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[alsa-devel] [Minutes] ELCE Audio mini conf.
by Liam Girdwood 01 Nov '15

01 Nov '15

I've written up the minutes here below (including Vinod's minutes after I left for my flight). Please do feel free to correct or add any missing items. If you intend to reply and discuss a topic in detail then it would probably be helpful to include the topic in the $SUBJECT (so that it's easy to track multiple topics that may arise from the minutes). Jaroslav, I have some PDF slides from a few delegates. Can I post them on to the Wiki ? This failed last time I tried after the Edinburgh audio conference (Wiki could not upload PDFs ??). Soundwire: Pierre ================= - Slides to be posted. - Shadow registers required for regamp. Mark: need to make sure we can commit changes, and rules for witting. Reuse of codec drivers since reg mapping should be similar to i2c regmaps. - Some devices wont support run time clock changes. Need compatible devices on bus. - Each device has unique ID set by HW. Device enumeration can be auto discovered. Can detect device but not capabilities. - Device capabilities can come from hard coded driver data or ACPI/DT data. Windows may use inf files for capabilities. - Define bus api for master "controller". - Grouped triggers can be used for start duplex streams for similtanious capture and playback. - DAI is physical connection, logical links need to be worked out with Routing config Mark: Redefining of ports/slots at runtime i.e. Leave up to userspace for handling supported configurations of ports. Energy efficiency: Alexander (Minutes also supplied by Alexander). ================================================================== Power consumption of a SandyBridge-based Sony laptop has been measured with a workload that does only audio playback. To ensure a reproducible workload, the whole system has been constructed by a script using debootstrap, and placed into an initramfs. Various versions of PulseAudio were compared to CRAS and to the use of ALSA hw device, at latencies 7 ms, 28 ms and 448 ms. Even in the best case (ALSA), going from 448 to 28 ms means using extra 300 mW. For PulseAudio, the overhead of using low latency is even higher, presumably because of a chatty protocol. In version 7, there was some work (srbchannel) with the aim to reduce this overhead, but there is still more way to go. There was a suggestion to use perf to analyze the problem further. It should be noted that latencies as high as 448 ms are not usable without the ability to rewind, which PulseAudio posesses (at the cost of more complex, and actually incorrect, code) and CRAS doesn't. So CRAS decision to never do rewinds costs it 300 mW (448 ms latency is usable only for benchmarking purposes). Note: at least two people in the audience misunderstood this statement as originally formulated (thinko?), and got it as "PulseAudio decision to do rewinds costs it 300 mW", which is of course wrong. The cost of rewinds is in code complexity, real user-reported bugs and "how to test this code path" questions from developers, which, in the opinion of Alexander Patrakov, may outweigh the gains in energy efficiency (Arun Raghavan disagrees). A proposal was made by P.-L. Bossart to introduce a hw param that the application would set if it intends to never do rewinds, with the implication that a SkyLake driver can use this information for codec-side power saving (by prefetching audio data and powering the link off). Arun Raghavan (presumably due to the above misunderstanding) disagreed and instead proposed to make the device non-rewindable if this means power savings. Alexander is in favor of the hw param, because it would be useful for CRAS, but harmful for the current PulseAudio (it would force abandoning high latency). A future version of PulseAudio could, though, use a separate DSP stream for a music application (if there is only one), and, because a new application never appears, and the hardware volume is precise enough, there would never be a need to rewind. This, though, would need eliminating the support for rewinds from the client API (the last two parameters for pa_stream_write()), which can be done by ignoring these parameters. Documentation: Takashi ====================== - Patches that introduce new functionality should also include documentation. - Caretaker required for docs. Who ? - Uploading materials, in central location, ask kernel.org. - Charles and Rakesh voluntered to write some docs. Alsa-lib: release schedule: Vinod. ================================== - Request for more frequent release. Release inline with 1 - 2 kernel releases. - Integrate tinycompress into alsa release. Testing/QA: Liam, Takashi, Mark, Pierre ======================================= - BAT tool: Now upstream, WiP for review comments docs and new features. - Mark doing brute force tool for kcontrol testing. Kcontrol status can be watched via debubFS. Multiple threads. - Takashi: Need unit testing codec drivers. Some codec vendors have tools for this but something generic would be good. QA unit test could use topology readback to check topology in userspace. - Pierre: Latency testing, USB device to toggle GPIO (to indicate stream start) and audio latency measured from this point. Available on 01.org latency tools. To be extended to BAT. HDA/gfx: relevant people not in room. ===================================== - Pierre: Hotplug DP over USB C required. Needs input from GFX folks. BATCH flag for USB: Arun. ========================= - Flag does not respond to reality, lets deprecate it. no users. - Dylan: need to know transfer size for CRAS (uses extra samples for buffering). - BATCH flag means period size transfers, applications that use new granularity API can ignore batch flag. Pierre: to implement. Splitting out controls: Takashi =============================== - Restricted access. Consensus to restrict access to some controls due to possibility of breaking HW at kernel level. i.e. prevent feeding digital Mic into HP amp to prevent speaker over heating. - Some clamping APIs already exist, but these should be extended to include DAPM. Simple card: ACPI support. Vinod ================================ - Clock drivers required alongside helper functions to help initialise DAI links etc. - Vinod to publish spec as RFC in Q4. - Doubts of using DMI names for differentiation. Topology: Liam, Mengdong ======================== - Proposals to use debugFS/ioctl/media controller for exporting topology data to userspace. Drawbacks with all three approaches, but are aimed at different end users i.e. debugFS aimed at kernel audio developers/integrators/testers and media aimed at userspace developers and end users. Will probably support both debugFS (or other kernel file) and media API. Mark to discuss at Media team in Seoul. ASoC internals to gather data for debugFS and Media API will be the same code so nothing blocking initial development. Tiny-fication. Keyon, Liam ========================== - IoT devices have tight memory footprints so audio kernel and userspace can be very limited. - Userspace is fine with tinyalsa. - Currently Keyon has worked on some IoT device options to disable ALSA functions. e.g. code saving of 13k for refinement, but some IoT options can break existing userpsace. - Suggestion is to research tinycompress API for general PCM usage. - Look at removing DAPM strings (will need topology dump tool to automate this though). - Some structures can also be shrunk. Android configuration consolidation =================================== - different configuration formats XML, ucm, This tool takes Android XML and converts to UCM - uses Sony Xperia - generates UCM file from XML - Android: Try to converge to a HAL, but diverges quickly. Quirks for modem etc cant be represented well. - Arun will send the link to his work HDA Restructuring: Vinod & Rakesh ================================= - Discussed HW and SW overview - Future HDA work and codec conversion discussion <dont have detailed notes here as I was talking> DPMST: Mengdong ================ - One DVI port can be used to transmit multiple independent streams. - Three display pipelines in HW, so at most 3 streams can be supported across all ports - Need to modify representation to have stream on device rather than per pin. - While moving the monitor can be incompatible so should be modified - display should tell us unplug/plug on move. Sound server reacts to it. - This notification thru Component fwk. - Various options are discussed. Needs more offline discussion on approach and to solve this issue ALSA Core Challenges: ====================== - ALSA Core locking is complicated. Core code is quite difficult to understand. - PCM linking makes things complex - Add documentation for locks. - Controls can be hidden in UI tools through iface_cards. - DPCM hidden PCMs should not be shown in usermode, hide them from Usermode Multi Channel ASoC Controls; ============================= - channel maps dont make sense for ASoC as asoc paths dont have channel information - conclusion: add channel maps on HDMI edge. - we can add multi-channel controls to the asoc Liam

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[alsa-devel] Applied "ASoC: topology: ABI - Rename dai_elems to pcm_elems in manifest" to the asoc tree
by Mark Brown 31 Oct '15

31 Oct '15

The patch ASoC: topology: ABI - Rename dai_elems to pcm_elems in manifest has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From bbb64300469d207f62c08ea7358c839991a52877 Mon Sep 17 00:00:00 2001 From: Mengdong Lin <mengdong.lin(a)intel.com> Date: Fri, 30 Oct 2015 15:13:16 +0800 Subject: [PATCH] ASoC: topology: ABI - Rename dai_elems to pcm_elems in manifest This field is the number of PCM objects (a pair of FE DAI and DAI link). Signed-off-by: Lin Mengdong <mengdong.lin(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/uapi/sound/asoc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 26539a7..c4cc1e4 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -243,7 +243,7 @@ struct snd_soc_tplg_manifest { __le32 control_elems; /* number of control elements */ __le32 widget_elems; /* number of widget elements */ __le32 graph_elems; /* number of graph elements */ - __le32 dai_elems; /* number of DAI elements */ + __le32 pcm_elems; /* number of PCM elements */ __le32 dai_link_elems; /* number of DAI link elements */ struct snd_soc_tplg_private priv; } __attribute__((packed)); -- 2.6.1

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Re: [alsa-devel] [PATCH 2/2] sound: soc: Add Cygnus audio driver
by Arnd Bergmann 31 Oct '15

31 Oct '15

On Saturday 31 October 2015 07:20:17 kbuild test robot wrote: > > 785 dev_dbg(rtd->cpu_dai->dev, "%s: size 0x%x @ 0x%p\n", > 786 __func__, size, buf->area); > Both reports are for the same bug: the "0x%x" should be "0x%zx". There is another bug in the same line: "0x%p" should be "%pK", to avoid the duplicate "0x" and to not leak the pointer when kptr_restrict is set. Arnd

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[alsa-devel] [PATCH 0/2] Add audio support for Broadcom Cygnus SoC
by Simran Rai 30 Oct '15

30 Oct '15

Hi, This patchset contains audio support for Broadcom's Cygnus SoC. It contains DT bindings, core audio driver and dts. The audio driver supports both capture and playback of Audio PCM samples over I2S/TDM interface and provides playback support over SPDIF interface. This patchset is derived from a previously submitted patchset: http://lkml.iu.edu/hypermail/linux/kernel/1503.3/05434.html This patchset has been tested on Cygnus wireless audio bcm958305K board. It is based on v4.3-rc5 and is available from github: repo: https://github.com/Broadcom/cygnus-linux/tree/cygnus-sound-v1 Simran Rai (2): dt-bindings: sound: Add DT bindings for Broadcom Cygnus audio sound: soc: Add Broadcom Cygnus audio driver. .../bindings/sound/brcm,cygnus-audio.txt | 63 + sound/soc/bcm/Kconfig | 18 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 903 ++++++++++++ sound/soc/bcm/cygnus-ssp.c | 1532 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 129 ++ 6 files changed, 2650 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h -- 1.7.9.5

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[alsa-devel] [PATCH 0/3] ALSA: firewire-digi00x: add another rawmidi character device for MIDI control ports
by Takashi Sakamoto 30 Oct '15

30 Oct '15

Hi, Digi 002/003 family uses two ways of MIDI transmission; by isochronous packet streaming and by asynchronous transmission. The former is for physical MIDI ports and the latter is for physical controls on device surface. Currently, all of these ports are handled by a set of operations in one ALSA rawmidi character device, thus: $ amidi -l Dir Device Name IO hw:1,0,0 Digi 002Rack control IO hw:1,0,1 Digi 002Rack MIDI 1 O hw:1,0,2 Digi 002Rack MIDI 2 Although, it's preferable to have two rawmidi character devices because these transmission ways are completely different and it's better to assign each set of operations. This patchset adds another rawmidi character device to ALSA digi00x driver. As a result, the driver register two rawmidi character device for each MIDI ports: $ amidi -l Dir Device Name IO hw:1,0,0 Digi 002Rack MIDI 1 O hw:1,0,1 Digi 002Rack MIDI 2 IO hw:1,1,0 Digi 002Rack control I'm sorry to post this patchset in this time, but next release is a first version with digi00x driver and it better to includes this intrusive patchset before releasing. Takashi Sakamoto (3): ALSA: firewire-digi00x: rename identifiers of MIDI operation for physical ports ALSA: firewire-digi00x: add MIDI operations for MIDI control port ALSA: firewire-digi00x: add another rawmidi character device for MIDI control ports sound/firewire/digi00x/digi00x-midi.c | 197 ++++++++++++++++++++++------------ 1 file changed, 130 insertions(+), 67 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH v2 0/3] Add support for Cirrus Logic CS47L24 and WM1831 codecs
by Richard Fitzgerald 30 Oct '15

30 Oct '15

mfd patch based on Lee Jones's for-next gpio patch based on Linus Walleij's for-next ASoC patch based on Mark Brown's for-next Lee/Mark - can one of you take all three patches (with maintainer acks) so that the patch set is kept together? Richard Fitzgerald (3): mfd: arizona: Support Cirrus Logic CS47L24 and WM1831 gpio: arizona: Support Cirrus Logic CS47L24 and WM1831 ASoC: cs47l24: Add driver for Cirrus Logic CS47L24 and WM1831 codecs Documentation/devicetree/bindings/mfd/arizona.txt | 4 +- drivers/gpio/gpio-arizona.c | 4 + drivers/mfd/Kconfig | 19 +- drivers/mfd/Makefile | 3 + drivers/mfd/arizona-core.c | 74 +- drivers/mfd/arizona-irq.c | 40 +- drivers/mfd/arizona-spi.c | 7 + drivers/mfd/arizona.h | 4 + drivers/mfd/cs47l24-tables.c | 1629 +++++++++++++++++++++ include/linux/mfd/arizona/core.h | 3 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs47l24.c | 1148 +++++++++++++++ sound/soc/codecs/cs47l24.h | 23 + 14 files changed, 2942 insertions(+), 26 deletions(-) create mode 100644 drivers/mfd/cs47l24-tables.c create mode 100644 sound/soc/codecs/cs47l24.c create mode 100644 sound/soc/codecs/cs47l24.h -- 1.9.1

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[alsa-devel] [PATCH alsa-lib] pcm: dmix: Handle slave PCM xrun and unexpected states properly
by Takashi Iwai 30 Oct '15

30 Oct '15

Currently, dmix & co plugins ignore the XRUN state of the slave PCM. It's (supposedly) because dmix deals with the PCM in a free-wheel mode, which is equivalent with XRUN. But, this difference (whether the correct freewheel or XRUN) should be done by the kernel, and we may have an XRUN state indeed (e.g. via xrun injection). This patch fixes this lack of behavior, to handle PCM xrun and does prepare when the slave PCM is in such a state. Also, the patch consolidates the prepare callback for all dmix, dsnoop and dshare plugins, and fix/cleanup a bit for dshare/dsnoop codes to align with dsnoop code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 22 ++++++++++++++++++++++ src/pcm/pcm_direct.h | 3 +++ src/pcm/pcm_dmix.c | 16 +++------------- src/pcm/pcm_dshare.c | 22 +++++++--------------- src/pcm/pcm_dsnoop.c | 23 +++++++---------------- 5 files changed, 42 insertions(+), 44 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 195fddf06cda..fd3877c30d04 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -807,6 +807,28 @@ int snd_pcm_direct_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map) return snd_pcm_set_chmap(dmix->spcm, map); } +int snd_pcm_direct_prepare(snd_pcm_t *pcm) +{ + snd_pcm_direct_t *dmix = pcm->private_data; + int err; + + switch (snd_pcm_state(dmix->spcm)) { + case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: + case SND_PCM_STATE_DISCONNECTED: + err = snd_pcm_prepare(dmix->spcm); + if (err < 0) + return err; + snd_pcm_start(dmix->spcm); + break; + } + snd_pcm_direct_check_interleave(dmix, pcm); + dmix->state = SND_PCM_STATE_PREPARED; + dmix->appl_ptr = dmix->last_appl_ptr = 0; + dmix->hw_ptr = 0; + return snd_pcm_direct_set_timer_params(dmix); +} + int snd_pcm_direct_resume(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 9b1ddbcf424a..611ad29a4821 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -224,6 +224,8 @@ struct snd_pcm_direct { snd1_pcm_direct_mmap #define snd_pcm_direct_munmap \ snd1_pcm_direct_munmap +#define snd_pcm_direct_prepare \ + snd1_pcm_direct_prepare #define snd_pcm_direct_resume \ snd1_pcm_direct_resume #define snd_pcm_direct_timer_stop \ @@ -304,6 +306,7 @@ int snd_pcm_direct_sw_params(snd_pcm_t *pcm, snd_pcm_sw_params_t * params); int snd_pcm_direct_channel_info(snd_pcm_t *pcm, snd_pcm_channel_info_t * info); int snd_pcm_direct_mmap(snd_pcm_t *pcm); int snd_pcm_direct_munmap(snd_pcm_t *pcm); +int snd_pcm_direct_prepare(snd_pcm_t *pcm); int snd_pcm_direct_resume(snd_pcm_t *pcm); int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix); void snd_pcm_direct_clear_timer_queue(snd_pcm_direct_t *dmix); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 58e4975d5225..b26a5c790e7e 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -450,9 +450,10 @@ static snd_pcm_state_t snd_pcm_dmix_state(snd_pcm_t *pcm) snd_pcm_state_t state; state = snd_pcm_state(dmix->spcm); switch (state) { + case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: - return state; case SND_PCM_STATE_DISCONNECTED: + dmix->state = state; return state; default: break; @@ -533,17 +534,6 @@ static int snd_pcm_dmix_hwsync(snd_pcm_t *pcm) } } -static int snd_pcm_dmix_prepare(snd_pcm_t *pcm) -{ - snd_pcm_direct_t *dmix = pcm->private_data; - - snd_pcm_direct_check_interleave(dmix, pcm); - dmix->state = SND_PCM_STATE_PREPARED; - dmix->appl_ptr = dmix->last_appl_ptr = 0; - dmix->hw_ptr = 0; - return snd_pcm_direct_set_timer_params(dmix); -} - static void reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) { dmix->slave_appl_ptr = dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; @@ -920,7 +910,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dmix_fast_ops = { .state = snd_pcm_dmix_state, .hwsync = snd_pcm_dmix_hwsync, .delay = snd_pcm_dmix_delay, - .prepare = snd_pcm_dmix_prepare, + .prepare = snd_pcm_direct_prepare, .reset = snd_pcm_dmix_reset, .start = snd_pcm_dmix_start, .drop = snd_pcm_dmix_drop, diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 02370dc7082d..58e47bbeac67 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -237,11 +237,14 @@ static int snd_pcm_dshare_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dshare_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; - switch (snd_pcm_state(dshare->spcm)) { + snd_pcm_state_t state; + state = snd_pcm_state(dshare->spcm); + switch (state) { + case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: - return SND_PCM_STATE_SUSPENDED; case SND_PCM_STATE_DISCONNECTED: - return SND_PCM_STATE_DISCONNECTED; + dshare->state = state; + return state; default: break; } @@ -296,17 +299,6 @@ static int snd_pcm_dshare_hwsync(snd_pcm_t *pcm) } } -static int snd_pcm_dshare_prepare(snd_pcm_t *pcm) -{ - snd_pcm_direct_t *dshare = pcm->private_data; - - snd_pcm_direct_check_interleave(dshare, pcm); - dshare->state = SND_PCM_STATE_PREPARED; - dshare->appl_ptr = dshare->last_appl_ptr = 0; - dshare->hw_ptr = 0; - return snd_pcm_direct_set_timer_params(dshare); -} - static int snd_pcm_dshare_reset(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; @@ -595,7 +587,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dshare_fast_ops = { .state = snd_pcm_dshare_state, .hwsync = snd_pcm_dshare_hwsync, .delay = snd_pcm_dshare_delay, - .prepare = snd_pcm_dshare_prepare, + .prepare = snd_pcm_direct_prepare, .reset = snd_pcm_dshare_reset, .start = snd_pcm_dshare_start, .drop = snd_pcm_dshare_drop, diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 8a2e87ad0ae1..576c35b111cd 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -201,12 +201,14 @@ static int snd_pcm_dsnoop_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dsnoop_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dsnoop = pcm->private_data; - switch (snd_pcm_state(dsnoop->spcm)) { + snd_pcm_state_t state; + state = snd_pcm_state(dsnoop->spcm); + switch (state) { + case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: - return SND_PCM_STATE_SUSPENDED; case SND_PCM_STATE_DISCONNECTED: - dsnoop->state = SNDRV_PCM_STATE_DISCONNECTED; - return -ENODEV; + dsnoop->state = state; + return state; default: break; } @@ -258,17 +260,6 @@ static int snd_pcm_dsnoop_hwsync(snd_pcm_t *pcm) } } -static int snd_pcm_dsnoop_prepare(snd_pcm_t *pcm) -{ - snd_pcm_direct_t *dsnoop = pcm->private_data; - - snd_pcm_direct_check_interleave(dsnoop, pcm); - dsnoop->state = SND_PCM_STATE_PREPARED; - dsnoop->appl_ptr = 0; - dsnoop->hw_ptr = 0; - return snd_pcm_direct_set_timer_params(dsnoop); -} - static int snd_pcm_dsnoop_reset(snd_pcm_t *pcm) { snd_pcm_direct_t *dsnoop = pcm->private_data; @@ -497,7 +488,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dsnoop_fast_ops = { .state = snd_pcm_dsnoop_state, .hwsync = snd_pcm_dsnoop_hwsync, .delay = snd_pcm_dsnoop_delay, - .prepare = snd_pcm_dsnoop_prepare, + .prepare = snd_pcm_direct_prepare, .reset = snd_pcm_dsnoop_reset, .start = snd_pcm_dsnoop_start, .drop = snd_pcm_dsnoop_drop, -- 2.6.2

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[alsa-devel] [PATCH 0/4] dmaengine: Add transfer termination synchronization support
by Lars-Peter Clausen 30 Oct '15

30 Oct '15

The DMAengine API has a long standing issue that is inherent to the API itself. For a client that calls dmaengine_terminate_all() it is not possible to properly synchronize the completion of any currently running complete callbacks to the current context. This means it is possible to end up with a use-after-free race condition if client frees resources that are accessed in a complete callback before the complete callback has finished running. This patch series introduces a new explicit synchronization primitive to the DMAengine API which allows clients to ensure that all complete callbacks have finished running. This allows them to safely free any resources that might be accessed in a complete callback. The series for now only implements synchronization support for a single driver and only updates single client to make use of the new API. If there is agreement on the general approach more will follow. - Lars Lars-Peter Clausen (4): dmaengine: Add transfer termination synchronization support dmaengine: virt-dma: Add synchronization helper function dmaengine: axi_dmac: Add synchronization support ALSA: pcm_dmaengine: Properly synchronize DMA on shutdown Documentation/dmaengine/client.txt | 38 ++++++++++++++- Documentation/dmaengine/provider.txt | 20 +++++++- drivers/dma/dma-axi-dmac.c | 8 ++++ drivers/dma/dmaengine.c | 5 +- drivers/dma/virt-dma.h | 13 ++++++ include/linux/dmaengine.h | 90 ++++++++++++++++++++++++++++++++++++ sound/core/pcm_dmaengine.c | 9 ++-- 7 files changed, 175 insertions(+), 8 deletions(-) -- 2.1.4

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Re: [alsa-devel] ASoC: Intel: Add Skylake HDA platform driver
by Vinod Koul 30 Oct '15

