72..8509d3c4366 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,5 +1,6 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := dma.o +snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-ac97-objs := ac97.o @@ -16,6 +17,7 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c new file mode 100644 index 00000000000..ebde0740ab1 --- /dev/null +++ b/sound/soc/samsung/idma.c @@ -0,0 +1,453 @@ +/* + * sound/soc/samsung/idma.c + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd. + * http://www.samsung.com + * + * I2S0's Internal DMA driver + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include +#include +#include +#include +#include +#include +#include + +#include "i2s.h" +#include "idma.h" +#include "dma.h" +#include "i2s-regs.h" + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +static const struct snd_pcm_hardware idma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_U24_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_IDMA_BUFFER, + .period_bytes_min = 128, + .period_bytes_max = MAX_IDMA_PERIOD, + .periods_min = 1, + .periods_max = 2, +}; + +struct idma_ctrl { + spinlock_t lock; + int state; + dma_addr_t start; + dma_addr_t pos; + dma_addr_t end; + dma_addr_t period; + dma_addr_t periodsz; + void *token; + void (*cb)(void *dt, int bytes_xfer); +}; + +static struct idma_info { + spinlock_t lock; + void __iomem *regs; + dma_addr_t lp_tx_addr; +} idma; + +static void idma_getpos(dma_addr_t *src) +{ + *src = idma.lp_tx_addr + + (readl(idma.regs + I2STRNCNT) & 0xffffff) * 4; +} + +static int idma_enqueue(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 val; + + spin_lock(&prtd->lock); + prtd->token = (void *) substream; + spin_unlock(&prtd->lock); + + /* Internal DMA Level0 Interrupt Address */ + val = idma.lp_tx_addr + prtd->periodsz; + writel(val, idma.regs + I2SLVL0ADDR); + + /* Start address0 of I2S internal DMA operation. */ + val = idma.lp_tx_addr; + writel(val, idma.regs + I2SSTR0); + + /* + * Transfer block size for I2S internal DMA. + * Should decide transfer size before start dma operation + */ + val = readl(idma.regs + I2SSIZE); + val &= ~(I2SSIZE_TRNMSK << I2SSIZE_SHIFT); + val |= (((runtime->dma_bytes >> 2) & + I2SSIZE_TRNMSK) << I2SSIZE_SHIFT); + writel(val, idma.regs + I2SSIZE); + + val = readl(idma.regs + I2SAHB); + val |= AHB_INTENLVL0; + writel(val, idma.regs + I2SAHB); + + return 0; +} + +static void idma_setcallbk(struct snd_pcm_substream *substream, + void (*cb)(void *, int)) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + spin_lock(&prtd->lock); + prtd->cb = cb; + spin_unlock(&prtd->lock); +} + +static void idma_control(int op) +{ + u32 val = readl(idma.regs + I2SAHB); + + spin_lock(&idma.lock); + + switch (op) { + case LPAM_DMA_START: + val |= (AHB_INTENLVL0 | AHB_DMAEN); + break; + case LPAM_DMA_STOP: + val &= ~(AHB_INTENLVL0 | AHB_DMAEN); + break; + default: + spin_unlock(&idma.lock); + return; + } + + writel(val, idma.regs + I2SAHB); + spin_unlock(&idma.lock); +} + +static void idma_done(void *id, int bytes_xfer) +{ + struct snd_pcm_substream *substream = id; + struct idma_ctrl *prtd = substream->runtime->private_data; + + if (prtd && (prtd->state & ST_RUNNING)) + snd_pcm_period_elapsed(substream); +} + +static int idma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 mod = readl(idma.regs + I2SMOD); + u32 ahb = readl(idma.regs + I2SAHB); + + ahb |= (AHB_DMARLD | AHB_INTMASK); + mod |= MOD_TXS_IDMA; + writel(ahb, idma.regs + I2SAHB); + writel(mod, idma.regs + I2SMOD); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->start = prtd->pos = runtime->dma_addr; + prtd->period = params_periods(params); + prtd->periodsz = params_period_bytes(params); + prtd->end = runtime->dma_addr + runtime->dma_bytes; + + idma_setcallbk(substream, idma_done); + + return 0; +} + +static int idma_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int idma_prepare(struct snd_pcm_substream *substream) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + prtd->pos = prtd->start; + + /* flush the DMA channel */ + idma_control(LPAM_DMA_STOP); + idma_enqueue(substream); + + return 0; +} + +static int idma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + idma_control(LPAM_DMA_START); + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + idma_control(LPAM_DMA_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t + idma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + dma_addr_t src; + unsigned long res; + + spin_lock(&prtd->lock); + + idma_getpos(&src); + res = src - prtd->start; + + spin_unlock(&prtd->lock); + + return bytes_to_frames(substream->runtime, res); +} + +static int idma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long size, offset; + int ret; + + /* From snd_pcm_lib_mmap_iomem */ + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_flags |= VM_IO; + size = vma->vm_end - vma->vm_start; + offset = vma->vm_pgoff << PAGE_SHIFT; + ret = io_remap_pfn_range(vma, vma->vm_start, + (runtime->dma_addr + offset) >> PAGE_SHIFT, + size, vma->vm_page_prot); + + return ret; +} + +static irqreturn_t iis_irq(int irqno, void *dev_id) +{ + struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; + u32 iiscon, iisahb, val, addr; + + iisahb = readl(idma.regs + I2SAHB); + iiscon = readl(idma.regs + I2SCON); + + val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; + + if (val) { + iisahb |= val; + writel(iisahb, idma.regs + I2SAHB); + + addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr; + addr += prtd->periodsz; + addr %= (prtd->end - prtd->start); + addr += idma.lp_tx_addr; + + writel(addr, idma.regs + I2SLVL0ADDR); + + if (prtd->cb) + prtd->cb(prtd->token, prtd->period); + } + + return IRQ_HANDLED; +} + +static int idma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &idma_hardware); + + prtd = kzalloc(sizeof(struct idma_ctrl), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); + if (ret < 0) { + pr_err("fail to claim i2s irq , ret = %d\n", ret); + kfree(prtd); + return ret; + } + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + return 0; +} + +static int idma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + + free_irq(IRQ_I2S0, prtd); + + if (!prtd) + pr_err("idma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static struct snd_pcm_ops idma_ops = { + .open = idma_open, + .close = idma_close, + .ioctl = snd_pcm_lib_ioctl, + .trigger = idma_trigger, + .pointer = idma_pointer, + .mmap = idma_mmap, + .hw_params = idma_hw_params, + .hw_free = idma_hw_free, + .prepare = idma_prepare, +}; + +static void idma_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (!substream) + return; + + buf = &substream->dma_buffer; + if (!buf->area) + return; + + iounmap(buf->area); + + buf->area = NULL; + buf->addr = 0; +} + +static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + /* Assign PCM buffer pointers */ + buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS; + buf->addr = idma.lp_tx_addr; + buf->bytes = idma_hardware.buffer_bytes_max; + buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes); + + return 0; +} + +static u64 idma_mask = DMA_BIT_MASK(32); + +static int idma_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &idma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->driver->playback.channels_min) + ret = preallocate_idma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + + return ret; +} + +void idma_reg_addr_init(void *regs, dma_addr_t addr) +{ + spin_lock_init(&idma.lock); + idma.regs = regs; + idma.lp_tx_addr = addr; +} + +struct snd_soc_platform_driver asoc_idma_platform = { + .ops = &idma_ops, + .pcm_new = idma_new, + .pcm_free = idma_free, +}; + +static int __devinit asoc_idma_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); +} + +static int __devexit asoc_idma_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver asoc_idma_driver = { + .driver = { + .name = "samsung-idma", + .owner = THIS_MODULE, + }, + + .probe = asoc_idma_platform_probe, + .remove = __devexit_p(asoc_idma_platform_remove), +}; + +static int __init asoc_idma_init(void) +{ + return platform_driver_register(&asoc_idma_driver); +} +module_init(asoc_idma_init); + +static void __exit asoc_idma_exit(void) +{ + platform_driver_unregister(&asoc_idma_driver); +} +module_exit(asoc_idma_exit); + +MODULE_AUTHOR("Jaswinder Singh, "); +MODULE_DESCRIPTION("Samsung ASoC IDMA Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h new file mode 100644 index 00000000000..48273216166 --- /dev/null +++ b/sound/soc/samsung/idma.h @@ -0,0 +1,26 @@ +/* + * sound/soc/samsung/idma.h + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd + * http://www.samsung.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __SND_SOC_SAMSUNG_IDMA_H_ +#define __SND_SOC_SAMSUNG_IDMA_H_ + +extern void idma_reg_addr_init(void *regs, dma_addr_t addr); + +/* dma_state */ +#define LPAM_DMA_STOP 0 +#define LPAM_DMA_START 1 + +#define MAX_IDMA_PERIOD (128 * 1024) +#define MAX_IDMA_BUFFER (160 * 1024) + +#endif /* __SND_SOC_SAMSUNG_IDMA_H_ */ -- cgit v1.2.3-70-g09d2 From 7ec41ee5ad5f716f67041c0d49014d0becb5332c Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 11 Aug 2011 15:44:57 +0300 Subject: ASoC: omap: Update e-mail address of Jarkko Nikula My gmail account got disabled and I'm not going to reopen it. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- MAINTAINERS | 2 +- include/sound/tlv320aic3x.h | 2 +- sound/soc/omap/n810.c | 4 ++-- sound/soc/omap/omap-mcbsp.c | 4 ++-- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.c | 4 ++-- sound/soc/omap/omap-pcm.h | 2 +- sound/soc/omap/rx51.c | 2 +- 8 files changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/MAINTAINERS b/MAINTAINERS index 51d42fbc8dc..46e3e6b9922 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4604,7 +4604,7 @@ F: arch/arm/mach-omap2/clockdomain2xxx_3xxx.c F: arch/arm/mach-omap2/clockdomain44xx.c OMAP AUDIO SUPPORT -M: Jarkko Nikula +M: Jarkko Nikula L: alsa-devel@alsa-project.org (subscribers-only) L: linux-omap@vger.kernel.org S: Maintained diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h index 99e0308bf2c..ffd9bc79310 100644 --- a/include/sound/tlv320aic3x.h +++ b/include/sound/tlv320aic3x.h @@ -1,7 +1,7 @@ /* * Platform data for Texas Instruments TLV320AIC3x codec * - * Author: Jarkko Nikula + * Author: Jarkko Nikula * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 83d213bfd3d..62e292f4931 