From 0dde4b573eee57799a6b07bc46705377deb0bc00 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 28 Mar 2018 02:14:22 +0000 Subject: ASoC: doc: replace codec to component Now we can replace Codec to Component. Let's do it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- Documentation/sound/soc/codec.rst | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/soc/codec.rst b/Documentation/sound/soc/codec.rst index f87612b94812..58f625fe3d39 100644 --- a/Documentation/sound/soc/codec.rst +++ b/Documentation/sound/soc/codec.rst @@ -179,12 +179,12 @@ i.e. static int wm8974_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf; + struct snd_soc_component *component = dai->component; + u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf; if (mute) - snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40); + snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40); else - snd_soc_write(codec, WM8974_DAC, mute_reg); + snd_soc_component_write(component, WM8974_DAC, mute_reg); return 0; } -- cgit v1.2.3-59-g8ed1b From d9f29e9ebe6ec7193c160ac3d38d5d1d6b55adda Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 8 May 2018 03:21:24 +0000 Subject: ASoC: convert platform explanation to component commit ef050bece1b55 ("ASoC: Remove platform code now everything is componentised") removed platform code, but it didn't care about platform documentation. This patch convert platform explanation to component Signed-off-by: Kuninori Morimoto Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- Documentation/sound/soc/platform.rst | 30 +++++++++++++----------------- 1 file changed, 13 insertions(+), 17 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/soc/platform.rst b/Documentation/sound/soc/platform.rst index d5574904d981..cc817078daa6 100644 --- a/Documentation/sound/soc/platform.rst +++ b/Documentation/sound/soc/platform.rst @@ -23,30 +23,26 @@ The platform DMA driver optionally supports the following ALSA operations:- }; The platform driver exports its DMA functionality via struct -snd_soc_platform_driver:- +snd_soc_component_driver:- :: - struct snd_soc_platform_driver { - char *name; + struct snd_soc_component_driver { + const char *name; - int (*probe)(struct platform_device *pdev); - int (*remove)(struct platform_device *pdev); - int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); - int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); + ... + int (*probe)(struct snd_soc_component *); + void (*remove)(struct snd_soc_component *); + int (*suspend)(struct snd_soc_component *); + int (*resume)(struct snd_soc_component *); /* pcm creation and destruction */ - int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *); + int (*pcm_new)(struct snd_soc_pcm_runtime *); void (*pcm_free)(struct snd_pcm *); - /* - * For platform caused delay reporting. - * Optional. - */ - snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, - struct snd_soc_dai *); - - /* platform stream ops */ - struct snd_pcm_ops *pcm_ops; + ... + const struct snd_pcm_ops *ops; + const struct snd_compr_ops *compr_ops; + ... }; Please refer to the ALSA driver documentation for details of audio DMA. -- cgit v1.2.3-59-g8ed1b From 09b83d107d8ac70ff8b0c368182e36a6e06b8cf4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 18 May 2018 12:18:59 +0200 Subject: ALSA: hda/conexant - Add hp-mic-fix model string Add "hp-mic-fix" model string for Conexant codecs so that user can test the quirk without recompiling. Signed-off-by: Takashi Iwai --- Documentation/sound/hd-audio/models.rst | 2 ++ sound/pci/hda/patch_conexant.c | 1 + 2 files changed, 3 insertions(+) (limited to 'Documentation/sound') diff --git a/Documentation/sound/hd-audio/models.rst b/Documentation/sound/hd-audio/models.rst index 1fee5a4f6660..7c2d37571af0 100644 --- a/Documentation/sound/hd-audio/models.rst +++ b/Documentation/sound/hd-audio/models.rst @@ -263,6 +263,8 @@ hp-dock HP dock support mute-led-gpio Mute LED control via GPIO +hp-mic-fix + Fix for headset mic pin on HP boxes STAC9200 ======== diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index dad7e9d5e5a6..dbf9910c5269 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -998,6 +998,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" }, { .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" }, { .id = CXT_FIXUP_MUTE_LED_GPIO, .name = "mute-led-gpio" }, + { .id = CXT_FIXUP_HP_MIC_NO_PRESENCE, .name = "hp-mic-fix" }, {} }; -- cgit v1.2.3-59-g8ed1b From f274baa49be67dd8a9f318cd95da6ef9f565d06b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 May 2018 13:01:17 +0200 Subject: ALSA: usb-audio: Allow non-vmalloc buffer for PCM buffers Currently, USB-audio driver allocates the PCM buffer via vmalloc(), as this serves merely as an intermediate buffer that is copied to each URB transfer buffer. This works well in general on x86, but on some archs this may result in cache coherency issues when mmap is used. OTOH, it works also on such arch unless mmap is used. This patch is a step for mitigating the inconvenience; a new module option "use_vmalloc" is provided so that user can choose to allocate the DMA coherent buffer instead of the existing vmalloc buffer. The drawback is that it'd be the standard dma_alloc_coherent() calls and the system would require contiguous pages on non-x86 archs. Note that it's a global option and not dynamically switchable since the buffer is pre-allocated at the probe time. In theory, it's possible to be switchable, but it'd be trickier and racier. As default use_vmalloc option is set to true, so that the old behavior is kept. For allowing the coherent mmap on ARM or MIPS, pass use_vmalloc=0 option explicitly. Reported-and-tested-by: Daniel Danzberger Signed-off-by: Takashi Iwai --- Documentation/sound/alsa-configuration.rst | 7 ++++ sound/usb/card.c | 4 ++ sound/usb/pcm.c | 59 +++++++++++++++++++++++++++--- sound/usb/pcm.