30 Oct '15

> > Hello Jeeja KP, > > > > This is a semi-automatic email about new static checker warnings. > > > > The patch a40e693c7f5e: "ASoC: Intel: Add Skylake HDA platform driver" > > from Jul 9, 2015, leads to the following Smatch complaint: > > > > sound/soc/intel/skylake/skl-pcm.c:802 skl_get_position() > > warn: variable dereferenced before check 'substream->runtime' (see > > line 792) > > > > sound/soc/intel/skylake/skl-pcm.c > > 791 struct snd_pcm_substream *substream = hstr->substream; > > 792 struct hdac_ext_bus *ebus = get_bus_ctx(substream); > > ^^^^^^^^^^^^^^^^^^^^^^ Dereference inside > > function. > > > > 793 unsigned int pos; > > 794 int delay; > > 795 > > 796 /* use the position buffer as default */ > > 797 pos = > > snd_hdac_stream_get_pos_posbuf(hdac_stream(hstream)); > > 798 > > 799 if (pos >= hdac_stream(hstream)->bufsize) > > 800 pos = 0; > > 801 > > 802 if (substream->runtime) { > > ^^^^^^^^^^^^^^^^^^ > > Checked too late. Right, will move ebus init here. Thanks for reporting this > > > > 803 delay = skl_get_delay_from_lpib(ebus, hstream, pos) > > 804 + codec_delay; > > > > regards, > > dan carpenter -- ~Vinod

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[alsa-devel] [PATCH 1/1] BAT: Add default device name for playback and capture
by han.lu＠intel.com 30 Oct '15

30 Oct '15

From: "Lu, Han" <han.lu(a)intel.com> If the playback device and the capture device was not set through '-D', '-P' or '-C' options, set the name of the two devices to "default". So by default, BAT will work through PulseAudio sound server on Ubuntu/Debian, and work though CRAS sound server on Chromium OS/Chrome OS. If the playback device or the capture device was set through '-D', '-P' or '-C' options by user, the device name "default" will not be set. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/bat/bat.c b/bat/bat.c index 086b9fa..4310be0 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -473,6 +473,10 @@ static int bat_init(struct bat *bat) return err; } + /* Set default playback and capture devices */ + if (bat->playback.device == NULL && bat->capture.device == NULL) + bat->playback.device = bat->capture.device = "default"; + /* Determine capture file */ if (bat->local) { bat->capture.file = bat->playback.file; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: fix typo in sizeof
by Vincent Stehlé 30 Oct '15

30 Oct '15

The size of the pointer to a data structure to send is erroneously passed to sst_ipc_tx_message_wait() as its tx_bytes argument. It should be given the size of the pointed skl_ipc_dxstate_info structure instead. Coincidentally, both the pointer and the structure have the same size of 8 bytes on a 64 bit machine, which "masks" the issue. Compiling for 32 bit reveals the issue more clearly. Fix the typo for correctness, and to make the code robust to future evolutions of the skl_ipc_dxstate_info structure size. This fixes the following coccicheck error: sound/soc/intel/skylake/skl-sst-ipc.c:641:8-14: ERROR: application of sizeof to pointer Signed-off-by: Vincent Stehlé <vincent.stehle(a)laposte.net> Cc: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Cc: Jeeja KP <jeeja.kp(a)intel.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: trivial(a)kernel.org --- sound/soc/intel/skylake/skl-sst-ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 937a0a3..9aee835 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -638,7 +638,7 @@ int skl_ipc_set_dx(struct sst_generic_ipc *ipc, u8 instance_id, dev_dbg(ipc->dev, "In %s primary =%x ext=%x\n", __func__, header.primary, header.extension); ret = sst_ipc_tx_message_wait(ipc, *ipc_header, - dx, sizeof(dx), NULL, 0); + dx, sizeof(*dx), NULL, 0); if (ret < 0) { dev_err(ipc->dev, "ipc: set dx failed, err %d\n", ret); return ret; -- 2.6.1

3 3

[alsa-devel] [PATCH 00/17] ASoC: Intel: Skylake: More fixes and updates
by Vinod Koul 30 Oct '15

30 Oct '15

Hi all, This series adds more minor fixes and some enhancements to the driver. Most of the fixes patches are single line count changes The enhancements are: - optimizing code and using module pin info for unbinding - supporting virtual dsp widgets like switches in topology - add pin formats and supports multiple heterogeneous pins Hardik T Shah (3): ASoC: Intel: Skylake: Add multiple pin formats ASoC: Intel: Skylake: Update the topology interface structure ASoC: Intel: Skylake: Add support for module GUIDs Jeeja KP (14): ASoC: Intel: Skylake: Fix to correct check for non DSP widget ASoC: Intel: Skylake: Fix not to ignore return value in be hw_params ASoC: Intel: Skylake: Fix to add 32 bit in update FE params ASoC: Intel: Skylake: Fix to ignore codec_mask check in probe ASoC: Intel: Skylake: Fix to ignore blob check if link type is HDA ASoC: Intel: Skylake: Fix support for multiple pins in a module ASoC: Intel: Skylake: Fix bit depth when querying the NHLT blob ASoC: Intel: Skylake: use module_pin info for unbind ASoC: Intel: Skylake: Add support for virtual dsp widgets ASoC: Intel: Skylake: Fix DSP pipe underrun/overrun issue ASoC: Intel: Skylake: Fix to remove be copier widget power check ASoC: Intel: Skylake: Ignore rate check for DMIC link ASoC: Intel: Skylake: Fix to remove channel_map calculation ASoC: Intel: Skylake: Fix PM behaviour sound/soc/intel/skylake/skl-messages.c | 189 +++++-------- sound/soc/intel/skylake/skl-nhlt.c | 19 +- sound/soc/intel/skylake/skl-pcm.c | 139 +++++----- sound/soc/intel/skylake/skl-topology.c | 387 ++++++++++++++------------- sound/soc/intel/skylake/skl-topology.h | 35 ++- sound/soc/intel/skylake/skl-tplg-interface.h | 90 +++++-- sound/soc/intel/skylake/skl.c | 63 +++-- sound/soc/intel/skylake/skl.h | 1 - 8 files changed, 483 insertions(+), 440 deletions(-) -- 2.4.3

2 40

[alsa-devel] [PATCH] snd-cs46xx: Fix suspend for all channels
by Ondrej Zary 30 Oct '15

30 Oct '15

snd_cs46xx_suspend() is missing snd_pcm_suspend_all calls for pcm_rear, pcm_center_lfe and pcm_iec958. There's also a copy/paste bug in snd_cs46xx_pcm_iec958(), overwriting pcm_rear in struct snd_cs46xx, leaving pcm_iec958 unused. Fix all this to allow all channels to survive suspend/resume. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/cs46xx/cs46xx_lib.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index e62c841..2706f27 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -1864,7 +1864,7 @@ int snd_cs46xx_pcm_iec958(struct snd_cs46xx *chip, int device) /* global setup */ pcm->info_flags = 0; strcpy(pcm->name, "CS46xx - IEC958"); - chip->pcm_rear = pcm; + chip->pcm_iec958 = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), 64*1024, 256*1024); @@ -3780,6 +3780,11 @@ static int snd_cs46xx_suspend(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); chip->in_suspend = 1; snd_pcm_suspend_all(chip->pcm); +#ifdef CONFIG_SND_CS46XX_NEW_DSP + snd_pcm_suspend_all(chip->pcm_rear); + snd_pcm_suspend_all(chip->pcm_center_lfe); + snd_pcm_suspend_all(chip->pcm_iec958); +#endif // chip->ac97_powerdown = snd_cs46xx_codec_read(chip, AC97_POWER_CONTROL); // chip->ac97_general_purpose = snd_cs46xx_codec_read(chip, BA0_AC97_GENERAL_PURPOSE); -- Ondrej Zary

2 1

[alsa-devel] [PATCH] snd-cs46xx: Fix Duplicate front for CS4294 and CS4298 codecs
by Ondrej Zary 30 Oct '15

30 Oct '15

Currently, the Duplicate front control is enabled only for two specific revisions of CS4294. Enable it for all CS4294 and CS4298 codecs instead. This fixes rear output on my VC-4620N sound card (CS4620 + CS4298 rev.5). Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 2a9f4a3..e62c841 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2528,7 +2528,7 @@ int snd_cs46xx_mixer(struct snd_cs46xx *chip, int spdif_device) #ifdef CONFIG_SND_CS46XX_NEW_DSP if (chip->nr_ac97_codecs == 1) { unsigned int id2 = chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]->id & 0xffff; - if (id2 == 0x592b || id2 == 0x592d) { + if ((id2 & 0xfff0) == 0x5920) { /* CS4294 and CS4298 */ err = snd_ctl_add(card, snd_ctl_new1(&snd_cs46xx_front_dup_ctl, chip)); if (err < 0) return err; -- Ondrej Zary

2 1

[alsa-devel] Different reg_default for rt5650 codec
by Bard Liao 30 Oct '15

30 Oct '15

Mark, rt5645.c support both rt5645 and rt5650 codec. And the default value of registers are not identical. So, we may need separate reg_defaults for rt5645 and rt5650. My idea is to assign a new reg_defaults once rt5650 is detected. Something like case RT5650_DEVICE_ID: + rt5645->regmap-> reg_defaults = rt5650_reg; + rt5645->regmap-> num_reg_defaults = ARRAY_SIZE(rt5650_reg) rt5645->codec_type = CODEC_TYPE_RT5650; break; Is that ok? Or should I use different regmap_config? There is one question if I use different regmap_config. We will know which codec it is by regmap_read. So, regmap is init before we know which codec it is. Assume rt5645_regmap is used and what should I do once rt5650 is detected? Thanks. Bard Liao Computer Peripherals Business Unit Realtek Semiconductor Corp. 886-3-578-0211 ext. 13334 bardliao(a)realtek.com

2 2

[alsa-devel] Applied "- filename : ASoC-rsrc-card-Clarify-compatible-value.patch
by Mark Brown 30 Oct '15

30 Oct '15

- name : ASoC: rsrc-card: Clarify compatible value" to the asoc tree In-Reply-To: <1446148795-20629-1-git-send-email-geert+renesas(a)glider.be> The patch - filename : ASoC-rsrc-card-Clarify-compatible-value.patch - name : ASoC: rsrc-card: Clarify compatible value has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 33833b9af9ec89356fbfcd9ff0679b136f20e12f Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven <geert+renesas(a)glider.be> Date: Thu, 29 Oct 2015 20:59:55 +0100 Subject: [PATCH] ASoC: rsrc-card: Clarify compatible value The compatible value can be board-specific, not SoC-specific. Add curly braces to indicate that the board type is optional. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt index 962748a..2b2caa2 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt @@ -4,8 +4,8 @@ Renesas Sampling Rate Convert Sound Card specifies audio DAI connections of SoC Required properties: -- compatible : "renesas,rsrc-card,<board>" - Examples with soctypes are: +- compatible : "renesas,rsrc-card{,<board>}" + Examples with boards are: - "renesas,rsrc-card" - "renesas,rsrc-card,lager" - "renesas,rsrc-card,koelsch" -- 2.6.1

1 0

[alsa-devel] Applied "- filename : ASoC-rt5677-use-active-low-logic-for-reset-pin.patch
by Mark Brown 30 Oct '15

30 Oct '15

- name : ASoC: rt5677: use 'active low' logic for reset pin" to the asoc tree In-Reply-To: <1446157919-27151-1-git-send-email-anatol.pomozov(a)gmail.com> The patch - filename : ASoC-rt5677-use-active-low-logic-for-reset-pin.patch - name : ASoC: rt5677: use 'active low' logic for reset pin has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7f7d276ef2f76f781d9867653d22ee96fc86edf2 Mon Sep 17 00:00:00 2001 From: Anatol Pomozov <anatol.pomozov(a)gmail.com> Date: Thu, 29 Oct 2015 15:31:59 -0700 Subject: [PATCH] ASoC: rt5677: use 'active low' logic for reset pin According to the datasheet RESET is active low pin, i.e. system goes to reset state when pin signal is low. The previous implementeation was assuming the pin is configured as 'active high' in DTS. Changle the gpio handling code and DTS configuration to 'active low'. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rt5677.txt | 2 +- sound/soc/codecs/rt5677.c | 8 ++++---- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index f070789..1b3c13d 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -18,7 +18,7 @@ Required properties: Optional properties: - realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin. -- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. +- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. Active low. - realtek,in1-differential - realtek,in2-differential diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index b4cd7e3..f73fd12 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4766,7 +4766,7 @@ static int rt5677_remove(struct snd_soc_codec *codec) regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); - gpiod_set_value_cansleep(rt5677->reset_pin, 0); + gpiod_set_value_cansleep(rt5677->reset_pin, 1); return 0; } @@ -4781,7 +4781,7 @@ static int rt5677_suspend(struct snd_soc_codec *codec) regcache_mark_dirty(rt5677->regmap); gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); - gpiod_set_value_cansleep(rt5677->reset_pin, 0); + gpiod_set_value_cansleep(rt5677->reset_pin, 1); } return 0; @@ -4793,7 +4793,7 @@ static int rt5677_resume(struct snd_soc_codec *codec) if (!rt5677->dsp_vad_en) { gpiod_set_value_cansleep(rt5677->pow_ldo2, 1); - gpiod_set_value_cansleep(rt5677->reset_pin, 1); + gpiod_set_value_cansleep(rt5677->reset_pin, 0); if (rt5677->pow_ldo2 || rt5677->reset_pin) msleep(10); @@ -5138,7 +5138,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, return ret; } rt5677->reset_pin = devm_gpiod_get_optional(&i2c->dev, - "realtek,reset", GPIOD_OUT_HIGH); + "realtek,reset", GPIOD_OUT_LOW); if (IS_ERR(rt5677->reset_pin)) { ret = PTR_ERR(rt5677->reset_pin); dev_err(&i2c->dev, "Failed to request RESET: %d\n", ret); -- 2.6.1

1 0

[alsa-devel] Applied "- filename : ASoC-rsnd-audio_clkout0-1-2-3-are-optinal-properties.patch
by Mark Brown 30 Oct '15

30 Oct '15

- name : ASoC: rsnd: audio_clkout0/1/2/3 are optinal properties" to the asoc tree In-Reply-To: <871tce16oa.wl%kuninori.morimoto.gx(a)renesas.com> The patch - filename : ASoC-rsnd-audio_clkout0-1-2-3-are-optinal-properties.patch - name : ASoC: rsnd: audio_clkout0/1/2/3 are optinal properties has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 61af9ed7475090b79e9305a4c52c9d3adb2d1784 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 29 Oct 2015 07:42:40 +0000 Subject: [PATCH] ASoC: rsnd: audio_clkout0/1/2/3 are optional properties Renesas sound driver can output Audio-clkout0/1/2/3, but these are optional properties for each board. Reported-by: Geert Uytterhoeven <geert+renesas(a)glider.be> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index c57cbd6..bbcb327 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -34,6 +34,8 @@ Required properties: see below for detail. - #sound-dai-cells : it must be 0 if your system is using single DAI it must be 1 if your system is using multi DAI + +Optional properties: - #clock-cells : it must be 0 if your system has audio_clkout it must be 1 if your system has audio_clkout0/1/2/3 - clock-frequency : for all audio_clkout0/1/2/3 -- 2.6.1

1 0

[alsa-devel] [PATCH V2 00/10] Add support for Imagination Technologies audio controllers
by Damien Horsley 30 Oct '15

30 Oct '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add drivers and binding documents for the following Imagination Technologies audio controllers: I2S Input Controller I2S Output Controller Parallel Output Controller SPDIF Input Controller SPDIF Output Controller These controllers are used in Pistachio SoC Changes in V2: - Sparse warnings fixed Damien.Horsley (10): ASoC: img: Add binding document for I2S input controller ASoC: img: Add driver for I2S input controller ASoC: img: Add binding document for I2S output controller ASoC: img: Add driver for I2S output controller ASoC: img: Add binding document for parallel output controller ASoC: img: Add driver for parallel output controller ASoC: img: Add binding document for SPDIF input controller ASoC: img: Add driver for SPDIF input controller ASoC: img: Add binding document for SPDIF output controller ASoC: img: Add driver for SPDIF output controller .../devicetree/bindings/sound/img,i2s-in.txt | 45 ++ .../devicetree/bindings/sound/img,i2s-out.txt | 49 ++ .../devicetree/bindings/sound/img,parallel-out.txt | 44 ++ .../devicetree/bindings/sound/img,spdif-in.txt | 41 ++ .../devicetree/bindings/sound/img,spdif-out.txt | 44 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/img/Kconfig | 44 ++ sound/soc/img/Makefile | 5 + sound/soc/img/img-i2s-in.c | 508 +++++++++++++ sound/soc/img/img-i2s-out.c | 552 ++++++++++++++ sound/soc/img/img-parallel-out.c | 373 ++++++++++ sound/soc/img/img-spdif-in.c | 799 +++++++++++++++++++++ sound/soc/img/img-spdif-out.c | 433 +++++++++++ 14 files changed, 2939 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,i2s-in.txt create mode 100644 Documentation/devicetree/bindings/sound/img,i2s-out.txt create mode 100644 Documentation/devicetree/bindings/sound/img,parallel-out.txt create mode 100644 Documentation/devicetree/bindings/sound/img,spdif-in.txt create mode 100644 Documentation/devicetree/bindings/sound/img,spdif-out.txt create mode 100644 sound/soc/img/Kconfig create mode 100644 sound/soc/img/Makefile create mode 100644 sound/soc/img/img-i2s-in.c create mode 100644 sound/soc/img/img-i2s-out.c create mode 100644 sound/soc/img/img-parallel-out.c create mode 100644 sound/soc/img/img-spdif-in.c create mode 100644 sound/soc/img/img-spdif-out.c -- 2.1.4

2 27

[alsa-devel] [PATCH] ASoC: rsrc-card: Clarify compatible value
by Geert Uytterhoeven 30 Oct '15

30 Oct '15

The compatible value can be board-specific, not SoC-specific. Add curly braces to indicate that the board type is optional. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt index 962748a8d9194dee..2b2caa281ce38abc 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt @@ -4,8 +4,8 @@ Renesas Sampling Rate Convert Sound Card specifies audio DAI connections of SoC Required properties: -- compatible : "renesas,rsrc-card,<board>" - Examples with soctypes are: +- compatible : "renesas,rsrc-card{,<board>}" + Examples with boards are: - "renesas,rsrc-card" - "renesas,rsrc-card,lager" - "renesas,rsrc-card,koelsch" -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5677: use 'active low' logic for reset pin
by Anatol Pomozov 29 Oct '15

29 Oct '15

According to the datasheet RESET is active low pin, i.e. system goes to reset state when pin signal is low. The previous implementeation was assuming the pin is configured as 'active high' in DTS. Changle the gpio handling code and DTS configuration to 'active low'. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5677.txt | 2 +- sound/soc/codecs/rt5677.c | 8 ++++---- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index f070789..1b3c13d 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -18,7 +18,7 @@ Required properties: Optional properties: - realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin. -- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. +- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. Active low. - realtek,in1-differential - realtek,in2-differential diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index b4cd7e3..f73fd12 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4766,7 +4766,7 @@ static int rt5677_remove(struct snd_soc_codec *codec) regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); - gpiod_set_value_cansleep(rt5677->reset_pin, 0); + gpiod_set_value_cansleep(rt5677->reset_pin, 1); return 0; } @@ -4781,7 +4781,7 @@ static int rt5677_suspend(struct snd_soc_codec *codec) regcache_mark_dirty(rt5677->regmap); gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); - gpiod_set_value_cansleep(rt5677->reset_pin, 0); + gpiod_set_value_cansleep(rt5677->reset_pin, 1); } return 0; @@ -4793,7 +4793,7 @@ static int rt5677_resume(struct snd_soc_codec *codec) if (!rt5677->dsp_vad_en) { gpiod_set_value_cansleep(rt5677->pow_ldo2, 1); - gpiod_set_value_cansleep(rt5677->reset_pin, 1); + gpiod_set_value_cansleep(rt5677->reset_pin, 0); if (rt5677->pow_ldo2 || rt5677->reset_pin) msleep(10); @@ -5138,7 +5138,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, return ret; } rt5677->reset_pin = devm_gpiod_get_optional(&i2c->dev, - "realtek,reset", GPIOD_OUT_HIGH); + "realtek,reset", GPIOD_OUT_LOW); if (IS_ERR(rt5677->reset_pin)) { ret = PTR_ERR(rt5677->reset_pin); dev_err(&i2c->dev, "Failed to request RESET: %d\n", ret); -- 2.6.0.rc2.230.g3dd15c0

1 0

[alsa-devel] Unknown PCM cards.pcm.front
by Rick Mann 29 Oct '15

29 Oct '15

In my custom cape for Beaglebone black, I'm able to play audio reasonably well. But for one situation when my C++ app (which plays via libao) opens an output, this is written to my app's output: "ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.front" Googling, I found this advice: http://alexpb.com/notes/articles/2012/11/14/error-when-playing-audio-on-ras… Which works. My question is, why do I get it only in this instance? Is it because I might be opening the audio device twice simultaneously? That's about the only thing that might set this situation apart from all the others. TIA, -- Rick Mann rmann(a)latencyzero.com

2 2

[alsa-devel] print audio ring buffer in kernel space
by Ankhit Vivekananda 29 Oct '15

29 Oct '15

Hello, I am trying to print the ring buffer used for DMA transfer using ALSA in kernel space. I was able to print the ring buffer in user space after calling the snd_pcm_readi() function. But, wanted to know as to which file I need to be looking at to print the hardware buffer in kernel space as I am trying to debug capturing audio using ALSA. I am using a MCBSP I2S on TM320DM365 processor and CQ93 on chip voice codec driver. Appreciate any help as I am new to ALSA. Thank you.