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula + * Contact: Jarkko Nikula * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -402,6 +402,6 @@ static void __exit n810_soc_exit(void) module_init(n810_soc_init); module_exit(n810_soc_exit); -MODULE_AUTHOR("Jarkko Nikula "); +MODULE_AUTHOR("Jarkko Nikula "); MODULE_DESCRIPTION("ALSA SoC Nokia N810"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 07b77235724..ebcc2d4d2b1 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula + * Contact: Jarkko Nikula * Peter Ujfalusi * * This program is free software; you can redistribute it and/or @@ -780,6 +780,6 @@ static void __exit snd_omap_mcbsp_exit(void) } module_exit(snd_omap_mcbsp_exit); -MODULE_AUTHOR("Jarkko Nikula "); +MODULE_AUTHOR("Jarkko Nikula "); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 9a7dedd6f5a..65cde9d3807 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula + * Contact: Jarkko Nikula * Peter Ujfalusi * * This program is free software; you can redistribute it and/or diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b2f5751edae..9b5c88ac35b 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula + * Contact: Jarkko Nikula * Peter Ujfalusi * * This program is free software; you can redistribute it and/or @@ -436,6 +436,6 @@ static void __exit snd_omap_pcm_exit(void) } module_exit(snd_omap_pcm_exit); -MODULE_AUTHOR("Jarkko Nikula "); +MODULE_AUTHOR("Jarkko Nikula "); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index a0ed1dbb52d..f95fe306417 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula + * Contact: Jarkko Nikula * Peter Ujfalusi * * This program is free software; you can redistribute it and/or diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 0aae998b654..893300a53ba 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -5,7 +5,7 @@ * * Contact: Peter Ujfalusi * Eduardo Valentin - * Jarkko Nikula + * Jarkko Nikula * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License -- cgit v1.2.3-70-g09d2 From f6b864a9071e21186476910613ec9913b56067a2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 12 Aug 2011 18:22:10 +0200 Subject: ASoC: Fix compile warning in wm8750.c MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit sound/soc/codecs/wm8750.c:784:2: warning: missing braces around initializer sound/soc/codecs/wm8750.c:784:2: warning: (near initialization for ‘wm8750_spi_ids[2].name’) It's because struct spi_device_id.name is a char array, not a pointer, while the driver initializes explicitly with 0. Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8750.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 82ac5fcaa2b..d0003cc3bcd 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -781,7 +781,7 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi) static const struct spi_device_id wm8750_spi_ids[] = { { "wm8750", 0 }, { "wm8987", 0 }, - { 0, 0 }, + { }, }; MODULE_DEVICE_TABLE(spi, wm8750_spi_ids); -- cgit v1.2.3-70-g09d2 From a115c72802c37351b6d87dfb62938d2ad440eef4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 4 Aug 2011 13:23:38 +0900 Subject: ASoC: Move WM8962 CLKREG_OVD earlier When the clocking registers are not overriden some of the registers are not writable. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8962.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 60d740ebeb5..28650edfdeb 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2927,10 +2927,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, WM8962_BIAS_ENA | 0x180); msleep(5); - - snd_soc_update_bits(codec, WM8962_CLOCKING2, - WM8962_CLKREG_OVD, - WM8962_CLKREG_OVD); } /* VMID 2*250k */ @@ -3868,6 +3864,10 @@ static int wm8962_probe(struct snd_soc_codec *codec) */ snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0); + /* Ensure we have soft control over all registers */ + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); + regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); if (pdata) { -- cgit v1.2.3-70-g09d2 From fd049755636a8b2cc084e088967dd566467ccebc Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Fri, 12 Aug 2011 17:52:59 +0300 Subject: ASoC: h1940: Fix compilation error due to missing header MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add linux/types.h to fix this compilation error: In file included from arch/arm/mach-s3c2410/include/mach/gpio-fns.h:27:0, from arch/arm/mach-s3c2410/include/mach/gpio.h:27, from /home/anarsoul/work/pda-linux/linux-next/arch/arm/include/asm/gpio.h:5, from include/linux/gpio.h:18, from sound/soc/samsung/rx1950_uda1380.c:20: arch/arm/plat-samsung/include/plat/gpio-cfg.h:29:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:30:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s5p_gpio_drvstr_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:57:2: error: expected specifier-qualifier-list before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:148:47: error: expected declaration specifiers or ‘...’ before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:156:24: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s3c_gpio_getpull’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:175:24: error: expected declaration specifiers or ‘...’ before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h: In function ‘s3c_gpio_cfgrange_nopull’: arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: ‘s3c_gpio_pull_t’ undeclared (first use in this function) arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: note: each undeclared identifier is reported only once for each function it appears in arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: expected ‘)’ before numeric constant arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: too many arguments to function ‘s3c_gpio_cfgall_range’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:174:12: note: declared here arch/arm/plat-samsung/include/plat/gpio-cfg.h: At top level: arch/arm/plat-samsung/include/plat/gpio-cfg.h:199:26: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s5p_gpio_get_drvstr’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:210:50: error: expected declaration specifiers or ‘...’ before ‘s5p_gpio_drvstr_t’ Signed-off-by: Vasily Khoruzhick Acked-by: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/samsung/h1940_uda1380.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 241f55d0066..c6c65892294 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -13,6 +13,7 @@ * */ +#include #include #include -- cgit v1.2.3-70-g09d2 From b8487928f5ca2976e4cb8d329943af849d2b6197 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Fri, 12 Aug 2011 17:53:00 +0300 Subject: ASoC: rx1950: Fix compilation error due to missing header MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add linux/types.h to fix this compilation error: In file included from arch/arm/mach-s3c2410/include/mach/gpio-fns.h:27:0, from arch/arm/mach-s3c2410/include/mach/gpio.h:27, from /home/anarsoul/work/pda-linux/linux-next/arch/arm/include/asm/gpio.h:5, from include/linux/gpio.h:18, from sound/soc/samsung/rx1950_uda1380.c:20: arch/arm/plat-samsung/include/plat/gpio-cfg.h:29:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:30:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s5p_gpio_drvstr_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:57:2: error: expected specifier-qualifier-list before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:148:47: error: expected declaration specifiers or ‘...’ before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:156:24: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s3c_gpio_getpull’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:175:24: error: expected declaration specifiers or ‘...’ before ‘s3c_gpio_pull_t’ arch/arm/plat-samsung/include/plat/gpio-cfg.h: In function ‘s3c_gpio_cfgrange_nopull’: arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: ‘s3c_gpio_pull_t’ undeclared (first use in this function) arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: note: each undeclared identifier is reported only once for each function it appears in arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: expected ‘)’ before numeric constant arch/arm/plat-samsung/include/plat/gpio-cfg.h:180:47: error: too many arguments to function ‘s3c_gpio_cfgall_range’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:174:12: note: declared here arch/arm/plat-samsung/include/plat/gpio-cfg.h: At top level: arch/arm/plat-samsung/include/plat/gpio-cfg.h:199:26: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘s5p_gpio_get_drvstr’ arch/arm/plat-samsung/include/plat/gpio-cfg.h:210:50: error: expected declaration specifiers or ‘...’ before ‘s5p_gpio_drvstr_t’ Signed-off-by: Vasily Khoruzhick Acked-by: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/samsung/rx1950_uda1380.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 1e574a5d440..bc8c1676459 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -17,6 +17,7 @@ * */ +#include #include #include -- cgit v1.2.3-70-g09d2 From da6094ea7d3c2295473d8f5134279307255d6ebf Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sun, 14 Aug 2011 11:31:16 +0200 Subject: ALSA: snd_usb_caiaq: track submitted output urbs The snd_usb_caiaq driver currently assumes that output urbs are serviced in time and doesn't track when and whether they are given back by the USB core. That usually works fine, but due to temporary limitations of the XHCI stack, we faced that urbs were submitted more than once with this approach. As it's no good practice to fire and forget urbs anyway, this patch introduces a proper bit mask to track which requests have been submitted and given back. That alone however doesn't make the driver work in case the host controller is broken and doesn't give back urbs at all, and the output stream will stop once all pre-allocated output urbs are consumed. But it does prevent crashes of the controller stack in such cases. See http://bugzilla.kernel.org/show_bug.cgi?id=40702 for more details. Signed-off-by: Daniel Mack Reported-and-tested-by: Matej Laitl Cc: Sarah Sharp Cc: stable@kernel.org Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 31 +++++++++++++++++++++++++++---- sound/usb/caiaq/device.h | 1 + 2 files changed, 28 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index aa52b3e13bb..2cf87f5afed 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -139,8 +139,12 @@ static void