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Re: [alsa-devel] [PATCH v2] asoc: davinci: let platform data define edma queue numbers
by Mark Brown 20 Jul '10

20 Jul '10

On Mon, Jul 19, 2010 at 12:31:16PM +0530, Sekhar Nori wrote: > Currently the EDMA queue to be used by for servicing ASP through > internal RAM is fixed to EDMAQ_0 and that to service internal RAM > from external RAM is fixed to EDMAQ_1. Applied, thanks. It's "ASoC", BTW.

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[alsa-devel] [PATCH 2/3] ASoC: multi-component - Support Samsung SoC(S5P) in I2Sv2
by Chanwoo Choi 20 Jul '10

20 Jul '10

This patch modify I2Sv2 driver to support Samsung SoC(S5PV210). Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/s3c-i2s-v2.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 6575410..f4fbc0e 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -32,7 +32,8 @@ #undef S3C_IIS_V2_SUPPORTED -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) \ + || defined(CONFIG_CPU_S5PV210) #define S3C_IIS_V2_SUPPORTED #endif -- 1.6.3.3

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[alsa-devel] [PATCH 3/3] ASoC: multi-component - Enable WM8993/4 common code to support WM8994
by Chanwoo Choi 20 Jul '10

20 Jul '10

This patch enable WM8993/4 common code which is used in WM8994 driver. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/codecs/Kconfig | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ea1f5ed..17e28eb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -85,8 +85,8 @@ config SND_SOC_ALL_CODECS config SND_SOC_WM_HUBS tristate - default y if SND_SOC_WM8993=y - default m if SND_SOC_WM8993=m + default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y + default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m config SND_SOC_AC97_CODEC tristate -- 1.6.3.3

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[alsa-devel] [PATCH] ASoC: da7210: Add HeadPhone Playback Volume control
by Kuninori Morimoto 20 Jul '10

20 Jul '10

HeadPhone Playback Volume control register of DA7210 has reserved area. This patch considered it as mute. This patch add HeadPhone Playback Volume 40 as default, because DA7210 doesn't have default volume. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/da7210.c | 37 +++++++++++++++++++++++++++++++++---- 1 files changed, 33 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 75af2d6..f0af2c8 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -145,6 +145,29 @@ #define DA7210_VERSION "0.0.1" +/* + * Playback Volume + * + * max : 0x3F (+15.0 dB) + * (1.5 dB step) + * min : 0x11 (-54.0 dB) + * mute : 0x10 + * reserved : 0x00 - 0x0F + * + * ** FIXME ** + * + * Reserved area are considered as "mute". + * -> min = -79.5 dB + */ +static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); + +static const struct snd_kcontrol_new da7210_snd_controls[] = { + + SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", + DA7210_HP_L_VOL, DA7210_HP_R_VOL, + 0, 0x3F, 0, hp_out_tlv), +}; + /* Codec private data */ struct da7210_priv { struct snd_soc_codec codec; @@ -227,10 +250,6 @@ static int da7210_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; if (is_play) { - /* PlayBack Volume 40 */ - snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); - snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); - /* Enable Out */ snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); @@ -572,6 +591,13 @@ static int da7210_init(struct da7210_priv *da7210) /* Activate all enabled subsystem */ da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); + /* + * HeadPhone PlayBack Volume 40 is given here + * because DA7210 doesn't have default volume. + */ + snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); + snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); + return ret; init_err: @@ -656,6 +682,9 @@ static int da7210_probe(struct platform_device *pdev) if (ret < 0) goto pcm_err; + snd_soc_add_controls(da7210_codec, da7210_snd_controls, + ARRAY_SIZE(da7210_snd_controls)); + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); pcm_err: -- 1.7.0.4

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[alsa-devel] [PATCH v2 - alsa-lib 0/3] Fixes for snd_tlv_convert_from_dB
by Peter Ujfalusi 20 Jul '10

20 Jul '10

Hello, Changes since the initial version: - In patch one the range check has been converted to be more intuitive Intro part from the first version: One cosmetic fix, and two real fixes for alsa-lib's tlv handling. There were two separate issue with the snd_tlv_convert_from_dB function: 1. When user asked for out of range dB value, and the control had SND_CTL_TLVT_DB_RANGE, the control's full range was not checked against the requested value, which resulted failure. 2. When there are 'holes' in the SND_CTL_TLVT_DB_RANGE array, snd_tlv_convert_from_dB will fail to find the apropriate raw value. For issue #2, I have sent patches for the drivers I maintain, but Mark Brown suggested to fix the issue in alsa-lib instead of converting the driver's non-overlapping to overlapping mapping: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-July/029483.html So the following series for alsa-lib, will fix the SND_CTL_TLVT_DB_RANGE handling. --- Peter Ujfalusi (3): tlv: Check out of range dB with SND_CTL_TLVT_DB_RANGE tlv: Handle 'holes' in SND_CTL_TLVT_DB_RANGE array tlv: Remove tailing tab after snd_ctl_get_dB_range function src/control/tlv.c | 20 ++++++++++++++++++-- 1 files changed, 18 insertions(+), 2 deletions(-)

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[alsa-devel] [PATCH 0/3] ASoC: twl4030/tpa6130a2: DB_RANGE mapping fixes
by Peter Ujfalusi 20 Jul '10