2 2

[alsa-devel] [PATCH v3 1/2] ASoC: wm9713: convert to regmap
by Robert Jarzmik 29 Oct '15

29 Oct '15

Convert the Wolfson WM9713 to regmap API. This will leverage all the regmap functions (debug, registers update, etc ...). As a bonus, this will pave the path to gpio chip introduction, and devicetree support. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v1: fix suspend/resume (that specific part is not tested yet) Since v2: split out the snd_soc_*() from regmap support --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/wm9713.c | 182 ++++++++++++++++++++++++++++------------------ 2 files changed, 114 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733ecd17f..ba306f8f3717 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -879,6 +879,7 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate + select REGMAP_AC97 # Amp config SND_SOC_LM4857 diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 4083a5130cbd..8e2cc1112e40 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -19,6 +19,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <linux/device.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> @@ -39,33 +40,15 @@ struct wm9713_priv { struct mutex lock; }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg); -static int ac97_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int val); - -/* - * WM9713 register cache - * Reg 0x3c bit 15 is used by touch driver. - */ -static const u16 wm9713_reg[] = { - 0x6174, 0x8080, 0x8080, 0x8080, - 0xc880, 0xe808, 0xe808, 0x0808, - 0x00da, 0x8000, 0xd600, 0xaaa0, - 0xaaa0, 0xaaa0, 0x0000, 0x0000, - 0x0f0f, 0x0040, 0x0000, 0x7f00, - 0x0405, 0x0410, 0xbb80, 0xbb80, - 0x0000, 0xbb80, 0x0000, 0x4523, - 0x0000, 0x2000, 0x7eff, 0xffff, - 0x0000, 0x0000, 0x0080, 0x0000, - 0x0000, 0x0000, 0xfffe, 0xffff, - 0x0000, 0x0000, 0x0000, 0xfffe, - 0x4000, 0x0000, 0x0000, 0x0000, - 0xb032, 0x3e00, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0006, - 0x0001, 0x0000, 0x574d, 0x4c13, -}; +static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + return snd_soc_read(codec, reg); +} +static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + return snd_soc_write(codec, reg, val); +} #define HPL_MIXER 0 #define HPR_MIXER 1 @@ -674,39 +657,97 @@ static const struct snd_soc_dapm_route wm9713_audio_map[] = { {"Capture Mono Mux", "Right", "Right Capture Source"}, }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg) +static bool wm9713_readable_reg(struct device *dev, unsigned int reg) { - struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || - reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || - reg == AC97_CD) - return soc_ac97_ops->read(wm9713->ac97, reg); - else { - reg = reg >> 1; - - if (reg >= (ARRAY_SIZE(wm9713_reg))) - return -EIO; - - return cache[reg]; + switch (reg) { + case AC97_RESET ... AC97_PCM_SURR_DAC_RATE: + case AC97_PCM_LR_ADC_RATE: + case AC97_CENTER_LFE_MASTER: + case AC97_SPDIF ... AC97_LINE1_LEVEL: + case AC97_GPIO_CFG ... 0x5c: + case AC97_CODEC_CLASS_REV ... AC97_PCI_SID: + case 0x74 ... AC97_VENDOR_ID2: + return true; + default: + return false; } } -static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) +static bool wm9713_writeable_reg(struct device *dev, unsigned int reg) { - struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); + switch (reg) { + case AC97_VENDOR_ID1: + case AC97_VENDOR_ID2: + return false; + default: + return wm9713_readable_reg(dev, reg); + } +} - u16 *cache = codec->reg_cache; - soc_ac97_ops->write(wm9713->ac97, reg, val); - reg = reg >> 1; - if (reg < (ARRAY_SIZE(wm9713_reg))) - cache[reg] = val; +static const struct reg_default wm9713_reg_defaults[] = { + { 0x02, 0x8080 }, /* Speaker Output Volume */ + { 0x04, 0x8080 }, /* Headphone Output Volume */ + { 0x06, 0x8080 }, /* Out3/OUT4 Volume */ + { 0x08, 0xc880 }, /* Mono Volume */ + { 0x0a, 0xe808 }, /* LINEIN Volume */ + { 0x0c, 0xe808 }, /* DAC PGA Volume */ + { 0x0e, 0x0808 }, /* MIC PGA Volume */ + { 0x10, 0x00da }, /* MIC Routing Control */ + { 0x12, 0x8000 }, /* Record PGA Volume */ + { 0x14, 0xd600 }, /* Record Routing */ + { 0x16, 0xaaa0 }, /* PCBEEP Volume */ + { 0x18, 0xaaa0 }, /* VxDAC Volume */ + { 0x1a, 0xaaa0 }, /* AUXDAC Volume */ + { 0x1c, 0x0000 }, /* Output PGA Mux */ + { 0x1e, 0x0000 }, /* DAC 3D control */ + { 0x20, 0x0f0f }, /* DAC Tone Control*/ + { 0x22, 0x0040 }, /* MIC Input Select & Bias */ + { 0x24, 0x0000 }, /* Output Volume Mapping & Jack */ + { 0x26, 0x7f00 }, /* Powerdown Ctrl/Stat*/ + { 0x28, 0x0405 }, /* Extended Audio ID */ + { 0x2a, 0x0410 }, /* Extended Audio Start/Ctrl */ + { 0x2c, 0xbb80 }, /* Audio DACs Sample Rate */ + { 0x2e, 0xbb80 }, /* AUXDAC Sample Rate */ + { 0x32, 0xbb80 }, /* Audio ADCs Sample Rate */ + { 0x36, 0x4523 }, /* PCM codec control */ + { 0x3a, 0x2000 }, /* SPDIF control */ + { 0x3c, 0xfdff }, /* Powerdown 1 */ + { 0x3e, 0xffff }, /* Powerdown 2 */ + { 0x40, 0x0000 }, /* General Purpose */ + { 0x42, 0x0000 }, /* Fast Power-Up Control */ + { 0x44, 0x0080 }, /* MCLK/PLL Control */ + { 0x46, 0x0000 }, /* MCLK/PLL Control */ + { 0x4c, 0xfffe }, /* GPIO Pin Configuration */ + { 0x4e, 0xffff }, /* GPIO Pin Polarity / Type */ + { 0x50, 0x0000 }, /* GPIO Pin Sticky */ + { 0x52, 0x0000 }, /* GPIO Pin Wake-Up */ + /* GPIO Pin Status */ + { 0x56, 0xfffe }, /* GPIO Pin Sharing */ + { 0x58, 0x4000 }, /* GPIO PullUp/PullDown */ + { 0x5a, 0x0000 }, /* Additional Functions 1 */ + { 0x5c, 0x0000 }, /* Additional Functions 2 */ + { 0x60, 0xb032 }, /* ALC Control */ + { 0x62, 0x3e00 }, /* ALC / Noise Gate Control */ + { 0x64, 0x0000 }, /* AUXDAC input control */ + { 0x74, 0x0000 }, /* Digitiser Reg 1 */ + { 0x76, 0x0006 }, /* Digitiser Reg 2 */ + { 0x78, 0x0001 }, /* Digitiser Reg 3 */ + { 0x7a, 0x0000 }, /* Digitiser Read Back */ +}; - return 0; -} +static const struct regmap_config wm9713_regmap_config = { + .reg_bits = 16, + .reg_stride = 2, + .val_bits = 16, + .max_register = 0x7e, + .cache_type = REGCACHE_RBTREE, + + .reg_defaults = wm9713_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm9713_reg_defaults), + .volatile_reg = regmap_ac97_default_volatile, + .readable_reg = wm9713_readable_reg, + .writeable_reg = wm9713_writeable_reg, +}; /* PLL divisors */ struct _pll_div { @@ -1158,6 +1199,9 @@ static int wm9713_soc_suspend(struct snd_soc_codec *codec) { u16 reg; + snd_soc_cache_sync(codec); + regcache_cache_bypass(codec->component.regmap, true); + /* Disable everything except touchpanel - that will be handled * by the touch driver and left disabled if touch is not in * use. */ @@ -1173,14 +1217,14 @@ static int wm9713_soc_suspend(struct snd_soc_codec *codec) static int wm9713_soc_resume(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - int i, ret; - u16 *cache = codec->reg_cache; + int ret; ret = snd_ac97_reset(wm9713->ac97, true, WM9713_VENDOR_ID, WM9713_VENDOR_ID_MASK); if (ret < 0) return ret; + regcache_cache_bypass(codec->component.regmap, false); snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY); /* do we need to re-start the PLL ? */ @@ -1188,14 +1232,10 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) wm9713_set_pll(codec, 0, wm9713->pll_in, 0); /* only synchronise the codec if warm reset failed */ - if (ret == 0) { - for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i += 2) { - if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || - i == AC97_EXTENDED_MSTATUS || i > 0x66) - continue; - soc_ac97_ops->write(wm9713->ac97, i, cache[i>>1]); - } - } + if (ret == 0) + regcache_mark_dirty(codec->component.regmap); + + snd_soc_cache_sync(codec); return ret; } @@ -1203,6 +1243,7 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) static int wm9713_soc_probe(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap; int reg; wm9713->ac97 = snd_soc_new_ac97_codec(codec, WM9713_VENDOR_ID, @@ -1210,6 +1251,14 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) if (IS_ERR(wm9713->ac97)) return PTR_ERR(wm9713->ac97); + regmap = devm_regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); + if (IS_ERR(regmap)) { + snd_soc_free_ac97_codec(wm9713->ac97); + return PTR_ERR(regmap); + } + + snd_soc_codec_init_regmap(codec, regmap); + /* unmute the adc - move to kcontrol */ reg = ac97_read(codec, AC97_CD) & 0x7fff; ac97_write(codec, AC97_CD, reg); @@ -1221,6 +1270,7 @@ static int wm9713_soc_remove(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); + snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(wm9713->ac97); return 0; } @@ -1230,13 +1280,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9713 = { .remove = wm9713_soc_remove, .suspend = wm9713_soc_suspend, .resume = wm9713_soc_resume, - .read = ac97_read, - .write = ac97_write, .set_bias_level = wm9713_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm9713_reg), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = wm9713_reg, .controls = wm9713_snd_ac97_controls, .num_controls = ARRAY_SIZE(wm9713_snd_ac97_controls), -- 2.1.4

3 6

[alsa-devel] [PATCH] ASoC: rsnd: audio_clkout0/1/2/3 are optinal properties
by Kuninori Morimoto 29 Oct '15

29 Oct '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas sound driver can output Audio-clkout0/1/2/3, but these are optinal properties for each board. Reported-by: Geert Uytterhoeven <geert+renesas(a)glider.be> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index c57cbd6..bbcb327 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -34,6 +34,8 @@ Required properties: see below for detail. - #sound-dai-cells : it must be 0 if your system is using single DAI it must be 1 if your system is using multi DAI + +Optional properties: - #clock-cells : it must be 0 if your system has audio_clkout it must be 1 if your system has audio_clkout0/1/2/3 - clock-frequency : for all audio_clkout0/1/2/3 -- 1.9.1

2 2

[alsa-devel] [PATCH] [RFC] ASoC: rsnd: Add missing initialization of ADG req_rate
by Geert Uytterhoeven 29 Oct '15

29 Oct '15

If the "clock-frequency" DT property is not found, req_rate is used uninitialized, and the "audio_clkout" clock will be created with an arbitrary clock rate. This uninitialized kernel stack data may leak to userspace through /sys/kernel/debug/clk/clk_summary, cfr. the value in the "rate" column: clock enable_cnt prepare_cnt rate accuracy phase -------------------------------------------------------------------- audio_clkout 0 0 4001836240 0 0 Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- Should the absence of this property be considered an error? Documentation/devicetree/bindings/sound/renesas,rsnd.txt states that both "#clock-cells" and "clock-frequency" are required properties, but none of the R-Car Gen1 and Gen2 DTSes seem to have them? --- sound/soc/sh/rcar/adg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 2a5b3a293cd243db..b123734f9fbdabd9 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -437,7 +437,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np = dev->of_node; u32 ckr, rbgx, rbga, rbgb; - u32 rate, req_rate, div; + u32 rate, req_rate = 0, div; uint32_t count = 0; unsigned long req_48kHz_rate, req_441kHz_rate; int i; -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: dpcm: Make BE prepare possible in suspend state
by Koro Chen 29 Oct '15

29 Oct '15

During suspend/resume, there is a flow that if a driver does not support SNDRV_PCM_INFO_RESUME, it will fail at snd_pcm_resume(), and user space can then issue SNDRV_PCM_IOCTL_PREPARE to let audio continue to play. However, in dpcm_be_dai_prepare() it only allows BEs to be prepared in state SND_SOC_DPCM_STATE_HW_PARAMS or SND_SOC_DPCM_STATE_STOP. The BE state will then stay in SND_SOC_DPCM_STATE_SUSPEND, consequently dpcm_be_dai_shutdown() is skipped in the end of playback and be_substream->runtime is not cleared while this runtime is actually freed by snd_pcm_detach_substream(). If another suspend comes, a NULL pointer dereference will happen in snd_pcm_suspend() when accessing BE substream's runtime. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/soc-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 3173958..5c41a58 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2117,7 +2117,8 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) continue; if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) continue; dev_dbg(be->dev, "ASoC: prepare BE %s\n", -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: rt5677: handle reset_pin logical state correctly
by Anatol Pomozov 29 Oct '15

29 Oct '15

According to the datasheet RESET is active low pin, i.e. system goes to reset state when pin is low. Handle logic state correctly - set reset_pin to logical 0 at boot time, and set it to logical 1 when we need to reset the chip. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5677.txt | 2 +- sound/soc/codecs/rt5677.c | 8 ++++---- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index f070789..1b3c13d 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -18,7 +18,7 @@ Required properties: Optional properties: - realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin. -- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. +- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. Active low. - realtek,in1-differential - realtek,in2-differential diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 3b46b35..433652f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4766,7 +4766,7 @@ static int rt5677_remove(struct snd_soc_codec *codec) if (rt5677->pow_ldo2) gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); if (rt5677->reset_pin) - gpiod_set_value_cansleep(rt5677->reset_pin, 0); + gpiod_set_value_cansleep(rt5677->reset_pin, 1); return 0; } @@ -4783,7 +4783,7 @@ static int rt5677_suspend(struct snd_soc_codec *codec) if (rt5677->pow_ldo2) gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); if (rt5677->reset_pin) - gpiod_set_value_cansleep(rt5677->reset_pin, 0); + gpiod_set_value_cansleep(rt5677->reset_pin, 1); } return 0; @@ -4797,7 +4797,7 @@ static int rt5677_resume(struct snd_soc_codec *codec) if (rt5677->pow_ldo2) gpiod_set_value_cansleep(rt5677->pow_ldo2, 1); if (rt5677->reset_pin) - gpiod_set_value_cansleep(rt5677->reset_pin, 1); + gpiod_set_value_cansleep(rt5677->reset_pin, 0); if (rt5677->pow_ldo2 || rt5677->reset_pin) msleep(10); @@ -5142,7 +5142,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, rt5677->pow_ldo2 = 0; } rt5677->reset_pin = devm_gpiod_get_optional(&i2c->dev, - "realtek,reset", GPIOD_OUT_HIGH); + "realtek,reset", GPIOD_OUT_LOW); if (IS_ERR(rt5677->reset_pin)) { ret = PTR_ERR(rt5677->reset_pin); dev_err(&i2c->dev, "Failed to request RESET: %d\n", ret); -- 2.4.3.573.g4eafbef

2 3

[alsa-devel] [PATCH] FSL_SSI: fix fsl_ssi interrupt reporting so that interrupts are only reported when they have been enabled.
by Caleb Crome 28 Oct '15

28 Oct '15

The current implementation reports all interrupt stats in /sys/kernel/debug/*.ssi/stats, even if the interrupt was not enabled. This is not the intention as per the comments. This fix keeps track of ever bit in the SIER register that has ever been enabled, which allows fsl_ssi_stats_show to actually only report those interrupt counts for interrupts that are enabled. Signed-off-by: Caleb Crome <caleb(a)crome.org> --- sound/soc/fsl/fsl_ssi.c | 15 +++++++++++++++ sound/soc/fsl/fsl_ssi.h | 1 + sound/soc/fsl/fsl_ssi_dbg.c | 8 +++----- 3 files changed, 19 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 73778c2..14d5ade 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -299,6 +299,18 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) return IRQ_HANDLED; } +// call fsl_ssi_dbg_update_sier_enbled every time sier is +// updated. it tracks which interrupts have ever been enabled. +// so that the print function can keep track. +void fsl_ssi_update_sier_enabled( + struct fsl_ssi_private *ssi_private) +{ + int sier; + struct regmap *regs = ssi_private->regs; + regmap_read(regs, CCSR_SSI_SIER, &sier); + ssi_private->dbg_stats.sier_ever_enabled |= sier; +} + /* * Enable/Disable all rx/tx config flags at once. */ @@ -312,6 +324,7 @@ static void fsl_ssi_rxtx_config(struct fsl_ssi_private *ssi_private, regmap_update_bits(regs, CCSR_SSI_SIER, vals->rx.sier | vals->tx.sier, vals->rx.sier | vals->tx.sier); + fsl_ssi_update_sier_enabled(ssi_private); regmap_update_bits(regs, CCSR_SSI_SRCR, vals->rx.srcr | vals->tx.srcr, vals->rx.srcr | vals->tx.srcr); @@ -404,6 +417,7 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable, */ if (enable) { regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier); + fsl_ssi_update_sier_enabled(ssi_private); regmap_update_bits(regs, CCSR_SSI_SRCR, vals->srcr, vals->srcr); regmap_update_bits(regs, CCSR_SSI_STCR, vals->stcr, vals->stcr); } else { @@ -431,6 +445,7 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable, regmap_update_bits(regs, CCSR_SSI_SRCR, srcr, 0); regmap_update_bits(regs, CCSR_SSI_STCR, stcr, 0); regmap_update_bits(regs, CCSR_SSI_SIER, sier, 0); + fsl_ssi_update_sier_enabled(ssi_private); } config_done: diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 5065105..a078c69 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -237,6 +237,7 @@ struct fsl_ssi_dbg { unsigned int tfe1; unsigned int tfe0; } stats; + unsigned int sier_ever_enabled; }; void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *ssi_dbg, u32 sisr); diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 5469ffb..1066620 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -82,13 +82,10 @@ void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *dbg, u32 sisr) dbg->stats.tfe0++; } -/* Show the statistics of a flag only if its interrupt is enabled. The - * compiler will optimze this code to a no-op if the interrupt is not - * enabled. - */ +/* Show the statistics of a flag only if its interrupt is enabled. */ #define SIER_SHOW(flag, name) \ do { \ - if (CCSR_SSI_SIER_##flag) \ + if (CCSR_SSI_SIER_##flag & sier) \ seq_printf(s, #name "=%u\n", ssi_dbg->stats.name); \ } while (0) @@ -102,6 +99,7 @@ void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *dbg, u32 sisr) static int fsl_ssi_stats_show(struct seq_file *s, void *unused) { struct fsl_ssi_dbg *ssi_dbg = s->private; + unsigned int sier = ssi_dbg->sier_ever_enabled; SIER_SHOW(RFRC_EN, rfrc); SIER_SHOW(TFRC_EN, tfrc); -- 2.1.4

3 5

[alsa-devel] [PATCH v2- ALSA 1/3] HDA - Allow hda_jack_tbl to use another nid for pin sense
by Raymond Yau 28 Oct '15

28 Oct '15

- add sense_nid to __snd_hda_jack_add_kctl for pin sense - update slave pin sense when master pin sense change - enable unsolicited event when widcap support UNSOL Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com> diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 366efbf..d81d65f 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -19,6 +19,7 @@ #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" +#include "hda_generic.h" /** * is_jack_detectable - Check whether the given pin is jack-detectable @@ -157,7 +158,14 @@ static void jack_detect_update(struct hda_codec *codec, if (jack->phantom_jack) jack->pin_sense = AC_PINSENSE_PRESENCE; else - jack->pin_sense = read_pin_sense(codec, jack->nid); + jack->pin_sense = read_pin_sense(codec, + jack->sense_nid ? jack->sense_nid : jack->nid); + + if (jack->slave_nid) { + struct hda_jack_tbl *slave = snd_hda_jack_tbl_get(codec, + jack->slave_nid); + slave->pin_sense = jack->pin_sense; + } /* A gating jack indicates the jack is invalid if gating is unplugged */ if (jack->gating_jack && !snd_hda_jack_detect(codec, jack->gating_jack)) @@ -269,11 +277,13 @@ snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, jack->jack_detect = 1; if (codec->jackpoll_interval > 0) return callback; /* No unsol if we're polling instead */ - err = snd_hda_codec_write_cache(codec, nid, 0, + if (is_jack_detectable(codec, nid)) { + err = snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | jack->tag); - if (err < 0) - return ERR_PTR(err); + if (err < 0) + return ERR_PTR(err); + } return callback; } EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable_callback); @@ -384,7 +394,7 @@ static void hda_free_jack_priv(struct snd_jack *jack) * will have the given name and index. */ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, bool phantom_jack) + const char *name, bool phantom_jack, hda_nid_t sense_nid) { struct hda_jack_tbl *jack; int err, state, type; @@ -403,6 +413,7 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, jack->phantom_jack = !!phantom_jack; jack->type = type; + jack->sense_nid = sense_nid; jack->jack->private_data = jack; jack->jack->private_free = hda_free_jack_priv; state = snd_hda_jack_detect(codec, nid); @@ -422,7 +433,7 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, const char *name) { - return __snd_hda_jack_add_kctl(codec, nid, name, false); + return __snd_hda_jack_add_kctl(codec, nid, name, false, nid); } EXPORT_SYMBOL_GPL(snd_hda_jack_add_kctl); @@ -451,7 +462,7 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, if (phantom_jack) /* Example final name: "Internal Mic Phantom Jack" */ strncat(name, " Phantom", sizeof(name) - strlen(name) - 1); - err = __snd_hda_jack_add_kctl(codec, nid, name, phantom_jack); + err = __snd_hda_jack_add_kctl(codec, nid, name, phantom_jack, nid); if (err < 0) return err; diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 387d309..ef3d162 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -41,6 +41,8 @@ struct hda_jack_tbl { hda_nid_t gated_jack; /* gated is dependent on this jack */ int type; struct snd_jack *jack; + hda_nid_t sense_nid; + hda_nid_t slave_nid; }; struct hda_jack_tbl * -- 2.5.0

3 11

Re: [alsa-devel] [PATCH 0/2] Remove .owner from spi_drivers
by Andrew F. Davis 28 Oct '15

28 Oct '15

On 10/23/2015 01:29 PM, Sebastian Reichel wrote: > Hi, > > On Fri, Oct 23, 2015 at 08:59:09AM -0500, Andrew F. Davis wrote: >> This series adds a helper macro to the SPI core that assigns the owner >> in spi_drivers so individual drivers do not have to do this, this is >> similar to other subsystems like platform_driver and i2c_driver that >> have made this change. > > Probably it makes sense to also update the related coccinelle > script: scripts/coccinelle/api/platform_no_drv_owner.cocci > > -- Sebastian > Makes sense, patch submitted. Thanks, Andrew

1 0

[alsa-devel] [PATCH] ALSA: DocBook: Add soc-ops.c and soc-compress.c
by Takashi Iwai 28 Oct '15

28 Oct '15

These have been missing in the template file although they provide good kernel doc comments. Let's add them. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/DocBook/alsa-driver-api.tmpl | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl index e94a10bb4a9e..53f439dcc94b 100644 --- a/Documentation/DocBook/alsa-driver-api.tmpl +++ b/Documentation/DocBook/alsa-driver-api.tmpl @@ -112,6 +112,8 @@ !Esound/soc/soc-devres.c !Esound/soc/soc-io.c !Esound/soc/soc-pcm.c +!Esound/soc/soc-ops.c +!Esound/soc/soc-compress.c </sect1> <sect1><title>ASoC DAPM API</title> !Esound/soc/soc-dapm.c -- 2.6.2

1 0

[alsa-devel] [PATCH] ALSA: hda - Add / fix kernel doc comments
by Takashi Iwai 28 Oct '15