stream_stop(struct snd_usb_caiaqdev *dev) for (i = 0; i < N_URBS; i++) { usb_kill_urb(dev->data_urbs_in[i]); - usb_kill_urb(dev->data_urbs_out[i]); + + if (test_bit(i, &dev->outurb_active_mask)) + usb_kill_urb(dev->data_urbs_out[i]); } + + dev->outurb_active_mask = 0; } static int snd_usb_caiaq_substream_open(struct snd_pcm_substream *substream) @@ -612,8 +616,8 @@ static void read_completed(struct urb *urb) { struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *dev; - struct urb *out; - int frame, len, send_it = 0, outframe = 0; + struct urb *out = NULL; + int i, frame, len, send_it = 0, outframe = 0; size_t offset = 0; if (urb->status || !info) @@ -624,7 +628,17 @@ static void read_completed(struct urb *urb) if (!dev->streaming) return; - out = dev->data_urbs_out[info->index]; + /* find an unused output urb that is unused */ + for (i = 0; i < N_URBS; i++) + if (test_and_set_bit(i, &dev->outurb_active_mask) == 0) { + out = dev->data_urbs_out[i]; + break; + } + + if (!out) { + log("Unable to find an output urb to use\n"); + goto requeue; + } /* read the recently received packet and send back one which has * the same layout */ @@ -655,8 +669,12 @@ static void read_completed(struct urb *urb) out->number_of_packets = outframe; out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); + } else { + struct snd_usb_caiaq_cb_info *oinfo = out->context; + clear_bit(oinfo->index, &dev->outurb_active_mask); } +requeue: /* re-submit inbound urb */ for (frame = 0; frame < FRAMES_PER_URB; frame++) { urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame; @@ -678,6 +696,8 @@ static void write_completed(struct urb *urb) dev->output_running = 1; wake_up(&dev->prepare_wait_queue); } + + clear_bit(info->index, &dev->outurb_active_mask); } static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) @@ -829,6 +849,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) if (!dev->data_cb_info) return -ENOMEM; + dev->outurb_active_mask = 0; + BUILD_BUG_ON(N_URBS > (sizeof(dev->outurb_active_mask) * 8)); + for (i = 0; i < N_URBS; i++) { dev->data_cb_info[i].dev = dev; dev->data_cb_info[i].index = i; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index b2b310194ff..3f9c6339ae9 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -96,6 +96,7 @@ struct snd_usb_caiaqdev { int input_panic, output_panic, warned; char *audio_in_buf, *audio_out_buf; unsigned int samplerates, bpp; + unsigned long outurb_active_mask; struct snd_pcm_substream *sub_playback[MAX_STREAMS]; struct snd_pcm_substream *sub_capture[MAX_STREAMS]; -- cgit v1.2.3-70-g09d2 From eade7b281c9fc18401b989c77d5e5e660b25a3b7 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 14 Aug 2011 22:43:01 -0400 Subject: ALSA: ac97: Add HP Compaq dc5100 SFF(PT003AW) to Headphone Jack Sense whitelist BugLink: https://bugs.launchpad.net/bugs/826081 The original reporter needs 'Headphone Jack Sense' enabled to have audible audio, so add his PCI SSID to the whitelist. Reported-and-tested-by: Muhammad Khurram Khan Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 200c9a1d48b..a872d0a8297 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1909,6 +1909,7 @@ static unsigned int ad1981_jacks_whitelist[] = { 0x103c0944, /* HP nc6220 */ 0x103c0934, /* HP nc8220 */ 0x103c006d, /* HP nx9105 */ + 0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */ 0x17340088, /* FSC Scenic-W */ 0 /* end */ }; -- cgit v1.2.3-70-g09d2 From d2b4c7bd7eabfaa2e3e5b8107d5eeb56ac879813 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 13 Aug 2011 19:15:01 +0800 Subject: ASoC: soc-jack: Fix checking return value of request_any_context_irq request_any_context_irq() returns a negative value on failure. On success, it returns either IRQC_IS_HARDIRQ or IRQC_IS_NESTED. Signed-off-by: Axel Lin Signed-off-by: Mark Brown Cc: stable@kernel.orG --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7c17b98d584..38b00131b2f 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -327,7 +327,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, IRQF_TRIGGER_FALLING, gpios[i].name, &gpios[i]); - if (ret) + if (ret < 0) goto err; if (gpios[i].wake) { -- cgit v1.2.3-70-g09d2 From 161d55c3ec4c7e26c96b11dc86caea0b3c9c6b0f Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 13 Aug 2011 11:33:08 +0800 Subject: ASoC: sta32x: Fix a memory leak if snd_soc_register_codec fails Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 409d89d1f34..fbd7eb9e61c 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -857,6 +857,7 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + kfree(sta32x); return ret; } -- cgit v1.2.3-70-g09d2 From bf545ed72f2eeac664695a8ea2199d9ddaef6020 Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 12 Aug 2011 18:04:10 -0400 Subject: ASoC: ad193x: fix registers definition fix dac word len mask and adc tdm fmt shift value Signed-off-by: Scott Jiang Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/ad193x.