20 Jul '10

Hello, There were 'holes' in the DB_RANGE and SCALE_ITEM mapping, which results failure, when user space requests dB value in these holes, since alsa-lib will fail to find a range, and can not round the requested dB to raw value. Fixing the drivers I'm maintaining to correctly fill the tlv struct. Peter --- Peter Ujfalusi (3): ASoC: TWL4030: Fix DB_RANGE tlv mapping for digital loopback ASoC: tpa6130a2: Fix DB_RANGE mapping for tpa6130a2 chip ASoC: tpa6130a2: Fix DB_RANGE mapping for tpa6140a2 chip sound/soc/codecs/tpa6130a2.c | 22 ++++++++++++++-------- sound/soc/codecs/twl4030.c | 4 ++-- 2 files changed, 16 insertions(+), 10 deletions(-)

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[alsa-devel] [PATCH v2] ASoC: da7210: Add HeadPhone Playback Volume control
by Kuninori Morimoto 20 Jul '10

20 Jul '10

HeadPhone Playback Volume control register of DA7210 has reserved area. This patch considered it as mute. This patch add HeadPhone Playback Volume 40 as default, because DA7210 doesn't have default volume. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 o based on ASoC's git tree (add #include <sound/tlv.h>) sound/soc/codecs/da7210.c | 38 ++++++++++++++++++++++++++++++++++---- 1 files changed, 34 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a83aa18..ec292cc 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -23,6 +23,7 @@ #include <sound/pcm_params.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "da7210.h" @@ -136,6 +137,29 @@ #define DA7210_VERSION "0.0.1" +/* + * Playback Volume + * + * max : 0x3F (+15.0 dB) + * (1.5 dB step) + * min : 0x11 (-54.0 dB) + * mute : 0x10 + * reserved : 0x00 - 0x0F + * + * ** FIXME ** + * + * Reserved area are considered as "mute". + * -> min = -79.5 dB + */ +static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); + +static const struct snd_kcontrol_new da7210_snd_controls[] = { + + SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", + DA7210_HP_L_VOL, DA7210_HP_R_VOL, + 0, 0x3F, 0, hp_out_tlv), +}; + /* Codec private data */ struct da7210_priv { struct snd_soc_codec codec; @@ -218,10 +242,6 @@ static int da7210_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; if (is_play) { - /* PlayBack Volume 40 */ - snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); - snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); - /* Enable Out */ snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); @@ -563,6 +583,13 @@ static int da7210_init(struct da7210_priv *da7210) /* Activate all enabled subsystem */ da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); + /* + * HeadPhone PlayBack Volume 40 is given here + * because DA7210 doesn't have default volume. + */ + snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40); + snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40); + return ret; init_err: @@ -647,6 +674,9 @@ static int da7210_probe(struct platform_device *pdev) if (ret < 0) goto pcm_err; + snd_soc_add_controls(da7210_codec, da7210_snd_controls, + ARRAY_SIZE(da7210_snd_controls)); + dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); pcm_err: -- 1.7.0.4

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[alsa-devel] [PATCH] Add conexant quirk for AMD based Lenovo G series machines
by Jerone Young 19 Jul '10

19 Jul '10

This is a follow on patch adds support for AMD based Lenovo G series machines, such as the Lenovo G555. Signed-off-by: Jerone Young <jerone.young(a)canonical.com> diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index c49030b..774edb9 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -3000,6 +3000,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G series (AMD)", CXT5066_IDEAPAD), {} };

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[alsa-devel] DMA Physical Addr Issue.
by Adish Kuvelker 19 Jul '10

19 Jul '10

Hello, I have allocated the DMA memory using the function "snd_pcm_lib_preallocate_pages_for_all" passing SNDRV_DMA_TYPE_DEV and "card->dev" as inputs to the function. Later, in the hw_params call out function I allocate pages to the requested DMA memory region using the function "snd_pcm_lib_malloc_pages". I play a wav file "Noise.wav" from command line by giving the following command ./aplay --file-type wav -f dat Noise.wav And, when I print the DMA region physical address in the "prepare" call out through "dma_addr" member of the runtime structure I get a weird decimal value of "-1036562432" as the physical address. Can anyone point where I might be going wrong in the whole DMA memory allocation? Regards; Aadish

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[alsa-devel] [PATCH 12/15] pci: riptide: check kzalloc() result
by Kulikov Vasiliy 19 Jul '10

19 Jul '10

If kzalloc() fails exit with -ENOMEM. Signed-off-by: Kulikov Vasiliy <segooon(a)gmail.com> --- sound/pci/riptide/riptide.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 59d7996..f64fb7d 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1615,7 +1615,10 @@ static int snd_riptide_playback_open(struct snd_pcm_substream *substream) chip->playback_substream[sub_num] = substream; runtime->hw = snd_riptide_playback; + data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; data->paths = lbus_play_paths[sub_num]; data->id = play_ids[sub_num]; data->source = play_sources[sub_num]; @@ -1635,7 +1638,10 @@ static int snd_riptide_capture_open(struct snd_pcm_substream *substream) chip->capture_substream = substream; runtime->hw = snd_riptide_capture; + data = kzalloc(sizeof(struct pcmhw), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; data->paths = lbus_rec_path; data->id = PADC; data->source = ACLNK2PADC; -- 1.7.0.4

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