28 Oct '15

Give some readable comment in kernel doc style for each exported function, as I promised in the previous meetings. While we're at it, fix the wrong comments, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_bus.c | 14 ++++++++++ sound/hda/hdac_device.c | 10 ++++--- sound/hda/hdac_i915.c | 70 +++++++++++++++++++++++++++++++++++++++++++++++++ sound/hda/hdac_regmap.c | 10 +++++++ sound/hda/hdac_stream.c | 3 ++- 5 files changed, 102 insertions(+), 5 deletions(-) diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 27c447e4fe5c..0e81ea89a596 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -172,6 +172,15 @@ static void process_unsol_events(struct work_struct *work) } } +/** + * snd_hdac_bus_add_device - Add a codec to bus + * @bus: HDA core bus + * @codec: HDA core device to add + * + * Adds the given codec to the list in the bus. The caddr_tbl array + * and codec_powered bits are updated, as well. + * Returns zero if success, or a negative error code. + */ int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec) { if (bus->caddr_tbl[codec->addr]) { @@ -188,6 +197,11 @@ int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec) } EXPORT_SYMBOL_GPL(snd_hdac_bus_add_device); +/** + * snd_hdac_bus_remove_device - Remove a codec from bus + * @bus: HDA core bus + * @codec: HDA core device to remove + */ void snd_hdac_bus_remove_device(struct hdac_bus *bus, struct hdac_device *codec) { diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index bbdb25f5bbb9..e361024eabb6 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -629,8 +629,10 @@ int snd_hdac_power_down_pm(struct hdac_device *codec) EXPORT_SYMBOL_GPL(snd_hdac_power_down_pm); #endif -/* - * Enable/disable the link power for a codec. +/** + * snd_hdac_link_power - Enable/disable the link power for a codec + * @codec: the codec object + * @bool: enable or disable the link power */ int snd_hdac_link_power(struct hdac_device *codec, bool enable) { @@ -1048,8 +1050,8 @@ int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid, } EXPORT_SYMBOL_GPL(snd_hdac_codec_write); -/* - * snd_hdac_check_power_state: check whether the actual power state matches +/** + * snd_hdac_check_power_state - check whether the actual power state matches * with the target state * * @hdac: the HDAC device diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 55c3df4458f7..8fef1b8d1fd8 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -23,6 +23,19 @@ static struct i915_audio_component *hdac_acomp; +/** + * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup + * @bus: HDA core bus + * @enable: enable or disable the wakeup + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with i915 graphics. + * + * This function should be called during the chip reset, also called at + * resume for updating STATESTS register read. + * + * Returns zero for success or a negative error code. + */ int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) { struct i915_audio_component *acomp = bus->audio_component; @@ -45,6 +58,19 @@ int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) } EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup); +/** + * snd_hdac_display_power - Power up / down the power refcount + * @bus: HDA core bus + * @enable: power up or down + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with i915 graphics. + * + * This function manages a refcount and calls the i915 get_power() and + * put_power() ops accordingly, toggling the codec wakeup, too. + * + * Returns zero for success or a negative error code. + */ int snd_hdac_display_power(struct hdac_bus *bus, bool enable) { struct i915_audio_component *acomp = bus->audio_component; @@ -71,6 +97,16 @@ int snd_hdac_display_power(struct hdac_bus *bus, bool enable) } EXPORT_SYMBOL_GPL(snd_hdac_display_power); +/** + * snd_hdac_get_display_clk - Get CDCLK in kHz + * @bus: HDA core bus + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with i915 graphics. + * + * This function queries CDCLK value in kHz from the graphics driver and + * returns the value. A negative code is returned in error. + */ int snd_hdac_get_display_clk(struct hdac_bus *bus) { struct i915_audio_component *acomp = bus->audio_component; @@ -134,6 +170,17 @@ static int hdac_component_master_match(struct device *dev, void *data) return !strcmp(dev->driver->name, "i915"); } +/** + * snd_hdac_i915_register_notifier - Register i915 audio component ops + * @aops: i915 audio component ops + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with i915 graphics. + * + * This function sets the given ops to be called by the i915 graphics driver. + * + * Returns zero for success or a negative error code. + */ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops) { if (WARN_ON(!hdac_acomp)) @@ -144,6 +191,18 @@ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops } EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); +/** + * snd_hdac_i915_init - Initialize i915 audio component + * @bus: HDA core bus + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with i915 graphics. + * + * This function initializes and sets up the audio component to communicate + * with i915 graphics driver. + * + * Returns zero for success or a negative error code. + */ int snd_hdac_i915_init(struct hdac_bus *bus) { struct component_match *match = NULL; @@ -187,6 +246,17 @@ out_err: } EXPORT_SYMBOL_GPL(snd_hdac_i915_init); +/** + * snd_hdac_i915_exit - Finalize i915 audio component + * @bus: HDA core bus + * + * This function is supposed to be used only by a HD-audio controller + * driver that needs the interaction with i915 graphics. + * + * This function releases the i915 audio component that has been used. + * + * Returns zero for success or a negative error code. + */ int snd_hdac_i915_exit(struct hdac_bus *bus) { struct device *dev = bus->dev; diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index b0ed870ffb88..eb8f7c30cb09 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -339,6 +339,12 @@ static const struct regmap_config hda_regmap_cfg = { .use_single_rw = true, }; +/** + * snd_hdac_regmap_init - Initialize regmap for HDA register accesses + * @codec: the codec object + * + * Returns zero for success or a negative error code. + */ int snd_hdac_regmap_init(struct hdac_device *codec) { struct regmap *regmap; @@ -352,6 +358,10 @@ int snd_hdac_regmap_init(struct hdac_device *codec) } EXPORT_SYMBOL_GPL(snd_hdac_regmap_init); +/** + * snd_hdac_regmap_init - Release the regmap from HDA codec + * @codec: the codec object + */ void snd_hdac_regmap_exit(struct hdac_device *codec) { if (codec->regmap) { diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index 8981159813ef..38990a77d7b7 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -426,7 +426,8 @@ int snd_hdac_stream_setup_periods(struct hdac_stream *azx_dev) } EXPORT_SYMBOL_GPL(snd_hdac_stream_setup_periods); -/* snd_hdac_stream_set_params - set stream parameters +/** + * snd_hdac_stream_set_params - set stream parameters * @azx_dev: HD-audio core stream for which parameters are to be set * @format_val: format value parameter * -- 2.6.2

1 0

[alsa-devel] [PATCH] ALSA: Constify ratden/ratnum constraints
by Lars-Peter Clausen 28 Oct '15

28 Oct '15

The ALSA core does not modify the constraints provided by a driver. Most constraint helper functions already take a const pointer to the constraint description, the exception at the moment being the ratden and ratnum constraints. Make those const as well, this allows a driver to declare them as const. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/pcm.h | 10 +++++----- sound/core/pcm_lib.c | 17 +++++++++-------- 2 files changed, 14 insertions(+), 13 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 4e6ad74..b0be092 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -265,12 +265,12 @@ struct snd_ratden { struct snd_pcm_hw_constraint_ratnums { int nrats; - struct snd_ratnum *rats; + const struct snd_ratnum *rats; }; struct snd_pcm_hw_constraint_ratdens { int nrats; - struct snd_ratden *rats; + const struct snd_ratden *rats; }; struct snd_pcm_hw_constraint_list { @@ -970,7 +970,7 @@ int snd_interval_list(struct snd_interval *i, unsigned int count, int snd_interval_ranges(struct snd_interval *i, unsigned int count, const struct snd_interval *list, unsigned int mask); int snd_interval_ratnum(struct snd_interval *i, - unsigned int rats_count, struct snd_ratnum *rats, + unsigned int rats_count, const struct snd_ratnum *rats, unsigned int *nump, unsigned int *denp); void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params); @@ -1000,11 +1000,11 @@ int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime, int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratnums *r); + const struct snd_pcm_hw_constraint_ratnums *r); int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratdens *r); + const struct snd_pcm_hw_constraint_ratdens *r); int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, unsigned int cond, unsigned int width, diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 05a3ca9..6b5a811 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -801,7 +801,7 @@ void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, * negative error code. */ int snd_interval_ratnum(struct snd_interval *i, - unsigned int rats_count, struct snd_ratnum *rats, + unsigned int rats_count, const struct snd_ratnum *rats, unsigned int *nump, unsigned int *denp) { unsigned int best_num, best_den; @@ -920,7 +920,8 @@ EXPORT_SYMBOL(snd_interval_ratnum); * negative error code. */ static int snd_interval_ratden(struct snd_interval *i, - unsigned int rats_count, struct snd_ratden *rats, + unsigned int rats_count, + const struct snd_ratden *rats, unsigned int *nump, unsigned int *denp) { unsigned int best_num, best_diff, best_den; @@ -1339,7 +1340,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_ranges); static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hw_constraint_ratnums *r = rule->private; + const struct snd_pcm_hw_constraint_ratnums *r = rule->private; unsigned int num = 0, den = 0; int err; err = snd_interval_ratnum(hw_param_interval(params, rule->var), @@ -1363,10 +1364,10 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratnums *r) + const struct snd_pcm_hw_constraint_ratnums *r) { return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_ratnums, r, + snd_pcm_hw_rule_ratnums, (void *)r, var, -1); } @@ -1375,7 +1376,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_ratnums); static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hw_constraint_ratdens *r = rule->private; + const struct snd_pcm_hw_constraint_ratdens *r = rule->private; unsigned int num = 0, den = 0; int err = snd_interval_ratden(hw_param_interval(params, rule->var), r->nrats, r->rats, &num, &den); @@ -1398,10 +1399,10 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratdens *r) + const struct snd_pcm_hw_constraint_ratdens *r) { return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_ratdens, r, + snd_pcm_hw_rule_ratdens, (void *)r, var, -1); } -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Disable 64bit address for Creative HDA controllers
by Takashi Iwai 28 Oct '15

28 Oct '15

We've had many reports that some Creative sound cards with CA0132 don't work well. Some reported that it starts working after reloading the module, while some reported it starts working when a 32bit kernel is used. All these facts seem implying that the chip fails to communicate when the buffer is located in 64bit address. This patch addresses these issues by just adding AZX_DCAPS_NO_64BIT flag to the corresponding PCI entries. I casually had a chance to test an SB Recon3D board, and indeed this seems helping. Although this hasn't been tested on all Creative devices, it's safer to assume that this restriction applies to the rest of them, too. So the flag is applied to all Creative entries. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c38c68f57938..61b8b75a3c80 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -334,6 +334,7 @@ enum { #define AZX_DCAPS_PRESET_CTHDA \ (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB |\ + AZX_DCAPS_NO_64BIT |\ AZX_DCAPS_4K_BDLE_BOUNDARY | AZX_DCAPS_SNOOP_OFF) /* @@ -2284,11 +2285,13 @@ static const struct pci_device_id azx_ids[] = { .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, .driver_data = AZX_DRIVER_CTX | AZX_DCAPS_CTX_WORKAROUND | + AZX_DCAPS_NO_64BIT | AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_POSFIX_LPIB }, #else /* this entry seems still valid -- i.e. without emu20kx chip */ { PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_CTX | AZX_DCAPS_CTX_WORKAROUND | + AZX_DCAPS_NO_64BIT | AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_POSFIX_LPIB }, #endif /* CM8888 */ -- 2.6.2

1 0

[alsa-devel] [PATCH] ASoC: rsnd: call rsnd_src_quit() from rsnd_src_quit_gen2()
by Kuninori Morimoto 28 Oct '15

28 Oct '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> 2d604e03("ASoC: rsnd: disable SRC.out only when stop timing") added rsnd_src_quit_gen2(), but it should call rsnd_src_quit() same as before. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index a710799..776b0ef 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -621,7 +621,7 @@ static int rsnd_src_quit_gen2(struct rsnd_mod *mod, /* stop both out/in */ rsnd_mod_write(mod, SRC_CTRL, 0); - return 0; + return rsnd_src_quit(mod, io, priv); } static void __rsnd_src_interrupt_gen2(struct rsnd_mod *mod, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: fixup rsnd_dmapp_stop() return value
by Kuninori Morimoto 28 Oct '15

28 Oct '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> 45a4394d03("ASoC: rsnd: use mod base common method on DMA phase3") Exchanged "void rsnd_dmapp_stop()" to "int rsnd_dmapp_stop()", but it returns inverted value. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index fc70e97..9917b98 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -368,11 +368,11 @@ static int rsnd_dmapp_stop(struct rsnd_mod *mod, for (i = 0; i < 1024; i++) { if (0 == rsnd_dmapp_read(dma, PDMACHCR)) - return -EIO; + return 0; udelay(1); } - return 0; + return -EIO; } static int rsnd_dmapp_start(struct rsnd_mod *mod, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl_esai: ETDR and TX0~5 registers are non volatile
by Zidan Wang 28 Oct '15

28 Oct '15

ETDR and TX0~5 registers are writable and not readable. So they are non volatile. Remove them from volatile list, and add default register value for them. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_esai.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index aab675a..dd5fde4 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -653,21 +653,28 @@ static const struct snd_soc_component_driver fsl_esai_component = { }; static const struct reg_default fsl_esai_reg_defaults[] = { - {0x8, 0x00000000}, - {0x10, 0x00000000}, - {0x18, 0x00000000}, - {0x98, 0x00000000}, - {0xd0, 0x00000000}, - {0xd4, 0x00000000}, - {0xd8, 0x00000000}, - {0xdc, 0x00000000}, - {0xe0, 0x00000000}, - {0xe4, 0x0000ffff}, - {0xe8, 0x0000ffff}, - {0xec, 0x0000ffff}, - {0xf0, 0x0000ffff}, - {0xf8, 0x00000000}, - {0xfc, 0x00000000}, + {REG_ESAI_ETDR, 0x00000000}, + {REG_ESAI_ECR, 0x00000000}, + {REG_ESAI_TFCR, 0x00000000}, + {REG_ESAI_RFCR, 0x00000000}, + {REG_ESAI_TX0, 0x00000000}, + {REG_ESAI_TX1, 0x00000000}, + {REG_ESAI_TX2, 0x00000000}, + {REG_ESAI_TX3, 0x00000000}, + {REG_ESAI_TX4, 0x00000000}, + {REG_ESAI_TX5, 0x00000000}, + {REG_ESAI_TSR, 0x00000000}, + {REG_ESAI_SAICR, 0x00000000}, + {REG_ESAI_TCR, 0x00000000}, + {REG_ESAI_TCCR, 0x00000000}, + {REG_ESAI_RCR, 0x00000000}, + {REG_ESAI_RCCR, 0x00000000}, + {REG_ESAI_TSMA, 0x0000ffff}, + {REG_ESAI_TSMB, 0x0000ffff}, + {REG_ESAI_RSMA, 0x0000ffff}, + {REG_ESAI_RSMB, 0x0000ffff}, + {REG_ESAI_PRRC, 0x00000000}, + {REG_ESAI_PCRC, 0x00000000}, }; static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg) @@ -705,17 +712,10 @@ static bool fsl_esai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_esai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { - case REG_ESAI_ETDR: case REG_ESAI_ERDR: case REG_ESAI_ESR: case REG_ESAI_TFSR: case REG_ESAI_RFSR: - case REG_ESAI_TX0: - case REG_ESAI_TX1: - case REG_ESAI_TX2: - case REG_ESAI_TX3: - case REG_ESAI_TX4: - case REG_ESAI_TX5: case REG_ESAI_RX0: case REG_ESAI_RX1: case REG_ESAI_RX2: -- 1.9.1

3 8

[alsa-devel] [PATCH] ASoC: wm8962: set ALC2 as non-volatile register
by Jiada Wang 28 Oct '15

28 Oct '15

Previously ALC2 register is set as a volatile register, declare it as one of ALC Coefficients register together with other non-volatile registers will cause issue, in case wm8962 has enter suspend mode, and cache_only flag is set, any attempt to read from ALC2 will fail. Because the 5 status bits in ALC2 aren't used anywhere nor are useful to end user, so this patch removes ALC2 register from volatile register list to make ALC2 be possible to be accessed when cache_only flag is set. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/codecs/wm8962.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 2fbc6ef..371ec72 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -131,7 +131,7 @@ static const struct reg_default wm8962_reg[] = { { 15, 0x6243 }, /* R15 - Software Reset */ { 17, 0x007B }, /* R17 - ALC1 */ - + { 18, 0x0000 }, /* R18 - ALC2 */ { 19, 0x1C32 }, /* R19 - ALC3 */ { 20, 0x3200 }, /* R20 - Noise Gate */ { 21, 0x00C0 }, /* R21 - Left ADC volume */ @@ -794,7 +794,6 @@ static bool wm8962_volatile_register(struct device *dev, unsigned int reg) case WM8962_CLOCKING1: case WM8962_CLOCKING2: case WM8962_SOFTWARE_RESET: - case WM8962_ALC2: case WM8962_THERMAL_SHUTDOWN_STATUS: case WM8962_ADDITIONAL_CONTROL_4: case WM8962_DC_SERVO_6: -- 2.4.5

3 2

[alsa-devel] [PATCH alsa-lib 1/2] pcm: drop unused sys/shm.h inclusion
by Mike Frysinger 27 Oct '15

27 Oct '15

From: Mike Frysinger <vapier(a)chromium.org> This header isn't used in these files, so stop trying to include it. This helps builds on systems that don't provide the header (such as Android/Bionic). Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- src/pcm/pcm.c | 1 - src/pcm/pcm_generic.c | 1 - src/pcm/pcm_hw.c | 1 - src/pcm/pcm_null.c | 1 - src/pcm/pcm_plugin.c | 1 - src/pcm/pcm_share.c | 1 - 6 files changed, 6 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index f5fc728..cbbc55a 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -635,7 +635,6 @@ playback devices. #include <signal.h> #include <ctype.h> #include <sys/poll.h> -#include <sys/shm.h> #include <sys/mman.h> #include <limits.h> #include "pcm_local.h" diff --git a/src/pcm/pcm_generic.c b/src/pcm/pcm_generic.c index 9b60591..4dbef08 100644 --- a/src/pcm/pcm_generic.c +++ b/src/pcm/pcm_generic.c @@ -26,7 +26,6 @@ * */ -#include <sys/shm.h> #include <sys/ioctl.h> #include <limits.h> #include "pcm_local.h" diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 232b197..66aec5c 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -36,7 +36,6 @@ #include <fcntl.h> #include <sys/ioctl.h> #include <sys/mman.h> -#include <sys/shm.h> #include "pcm_local.h" #include "../control/control_local.h" #include "../timer/timer_local.h" diff --git a/src/pcm/pcm_null.c b/src/pcm/pcm_null.c index 0529820..5e63caa 100644 --- a/src/pcm/pcm_null.c +++ b/src/pcm/pcm_null.c @@ -28,7 +28,6 @@ #include <byteswap.h> #include <limits.h> -#include <sys/shm.h> #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index c19e2f1..d007e8c 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -82,7 +82,6 @@ pcm.rate44100Hz { */ -#include <sys/shm.h> #include <limits.h> #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_share.c b/src/pcm/pcm_share.c index 4b68f1f..5d8aaf2 100644 --- a/src/pcm/pcm_share.c +++ b/src/pcm/pcm_share.c @@ -35,7 +35,6 @@ #include <math.h> #include <sys/socket.h> #include <sys/poll.h> -#include <sys/shm.h> #include <pthread.h> #include "pcm_local.h" -- 2.5.2

1 1

[alsa-devel] [PATCH alsa-lib] move page helpers to common code
by Mike Frysinger 27 Oct '15

27 Oct '15

From: Mike Frysinger <vapier(a)chromium.org> If you build with --disable-pcm, the rawmidi code fails to link because it uses the page_size helper declared in local.h, but the page_* funcs are in the pcm mmap module. Move these funcs into conf.c so that they are always available to the rest of the system. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- src/conf.c | 36 ++++++++++++++++++++++++++++++++++++ src/pcm/pcm_mmap.c | 36 ------------------------------------ 2 files changed, 36 insertions(+), 36 deletions(-) diff --git a/src/conf.c b/src/conf.c index 91fca25..a4f2a4f 100644 --- a/src/conf.c +++ b/src/conf.c @@ -4876,3 +4876,39 @@ static void _snd_config_end(void) files_info_count = 0; } #endif + +size_t page_size(void) +{ + long s = sysconf(_SC_PAGE_SIZE); + assert(s > 0); + return s; +} + +size_t page_align(size_t size) +{ + size_t r; + long psz = page_size(); + r = size % psz; + if (r) + return size + psz - r; + return size; +} + +size_t page_ptr(size_t object_offset, size_t object_size, size_t *offset, size_t *mmap_offset) +{ + size_t r; + long psz = page_size(); + assert(offset); + assert(mmap_offset); + *mmap_offset = object_offset; + object_offset %= psz; + *mmap_offset -= object_offset; + object_size += object_offset; + r = object_size % psz; + if (r) + r = object_size + psz - r; + else + r = object_size; + *offset = object_offset; + return r; +} diff --git a/src/pcm/pcm_mmap.c b/src/pcm/pcm_mmap.c index 83e74e5..470bd04 100644 --- a/src/pcm/pcm_mmap.c +++ b/src/pcm/pcm_mmap.c @@ -26,42 +26,6 @@ #include <sys/shm.h> #include "pcm_local.h" -size_t page_size(void) -{ - long s = sysconf(_SC_PAGE_SIZE); - assert(s > 0); - return s; -} - -size_t page_align(size_t size) -{ - size_t r; - long psz = page_size(); - r = size % psz; - if (r) - return size + psz - r; - return size; -} - -size_t page_ptr(size_t object_offset, size_t object_size, size_t *offset, size_t *mmap_offset) -{ - size_t r; - long psz = page_size(); - assert(offset); - assert(mmap_offset); - *mmap_offset = object_offset; - object_offset %= psz; - *mmap_offset -= object_offset; - object_size += object_offset; - r = object_size % psz; - if (r) - r = object_size + psz - r; - else - r = object_size; - *offset = object_offset; - return r; -} - void snd_pcm_mmap_appl_backward(snd_pcm_t *pcm, snd_pcm_uframes_t frames) { snd_pcm_sframes_t appl_ptr = *pcm->appl.ptr; -- 2.5.2

1 0

[alsa-devel] [PATCH] ASoC: sun4i-codec: use consistent names for PA controls
by Adam Sampson 27 Oct '15

27 Oct '15

The power amplifier for the headphone output is called "the PA" and "the headphone amplifier" in Allwinner's documentation for the A10 and A20. sun4i-codec calls it "PA" in some places and "Pre-Amplifier" (which isn't really accurate) in others, leading to user-visible controls with different names referring to the same device. When this driver implements audio input, it'll also need to expose controls for the line and mic input preamps, so just referring to "the Pre-Amplifier" will be ambiguous. Change it to use "PA" consistently for the power amplifier. Signed-off-by: Adam Sampson <ats(a)offog.org> --- sound/soc/sunxi/sun4i-codec.c | 23 ++++++++++++----------- 1 file changed, 12 insertions(+), 11 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index bcbf4da..ff3304d 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -2,6 +2,7 @@ * Copyright 2014 Emilio López <emilio(a)elopez.com.ar> * Copyright 2014 Jon Smirl <jonsmirl(a)gmail.com> * Copyright 2015 Maxime Ripard <maxime.ripard(a)free-electrons.com> + * Copyright 2015 Adam Sampson <ats(a)offog.org> * * Based on the Allwinner SDK driver, released under the GPL. * @@ -452,12 +453,12 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0), - /* Pre-Amplifier */ - SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL, + /* Headphone output power amplifier */ + SND_SOC_DAPM_MIXER("PA", SUN4I_CODEC_ADC_ACTL, SUN4I_CODEC_ADC_ACTL_PA_EN, 0, sun4i_codec_pa_mixer_controls, ARRAY_SIZE(sun4i_codec_pa_mixer_controls)), - SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("PA Mute", SND_SOC_NOPM, 0, 0, &sun4i_codec_pa_mute), SND_SOC_DAPM_OUTPUT("HP Right"), @@ -480,16 +481,16 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { { "Left Mixer", NULL, "Mixer Enable" }, { "Left Mixer", "Left DAC Playback Switch", "Left DAC" }, - /* Pre-Amplifier Mixer Routes */ - { "Pre-Amplifier", "Mixer Playback Switch", "Left Mixer" }, - { "Pre-Amplifier", "Mixer Playback Switch", "Right Mixer" }, - { "Pre-Amplifier", "DAC Playback Switch", "Left DAC" }, - { "Pre-Amplifier", "DAC Playback Switch", "Right DAC" }, + /* PA Mixer Routes */ + { "PA", "Mixer Playback Switch", "Left Mixer" }, + { "PA", "Mixer Playback Switch", "Right Mixer" }, + { "PA", "DAC Playback Switch", "Left DAC" }, + { "PA", "DAC Playback Switch", "Right DAC" }, /* PA -> HP path */ - { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, - { "HP Right", NULL, "Pre-Amplifier Mute" }, - { "HP Left", NULL, "Pre-Amplifier Mute" }, + { "PA Mute", "Switch", "PA" }, + { "HP Right", NULL, "PA Mute" }, + { "HP Left", NULL, "PA Mute" }, }; static struct snd_soc_codec_driver sun4i_codec_codec = { -- 2.1.4