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 9747b549787..c1029d242c8 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -34,7 +34,7 @@ #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) #define AD193X_DAC_CTRL2 0x804 -#define AD193X_DAC_WORD_LEN_MASK 0xC +#define AD193X_DAC_WORD_LEN_MASK 0x18 #define AD193X_DAC_MASTER_MUTE 1 #define AD193X_DAC_CHNL_MUTE 0x805 #define AD193X_DACL1_MUTE 0 @@ -63,7 +63,7 @@ #define AD193X_ADC_CTRL1 0x80f #define AD193X_ADC_SERFMT_MASK 0x60 #define AD193X_ADC_SERFMT_STEREO (0 << 5) -#define AD193X_ADC_SERFMT_TDM (1 << 2) +#define AD193X_ADC_SERFMT_TDM (1 << 5) #define AD193X_ADC_SERFMT_AUX (2 << 5) #define AD193X_ADC_WORD_LEN_MASK 0x3 #define AD193X_ADC_CTRL2 0x810 -- cgit v1.2.3-70-g09d2 From 95c93d8525ebce1024bda7316f602ae45c36cd6f Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 12 Aug 2011 18:04:11 -0400 Subject: ASoC: ad193x: fix dac word len setting dac word len value should left shift before setting Signed-off-by: Scott Jiang Acked-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/ad193x.c | 3 ++- sound/soc/codecs/ad193x.h | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 2374ca5ffe6..f1a8be58255 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -307,7 +307,8 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); reg = snd_soc_read(codec, AD193X_DAC_CTRL2); - reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len; + reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) + | (word_len << AD193X_DAC_WORD_LEN_SHFT); snd_soc_write(codec, AD193X_DAC_CTRL2, reg); reg = snd_soc_read(codec, AD193X_ADC_CTRL1); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index c1029d242c8..cccc2e8e5fb 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -34,6 +34,7 @@ #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) #define AD193X_DAC_CTRL2 0x804 +#define AD193X_DAC_WORD_LEN_SHFT 3 #define AD193X_DAC_WORD_LEN_MASK 0x18 #define AD193X_DAC_MASTER_MUTE 1 #define AD193X_DAC_CHNL_MUTE 0x805 -- cgit v1.2.3-70-g09d2 From 25ea524bed0202f823a0adcbbda68e86a22e3670 Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 12 Aug 2011 18:04:12 -0400 Subject: ASoC: ad193x: fix system clock system clock is 24.576MHz instead of 12.288MHz Signed-off-by: Scott Jiang Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ad193x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index d6651c033cb..a118a0fb9d8 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -56,7 +56,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 48000: - clk = 12288000; + clk = 24576000; break; } -- cgit v1.2.3-70-g09d2 From 396a2e79cdbd562bf7ea48132f8d3ba8304109b2 Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 12 Aug 2011 18:04:13 -0400 Subject: ASoC: Add spi hw read function for 16 addr 8 data mode for ad193x fix [This will be used by the ad193x driver to fix the fact that the original author of the driver put a bodge for their particular chip into a the generic ASoC register I/O abstraction layer which looked like an obvious bug which ended up getting fixed in 3.0. Sadly there were no comments documenting what was going on. A minimally invasive correction to the driver is to remove the register cache support and go direct to the hardware all the time so we're adding a new feature -- broonie] Signed-off-by: Scott Jiang Signed-off-by: Mark Brown --- sound/soc/soc-io.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index cca490c8058..a62f7dd4ba9 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -205,6 +205,25 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, #define snd_soc_16_8_read_i2c NULL #endif +#if defined(CONFIG_SPI_MASTER) +static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec, + unsigned int r) +{ + struct spi_device *spi = codec->control_data; + + const u16 reg = cpu_to_be16(r | 0x100); + u8 data; + int ret; + + ret = spi_write_then_read(spi, ®, 2, &data, 1); + if (ret < 0) + return 0; + return data; +} +#else +#define snd_soc_16_8_read_spi NULL +#endif + static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -295,6 +314,7 @@ static struct { int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); + unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { { .addr_bits = 4, .data_bits = 12, @@ -318,6 +338,7 @@ static struct { .addr_bits = 16, .data_bits = 8, .write = snd_soc_16_8_write, .i2c_read = snd_soc_16_8_read_i2c, + .spi_read = snd_soc_16_8_read_spi, }, { .addr_bits = 16, .data_bits = 16, @@ -383,6 +404,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, #ifdef CONFIG_SPI_MASTER codec->hw_write = do_spi_write; #endif + if (io_types[i].spi_read) + codec->hw_read = io_types[i].spi_read; codec->control_data = container_of(codec->dev, struct spi_device, -- cgit v1.2.3-70-g09d2 From 0cc62e926324d4f3bd02d378baafbe73164fca35 Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 12 Aug 2011 18:04:14 -0400 Subject: ASoC: ad193x: remove cache support asoc cache layer can't support this kind of spi registers well. remove cache support and read/write registers directly Signed-off-by: Scott Jiang Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index f1a8be58255..eedb6f5e582 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -27,11 +27,6 @@ struct ad193x_priv { int sysclk; }; -/* ad193x register cache & default register settings */ -static const u8 ad193x_reg[AD193X_NUM_REGS] = { - 0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0, -}; - /* * AD193X volume/mute/de-emphasis etc. controls */ @@ -390,9 +385,6 @@ static int ad193x_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_ad193x = { .probe = ad193x_probe, - .reg_cache_default = ad193x_reg, - .reg_cache_size = AD193X_NUM_REGS, - .reg_word_size = sizeof(u16), }; #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-70-g09d2 From c503ad466da44ca23c658986629bf7a2e2eabbb7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Aug 2011 14:23:20 +0200 Subject: ALSA: hda - Fix duplicated capture-volume creation for ALC268 models Fix the duplicated creation of capture-mixer elements for some static ALC268 configurations. The capture mixers must be put to cap_mixer field instead of mixers array. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc268_quirks.c | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc268_quirks.c b/sound/pci/hda/alc268_quirks.c index be58bf2f3ae..2e5876ce71f 100644 --- a/sound/pci/hda/alc268_quirks.c +++ b/sound/pci/hda/alc268_quirks.c @@ -476,8 +476,8 @@ static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = { static const struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { - .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer, - alc268_capture_nosrc_mixer }, + .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_nosrc_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc267_quanta_il1_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -492,8 +492,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_3ST] = { - .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_base_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), .dac_nids = alc268_dac_nids, @@ -507,8 +507,8 @@ static const struct alc_config_preset alc268_presets[] = { .input_mux = &alc268_capture_source, }, [ALC268_TOSHIBA] = { - .mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_toshiba_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_toshiba_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -525,8 +525,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_ACER] = { - .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_acer_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -543,8 +543,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_ACER_DMIC] = { - .mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_acer_dmic_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -561,9 +561,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_ACER_ASPIRE_ONE] = { - .mixers = { alc268_acer_aspire_one_mixer, - alc268_beep_mixer, - alc268_capture_nosrc_mixer }, + .mixers = { alc268_acer_aspire_one_mixer, alc268_beep_mixer}, + .cap_mixer = alc268_capture_nosrc_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_aspire_one_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -579,8 +578,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_DELL] = { - .mixers = { alc268_dell_mixer, alc268_beep_mixer, - alc268_capture_nosrc_mixer }, + .mixers = { alc268_dell_mixer, alc268_beep_mixer}, + .cap_mixer = alc268_capture_nosrc_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_dell_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -596,8 +595,8 @@ static const struct alc_config_preset alc268_presets[] = { .init_hook = alc_inithook, }, [ALC268_ZEPTO] = { - .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, - alc268_beep_mixer }, + .mixers = { alc268_base_mixer, alc268_beep_mixer }, + .cap_mixer = alc268_capture_alt_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_toshiba_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -616,7 +615,8 @@ static const struct alc_config_preset alc268_presets[] = { }, #ifdef CONFIG_SND_DEBUG [ALC268_TEST] = { - .mixers = { alc268_test_mixer, alc268_capture_mixer }, + .mixers = { alc268_test_mixer }, + .cap_mixer = alc268_capture_mixer, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_volume_init_verbs, alc268_beep_init_verbs }, -- cgit v1.2.3-70-g09d2 From 25b7679136fd85b1e5197e36a0ca126163e89590 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Wed, 17 Aug 2011 09:20:01 +0200 Subject: ASoC: Fix check for symmetric rate enforcement The ASoC core tries to not enforce symmetric rates when two streams open simultaneously. It does so by checking rtd->rate being zero. This works exactly once after booting because it is not set to zero again when the streams close. Fix this by setting rtd->rate when no active stream is left. [This leads to lots of warnings about not enforcing the symmetry in some situations as there's a race in the userspace API where we know we've got two applications but don't know what rates they want to set. -- broonie ] Signed-off-by: Sascha Hauer Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b5759397afa..2879c883eeb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -290,6 +290,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; + if (!cpu_dai->active && !codec_dai->active) + rtd->rate = 0; + /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ -- cgit v1.2.3-70-g09d2 From 3fe45aeaf2033c9eaa5028ed5ba68b466008876f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 18 Aug 2011 15:13:17 +0200 Subject: ALSA: hda - Add "PCM" volume to vmaster slave list The new parser may use "PCM" volume, but it was missing the vmaster slave