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 2 + 6 files changed, 70 insertions(+), 5 deletions(-) (limited to 'Documentation/sound') diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index aed6b4fb8e46..b1052e18292d 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2224,6 +2224,13 @@ quirk_alias Quirk alias list, pass strings like ``0123abcd:5678beef``, which applies the existing quirk for the device 5678:beef to a new device 0123:abcd. +use_vmalloc + Use vmalloc() for allocations of the PCM buffers (default: yes). + For architectures with non-coherent memory like ARM or MIPS, the + mmap access may give inconsistent results with vmalloc'ed + buffers. If mmap is used on such architectures, turn off this + option, so that the DMA-coherent buffers are allocated and used + instead. This module supports multiple devices, autoprobe and hotplugging. diff --git a/sound/usb/card.c b/sound/usb/card.c index c80224807e8f..a1ed798a1c6b 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -86,6 +86,8 @@ static bool ignore_ctl_error; static bool autoclock = true; static char *quirk_alias[SNDRV_CARDS]; +bool snd_usb_use_vmalloc = true; + module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); module_param_array(id, charp, NULL, 0444); @@ -105,6 +107,8 @@ module_param(autoclock, bool, 0444); MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes)."); module_param_array(quirk_alias, charp, NULL, 0444); MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef."); +module_param_named(use_vmalloc, snd_usb_use_vmalloc, bool, 0444); +MODULE_PARM_DESC(use_vmalloc, "Use vmalloc for PCM intermediate buffers (default: yes)."); /* * we keep the snd_usb_audio_t instances by ourselves for merging diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 897a2cbef6de..78d1cad08a0a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -728,7 +728,11 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; - ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + if (snd_usb_use_vmalloc) + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + else + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (ret < 0) return ret; @@ -781,7 +785,11 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) snd_usb_endpoint_deactivate(subs->data_endpoint); snd_usb_unlock_shutdown(subs->stream->chip); } - return snd_pcm_lib_free_vmalloc_buffer(substream); + + if (snd_usb_use_vmalloc) + return snd_pcm_lib_free_vmalloc_buffer(substream); + else + return snd_pcm_lib_free_pages(substream); } /* @@ -1702,9 +1710,50 @@ static const struct snd_pcm_ops snd_usb_capture_ops = { .mmap = snd_pcm_lib_mmap_vmalloc, }; +static const struct snd_pcm_ops snd_usb_playback_dev_ops = { + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_usb_hw_params, + .hw_free = snd_usb_hw_free, + .prepare = snd_usb_pcm_prepare, + .trigger = snd_usb_substream_playback_trigger, + .pointer = snd_usb_pcm_pointer, + .page = snd_pcm_sgbuf_ops_page, +}; + +static const struct snd_pcm_ops snd_usb_capture_dev_ops = { + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_usb_hw_params, + .hw_free = snd_usb_hw_free, + .prepare = snd_usb_pcm_prepare, + .trigger = snd_usb_substream_capture_trigger, + .pointer = snd_usb_pcm_pointer, + .page = snd_pcm_sgbuf_ops_page, +}; + void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream) { - snd_pcm_set_ops(pcm, stream, - stream == SNDRV_PCM_STREAM_PLAYBACK ? - &snd_usb_playback_ops : &snd_usb_capture_ops); + const struct snd_pcm_ops *ops; + + if (snd_usb_use_vmalloc) + ops = stream == SNDRV_PCM_STREAM_PLAYBACK ? + &snd_usb_playback_ops : &snd_usb_capture_ops; + else + ops = stream == SNDRV_PCM_STREAM_PLAYBACK ? + &snd_usb_playback_dev_ops : &snd_usb_capture_dev_ops; + snd_pcm_set_ops(pcm, stream, ops); +} + +void snd_usb_preallocate_buffer(struct snd_usb_substream *subs) +{ + struct snd_pcm *pcm = subs->stream->pcm; + struct snd_pcm_substream *s = pcm->streams[subs->direction].substream; + struct device *dev = subs->dev->bus->controller; + + if (!snd_usb_use_vmalloc) + snd_pcm_lib_preallocate_pages(s, SNDRV_DMA_TYPE_DEV_SG, + dev, 64*1024, 512*1024); } diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index 35740d5ef268..f77ec58bf1a1 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -10,6 +10,7 @@ void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream); int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt); +void snd_usb_preallocate_buffer(struct snd_usb_substream *subs); #endif /* __USBAUDIO_PCM_H */ diff --git a/sound/usb/stream.c b/sound/usb/stream.c index d16e1c23f4e9..729afd808cc4 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -106,6 +106,8 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->ep_num = fp->endpoint; if (fp->channels > subs->channels_max) subs->channels_max = fp->channels; + + snd_usb_preallocate_buffer(subs); } /* kctl callbacks for usb-audio channel maps */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 7b28cbde22c0..b9faeca645fd 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -127,4 +127,6 @@ struct snd_usb_audio_quirk { int snd_usb_lock_shutdown(struct snd_usb_audio *chip); void snd_usb_unlock_shutdown(struct snd_usb_audio *chip); +extern bool snd_usb_use_vmalloc; + #endif /* __USBAUDIO_H */ -- cgit v1.2.3-59-g8ed1b