2 1

[alsa-devel] [PATCH v2] ASoC: wm9713: convert to regmap
by Robert Jarzmik 27 Oct '15

27 Oct '15

Convert the Wolfson WM9713 to regmap API. This will leverage all the regmap functions (debug, registers update, etc ...). As a bonus, this will pave the path to gpio chip introduction, and devicetree support. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v1: fix suspend/resume (that specific part is not tested yet) --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/wm9713.c | 296 +++++++++++++++++++++++----------------------- 2 files changed, 152 insertions(+), 145 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733ecd17f..ba306f8f3717 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -879,6 +879,7 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate + select REGMAP_AC97 # Amp config SND_SOC_LM4857 diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 4083a5130cbd..49f2f41dc553 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -19,6 +19,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <linux/device.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> @@ -39,34 +40,6 @@ struct wm9713_priv { struct mutex lock; }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg); -static int ac97_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int val); - -/* - * WM9713 register cache - * Reg 0x3c bit 15 is used by touch driver. - */ -static const u16 wm9713_reg[] = { - 0x6174, 0x8080, 0x8080, 0x8080, - 0xc880, 0xe808, 0xe808, 0x0808, - 0x00da, 0x8000, 0xd600, 0xaaa0, - 0xaaa0, 0xaaa0, 0x0000, 0x0000, - 0x0f0f, 0x0040, 0x0000, 0x7f00, - 0x0405, 0x0410, 0xbb80, 0xbb80, - 0x0000, 0xbb80, 0x0000, 0x4523, - 0x0000, 0x2000, 0x7eff, 0xffff, - 0x0000, 0x0000, 0x0080, 0x0000, - 0x0000, 0x0000, 0xfffe, 0xffff, - 0x0000, 0x0000, 0x0000, 0xfffe, - 0x4000, 0x0000, 0x0000, 0x0000, - 0xb032, 0x3e00, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0006, - 0x0001, 0x0000, 0x574d, 0x4c13, -}; - #define HPL_MIXER 0 #define HPR_MIXER 1 @@ -220,18 +193,15 @@ static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - u16 status, rate; if (WARN_ON(event != SND_SOC_DAPM_PRE_PMD)) return -EINVAL; /* Gracefully shut down the voice interface. */ - status = ac97_read(codec, AC97_EXTENDED_MID) | 0x1000; - rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF; - ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0f00, 0x0200); schedule_timeout_interruptible(msecs_to_jiffies(1)); - ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00); - ac97_write(codec, AC97_EXTENDED_MID, status); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0f00, 0x0f00); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0x1000, 0x1000); return 0; } @@ -674,39 +644,97 @@ static const struct snd_soc_dapm_route wm9713_audio_map[] = { {"Capture Mono Mux", "Right", "Right Capture Source"}, }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg) +static bool wm9713_readable_reg(struct device *dev, unsigned int reg) { - struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || - reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || - reg == AC97_CD) - return soc_ac97_ops->read(wm9713->ac97, reg); - else { - reg = reg >> 1; - - if (reg >= (ARRAY_SIZE(wm9713_reg))) - return -EIO; - - return cache[reg]; + switch (reg) { + case AC97_RESET ... AC97_PCM_SURR_DAC_RATE: + case AC97_PCM_LR_ADC_RATE: + case AC97_CENTER_LFE_MASTER: + case AC97_SPDIF ... AC97_LINE1_LEVEL: + case AC97_GPIO_CFG ... 0x5c: + case AC97_CODEC_CLASS_REV ... AC97_PCI_SID: + case 0x74 ... AC97_VENDOR_ID2: + return true; + default: + return false; } } -static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) +static bool wm9713_writeable_reg(struct device *dev, unsigned int reg) { - struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); + switch (reg) { + case AC97_VENDOR_ID1: + case AC97_VENDOR_ID2: + return false; + default: + return wm9713_readable_reg(dev, reg); + } +} - u16 *cache = codec->reg_cache; - soc_ac97_ops->write(wm9713->ac97, reg, val); - reg = reg >> 1; - if (reg < (ARRAY_SIZE(wm9713_reg))) - cache[reg] = val; +static const struct reg_default wm9713_reg_defaults[] = { + { 0x02, 0x8080 }, /* Speaker Output Volume */ + { 0x04, 0x8080 }, /* Headphone Output Volume */ + { 0x06, 0x8080 }, /* Out3/OUT4 Volume */ + { 0x08, 0xc880 }, /* Mono Volume */ + { 0x0a, 0xe808 }, /* LINEIN Volume */ + { 0x0c, 0xe808 }, /* DAC PGA Volume */ + { 0x0e, 0x0808 }, /* MIC PGA Volume */ + { 0x10, 0x00da }, /* MIC Routing Control */ + { 0x12, 0x8000 }, /* Record PGA Volume */ + { 0x14, 0xd600 }, /* Record Routing */ + { 0x16, 0xaaa0 }, /* PCBEEP Volume */ + { 0x18, 0xaaa0 }, /* VxDAC Volume */ + { 0x1a, 0xaaa0 }, /* AUXDAC Volume */ + { 0x1c, 0x0000 }, /* Output PGA Mux */ + { 0x1e, 0x0000 }, /* DAC 3D control */ + { 0x20, 0x0f0f }, /* DAC Tone Control*/ + { 0x22, 0x0040 }, /* MIC Input Select & Bias */ + { 0x24, 0x0000 }, /* Output Volume Mapping & Jack */ + { 0x26, 0x7f00 }, /* Powerdown Ctrl/Stat*/ + { 0x28, 0x0405 }, /* Extended Audio ID */ + { 0x2a, 0x0410 }, /* Extended Audio Start/Ctrl */ + { 0x2c, 0xbb80 }, /* Audio DACs Sample Rate */ + { 0x2e, 0xbb80 }, /* AUXDAC Sample Rate */ + { 0x32, 0xbb80 }, /* Audio ADCs Sample Rate */ + { 0x36, 0x4523 }, /* PCM codec control */ + { 0x3a, 0x2000 }, /* SPDIF control */ + { 0x3c, 0xfdff }, /* Powerdown 1 */ + { 0x3e, 0xffff }, /* Powerdown 2 */ + { 0x40, 0x0000 }, /* General Purpose */ + { 0x42, 0x0000 }, /* Fast Power-Up Control */ + { 0x44, 0x0080 }, /* MCLK/PLL Control */ + { 0x46, 0x0000 }, /* MCLK/PLL Control */ + { 0x4c, 0xfffe }, /* GPIO Pin Configuration */ + { 0x4e, 0xffff }, /* GPIO Pin Polarity / Type */ + { 0x50, 0x0000 }, /* GPIO Pin Sticky */ + { 0x52, 0x0000 }, /* GPIO Pin Wake-Up */ + /* GPIO Pin Status */ + { 0x56, 0xfffe }, /* GPIO Pin Sharing */ + { 0x58, 0x4000 }, /* GPIO PullUp/PullDown */ + { 0x5a, 0x0000 }, /* Additional Functions 1 */ + { 0x5c, 0x0000 }, /* Additional Functions 2 */ + { 0x60, 0xb032 }, /* ALC Control */ + { 0x62, 0x3e00 }, /* ALC / Noise Gate Control */ + { 0x64, 0x0000 }, /* AUXDAC input control */ + { 0x74, 0x0000 }, /* Digitiser Reg 1 */ + { 0x76, 0x0006 }, /* Digitiser Reg 2 */ + { 0x78, 0x0001 }, /* Digitiser Reg 3 */ + { 0x7a, 0x0000 }, /* Digitiser Read Back */ +}; - return 0; -} +static const struct regmap_config wm9713_regmap_config = { + .reg_bits = 16, + .reg_stride = 2, + .val_bits = 16, + .max_register = 0x7e, + .cache_type = REGCACHE_RBTREE, + + .reg_defaults = wm9713_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm9713_reg_defaults), + .volatile_reg = regmap_ac97_default_volatile, + .readable_reg = wm9713_readable_reg, + .writeable_reg = wm9713_writeable_reg, +}; /* PLL divisors */ struct _pll_div { @@ -793,10 +821,8 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, /* turn PLL off ? */ if (freq_in == 0) { /* disable PLL power and select ext source */ - reg = ac97_read(codec, AC97_HANDSET_RATE); - ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080); - reg = ac97_read(codec, AC97_EXTENDED_MID); - ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0080, 0x0080); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0x0200, 0x0200); wm9713->pll_in = 0; return 0; } @@ -806,7 +832,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, if (pll_div.k == 0) { reg = (pll_div.n << 12) | (pll_div.lf << 11) | (pll_div.divsel << 9) | (pll_div.divctl << 8); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); } else { /* write the fractional k to the reg 0x46 pages */ reg2 = (pll_div.n << 12) | (pll_div.lf << 11) | (1 << 10) | @@ -814,33 +840,31 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, /* K [21:20] */ reg = reg2 | (0x5 << 4) | (pll_div.k >> 20); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); /* K [19:16] */ reg = reg2 | (0x4 << 4) | ((pll_div.k >> 16) & 0xf); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); /* K [15:12] */ reg = reg2 | (0x3 << 4) | ((pll_div.k >> 12) & 0xf); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); /* K [11:8] */ reg = reg2 | (0x2 << 4) | ((pll_div.k >> 8) & 0xf); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); /* K [7:4] */ reg = reg2 | (0x1 << 4) | ((pll_div.k >> 4) & 0xf); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); reg = reg2 | (0x0 << 4) | (pll_div.k & 0xf); /* K [3:0] */ - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); } /* turn PLL on and select as source */ - reg = ac97_read(codec, AC97_EXTENDED_MID); - ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff); - reg = ac97_read(codec, AC97_HANDSET_RATE); - ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0x0200, 0x0000); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0080, 0x0000); wm9713->pll_in = freq_in; /* wait 10ms AC97 link frames for the link to stabilise */ @@ -863,10 +887,10 @@ static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai, int tristate) { struct snd_soc_codec *codec = codec_dai->codec; - u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0x9fff; if (tristate) - ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, + 0x6000, 0x0000); return 0; } @@ -879,36 +903,30 @@ static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; - u16 reg; switch (div_id) { case WM9713_PCMCLK_DIV: - reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff; - ac97_write(codec, AC97_HANDSET_RATE, reg | div); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0f00, div); break; case WM9713_CLKA_MULT: - reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffd; - ac97_write(codec, AC97_HANDSET_RATE, reg | div); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0002, div); break; case WM9713_CLKB_MULT: - reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffb; - ac97_write(codec, AC97_HANDSET_RATE, reg | div); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0004, div); break; case WM9713_HIFI_DIV: - reg = ac97_read(codec, AC97_HANDSET_RATE) & 0x8fff; - ac97_write(codec, AC97_HANDSET_RATE, reg | div); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x7000, div); break; case WM9713_PCMBCLK_DIV: - reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xf1ff; - ac97_write(codec, AC97_CENTER_LFE_MASTER, reg | div); + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, 0x0e00, div); break; case WM9713_PCMCLK_PLL_DIV: - reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; - ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x60 | div); + snd_soc_update_bits(codec, AC97_LINE1_LEVEL, + 0x007f, div | 0x60); break; case WM9713_HIFI_PLL_DIV: - reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; - ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x70 | div); + snd_soc_update_bits(codec, AC97_LINE1_LEVEL, + 0x007f, div | 0x70); break; default: return -EINVAL; @@ -921,7 +939,7 @@ static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u16 gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffc5; + u16 gpio = snd_soc_read(codec, AC97_GPIO_CFG) & 0xffc5; u16 reg = 0x8000; /* clock masters */ @@ -974,8 +992,8 @@ static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, break; } - ac97_write(codec, AC97_GPIO_CFG, gpio); - ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + snd_soc_write(codec, AC97_GPIO_CFG, gpio); + snd_soc_write(codec, AC97_CENTER_LFE_MASTER, reg); return 0; } @@ -984,24 +1002,24 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3; + /* enable PCM interface in master mode */ switch (params_width(params)) { case 16: break; case 20: - reg |= 0x0004; + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, + 0x000c, 0x0004); break; case 24: - reg |= 0x0008; + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, + 0x000c, 0x0008); break; case 32: - reg |= 0x000c; + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, + 0x000c, 0x000c); break; } - - /* enable PCM interface in master mode */ - ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); return 0; } @@ -1011,17 +1029,15 @@ static int ac97_hifi_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; int reg; - u16 vra; - vra = ac97_read(codec, AC97_EXTENDED_STATUS); - ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x0001, 0x0001); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = AC97_PCM_FRONT_DAC_RATE; else reg = AC97_PCM_LR_ADC_RATE; - return ac97_write(codec, reg, runtime->rate); + return snd_soc_write(codec, reg, runtime->rate); } static int ac97_aux_prepare(struct snd_pcm_substream *substream, @@ -1029,17 +1045,14 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; - u16 vra, xsle; - vra = ac97_read(codec, AC97_EXTENDED_STATUS); - ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); - xsle = ac97_read(codec, AC97_PCI_SID); - ac97_write(codec, AC97_PCI_SID, xsle | 0x8000); + snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x0001, 0x0001); + snd_soc_update_bits(codec, AC97_PCI_SID, 0x8000, 0x8000); if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; - return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); + return snd_soc_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); } #define WM9713_RATES (SNDRV_PCM_RATE_8000 | \ @@ -1128,27 +1141,23 @@ static struct snd_soc_dai_driver wm9713_dai[] = { static int wm9713_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: /* enable thermal shutdown */ - reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff; - ac97_write(codec, AC97_EXTENDED_MID, reg); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0xe400, 0x0000); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* enable master bias and vmid */ - reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff; - ac97_write(codec, AC97_EXTENDED_MID, reg); - ac97_write(codec, AC97_POWERDOWN, 0x0000); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0xc400, 0x0000); + snd_soc_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_OFF: /* disable everything including AC link */ - ac97_write(codec, AC97_EXTENDED_MID, 0xffff); - ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); - ac97_write(codec, AC97_POWERDOWN, 0xffff); + snd_soc_write(codec, AC97_EXTENDED_MID, 0xffff); + snd_soc_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); + snd_soc_write(codec, AC97_POWERDOWN, 0xffff); break; } return 0; @@ -1156,16 +1165,17 @@ static int wm9713_set_bias_level(struct snd_soc_codec *codec, static int wm9713_soc_suspend(struct snd_soc_codec *codec) { - u16 reg; + regcache_cache_only(codec->component.regmap, true); + snd_soc_cache_sync(codec); /* Disable everything except touchpanel - that will be handled * by the touch driver and left disabled if touch is not in * use. */ - reg = ac97_read(codec, AC97_EXTENDED_MID); - ac97_write(codec, AC97_EXTENDED_MID, reg | 0x7fff); - ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); - ac97_write(codec, AC97_POWERDOWN, 0x6f00); - ac97_write(codec, AC97_POWERDOWN, 0xffff); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0x7fff, + 0x7fff); + snd_soc_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); + snd_soc_write(codec, AC97_POWERDOWN, 0x6f00); + snd_soc_write(codec, AC97_POWERDOWN, 0xffff); return 0; } @@ -1173,8 +1183,7 @@ static int wm9713_soc_suspend(struct snd_soc_codec *codec) static int wm9713_soc_resume(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - int i, ret; - u16 *cache = codec->reg_cache; + int ret; ret = snd_ac97_reset(wm9713->ac97, true, WM9713_VENDOR_ID, WM9713_VENDOR_ID_MASK); @@ -1189,12 +1198,8 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) /* only synchronise the codec if warm reset failed */ if (ret == 0) { - for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i += 2) { - if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || - i == AC97_EXTENDED_MSTATUS || i > 0x66) - continue; - soc_ac97_ops->write(wm9713->ac97, i, cache[i>>1]); - } + regcache_cache_only(codec->component.regmap, false); + snd_soc_cache_sync(codec); } return ret; @@ -1203,16 +1208,23 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) static int wm9713_soc_probe(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - int reg; + struct regmap *regmap; wm9713->ac97 = snd_soc_new_ac97_codec(codec, WM9713_VENDOR_ID, WM9713_VENDOR_ID_MASK); if (IS_ERR(wm9713->ac97)) return PTR_ERR(wm9713->ac97); + regmap = devm_regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); + if (IS_ERR(regmap)) { + snd_soc_free_ac97_codec(wm9713->ac97); + return PTR_ERR(regmap); + } + + snd_soc_codec_init_regmap(codec, regmap); + /* unmute the adc - move to kcontrol */ - reg = ac97_read(codec, AC97_CD) & 0x7fff; - ac97_write(codec, AC97_CD, reg); + snd_soc_update_bits(codec, AC97_CD, 0x7fff, 0x0000); return 0; } @@ -1230,13 +1242,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9713 = { .remove = wm9713_soc_remove, .suspend = wm9713_soc_suspend, .resume = wm9713_soc_resume, - .read = ac97_read, - .write = ac97_write, .set_bias_level = wm9713_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm9713_reg), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = wm9713_reg, .controls = wm9713_snd_ac97_controls, .num_controls = ARRAY_SIZE(wm9713_snd_ac97_controls), -- 2.1.4

3 5

[alsa-devel] alsa audio open error
by Srinivasan S 27 Oct '15

27 Oct '15

Dear Alsa community, I facing the below error, am using am335x & ti sdk 8 Something wrong when execute arecord & aplay : arecord and aplay does not work for the first time & it works after throwing error after 5 times Couple of times I get the below error: root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono Playing WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono ^CAborted by signal Interrupt... Aborted by signal Interrupt... root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 ALSA lib /home/jenkins/amsdk-nightly-build/build-CORTEX_1/arago-tmp-external-linaro-toolchain/work/cortexa8t2hf-vfp-neon-oe-linux-gnueabi/alsa-lib/1.0.27.2-r0/alsa-lib-1.0.27.2/src/pcm/pcm_dsnoop.c:618:(snd_pcm_dsnoop_open) unable to open slave arecord: main:722: audio open error: Invalid argument aplay: playback:2715: read error root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 ALSA lib /home/jenkins/amsdk-nightly-build/build-CORTEX_1/arago-tmp-external-linaro-toolchain/work/cortexa8t2hf-vfp-neon-oe-linux-gnueabi/alsa-lib/1.0.27.2-r0/alsa-lib-1.0.27.2/src/pcm/pcm_dshare.c:710:(snd_pcm_dshare_open) unable to open slave aplay: main:722: audio open error: Invalid argument Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 ALSA lib /home/jenkins/amsdk-nightly-build/build-CORTEX_1/arago-tmp-external-linaro-toolchain/work/cortexa8t2hf-vfp-neon-oe-linux-gnueabi/alsa-lib/1.0.27.2-r0/alsa-lib-1.0.27.2/src/pcm/pcm_dshare.c:710:(snd_pcm_dshare_open) unable to open slave aplay: main:722: audio open error: Invalid argument Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 ALSA lib /home/jenkins/amsdk-nightly-build/build-CORTEX_1/arago-tmp-external-linaro-toolchain/work/cortexa8t2hf-vfp-neon-oe-linux-gnueabi/alsa-lib/1.0.27.2-r0/alsa-lib-1.0.27.2/src/pcm/pcm_dshare.c:710:(snd_pcm_dshare_open) unable to open slave aplay: main:722: audio open error: Invalid argument Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 ALSA lib /home/jenkins/amsdk-nightly-build/build-CORTEX_1/arago-tmp-external-linaro-toolchain/work/cortexa8t2hf-vfp-neon-oe-linux-gnueabi/alsa-lib/1.0.27.2-r0/alsa-lib-1.0.27.2/src/pcm/pcm_dshare.c:710:(snd_pcm_dshare_open) unable to open slave aplay: main:722: audio open error: Invalid argument Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono root@lifeline:/# arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono Playing WAVE 'stdin' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono But there is no answer... Could someone give me some clues? Kindly do the needful as early as possible Many Thanks in advance

1 0

[alsa-devel] No sound after suspend-to-ram
by George Gugulea 27 Oct '15

27 Oct '15

After a fresh reboot, the sound works ok but if I put the computer to sleep (pm-suspend), after it wakes up there is no sound anymore. I did try 'alsa force-reload' after suspend but still no sound. After 'pm-hibernate' sound is back again. I found an old thread here: https://bugzilla.kernel.org/show_bug.cgi?id=49421 where Michael seems to have the same problem which is still unsolved; maybe I can help with more info. I attached the 'alsa-info.sh' output before and after suspend: 'alsa-info.txt.before' and 'alsa-info.txt.after' (as a patch to stay below 80k mail size rule: diff -u alsa-info.txt.before alsa-info.txt.after > alsa-info.txt.patch) The only difference between before and after is this one: @@ -241,7 +241,7 @@ Node 0x11 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x11, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x12 0x12] - Converter: stream=4, channel=0 + Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 The PC is here: http://www.dell.com/us/p/xps-27-2720-aio/pd?ref=PD_OC, but the 2y older version with no touch If I boot into windows (8.1) everything works great, including sound (even after suspend) -- gg

3 15

[alsa-devel] Using snd_pcm_set_params() on a capture handle
by Erik de Castro Lopo 27 Oct '15