list, thus "Master" volume didn't control it. Reference: https://bugzilla.kernel.org/show_bug.cgi?id=41342 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9a1aa09f47f..fcb11af9ad2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1784,6 +1784,7 @@ static const char * const alc_slave_vols[] = { "Speaker Playback Volume", "Mono Playback Volume", "Line-Out Playback Volume", + "PCM Playback Volume", NULL, }; @@ -1798,6 +1799,7 @@ static const char * const alc_slave_sws[] = { "Mono Playback Switch", "IEC958 Playback Switch", "Line-Out Playback Switch", + "PCM Playback Switch", NULL, }; -- cgit v1.2.3-70-g09d2 From e574044acbad7421879270a80acd337459c94cc8 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 18 Aug 2011 15:02:47 +0300 Subject: ASoC: omap: Fix build errors in ams-delta Fix "error: too few arguments to function 'ams_delta_set_bias_level'" build errors in ams-delta.c that were introduced after commit d4c6005 ("ASoC: Add context parameter to card DAPM callbacks") by adding dapm context to ams_delta_set_bias_level calls. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/ams-delta.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 30fe0d0efe1..0aa475f92ef 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -514,7 +514,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) } /* Set codec bias level */ - ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY); + ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY); /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ @@ -649,7 +649,9 @@ static void __exit ams_delta_module_exit(void) ams_delta_hook_switch_gpios); /* Keep modem power on */ - ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY); + ams_delta_set_bias_level(&ams_delta_audio_card, + &ams_delta_audio_card.rtd[0].codec->dapm, + SND_SOC_BIAS_STANDBY); platform_device_unregister(cx20442_platform_device); platform_device_unregister(ams_delta_audio_platform_device); -- cgit v1.2.3-70-g09d2 From 38b65190c6ab0be8ce7cff69e734ca5b5e7fa309 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Aug 2011 07:55:10 +0200 Subject: ALSA: usb-audio - Fix missing mixer dB information The recent fix for testing dB range at the mixer creation time seems to cause regressions in some devices. In such devices, reading the dB info at probing time gives an error, thus both dBmin and dBmax are still zero, and TLV flag isn't set although the later read of dB info succeeds. This patch adds a workaround for such a case by assuming that the later read will succeed. In future, a similar test should be performed in a case where a wrong dB range is seen even in the later read. Signed-off-by: Takashi Iwai Cc: --- sound/usb/mixer.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c04d7c71ac8..cdd19d7fe50 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -152,6 +152,7 @@ static inline void check_mapped_dB(const struct usbmix_name_map *p, if (p && p->dB) { cval->dBmin = p->dB->min; cval->dBmax = p->dB->max; + cval->initialized = 1; } } @@ -1092,7 +1093,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, " Switch" : " Volume"); if (control == UAC_FU_VOLUME) { check_mapped_dB(map, cval); - if (cval->dBmin < cval->dBmax) { + if (cval->dBmin < cval->dBmax || !cval->initialized) { kctl->tlv.c = mixer_vol_tlv; kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ | -- cgit v1.2.3-70-g09d2 From b6acf013bdc6f6ff9643030add85832d44034a28 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 20 Aug 2011 09:14:45 +0200 Subject: ALSA: hda - Don't spew too many ELD errors Currently HD-audio driver shows the all error ELD byte as an error in the kernel message. This is annoying when the video driver doesn't set the correct ELD from the beginning. e.g. radeon sends a zero-byte data, but we still check ELD with the fixed 128 byte as a workaround for some broken devices, it spews 128-times errors. For avoiding this, the driver aborts reading when the first byte is invalid. In such a case, the whole data is certainly invalid. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 31 +++++++++++++++++++------------ 1 file changed, 19 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 28ce17d09c3..c34f730f481 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -144,25 +144,17 @@ static int cea_sampling_frequencies[8] = { SNDRV_PCM_RATE_192000, /* 7: 192000Hz */ }; -static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid, +static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid, int byte_index) { unsigned int val; val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_ELDD, byte_index); - #ifdef BE_PARANOID printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val); #endif - - if ((val & AC_ELDD_ELD_VALID) == 0) { - snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n", - byte_index); - val = 0; - } - - return val & AC_ELDD_ELD_DATA; + return val; } #define GRAB_BITS(buf, byte, lowbit, bits) \ @@ -344,11 +336,26 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, if (!buf) return -ENOMEM; - for (i = 0; i < size; i++) - buf[i] = hdmi_get_eld_byte(codec, nid, i); + for (i = 0; i < size; i++) { + unsigned int val = hdmi_g


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