March 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] compress offload patch
by Eric Laurent 04 Mar '16

04 Mar '16

Takashi, Vinod, Can you help with merging the attached patch for compress offload core upstream? Thanks. Eric.

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[alsa-devel] [PATCH 1/2] ALSA: compress: Add SNDRV_PCM_STATE_PREPARED state explanation
by Vinod Koul 04 Mar '16

04 Mar '16

Stream states were explained in the code comments but SNDRV_PCM_STATE_PREPARED was missed so add it Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/core/compress_offload.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 7fac3cae8abd..51a8eb495f54 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -74,6 +74,9 @@ struct snd_compr_file { * SNDRV_PCM_STATE_SETUP: When stream has been initialized. This is done by * calling SNDRV_COMPRESS_SET_PARAMS. running streams will come to this * state at stop by calling SNDRV_COMPRESS_STOP, or at end of drain. + * SNDRV_PCM_STATE_PREPARED: When a stream has been written to (for + * playback only). User after setting up stream writes the data buffer + * before starting the stream. * SNDRV_PCM_STATE_RUNNING: When stream has been started and is * decoding/encoding and rendering/capturing data. * SNDRV_PCM_STATE_DRAINING: When stream is draining current data. This is done -- 1.9.1

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[alsa-devel] USB audio2.0 status interrupt endpoint
by Puneet Sharma 04 Mar '16

04 Mar '16

Hello all, I am using the uac2 kernel driver(drivers/usb/gadget/f_uac2.c) having kernel version 3.10.31 for an embedded board. The board is presented as a USB device to a Linux PC having USB host controller in it. I have added an interrupt endpoint in the uac2 kernel driver to support the status interrupt endpoint request as per usb audio 2.0 specification. For the interrupt transfer to happen, I prepared the interrupt packet structure as per audio 2.0 specification and queued the endpoint to transfer it to the host driver. The interrupt structure is prepared as follows: bInfo : 0 (Interface request) bAttribute : 0x1 (CUR request) wValue : 0x200 (VOLUME feature unit control selector) wIndex : 0x700 (FEATURE UNIT ID) As per my understanding, when the above interrupt endpoint is transferred to host in response to Interrupt IN request, the host should send me a GET CUR volume class-specific request to further qualify the cause of interrupt. However, I am not getting any class-specific request from host in response to the status interrupt transferred to host. My query is do i need to implement a mixer interface in the uac2 kernel driver to get the request or i am doing something wrong in the process. I am stuck onto this and cannot proceed further. Thanks Puneet

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[alsa-devel] writing an alsa driver
by Michele Curti 04 Mar '16

04 Mar '16

Hi all, I'm going to *try* to "write" a driver for the sound card on my new laptop, an asus x205ta. To be fair I have almost no experience but I'm hoping that all the code is already present and it only needs some adaptations. The cpu is a intel bay trail z3735f, the sound card is a realtek rt5648 that has the same product Id of the rt5645 (0x6308). I see in kernel sources: - the codec driver sound/soc/codecs/rt5645.c - a baytrail board that uses the rt5640 sound/soc/intel/boards/byt-rt5640.c - a braswell board that uses the rt5645 sound/soc/intel/boards/cht_bsw_rt5645.c What do you think? Is it feasible, given a good amount of free-time? :p Thanks, Michele

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[alsa-devel] [PATCH v2] ASoC: pcm3168a: remove unused variable
by Sudip Mukherjee 04 Mar '16

04 Mar '16

From: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> While building we were getting build warning about: sound/soc/codecs/pcm3168a.c:403:21: warning: variable 'channels' set but not used The variable channels were being assigned some value but that was never reused. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- v2: From: address changed sound/soc/codecs/pcm3168a.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 0c7248a..992a77e 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -400,13 +400,12 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, struct pcm3168a_priv *pcm3168a = snd_soc_codec_get_drvdata(codec); bool tx, master_mode; u32 val, mask, shift, reg; - unsigned int rate, channels, fmt, ratio, max_ratio; + unsigned int rate, fmt, ratio, max_ratio; int i, min_frame_size; snd_pcm_format_t format; rate = params_rate(params); format = params_format(params); - channels = params_channels(params); ratio = pcm3168a->sysclk / rate; -- 1.9.1

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[alsa-devel] [PATCH] Make alsa asound.h header more portable.
by Thomas Klausner 04 Mar '16

04 Mar '16

From: Thomas Klausner <wiz(a)NetBSD.org> This header is used as-is in the alsa-lib userland library, which is portable to other operating systems. For this reason, include linux/types.h only on Linux systems. Add sys/ioctl.h for _IOR/_IOW/etc. (works at least on *BSD and Solaris). Signed-off-by: Thomas Klausner <wiz(a)NetBSD.org> --- include/uapi/sound/asound.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index a82108e..3d581b5 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -23,7 +23,11 @@ #ifndef _UAPI__SOUND_ASOUND_H #define _UAPI__SOUND_ASOUND_H +#ifdef __linux__ #include <linux/types.h> +#else +#include <sys/ioctl.h> +#endif #ifndef __KERNEL__ #include <stdlib.h> -- 2.7.1