27 Oct '15

Hi all, I've got a simple PCM capture test program below, which is basically a slightly tweak version of something from: http://equalarea.com/paul/alsa-audio.html The program uses two different methods of setting up the capture handle; one that works (and is complex) and one that doesn't but seems like it should (and is simple). I'm trying to figure out why the simple one (using snd_pcm_set_params) doesn't work. The program can be compiled and run in two ways: gcc -Wall -O2 -DUSE_SET_PARAMS=1 alsa_capture.c -lasound -o alsa_capture ./alsa_capture which fails with: Read from audio interface failed (Input/output error) and: gcc -Wall -O2 -DUSE_SET_PARAMS=0 alsa_capture.c -lasound -o alsa_capture ./alsa_capture which works as expected. Questions: * Is snd_pcm_set_params() supposed to work on capture handles (I've used it successfully on playback handles)? * Is my use of snd_pcm_set_params() missing something and if so what? * If snd_pcm_set_params() is not supposed to work on capture handles would it be possible to update the documentation to reflect that? Thanks, Erik //--------------------------------------------- #include <stdio.h> #include <stdlib.h> #include <alsa/asoundlib.h> #include <alsa/pcm.h> int main (int argc, char *argv[]) { int i; int err; char *buffer; int buffer_frames = 128; unsigned int rate = 44100; snd_pcm_t *capture_handle; snd_pcm_format_t format = SND_PCM_FORMAT_S32_LE; char* device = "default"; if (argc > 1) device = argv[1]; if ((err = snd_pcm_open (&capture_handle, device, SND_PCM_STREAM_CAPTURE, 0)) < 0) { printf ("Cannot open audio device %s (%s)\n", device, snd_strerror (err)); exit (1) ; } if (USE_SET_PARAMS) { if ((err = snd_pcm_set_params (capture_handle, format, SND_PCM_ACCESS_RW_INTERLEAVED, 2, rate, 0, 500000)) < 0) { /* 0.5sec */ printf ("Capture open error: %s\n", snd_strerror (err)); exit (1) ; } } else { snd_pcm_hw_params_t *hw_params; if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) { printf ("Cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)); exit (1) ; } if ((err = snd_pcm_hw_params_any (capture_handle, hw_params)) < 0) { printf ("Cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)); exit (1) ; } if ((err = snd_pcm_hw_params_set_access (capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { printf ("cannot set access type (%s)\n", snd_strerror (err)); exit (1) ; } if ((err = snd_pcm_hw_params_set_format (capture_handle, hw_params, format)) < 0) { printf ("Cannot set sample format (%s)\n", snd_strerror (err)); exit (1) ; } if ((err = snd_pcm_hw_params_set_rate_near (capture_handle, hw_params, &rate, 0)) < 0) { printf ("Cannot set sample rate (%s)\n", snd_strerror (err)); exit (1) ; } if ((err = snd_pcm_hw_params_set_channels (capture_handle, hw_params, 2)) < 0) { printf ("cannot set channel count (%s)\n", snd_strerror (err)); exit (1) ; } if ((err = snd_pcm_hw_params (capture_handle, hw_params)) < 0) { printf ("cannot set parameters (%s)\n", snd_strerror (err)); exit (1) ; } snd_pcm_hw_params_free (hw_params); } buffer = malloc (buffer_frames * snd_pcm_format_width(format) / 8 * 2); if ((err = snd_pcm_prepare (capture_handle)) < 0) { printf ("cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1) ; } for (i = 0; i < 10; ++i) { if ((err = snd_pcm_readi (capture_handle, buffer, buffer_frames)) != buffer_frames) { printf ("Read from audio interface failed (%s)\n", snd_strerror (err)); exit (1) ; break; } printf("read %d done\n", i); } free(buffer); snd_pcm_close (capture_handle); puts ("Audio interface closed"); return 0; } -- ---------------------------------------------------------------------- Erik de Castro Lopo http://www.mega-nerd.com/

4 10

[alsa-devel] [PATCH 0/18] ASoC: rsnd: use mod base common method on DMA/CMD/SSIU/SSI-parent
by Kuninori Morimoto 27 Oct '15

27 Oct '15

Hi Mark Renesas sound has (SSI/SSIU/SRC/CTU/MIX/DVC/CMD/AudioDMAC/AudioDMACpp). Here, SSI/SRC/CTU/MIX/DVC are implemented as "module". But, SSI parent / SSIU are implemented as part of SSI, CMD is implemented as part of CTU/MIX/DVC AudioDMAC/AudioDMACpp are implemented as part of SSI/SRC. because I assumed that SSI/SRC/CTU/MIX/DVC (which are implemented as module) are MAIN modules, and others are option. But, current style is 1) ununderstandable each IPs setting order when debugging, because debug message can't indicate invisible "modules". 2) there is no flexibility for each devices. This patch-set makes DMA/CMD/SSIU/SSI-parent as common module. All devices are controlled as common module now. 1) - 12) prepare for each device 13) - 18) mod base common method patches. Becase DMA was implemented as part of SSI/SRC, it was difficult to separate it in 1 patch. It used 3 phases here Kuninori Morimoto (18): 1) ASoC: rsnd: DMA become SSI/SRC member 2) ASoC: rsnd: DMA related definition goes to dma.c 3) ASoC: rsnd: rename rsnd_dma_init() to rsnd_dma_attach() 4) ASoC: rsnd: enable to use rsnd_dai_connect() from each mod 5) ASoC: rsnd: remove all modules when DMA fail case 6) ASoC: rsnd: fixup rsnd_dma_of_path method for mod base common method 7) ASoC: rsnd: move rsnd_src_ssi_irq_enable/disable() to ssi.c 8) ASoC: rsnd: disable SRC.out only when stop timing 9) ASoC: rsnd: rsnd_dai_stream has each mod's status insted of rsnd_mod 10) ASoC: rsnd: Don't stop HW even if a large number of errors occur 11) ASoC: rsnd: avoid pointless loop in rsnd_mod_interrupt() 12) ASoC: rsnd: use common rsnd_ssi_status_xxx() 13) ASoC: rsnd: use mod base common method on DMA phase1 14) ASoC: rsnd: use mod base common method on DMA phase2 15) ASoC: rsnd: use mod base common method on DMA phase3 16) ASoC: rsnd: use mod base common method on CMD 17) ASoC: rsnd: use mod base common method on SSIU 18) ASoC: rsnd: use mod base common method on SSI-parent sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/cmd.c | 153 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/core.c | 155 +++++++++++++------------------------------------ sound/soc/sh/rcar/ctu.c | 8 +++ sound/soc/sh/rcar/dma.c | 240 +++++++++++++++++++++++++++++++++++++++++++++------------------------------ sound/soc/sh/rcar/dvc.c | 18 ++++-- sound/soc/sh/rcar/mix.c | 10 +++- sound/soc/sh/rcar/rsnd.h | 88 +++++++++++++++------------- sound/soc/sh/rcar/src.c | 179 ++++++++++++-------------------------------------------- sound/soc/sh/rcar/ssi.c | 423 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++-------------------------------------------------------------- sound/soc/sh/rcar/ssiu.c | 181 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 11 files changed, 860 insertions(+), 597 deletions(-)

2 36

[alsa-devel] [PATCH 0/5] ALSA: firewire-tascam: add MIDI functionality
by Takashi Sakamoto 27 Oct '15

27 Oct '15

Hi, TASCAM FireWire series driver may be newly available in Linux 4.4. For the driver, this patchset adds ALSA MIDI ports to communicate to physical MIDI ports. This patchset includes some fixes, and updates my former post: [alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096739.html Changes: * support system exclusive messages * drop virtual MIDI ports * drop MMAP support via hwdep interface for status and control message Although TASCAM FireWire series has physical controls, this patchset doesn't support them. I think some facilities are required to enable this functionality: * handling MIDI messages on virtual MIDI ports by this driver * handling status and control messages in isochronous packet by this driver * parsing the status and control messages by userspace application * supporting assignment of physical controls to MIDI ports by userspace application * turn on/off LEDs on device by userspace application These work are apparently beyond my current effort. Takashi Sakamoto (5): ALSA: firewire-tascam: add support for incoming MIDI messages by asynchronous transaction ALSA: firewire-tascam: add support for outgoing MIDI messages by asynchronous transaction ALSA: firewire-tascam: add support for MIDI functionality ALSA: firewire-tascam: Turn on/off FireWire LED ALSA: firewire-tascam: change device probing processing sound/firewire/tascam/Makefile | 3 +- sound/firewire/tascam/tascam-midi.c | 135 +++++++++++++ sound/firewire/tascam/tascam-transaction.c | 293 +++++++++++++++++++++++++++++ sound/firewire/tascam/tascam.c | 89 +++++---- sound/firewire/tascam/tascam.h | 30 +++ 5 files changed, 503 insertions(+), 47 deletions(-) create mode 100644 sound/firewire/tascam/tascam-midi.c create mode 100644 sound/firewire/tascam/tascam-transaction.c -- 2.1.4

4 25

[alsa-devel] [PATCH v2 0/4] Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 26 Oct '15

26 Oct '15

From: Marcus Cooper <codekipper(a)gmail.com> This patch set adds support for the Allwinner SPDIF transceiver as present on the A10, A20 and A31 SoC boards. For now just the SPDIF transmitter has been tested on a Mele A2000. In order for this patch set to be functional we require audio clock patches which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery This is version 2 of the patch series. After Mark's initial comments I realised that I shouldn't of blindly copied the register set-up from the original SDK that was released. It doesn't help also that the datasheet that finally emerged lacked a serious amount of detail. If anybody does recognise the IP used for the block then please speak out. In short, much of the code that was in the first release is no longer here. I'm still unsure about the machine driver setup. Simple card seems a bit overkill and what I've added is pretty much a copy of what is done in imx-spdif. I'm not planning on arguing the case for my method so if you can advise then I'm all ears. Thanks in advance, CK --- Changes since v1: - Moved sunxi-machine-spdif.c to seperate patch - replaced sunxi in naming scheme with sun4i in the sun4i-spdif driver. - split tx controller into seperate enable/disable functions - moved setclk and setfmt functionality into hw params - added support for mono signals. - cleaned up probe clock set up. - removed all writes to transmit status registers. - removed of_id - removed power management code. - Added support for more rates. --- Marcus Cooper (4): dt-bindings: add sunxi SPDIF transceiver bindings dt-binding: Add sunxi S/PDIF machine driver ASoC: sunxi: Add S/PDIF machine driver. ASOC: sunxi: Add support for the spdif block .../devicetree/bindings/sound/sunxi,spdif.txt | 49 ++ .../bindings/sound/sunxi-audio-spdif.txt | 36 ++ sound/soc/sunxi/Kconfig | 12 + sound/soc/sunxi/Makefile | 4 + sound/soc/sunxi/sun4i-spdif.c | 612 +++++++++++++++++++++ sound/soc/sunxi/sunxi-machine-spdif.c | 108 ++++ 6 files changed, 821 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/sunxi-audio-spdif.txt create mode 100644 sound/soc/sunxi/sun4i-spdif.c create mode 100644 sound/soc/sunxi/sunxi-machine-spdif.c -- 1.9.1

4 17

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Convert leftover pr_info() and pr_err()
by Takashi Iwai 26 Oct '15

26 Oct '15

Use codec_*() macro instead of pr_*() for improving the log information. The current form even slips from alsa-info.sh. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 3a02e5c14d09..f8a12ca477f1 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2673,13 +2673,13 @@ static bool dspload_wait_loaded(struct hda_codec *codec) do { if (dspload_is_loaded(codec)) { - pr_info("ca0132 DOWNLOAD OK :-) DSP IS RUNNING.\n"); + codec_info(codec, "ca0132 DSP downloaded and running\n"); return true; } msleep(20); } while (time_before(jiffies, timeout)); - pr_err("ca0132 DOWNLOAD FAILED!!! DSP IS NOT RUNNING.\n"); + codec_err(codec, "ca0132 failed to download DSP\n"); return false; } @@ -4375,7 +4375,7 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) { - pr_err("ca0132 dspload_image failed.\n"); + codec_err(codec, "ca0132 DSP load image failed\n"); goto exit_download; } -- 2.6.2

1 0

Re: [alsa-devel] BAT: restructuring code
by Caleb Crome 26 Oct '15

26 Oct '15

On Fri, Oct 23, 2015 at 3:47 AM, Liam Girdwood <liam.r.girdwood(a)linux.intel.com> wrote: > Hi Caleb, > > I have no problem as long as we can still support Android (via tinyalsa) > and Chrome (via alsa-lib) and also tinycompress. You will need to > justify any changes you make in your commit message. > > Btw, probably best to also CC alsa-devel mailing list for these sort of > questions too as other folks doing BAT patches may also be interested. > > Fwiw, a structure like your proposing could also be used for SIMD based > optimisations too (if we ever need them). > > Liam > > On Thu, 2015-10-22 at 17:23 -0700, Caleb Crome wrote: >> Hi guys, >> since i've got several different tests to write, I'm basically >> rewriting a meaningful section of the code to make the >> init/generate/playout/analyze modular. >> >> To that end, I'm creating a struct like this: >> >> >> struct bat_ops { >> (void *)(*init)(int channels, int format, float sample_rate, float >> *frequencies) >> // generate_playout_data is called every period to, well, generate >> the data to be played. >> int (*generate_playout_data)(void *priv, int frames, void *buffer); >> // process_capture_data is called every period to do something >> with the data. >> int (*process_input_data)(void *priv, int frames, const void *buffer); >> // analyze is called at the end of the whole capture time. Can be >> used to analyze and report results. >> int (*analyze)(void *priv); >> void (*free)(**priv); >> }; >> >> Then, we can create a separate ops struct for each type of test, i.e. >> a generated sine wave, reading from a file, ramp test, etc. >> >> The alsa code is fantastically complex, so I'm not touching any of >> that. :-/ Just splitting out the generate, capture, and analysis to >> be modular. >> >> Let me know if you think this is going to be a problem. >> >> -Caleb > > In order to support the molecularity, I need the ability for each module (i.e. test type) to accept its own arguments. In order to accomplish this, I think it would be better to use argp instead of getopt_long. Is it okay if I switch it over? Will that break android? Basically, I need the general options like sample rate, channels and format to be in the first set of arguments, but then be able to call a second parser to manage the module's options, like sine frequency (for the sine based test), or impulse response length (for an impulse response test). Perhaps there's a way to do it with getopt_long, but it doesn't look too simple because getopt_long requires all the options to be already defined before parsing starts, whereas argp can bring in new parsers (from the ops struct previously described). Any thoughts? -Caleb

2 2

[alsa-devel] [GIT PULL] ASoC updates for v4.3
by Mark Brown 26 Oct '15

26 Oct '15

The following changes since commit 32b88194f71d6ae7768a29f87fbba454728273ee: Linux 4.3-rc7 (2015-10-25 10:39:47 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.3-rc6 for you to fetch changes up to bb9a13a0209c56cdf27d125a1f2f6f34378c64f4: Merge remote-tracking branches 'asoc/topic/wm8904', 'asoc/topic/wm8955' and 'asoc/topic/wm8960' into asoc-next (2015-10-26 11:16:20 +0900) ---------------------------------------------------------------- ASoC: Updates for v4.4 Not much core work here, a few small tweaks to interfaces but mainly the changes here are driver ones. Highlights include: - Updates to the topology userspace interface - Big updates to the Renesas support from Morimoto-san - Most of the support for Intel Sky Lake systems. - New drivers for Asahi Kasei Microdevices AK4613, Allwinnner A10, Cirrus Logic WM8998, Dialog DA7219, Nuvoton NAU8825 and Rockchip S/PDIF. ---------------------------------------------------------------- Also a new driver for the Atmel Class D speaker drivers which I forgot in the tag message, sorry. Adam Thomson (11): ASoC: Add SOC_DOUBLE_R_EXT ASoC: codecs: Add da7219 codec driver ASoC: da7219: Add bindings documentation for DA7219 audio codec ASoC: da7219: Add entry to cover DA7219 bindings document ASoC: da7219: Improve error handling for regulator supplies ASoC: da7219: Use of_match_ptr() when assigning match table ASoC: da7219: Improve error checking of mclk enable/disable ASoC: da7213: Add DT support to codec driver ASoC: da7213: Add support to handle mclk data provided to driver ASoC: da7213: Add bindings documentation for codec driver ASoC: da7213: Correct units description of dmic-clkrate Anatol Pomozov (2): ASoC: nau8825: Add driver for headset chip Nuvoton 8825 ASoC: Document DAI signal polarity Andy Shevchenko (1): ASoC: Intel: use dw_dmac autoconfiguration Axel Lin (20): ASoC: mediatek: Convert to devm_snd_soc_register_card ASoC: au1x: Convert to devm_snd_soc_register_card ASoC: blackfin: Convert to devm_snd_soc_register_card ASoC: mxs-sgtl5000: Convert to devm_snd_soc_register_card ASoC: pxa: Convert to devm_snd_soc_register_card ASoC: sh: siu_dai: Convert to use resource managed APIs ASoC: intel: broadwell: Convert to devm_snd_soc_register_card ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM ASoC: rockchip: Fix missing CLKDEV_LOOKUP dependency ASoC: rl6347a: Clean up unneeded inclusion of header files ASoC: ad193x-spi: Fixup ad193x_spi_id table ASoC: rt286: Fix run time error while modifying const data ASoC: rt298: Make rt298_index_def const ASoC: rockchip: spdif: Convert to use devm_snd_dmaengine_pcm_register ASoC: ad193x-spi: Add adau1328 to ad193x_spi_id table ASoC: ad193x: Drop .volatile_reg implementation ASoC: ssm2518: Drop .volatile_reg implementation ASoC: da7219: Fix da7219->alc_en state when enabling ALC ASoC: Intel: Skylake: Convert to devm_snd_soc_register_card ASoC: rt5645: Sort the order for register bit defines Bard Liao (1): ASoC: rt298: set register non-volatile by default Ben Zhang (3): ASoC: rt5645: Allow 4 channel recording on AIF1 ASoC: nau8825: Add FLL configuration ASoC: nau8825: Show device properties Charles Keepax (4): ASoC: arizona: Add default cases for event switches ASoC: arizona: Add utility function to check if an input is analog ASoC: Add SOC_SINGLE_RANGE_EXT_TLV macro ASoC: wm5110: Add additional analogue input enable for early revs Cyrille Pitchen (1): ASoC: ad193x: add support to ad1934 Dan Carpenter (7): ASoC: wm8904: harmless underflow in wm8904_put_deemph() ASoC: wm8731: harmless underflow in wm8731_put_deemph() ASoC: wm8960: harmless underflow in wm8960_put_deemph() ASoC: wm8903: harmless underflow in wm8903_put_deemph() ASoC: wm8955: harmless underflow in wm8955_put_deemph() ASoC: es8328: harmless underflow in es8328_put_deemph() ASoC: wm2000: a couple harmless underflows Dan Williams (1): ASoC: Intel: switch from ioremap_cache to memremap Daniel Mack (1): ASoC: pxa: pxa-pcm-lib: switch over to snd-soc-dmaengine-pcm Emilio López (1): ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs Fang, Yang A (1): ASoC: nau8825: add acpi match ID Geert Uytterhoeven (1): ASoC: rsnd: Remove obsolete platform data support Jarkko Nikula (1): spi: atmel: Remove needless bits_per_word and speed_hz tests Jeeja KP (19): ASoC: Intel: Skylake: Remove unused CPU dais ASoC: Intel: Skylake: Add pipe and modules handlers ASoC: Intel: Skylake: Add module configuration helpers ASoC: Intel: Skylake: Initialize and load DSP controls ASoC: Intel: Skylake: Add DSP support and enable it ASoC: Intel: Skylake: Initialize NHLT table ASoC: Intel: Skylake: Check CPA bit in DSP core power down ASoC: Intel: Skylake: Verify the status bit before handling interrupt ASoC: Intel: Skylake: Correct the runtime handler behaviour ASoC: Intel: Skylake: power down all link in suspend ASoC: dapm: Add kcontrol support for PGAs ASoC: dapm: Add startup & shutdown for dai_links ASoC: Intel: Skylake: Fix return for pm_runtime_get_sync() ASoC: Intel: Skylake: Remove BE copier blob memcpy ASoC: Intel: Skylake: Update for ssp node index in copier cfg ASoC: Intel: Skylake: Add support to disable module notifications ASoC: Intel: Skylake: Fix to check return value of dsp init ASoC: Intel: Skylake: Fix to set Non gateway copier cfg ASoC: Intel: Skylake: Add support to topology for module static pin Jie Yang (2): ASoC: compress: add config item for soc-compress to make it compiled only when needed ASoC: Intel: sst: only use sst-firmware when DW DMAC is available John Lin (3): ASoC: rt5645: Add dmi for Broadwell ASoC: rt5645: Add jd_invert for Broadwell ASoC: rt5645: Add dmi "Google Reks" for chrome Julia Lawall (1): ASoC: qcom: change PTR_ERR argument Jyri Sarha (6): ASoC: tlv320aic3x: Improve tdm support ASoC: davinci-mcasp: Add set_tdm_slots() support ASoC: simple-card: Add tdm slot mask support to simple-card ASoC: soc-core: Fix sparse warning in be32_to_cpup() call ASoC: davinci-mcasp: Get rid of bclk_lrclk_ratio in private data ASoC: hdmi: Remove obsolete dummy HDMI codec Kuninori Morimoto (31): ASoC: ak4642: use *dev on ak4642_i2c_probe() ASoC: ak4642: enable to use MCKO as fixed rate output pin on DT ASoC: rsnd: tidyup rsnd_dma_to_xxx() position ASoC: rsnd: rename rsnd_src_pcm_new() to rsnd_src_pcm_new_gen2() ASoC: rsnd: remove unneeded sh_clk header ASoC: rsnd: add missing #sound-dai-cells explain on Document ASoC: rsnd: Add Gen3 initial support ASoC: rsnd: add rsnd_mod_get() macro and use it ASoC: rsnd: ADG uses mod base common method ASoC: rsnd: add common mod confirm method ASoC: rsnd: ADG initial setup on rsnd_adg_ssi_clk_init() ASoC: rsnd: tidyup ADG clock calculate method ASoC: rsnd: tidyup ADG debug message for clock selection ASoC: rsnd: adg: ignore undefined clock error ASoC: rsnd: add AUDIO_CLKOUT support ASoC: rsnd: SND_SOC_RCAR depends on COMMON_CLK ASoC: ak4642: exchange macro name to avoid redefinition ASoC: rsnd: fixup clkout_name[] index error for single AUDIO_CLKOUT SoC ASoC: add ak4613 support ASoC: ak4613: add Digital Playback Volume support ASoC: rsnd: Announce for removing Gen1 SRU support ASoC: rsnd: Gen1 probe is not error ASoC: rsnd: fixup print debug message after read ASoC: rsnd: fixup struct rsnd_gen::res array size ASoC: rsnd: fixup devm_request_irq() option on ssi.c ASoC: rsnd: fixup rsnd_mod_call() behavior for debug ASoC: rsnd: fixup rsnd_dai_call() behavior for unimplemented method ASoC: rsnd: fixup rsnd_dai_call() behavior for .stop/.quit ASoC: rsnd: rename rsnd_mod_hw_start/stop to rsnd_mod_power_on/off ASoC: rsnd: remove unused rsnd_dma_to_ssi() macro ASoC: rsnd: remove duplicate parameter from rsnd_ssi_xxx() Lars-Peter Clausen (4): ASoC: Let snd_soc_limit_volume() take a snd_soc_card ASoC: h1940_uda1380: Constify rate constraints ASoC: rx1950_uda1380: Constify rate constraints ASoC: fsl_sai: Constify rate constraints Luis de Bethencourt (8): ASoC: atmel_wm8904: Fix module autoload for OF platform driver ASoC: fsl-asoc-card: Fix module autoload for OF platform driver ASoC: fsl_sai: Fix module autoload for OF platform driver ASoC: jz4740: Fix module autoload for OF platform driver ASoC: kirkwood: Fix module autoload for OF platform driver ASoC: pxa: Fix module autoload for OF platform drivers ASoC: mop500: Fix module autoload for OF platform driver ASoC: ux500: Fix module autoload for OF platform driver Maciej S. Szmigiero (1): ASoC: fsl-asoc-card: add AC'97 support Mark Brown (27): Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branches 'asoc/topic/ak4613', 'asoc/topic/ak4642', 'asoc/topic/arizona', 'asoc/topic/atmel' and 'asoc/topic/au1x' into asoc-next Merge remote-tracking branches 'asoc/topic/blackfin', 'asoc/topic/davinci', 'asoc/topic/fsl', 'asoc/topic/hdmi' and 'asoc/topic/intel' into asoc-next Merge remote-tracking branches 'asoc/topic/jz4740', 'asoc/topic/kirkwood', 'asoc/topic/mtk', 'asoc/topic/multi' and 'asoc/topic/mxs' into asoc-next Merge remote-tracking branches 'asoc/topic/pxa', 'asoc/topic/qcom', 'asoc/topic/rcar' and 'asoc/topic/rockchip' into asoc-next Merge remote-tracking branches 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/spear', 'asoc/topic/sunxi' and 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Merge tag 'asoc-fix-v4.3-rc2' into asoc-pxa Merge branch 'fix/rt298' of git://git.kernel.org/.../broonie/sound into asoc-rt298 Merge branch 'topic/fsl-mega-fast' of git://git.kernel.org/.../broonie/sound into asoc-fsl Merge tag 'asoc-v4.3-rc2' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/doc' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branches 'asoc/topic/ad193x' and 'asoc/topic/arizona' into asoc-next Merge remote-tracking branches 'asoc/topic/atmel-classd' and 'asoc/topic/da7213' into asoc-next Merge remote-tracking branch 'asoc/topic/da7219' into asoc-next Merge remote-tracking branches 'asoc/topic/dwc', 'asoc/topic/es8328', 'asoc/topic/fsl' and 'asoc/topic/fsl-card' into asoc-next Merge remote-tracking branches 'asoc/topic/imx' and 'asoc/topic/intel' into asoc-next Merge remote-tracking branches 'asoc/topic/nau8825' and 'asoc/topic/pxa' into asoc-next Merge remote-tracking branches 'asoc/topic/rcar', 'asoc/topic/rl6347a', 'asoc/topic/rockchip' and 'asoc/topic/rt286' into asoc-next Merge remote-tracking branches 'asoc/topic/rt298', 'asoc/topic/rt5640', 'asoc/topic/samsung' and 'asoc/topic/sh' into asoc-next Merge remote-tracking branches 'asoc/topic/ssm2518' and 'asoc/topic/sunxi' into asoc-next Merge remote-tracking branches 'asoc/topic/topology', 'asoc/topic/wm2000', 'asoc/topic/wm8731' and 'asoc/topic/wm8903' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8904', 'asoc/topic/wm8955' and 'asoc/topic/wm8960' into asoc-next Maruthi Srinivas Bayyavarapu (2): ASoC: dwc: support dw i2s in slave mode ASoC: dwc: add check for master/slave format Masanari Iida (1): ASoC: sh: Fit typo in Kconfig Maxime Ripard (4): ASoC: sunxi: Add the Allwinner A10 codec bindings ASoC: sun4i-codec: pass through clk_set_rate error ASoC: sun4i-codec: Remove redundant SND_PCM_RATE_KNOT ASoC: sun4i-codec: Remove the routing property Mengdong Lin (2): ASoC: topology: ABI - Change stream formats to a bitwise flag ASoC: topology: ABI - Remove tdm_slot & dai_fmt from snd_soc_tplg_stream Mythri P K (1): ASoC: dapm: Add snd_soc_dapm_kcontrol_widget() Nicolin Chen (1): ASoC: fsl: Use #ifdef instead of #if for CONFIG_PM_SLEEP Oder Chiou (7): ASoC: rt5645: Remove the repeated definitions ASoC: rt5645: Add the control of ClassD modulator Speaker Gain Ratio ASoC: rt5645: Prevent the weird sound of the headphone while rebooting ASoC: rt5645: Recheck the jack detect status after resuming from S3 ASoC: rt5640: Revise the input pin name of IN1 and IN2 in document of the devicetree ASoC: rt5640: Fill up the IN3's support ASoC: rt5645: Add the HWEQ for the speaker output Omair M Abdullah (1): ASoC: Intel: Skylake - Add Skylake RT286 I2S machine driver Ricard Wanderlof (1): ASoC: Handle multiple codecs with split playback / capture Richard Fitzgerald (2): ASoC: arizona: add defines for single-input gain control ASoC: wm8998: Initial WM8998 codec driver Sjoerd Simons (4): ASoC: rockchip: namespace rockchip i2s module name ASoC: dt-bindings: add rockchip tranceiver bindings ASoC: rockchip: Add rockchip SPDIF transceiver driver ASoC: rockchip: Drop unneeded properties rockchip i2s/spdif bindings Songjun Wu (2): ASoC: atmel-classd: add the Audio Class D Amplifier ASoC: atmel-classd: DT binding for Class D audio amplifier driver Subhransu S. Prusty (2): ASoC: Intel: Skylake: Fix to use correct macros for the path iteration ASoC: Intel: Skylake: Fix to fill all sink/source pipe params Sudip Mukherjee (2): ASoC: Intel: remove unused function ASoC: rt5645: fix build warning Sugar Zhang (3): ASoC: rockchip: i2s: add 8 channels capture support ASoC: rockchip: add capture property ASoC: rockchip: i2s: share tx/rx lrck when symmetric_rates enabled Vedang Patel (6): ASoC: topology: ABI - Add PCM Support and bump ABI version to 4 ASoC: topology: ABI - Remove unused struct snd_soc_tplg_stream_config ASoC: topology: ABI - Use __le32 instead of __u32 in snd_soc_tplg_dapm_widget ASoC: topology: ABI - Add name element to snd_soc_tplg_stream ASoC: topology: ABI - Add the type for BE DAI link ASoC: topology: ABI - Add configuration for BE & Codec-Codec DAI Links Vinod Koul (4): ASoC: Intel: Skylake: add DSP platform widget event handlers ASoC: Intel: Skylake: Add FE and BE hw_params handling ASoC: Intel: Skylake: Add topology core init and handlers ASoC: Intel: Skylake: Modify the log level Zidan Wang (5): ASoC: imx-spdif: add snd_soc_pm_ops for spdif machine driver ASoC: fsl_sai: Add driver suspend and resume to support MEGA Fast ASoC: fsl_spdif: Add driver suspend and resume to support MEGA Fast ASoC: fsl_ssi: Add driver suspend and resume to support MEGA Fast ASoC: fsl_esai: Add driver suspend and resume to support MEGA Fast kbuild test robot (2): ASoC: sunxi: fix platform_no_drv_owner.cocci warnings ASoC: nau8825: fix platform_no_drv_owner.cocci warnings Documentation/devicetree/bindings/sound/ak4613.txt | 17 + Documentation/devicetree/bindings/sound/ak4642.txt | 22 +- .../devicetree/bindings/sound/atmel-classd.txt | 52 + Documentation/devicetree/bindings/sound/da7213.txt | 41 + Documentation/devicetree/bindings/sound/da7219.txt | 106 ++ .../devicetree/bindings/sound/fsl-asoc-card.txt | 10 +- .../devicetree/bindings/sound/nau8825.txt | 102 + .../devicetree/bindings/sound/renesas,rsnd.txt | 7 + .../devicetree/bindings/sound/rockchip-i2s.txt | 6 +- .../devicetree/bindings/sound/rockchip-spdif.txt | 40 + Documentation/devicetree/bindings/sound/rt5640.txt | 9 +- .../devicetree/bindings/sound/sun4i-codec.txt | 27 + .../devicetree/bindings/sound/tdm-slot.txt | 11 +- MAINTAINERS | 1 + drivers/spi/spi-atmel.c | 21 +- include/sound/da7213.h | 3 - include/sound/da7219-aad.h | 99 + include/sound/da7219.h | 55 + include/sound/designware_i2s.h | 2 + include/sound/pxa2xx-lib.h | 1 - include/sound/rt5640.h | 3 +- include/sound/rt5645.h | 2 + include/sound/simple_card.h | 2 + include/sound/soc-dai.h | 19 +- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 27 +- include/uapi/sound/asoc.h | 76 +- sound/arm/pxa2xx-ac97.c | 13 +- sound/arm/pxa2xx-pcm-lib.c | 201 +- sound/arm/pxa2xx-pcm.c | 12 +- sound/arm/pxa2xx-pcm.h | 2 - sound/soc/Kconfig | 6 +- sound/soc/Makefile | 4 +- sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 679 +++++++ sound/soc/atmel/atmel-classd.h | 120 ++ sound/soc/atmel/atmel_wm8904.c | 1 + sound/soc/au1x/db1000.c | 10 +- sound/soc/au1x/db1200.c | 10 +- sound/soc/blackfin/bf5xx-ad1836.c | 11 +- sound/soc/blackfin/bfin-eval-adau1373.c | 12 +- sound/soc/blackfin/bfin-eval-adau1701.c | 12 +- sound/soc/blackfin/bfin-eval-adav80x.c | 12 +- sound/soc/codecs/Kconfig | 23 +- sound/soc/codecs/Makefile | 10 +- sound/soc/codecs/ad193x-i2c.c | 8 +- sound/soc/codecs/ad193x-spi.c | 16 +- sound/soc/codecs/ad193x.c | 128 +- sound/soc/codecs/ad193x.h | 9 +- sound/soc/codecs/ak4613.c | 497 +++++ sound/soc/codecs/ak4642.c | 153 +- sound/soc/codecs/arizona.c | 49 + sound/soc/codecs/arizona.h | 14 +- sound/soc/codecs/da7213.c | 190 +- sound/soc/codecs/da7213.h | 8 +- sound/soc/codecs/da7219-aad.c | 823 ++++++++ sound/soc/codecs/da7219-aad.h | 212 +++ sound/soc/codecs/da7219.c | 1955 ++++++++++++++++++++ sound/soc/codecs/da7219.h | 820 ++++++++ sound/soc/codecs/es8328.c | 2 +- sound/soc/codecs/hdmi.c | 109 -- sound/soc/codecs/nau8825.c | 1309 +++++++++++++ sound/soc/codecs/nau8825.h | 341 ++++ sound/soc/codecs/rl6347a.c | 19 +- sound/soc/codecs/rl6347a.h | 2 + sound/soc/codecs/rt286.c | 9 +- sound/soc/codecs/rt298.c | 11 +- sound/soc/codecs/rt5640.c | 22 +- sound/soc/codecs/rt5645.c | 172 +- sound/soc/codecs/rt5645.h | 30 +- sound/soc/codecs/ssm2518.c | 6 - sound/soc/codecs/tlv320aic3x.c | 30 +- sound/soc/codecs/wm2000.c | 4 +- sound/soc/codecs/wm5110.c | 187 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8998.c | 1430 ++++++++++++++ sound/soc/codecs/wm8998.h | 23 + sound/soc/davinci/davinci-mcasp.c | 305 +-- sound/soc/dwc/designware_i2s.c | 123 +- sound/soc/fsl/fsl-asoc-card.c | 141 +- sound/soc/fsl/fsl_esai.c | 88 + sound/soc/fsl/fsl_sai.c | 38 +- sound/soc/fsl/fsl_spdif.c | 73 + sound/soc/fsl/fsl_ssi.c | 102 + sound/soc/fsl/imx-spdif.c | 1 + sound/soc/generic/simple-card.c | 8 +- sound/soc/intel/Kconfig | 15 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 19 +- sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/broadwell.c | 9 +- sound/soc/intel/boards/skl_rt286.c | 259 +++ sound/soc/intel/common/Makefile | 6 +- sound/soc/intel/common/sst-dsp-priv.h | 1 + sound/soc/intel/common/sst-dsp.c | 2 + sound/soc/intel/common/sst-dsp.h | 2 + sound/soc/intel/common/sst-firmware.c | 10 +- sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-messages.c | 44 +- sound/soc/intel/skylake/skl-nhlt.c | 10 +- sound/soc/intel/skylake/skl-pcm.c | 176 +- sound/soc/intel/skylake/skl-sst-dsp.c | 7 +- sound/soc/intel/skylake/skl-sst-ipc.c | 12 + sound/soc/intel/skylake/skl-sst-ipc.h | 1 + sound/soc/intel/skylake/skl-sst.c | 56 +- sound/soc/intel/skylake/skl-topology.c | 1252 +++++++++++++ sound/soc/intel/skylake/skl-topology.h | 36 +- sound/soc/intel/skylake/skl-tplg-interface.h | 83 + sound/soc/intel/skylake/skl.c | 32 +- sound/soc/intel/skylake/skl.h | 17 +- sound/soc/jz4740/jz4740-i2s.c | 1 + sound/soc/kirkwood/armada-370-db.c | 1 + sound/soc/mediatek/mt8173-max98090.c | 11 +- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 11 +- sound/soc/mxs/mxs-sgtl5000.c | 6 +- sound/soc/omap/rx51.c | 2 +- sound/soc/pxa/brownstone.c | 9 +- sound/soc/pxa/corgi.c | 11 +- sound/soc/pxa/e740_wm9705.c | 5 +- sound/soc/pxa/e750_wm9705.c | 5 +- sound/soc/pxa/e800_wm9712.c | 5 +- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/imote2.c | 11 +- sound/soc/pxa/mioa701_wm9713.c | 11 +- sound/soc/pxa/palm27x.c | 9 +- sound/soc/pxa/poodle.c | 11 +- sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/pxa/pxa2xx-ac97.c | 49 +- sound/soc/pxa/pxa2xx-i2s.c | 3 + sound/soc/pxa/pxa2xx-pcm.c | 33 +- sound/soc/pxa/spitz.c | 5 +- sound/soc/pxa/tosa.c | 5 +- sound/soc/pxa/ttc-dkb.c | 12 +- sound/soc/qcom/lpass-cpu.c | 3 +- sound/soc/rockchip/Kconfig | 12 +- sound/soc/rockchip/Makefile | 6 +- sound/soc/rockchip/rockchip_i2s.c | 48 +- sound/soc/rockchip/rockchip_i2s.h | 16 + sound/soc/rockchip/rockchip_spdif.c | 405 ++++ sound/soc/rockchip/rockchip_spdif.h | 63 + sound/soc/samsung/h1940_uda1380.c | 5 +- sound/soc/samsung/rx1950_uda1380.c | 5 +- sound/soc/sh/Kconfig | 3 +- sound/soc/sh/rcar/adg.c | 306 ++- sound/soc/sh/rcar/core.c | 66 +- sound/soc/sh/rcar/ctu.c | 16 +- sound/soc/sh/rcar/dma.c | 2 +- sound/soc/sh/rcar/dvc.c | 16 +- sound/soc/sh/rcar/gen.c | 14 +- sound/soc/sh/rcar/mix.c | 16 +- {include/sound => sound/soc/sh/rcar}/rcar_snd.h | 1 - sound/soc/sh/rcar/rsnd.h | 29 +- sound/soc/sh/rcar/src.c | 27 +- sound/soc/sh/rcar/ssi.c | 113 +- sound/soc/sh/siu_dai.c | 85 +- sound/soc/soc-compress.c | 12 +- sound/soc/soc-core.c | 29 +- sound/soc/soc-dapm.c | 59 +- sound/soc/soc-ops.c | 8 +- sound/soc/soc-pcm.c | 49 + sound/soc/soc-topology.c | 4 +- sound/soc/sunxi/Kconfig | 11 + sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-codec.c | 712 +++++++ sound/soc/ux500/mop500.c | 1 + sound/soc/ux500/ux500_msp_dai.c | 1 + 170 files changed, 14388 insertions(+), 1422 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt create mode 100644 Documentation/devicetree/bindings/sound/da7213.txt create mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt create mode 100644 include/sound/da7219-aad.h create mode 100644 include/sound/da7219.h create mode 100644 sound/soc/atmel/atmel-classd.c create mode 100644 sound/soc/atmel/atmel-classd.h create mode 100644 sound/soc/codecs/ak4613.c create mode 100644 sound/soc/codecs/da7219-aad.c create mode 100644 sound/soc/codecs/da7219-aad.h create mode 100644 sound/soc/codecs/da7219.c create mode 100644 sound/soc/codecs/da7219.h delete mode 100644 sound/soc/codecs/hdmi.c create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h create mode 100644 sound/soc/codecs/wm8998.c create mode 100644 sound/soc/codecs/wm8998.h create mode 100644 sound/soc/intel/boards/skl_rt286.c create mode 100644 sound/soc/intel/skylake/skl-topology.c create mode 100644 sound/soc/rockchip/rockchip_spdif.c create mode 100644 sound/soc/rockchip/rockchip_spdif.h rename {include/sound => sound/soc/sh/rcar}/rcar_snd.h (98%) create mode 100644 sound/soc/sunxi/Kconfig create mode 100644 sound/soc/sunxi/Makefile create mode 100644 sound/soc/sunxi/sun4i-codec.c

2 1

Re: [alsa-devel] Delay issue in ALSA in connecting GSM call
by Srinivasan S 26 Oct '15

26 Oct '15

Dear Clemens, Am awaiting, Could you please provide your valuable guidance or inputs w.r.t the below issue Kindly do the needful as early as possible Many Thanks in advance ________________________________ From: Srinivasan S Sent: Friday, October 23, 2015 2:40 PM To: Clemens Ladisch Cc: alsa-devel(a)alsa-project.org; linux-audio-dev(a)lists.linuxaudio.org Subject: Delay issue in ALSA in connecting GSM call Dear Clemens, I am using the above ALSA application to connect GSM two way call, I see that there seems to be a delay of 3 seconds, when GSM call is made, that ie., it takes 3 seconds to hear on the speaker of our custom board when ever we talk from GSM & the delay is same when ever we talk from board MIC to speaker of GSM But I tried to do loopback using the below command instead of using the above attached ALSA application, but still the delay issue persists arecord -D VINL8 -f S16_LE | aplay -D VOUTR8 Could you please let me know how this delay issue can be resolved, Could you please provide any inputs or guidance w.r.t this issue Any help would be greatly appreciated Kindly do the needful as early as possible Many Thanks in advance

1 0

[alsa-devel] [PATCH] BAT: Change maximum number of channels from 2 to 64
by Caleb Crome 26 Oct '15

26 Oct '15

commit 23521f2e04df7ae39ccdbc7b4a9aa59220b56495 Author: Caleb Crome <ccrome(a)gmail.com> Date: Tue Oct 20 20:57:57 2015 +0000 Change maximum number of channels from 2 to 64. Added a round-robbin frequency assignment from the command line so that if you specify -F 100:200:250 on the command line, then all subsequent channels will get assigned frequencies like this first set. So by specifying -c 6 -F 100:200:250 the resulting frequencies for the 6 channels will be the same as if you specified -c 6 -F 100:200:250:100:200:250 diff --git a/bat/bat.c b/bat/bat.c index 56cfaf6..57c9db8 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -72,16 +72,31 @@ err_exit: static void get_sine_frequencies(struct bat *bat, char *freq) { - char *tmp1; - - tmp1 = strchr(freq, ':'); - if (tmp1 == NULL) { - bat->target_freq[1] = bat->target_freq[0] = atof(optarg); - } else { - *tmp1 = '\0'; - bat->target_freq[0] = atof(optarg); - bat->target_freq[1] = atof(tmp1 + 1); + char *tmp0, *tmp1; + int ncolons; + int nfreqs; + int i; + + ncolons = 0; + tmp1 = freq; + while ((tmp1 = strchr(tmp1, ':'))) { + tmp1++; + ncolons ++; } + nfreqs = ncolons+1; + + tmp0 = freq; + tmp1 = strchr(tmp0, ':'); + for (i = 0; i < MAX_CHANNELS && i < nfreqs; i++) { + if (i+1 < nfreqs) + *tmp1 = '\0'; + bat->target_freq[i] = atof(tmp0); + tmp0 = tmp1+1; + if (i+1 < nfreqs) + tmp1 = strchr(tmp0, ':'); + } + for (i = nfreqs; i < MAX_CHANNELS; i++) + bat->target_freq[i] = bat->target_freq[i % nfreqs]; } static void get_format(struct bat *bat, char *optarg) @@ -301,6 +316,7 @@ _("Usage: bat [-options]...\n" static void set_defaults(struct bat *bat) { + int i; memset(bat, 0, sizeof(struct bat)); /* Set default values */ @@ -312,8 +328,9 @@ static void set_defaults(struct bat *bat) bat->convert_float_to_sample = convert_float_to_int16; bat->convert_sample_to_double = convert_int16_to_double; bat->frames = bat->rate * 2; - bat->target_freq[0] = 997.0; - bat->target_freq[1] = 997.0; + for (i = 0; i < MAX_CHANNELS; i++) { + bat->target_freq[i] = 997.0; + } bat->sigma_k = 3.0; bat->playback.device = NULL; bat->capture.device = NULL; diff --git a/bat/common.h b/bat/common.h index c04452d..6d5e46e 100644 --- a/bat/common.h +++ b/bat/common.h @@ -30,7 +30,7 @@ #define WAV_DATA COMPOSE('d', 'a', 't', 'a') #define WAV_FORMAT_PCM 1 /* PCM WAVE file encoding */ -#define MAX_CHANNELS 2 +#define MAX_CHANNELS 64 #define MIN_CHANNELS 1 #define MAX_PEAKS 10 #define MAX_FRAMES (10 * 1024 * 1024)

3 4

[alsa-devel] [PATCH] ASoC: fsl_sai: fix Rx synchrounous mode
by Stefan Agner 26 Oct '15

26 Oct '15

When using the Rx clock for both, transmitter and receiver, the transmitter needs to be set to synchronous with receiver. This reverts 855675f6e6a6 ("ASoC: fsl_sai: Set SYNC bit of TCR2 to Asynchronous Mode"), which, judiging from the commit log, seems to mixed up between the two synchronous modes: The boolean sai->synchronous[TX] is indicating wheather the SAI should work in Rx synchronous mode (sync Tx with Rx), hence if the value is true, the SYNC field of TCR2 needs to be set to 0x1 ("Synchronous with receiver"). Signed-off-by: Stefan Agner <stefan(a)agner.ch> --- Hi Nicolin, I got this reported from a customer, he tried to use the device tree property "fsl,sai-synchronous-rx" according to the description but it failed. In a quick test, setting the SYNC field of TCR2 to 0x1 helped in his case. So I think this here is right. But given that you chaged the very same line can you review that again and give me a Ack/Nack? -- Stefan sound/soc/fsl/fsl_sai.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a18fd92..1f0e552 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -454,7 +454,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx. * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx. */ - regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, + sai->synchronous[TX] ? FSL_SAI_CR2_SYNC : 0); regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0); -- 2.6.1

3 2

[alsa-devel] [PATCH] ASoC: rt5645: Sort the order for register bit defines
by Axel Lin 26 Oct '15