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[alsa-devel] [PATCH 1/2] ALSA: hda - hdmi add wmb barrier for audio component
by libin.yang＠linux.intel.com 04 Mar '16

04 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> To make sure audio_ptr is set before intel_audio_codec_enable() or intel_audio_codec_disable() calling pin_eld_notify(), this patch adds wmb barrier to prevent optimizing. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index aa2d6a2..b5190e0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2939,6 +2939,11 @@ static int patch_generic_hdmi(struct hda_codec *codec) if (codec_has_acomp(codec)) { codec->depop_delay = 0; spec->i915_audio_ops.audio_ptr = codec; + /* intel_audio_codec_enable() or intel_audio_codec_disable() + * will call pin_eld_notify with using audio_ptr pointer + * We need make sure audio_ptr is really setup + */ + wmb(); spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; snd_hdac_i915_register_notifier(&spec->i915_audio_ops); } -- 1.9.1

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Re: [alsa-devel] [PATCH v3] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Nicolin Chen 04 Mar '16

04 Mar '16

On Fri, Mar 04, 2016 at 07:15:44AM +0000, Zidan Wang wrote: > > cpu-dai1 : The phandle of an CPU DAI controller for AIF1 > > cpu-dai2 : The phandle of an CPU DAI controller for AIF2 > > cpu-dai3 : The phandle of an CPU DAI controller for AIF3 > > > > > + data->card.num_links = DAI_LINK_NUM; > > > + data->card.dai_link = data->dai_link; > > > > Then, rather than hard coding the total DAI link number and creating > > useless dummy links, you may create the links dynamically according to > > the presence of the three properties above. > > I don't know how to using cpu-dai1/2/3. There don't have voice and bt controller on the IMX cpu, don't have such cpu nodes. > Do you means our hardware connection are wrong, and we must have voice and bt controller on the cpu? Okay. I was trying to give you an excuse to have all three aifs included, as I thought you might eventually need them to *connect* something via DT. If you think you don't have any idea to pre-allocate these links, you probably should ignore my suggestion here and just follow Mark's comments: [ Why are you mapping in dummy DAIs? If these devices aren't connected then they're not connected and you shouldn't represent them. If they are connected to something then describe those connections, possibly in followup patches if you have other devices you need to support upstream first. ]

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[alsa-devel] [PATCH] ASoC: rt5616: allow to build with CONFIG_SND_SOC_RT5616
by Caesar Wang 04 Mar '16

04 Mar '16

This patch fixes that hasn't built the rt5616 driver with 'CONFIG_SND_SOC_RT5616=y' in .config. The tristate is the prompt on the 'make menuconfig', in other words, that can't show the prompt and select it if we don't say what's the tristate. Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4383966..14eb5f3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -623,7 +623,7 @@ config SND_SOC_RT5514 tristate config SND_SOC_RT5616 - tristate + tristate "Realtek RT5616 CODEC" config SND_SOC_RT5631 tristate "Realtek ALC5631/RT5631 CODEC" -- 1.9.1

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[alsa-devel] [PATCH] sound/usb: Use meaninful names for goto labels
by Shuah Khan 04 Mar '16

04 Mar '16

Fix to use meaningful names instead of numbered goto labels Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- sound/usb/media.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/usb/media.c b/sound/usb/media.c index b87d5b3..ab6a642 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -84,21 +84,21 @@ int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, ret = media_device_register_entity(mctl->media_dev, &mctl->media_entity); if (ret) - goto err1; + goto free_mctl; mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, MAJOR(pcm_dev->devt), MINOR(pcm_dev->devt)); if (!mctl->intf_devnode) { ret = -ENOMEM; - goto err2; + goto unregister_entity; } mctl->intf_link = media_create_intf_link(&mctl->media_entity, &mctl->intf_devnode->intf, MEDIA_LNK_FL_ENABLED); if (!mctl->intf_link) { ret = -ENOMEM; - goto err3; + goto devnode_remove; } /* create link between mixer and audio */ @@ -109,7 +109,7 @@ int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, &mctl->media_entity, 0, MEDIA_LNK_FL_ENABLED); if (ret) - goto err4; + goto remove_intf_link; break; } } @@ -117,13 +117,13 @@ int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, subs->media_ctl = mctl; return 0; -err4: +remove_intf_link: media_remove_intf_link(mctl->intf_link); -err3: +devnode_remove: media_devnode_remove(mctl->intf_devnode); -err2: +unregister_entity: media_device_unregister_entity(&mctl->media_entity); -err1: +free_mctl: kfree(mctl); return ret; } -- 2.5.0

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