26 Oct '15

So we have consistent order for register bit defines. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5645.h | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 13ac732..093e46d 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -621,14 +621,14 @@ #define RT5645_G_OM_L_SM_L_SFT 6 #define RT5645_M_BST1_L_SM_L (0x1 << 5) #define RT5645_M_BST1_L_SM_L_SFT 5 +#define RT5645_M_BST3_L_SM_L (0x1 << 4) +#define RT5645_M_BST3_L_SM_L_SFT 4 #define RT5645_M_IN_L_SM_L (0x1 << 3) #define RT5645_M_IN_L_SM_L_SFT 3 -#define RT5645_M_DAC_L1_SM_L (0x1 << 1) -#define RT5645_M_DAC_L1_SM_L_SFT 1 #define RT5645_M_DAC_L2_SM_L (0x1 << 2) #define RT5645_M_DAC_L2_SM_L_SFT 2 -#define RT5645_M_BST3_L_SM_L (0x1 << 4) -#define RT5645_M_BST3_L_SM_L_SFT 4 +#define RT5645_M_DAC_L1_SM_L (0x1 << 1) +#define RT5645_M_DAC_L1_SM_L_SFT 1 /* SPK Right Mixer Control (0x47) */ #define RT5645_G_RM_R_SM_R_MASK (0x3 << 14) @@ -643,14 +643,14 @@ #define RT5645_G_OM_R_SM_R_SFT 6 #define RT5645_M_BST2_R_SM_R (0x1 << 5) #define RT5645_M_BST2_R_SM_R_SFT 5 +#define RT5645_M_BST3_R_SM_R (0x1 << 4) +#define RT5645_M_BST3_R_SM_R_SFT 4 #define RT5645_M_IN_R_SM_R (0x1 << 3) #define RT5645_M_IN_R_SM_R_SFT 3 -#define RT5645_M_DAC_R1_SM_R (0x1 << 1) -#define RT5645_M_DAC_R1_SM_R_SFT 1 #define RT5645_M_DAC_R2_SM_R (0x1 << 2) #define RT5645_M_DAC_R2_SM_R_SFT 2 -#define RT5645_M_BST3_R_SM_R (0x1 << 4) -#define RT5645_M_BST3_R_SM_R_SFT 4 +#define RT5645_M_DAC_R1_SM_R (0x1 << 1) +#define RT5645_M_DAC_R1_SM_R_SFT 1 /* SPOLMIX Control (0x48) */ #define RT5645_M_DAC_L1_SPM_L (0x1 << 15) @@ -675,12 +675,12 @@ #define RT5645_SPK_G_CLSD_SFT 0 /* Mono Output Mixer Control (0x4c) */ +#define RT5645_G_MONOMIX_MASK (0x1 << 10) +#define RT5645_G_MONOMIX_SFT 10 #define RT5645_M_OV_L_MM (0x1 << 9) #define RT5645_M_OV_L_MM_SFT 9 #define RT5645_M_DAC_L2_MA (0x1 << 8) #define RT5645_M_DAC_L2_MA_SFT 8 -#define RT5645_G_MONOMIX_MASK (0x1 << 10) -#define RT5645_G_MONOMIX_SFT 10 #define RT5645_M_BST2_MM (0x1 << 4) #define RT5645_M_BST2_MM_SFT 4 #define RT5645_M_DAC_R1_MM (0x1 << 3) -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: fsl: Use #ifdef instead of #if for CONFIG_PM_SLEEP
by Nicolin Chen 26 Oct '15

26 Oct '15

Change them to #ifdef as CONFIG_PM_SLEEP might not be defined at all. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index aab675a..59f234e 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -900,7 +900,7 @@ static const struct of_device_id fsl_esai_dt_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_esai_dt_ids); -#if CONFIG_PM_SLEEP +#ifdef CONFIG_PM_SLEEP static int fsl_esai_suspend(struct device *dev) { struct fsl_esai *esai = dev_get_drvdata(dev); diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index f652f2e..15173e9 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -807,7 +807,7 @@ static const struct of_device_id fsl_sai_ids[] = { }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -#if CONFIG_PM_SLEEP +#ifdef CONFIG_PM_SLEEP static int fsl_sai_suspend(struct device *dev) { struct fsl_sai *sai = dev_get_drvdata(dev); -- 1.9.1

2 1

Re: [alsa-devel] [PATCH 0/2] Remove .owner from spi_drivers
by Jonathan Cameron 25 Oct '15

25 Oct '15

1 0

[alsa-devel] [PATCH] ASoC: wm9713: convert to regmap
by Robert Jarzmik 24 Oct '15

24 Oct '15

Convert the Wolfson WM9713 to regmap API. This will leverage all the regmap functions (debug, registers update, etc ...). As a bonus, this will pave the path to gpio chip introduction, and devicetree support. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/wm9713.c | 284 ++++++++++++++++++++++++---------------------- 2 files changed, 147 insertions(+), 138 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733ecd17f..ba306f8f3717 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -879,6 +879,7 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate + select REGMAP_AC97 # Amp config SND_SOC_LM4857 diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 4083a5130cbd..18f371b2add5 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -19,6 +19,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <linux/device.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> @@ -39,34 +40,6 @@ struct wm9713_priv { struct mutex lock; }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg); -static int ac97_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int val); - -/* - * WM9713 register cache - * Reg 0x3c bit 15 is used by touch driver. - */ -static const u16 wm9713_reg[] = { - 0x6174, 0x8080, 0x8080, 0x8080, - 0xc880, 0xe808, 0xe808, 0x0808, - 0x00da, 0x8000, 0xd600, 0xaaa0, - 0xaaa0, 0xaaa0, 0x0000, 0x0000, - 0x0f0f, 0x0040, 0x0000, 0x7f00, - 0x0405, 0x0410, 0xbb80, 0xbb80, - 0x0000, 0xbb80, 0x0000, 0x4523, - 0x0000, 0x2000, 0x7eff, 0xffff, - 0x0000, 0x0000, 0x0080, 0x0000, - 0x0000, 0x0000, 0xfffe, 0xffff, - 0x0000, 0x0000, 0x0000, 0xfffe, - 0x4000, 0x0000, 0x0000, 0x0000, - 0xb032, 0x3e00, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0006, - 0x0001, 0x0000, 0x574d, 0x4c13, -}; - #define HPL_MIXER 0 #define HPR_MIXER 1 @@ -220,18 +193,15 @@ static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - u16 status, rate; if (WARN_ON(event != SND_SOC_DAPM_PRE_PMD)) return -EINVAL; /* Gracefully shut down the voice interface. */ - status = ac97_read(codec, AC97_EXTENDED_MID) | 0x1000; - rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF; - ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0f00, 0x0200); schedule_timeout_interruptible(msecs_to_jiffies(1)); - ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00); - ac97_write(codec, AC97_EXTENDED_MID, status); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0f00, 0x0f00); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0x1000, 0x1000); return 0; } @@ -674,39 +644,97 @@ static const struct snd_soc_dapm_route wm9713_audio_map[] = { {"Capture Mono Mux", "Right", "Right Capture Source"}, }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg) +static bool wm9713_readable_reg(struct device *dev, unsigned int reg) { - struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || - reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || - reg == AC97_CD) - return soc_ac97_ops->read(wm9713->ac97, reg); - else { - reg = reg >> 1; - - if (reg >= (ARRAY_SIZE(wm9713_reg))) - return -EIO; - - return cache[reg]; + switch (reg) { + case AC97_RESET ... AC97_PCM_SURR_DAC_RATE: + case AC97_PCM_LR_ADC_RATE: + case AC97_CENTER_LFE_MASTER: + case AC97_SPDIF ... AC97_LINE1_LEVEL: + case AC97_GPIO_CFG ... 0x5c: + case AC97_CODEC_CLASS_REV ... AC97_PCI_SID: + case 0x74 ... AC97_VENDOR_ID2: + return true; + default: + return false; } } -static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) +static bool wm9713_writeable_reg(struct device *dev, unsigned int reg) { - struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); + switch (reg) { + case AC97_VENDOR_ID1: + case AC97_VENDOR_ID2: + return false; + default: + return wm9713_readable_reg(dev, reg); + } +} - u16 *cache = codec->reg_cache; - soc_ac97_ops->write(wm9713->ac97, reg, val); - reg = reg >> 1; - if (reg < (ARRAY_SIZE(wm9713_reg))) - cache[reg] = val; +static const struct reg_default wm9713_reg_defaults[] = { + { 0x02, 0x8080 }, /* Speaker Output Volume */ + { 0x04, 0x8080 }, /* Headphone Output Volume */ + { 0x06, 0x8080 }, /* Out3/OUT4 Volume */ + { 0x08, 0xc880 }, /* Mono Volume */ + { 0x0a, 0xe808 }, /* LINEIN Volume */ + { 0x0c, 0xe808 }, /* DAC PGA Volume */ + { 0x0e, 0x0808 }, /* MIC PGA Volume */ + { 0x10, 0x00da }, /* MIC Routing Control */ + { 0x12, 0x8000 }, /* Record PGA Volume */ + { 0x14, 0xd600 }, /* Record Routing */ + { 0x16, 0xaaa0 }, /* PCBEEP Volume */ + { 0x18, 0xaaa0 }, /* VxDAC Volume */ + { 0x1a, 0xaaa0 }, /* AUXDAC Volume */ + { 0x1c, 0x0000 }, /* Output PGA Mux */ + { 0x1e, 0x0000 }, /* DAC 3D control */ + { 0x20, 0x0f0f }, /* DAC Tone Control*/ + { 0x22, 0x0040 }, /* MIC Input Select & Bias */ + { 0x24, 0x0000 }, /* Output Volume Mapping & Jack */ + { 0x26, 0x7f00 }, /* Powerdown Ctrl/Stat*/ + { 0x28, 0x0405 }, /* Extended Audio ID */ + { 0x2a, 0x0410 }, /* Extended Audio Start/Ctrl */ + { 0x2c, 0xbb80 }, /* Audio DACs Sample Rate */ + { 0x2e, 0xbb80 }, /* AUXDAC Sample Rate */ + { 0x32, 0xbb80 }, /* Audio ADCs Sample Rate */ + { 0x36, 0x4523 }, /* PCM codec control */ + { 0x3a, 0x2000 }, /* SPDIF control */ + { 0x3c, 0xfdff }, /* Powerdown 1 */ + { 0x3e, 0xffff }, /* Powerdown 2 */ + { 0x40, 0x0000 }, /* General Purpose */ + { 0x42, 0x0000 }, /* Fast Power-Up Control */ + { 0x44, 0x0080 }, /* MCLK/PLL Control */ + { 0x46, 0x0000 }, /* MCLK/PLL Control */ + { 0x4c, 0xfffe }, /* GPIO Pin Configuration */ + { 0x4e, 0xffff }, /* GPIO Pin Polarity / Type */ + { 0x50, 0x0000 }, /* GPIO Pin Sticky */ + { 0x52, 0x0000 }, /* GPIO Pin Wake-Up */ + /* GPIO Pin Status */ + { 0x56, 0xfffe }, /* GPIO Pin Sharing */ + { 0x58, 0x4000 }, /* GPIO PullUp/PullDown */ + { 0x5a, 0x0000 }, /* Additional Functions 1 */ + { 0x5c, 0x0000 }, /* Additional Functions 2 */ + { 0x60, 0xb032 }, /* ALC Control */ + { 0x62, 0x3e00 }, /* ALC / Noise Gate Control */ + { 0x64, 0x0000 }, /* AUXDAC input control */ + { 0x74, 0x0000 }, /* Digitiser Reg 1 */ + { 0x76, 0x0006 }, /* Digitiser Reg 2 */ + { 0x78, 0x0001 }, /* Digitiser Reg 3 */ + { 0x7a, 0x0000 }, /* Digitiser Read Back */ +}; - return 0; -} +static const struct regmap_config wm9713_regmap_config = { + .reg_bits = 16, + .reg_stride = 2, + .val_bits = 16, + .max_register = 0x7e, + .cache_type = REGCACHE_RBTREE, + + .reg_defaults = wm9713_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm9713_reg_defaults), + .volatile_reg = regmap_ac97_default_volatile, + .readable_reg = wm9713_readable_reg, + .writeable_reg = wm9713_writeable_reg, +}; /* PLL divisors */ struct _pll_div { @@ -793,10 +821,8 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, /* turn PLL off ? */ if (freq_in == 0) { /* disable PLL power and select ext source */ - reg = ac97_read(codec, AC97_HANDSET_RATE); - ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080); - reg = ac97_read(codec, AC97_EXTENDED_MID); - ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0080, 0x0080); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0x0200, 0x0200); wm9713->pll_in = 0; return 0; } @@ -806,7 +832,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, if (pll_div.k == 0) { reg = (pll_div.n << 12) | (pll_div.lf << 11) | (pll_div.divsel << 9) | (pll_div.divctl << 8); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); } else { /* write the fractional k to the reg 0x46 pages */ reg2 = (pll_div.n << 12) | (pll_div.lf << 11) | (1 << 10) | @@ -814,33 +840,31 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, /* K [21:20] */ reg = reg2 | (0x5 << 4) | (pll_div.k >> 20); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); /* K [19:16] */ reg = reg2 | (0x4 << 4) | ((pll_div.k >> 16) & 0xf); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); /* K [15:12] */ reg = reg2 | (0x3 << 4) | ((pll_div.k >> 12) & 0xf); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); /* K [11:8] */ reg = reg2 | (0x2 << 4) | ((pll_div.k >> 8) & 0xf); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); /* K [7:4] */ reg = reg2 | (0x1 << 4) | ((pll_div.k >> 4) & 0xf); - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); reg = reg2 | (0x0 << 4) | (pll_div.k & 0xf); /* K [3:0] */ - ac97_write(codec, AC97_LINE1_LEVEL, reg); + snd_soc_write(codec, AC97_LINE1_LEVEL, reg); } /* turn PLL on and select as source */ - reg = ac97_read(codec, AC97_EXTENDED_MID); - ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff); - reg = ac97_read(codec, AC97_HANDSET_RATE); - ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0x0200, 0x0000); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0080, 0x0000); wm9713->pll_in = freq_in; /* wait 10ms AC97 link frames for the link to stabilise */ @@ -863,10 +887,10 @@ static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai, int tristate) { struct snd_soc_codec *codec = codec_dai->codec; - u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0x9fff; if (tristate) - ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, + 0x6000, 0x0000); return 0; } @@ -879,36 +903,30 @@ static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; - u16 reg; switch (div_id) { case WM9713_PCMCLK_DIV: - reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff; - ac97_write(codec, AC97_HANDSET_RATE, reg | div); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0f00, div); break; case WM9713_CLKA_MULT: - reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffd; - ac97_write(codec, AC97_HANDSET_RATE, reg | div); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0002, div); break; case WM9713_CLKB_MULT: - reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffb; - ac97_write(codec, AC97_HANDSET_RATE, reg | div); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x0004, div); break; case WM9713_HIFI_DIV: - reg = ac97_read(codec, AC97_HANDSET_RATE) & 0x8fff; - ac97_write(codec, AC97_HANDSET_RATE, reg | div); + snd_soc_update_bits(codec, AC97_HANDSET_RATE, 0x7000, div); break; case WM9713_PCMBCLK_DIV: - reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xf1ff; - ac97_write(codec, AC97_CENTER_LFE_MASTER, reg | div); + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, 0x0e00, div); break; case WM9713_PCMCLK_PLL_DIV: - reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; - ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x60 | div); + snd_soc_update_bits(codec, AC97_LINE1_LEVEL, + 0x007f, div | 0x60); break; case WM9713_HIFI_PLL_DIV: - reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; - ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x70 | div); + snd_soc_update_bits(codec, AC97_LINE1_LEVEL, + 0x007f, div | 0x70); break; default: return -EINVAL; @@ -921,7 +939,7 @@ static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u16 gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffc5; + u16 gpio = snd_soc_read(codec, AC97_GPIO_CFG) & 0xffc5; u16 reg = 0x8000; /* clock masters */ @@ -974,8 +992,8 @@ static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, break; } - ac97_write(codec, AC97_GPIO_CFG, gpio); - ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + snd_soc_write(codec, AC97_GPIO_CFG, gpio); + snd_soc_write(codec, AC97_CENTER_LFE_MASTER, reg); return 0; } @@ -984,24 +1002,24 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3; + /* enable PCM interface in master mode */ switch (params_width(params)) { case 16: break; case 20: - reg |= 0x0004; + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, + 0x000c, 0x0004); break; case 24: - reg |= 0x0008; + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, + 0x000c, 0x0008); break; case 32: - reg |= 0x000c; + snd_soc_update_bits(codec, AC97_CENTER_LFE_MASTER, + 0x000c, 0x000c); break; } - - /* enable PCM interface in master mode */ - ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); return 0; } @@ -1011,17 +1029,15 @@ static int ac97_hifi_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; int reg; - u16 vra; - vra = ac97_read(codec, AC97_EXTENDED_STATUS); - ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x0001, 0x0001); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = AC97_PCM_FRONT_DAC_RATE; else reg = AC97_PCM_LR_ADC_RATE; - return ac97_write(codec, reg, runtime->rate); + return snd_soc_write(codec, reg, runtime->rate); } static int ac97_aux_prepare(struct snd_pcm_substream *substream, @@ -1029,17 +1045,14 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; - u16 vra, xsle; - vra = ac97_read(codec, AC97_EXTENDED_STATUS); - ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); - xsle = ac97_read(codec, AC97_PCI_SID); - ac97_write(codec, AC97_PCI_SID, xsle | 0x8000); + snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x0001, 0x0001); + snd_soc_update_bits(codec, AC97_PCI_SID, 0x8000, 0x8000); if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; - return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); + return snd_soc_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); } #define WM9713_RATES (SNDRV_PCM_RATE_8000 | \ @@ -1128,27 +1141,23 @@ static struct snd_soc_dai_driver wm9713_dai[] = { static int wm9713_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: /* enable thermal shutdown */ - reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff; - ac97_write(codec, AC97_EXTENDED_MID, reg); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0xe400, 0x0000); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* enable master bias and vmid */ - reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff; - ac97_write(codec, AC97_EXTENDED_MID, reg); - ac97_write(codec, AC97_POWERDOWN, 0x0000); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0xc400, 0x0000); + snd_soc_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_OFF: /* disable everything including AC link */ - ac97_write(codec, AC97_EXTENDED_MID, 0xffff); - ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); - ac97_write(codec, AC97_POWERDOWN, 0xffff); + snd_soc_write(codec, AC97_EXTENDED_MID, 0xffff); + snd_soc_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); + snd_soc_write(codec, AC97_POWERDOWN, 0xffff); break; } return 0; @@ -1156,16 +1165,14 @@ static int wm9713_set_bias_level(struct snd_soc_codec *codec, static int wm9713_soc_suspend(struct snd_soc_codec *codec) { - u16 reg; - /* Disable everything except touchpanel - that will be handled * by the touch driver and left disabled if touch is not in * use. */ - reg = ac97_read(codec, AC97_EXTENDED_MID); - ac97_write(codec, AC97_EXTENDED_MID, reg | 0x7fff); - ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); - ac97_write(codec, AC97_POWERDOWN, 0x6f00); - ac97_write(codec, AC97_POWERDOWN, 0xffff); + snd_soc_update_bits(codec, AC97_EXTENDED_MID, 0x7fff, + 0x7fff); + snd_soc_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); + snd_soc_write(codec, AC97_POWERDOWN, 0x6f00); + snd_soc_write(codec, AC97_POWERDOWN, 0xffff); return 0; } @@ -1203,16 +1210,23 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) static int wm9713_soc_probe(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - int reg; + struct regmap *regmap; wm9713->ac97 = snd_soc_new_ac97_codec(codec, WM9713_VENDOR_ID, WM9713_VENDOR_ID_MASK); if (IS_ERR(wm9713->ac97)) return PTR_ERR(wm9713->ac97); + regmap = devm_regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config); + if (IS_ERR(regmap)) { + snd_soc_free_ac97_codec(wm9713->ac97); + return PTR_ERR(regmap); + } + + snd_soc_codec_init_regmap(codec, regmap); + /* unmute the adc - move to kcontrol */ - reg = ac97_read(codec, AC97_CD) & 0x7fff; - ac97_write(codec, AC97_CD, reg); + snd_soc_update_bits(codec, AC97_CD, 0x7fff, 0x0000); return 0; } @@ -1230,13 +1244,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9713 = { .remove = wm9713_soc_remove, .suspend = wm9713_soc_suspend, .resume = wm9713_soc_resume, - .read = ac97_read, - .write = ac97_write, .set_bias_level = wm9713_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm9713_reg), - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, - .reg_cache_default = wm9713_reg, .controls = wm9713_snd_ac97_controls, .num_controls = ARRAY_SIZE(wm9713_snd_ac97_controls), -- 2.1.4

1 0

[alsa-devel] [PATCH 1/2] BAT: eliminate compiler warnings due to printf type mismatch
by Caleb Crome 24 Oct '15

24 Oct '15

Explict type cast to (size_t) to eliminate compiler warnings. Signed-off-by: Caleb Crome <caleb(a)crome.org> --- bat/alsa.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/bat/alsa.c b/bat/alsa.c index 582c604..b2aa34b 100644 --- a/bat/alsa.c +++ b/bat/alsa.c @@ -161,7 +161,7 @@ static int set_snd_pcm_params(struct bat *bat, struct pcm_container *sndpcm) if (err < 0) { fprintf(bat->err, _("Get parameter from device error: ")); fprintf(bat->err, _("period size: %zd %s: %s(%d)\n"), - sndpcm->period_size, + (size_t) sndpcm->period_size, device_name, snd_strerror(err), err); return err; } @@ -170,7 +170,7 @@ static int set_snd_pcm_params(struct bat *bat, struct pcm_container *sndpcm) if (err < 0) { fprintf(bat->err, _("Get parameter from device error: ")); fprintf(bat->err, _("buffer size: %zd %s: %s(%d)\n"), - sndpcm->buffer_size, + (size_t) sndpcm->buffer_size, device_name, snd_strerror(err), err); return err; } @@ -178,7 +178,7 @@ static int set_snd_pcm_params(struct bat *bat, struct pcm_container *sndpcm) if (sndpcm->period_size == sndpcm->buffer_size) { fprintf(bat->err, _("Invalid parameters: can't use period ")); fprintf(bat->err, _("equal to buffer size (%zd)\n"), - sndpcm->period_size); + (size_t) sndpcm->period_size); return -EINVAL; } -- 2.6.1

2 4

[alsa-devel] [PATCH 0/9] ASoC: Intel: Skylake: Fixes and updates
by Vinod Koul 24 Oct '15

24 Oct '15

This series adds various fixes thru the driver. It also updates ssp node structure to latest FW interface. Last patch add support for static pin connections between modules, which is useful for intra pipe connections Jeeja KP (7): ASoC: Intel: Skylake: Fix return for pm_runtime_get_sync() ASoC: Intel: Skylake: Remove BE copier blob memcpy ASoC: Intel: Skylake: Update for ssp node index in copier cfg ASoC: Intel: Skylake: Add support to disable module notifications ASoC: Intel: Skylake: Fix to check return value of dsp init ASoC: Intel: Skylake: Fix to set Non gateway copier cfg ASoC: Intel: Skylake: Add support to topology for module static pin Subhransu S. Prusty (2): ASoC: Intel: Skylake: Fix to use correct macros for the path iteration ASoC: Intel: Skylake: Fix to fill all sink/source pipe params sound/soc/intel/skylake/skl-messages.c | 44 +++++++++++++++-- sound/soc/intel/skylake/skl-pcm.c | 2 +- sound/soc/intel/skylake/skl-topology.c | 74 +++++++++++++++++----------- sound/soc/intel/skylake/skl-topology.h | 8 ++- sound/soc/intel/skylake/skl-tplg-interface.h | 7 ++- 5 files changed, 95 insertions(+), 40 deletions(-) -- 1.9.1

2 20

[alsa-devel] [PATCH 1/2] BAT: Add missing locale.h header
by Caleb Crome 24 Oct '15

24 Oct '15

Missing include became apparent when -O3 was not specified. Added the header to remove error messages during debug compile. Signed-off-by: Caleb Crome <caleb(a)crome.org> --- bat/bat.c | 1 + 1 file changed, 1 insertion(+) diff --git a/bat/bat.c b/bat/bat.c index 56cfaf6..086b9fa 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -23,6 +23,7 @@ #include <getopt.h> #include <math.h> #include <limits.h> +#include <locale.h> #include "aconfig.h" #include "gettext.h" -- 2.6.1

2 2

[alsa-devel] [PATCH 0/2] ALSA: fireworks: refactoring of substream counting
by Takashi Sakamoto 24 Oct '15

24 Oct '15

Hi, This patchset is for refactoring of substream counting. Currently the counting has atomic_t type. When utilizing lock by mutex in snd_efw structure, the counting can have simpler type, unsigned int. Takashi Sakamoto (2): fireworks: move mutex from function callees to callers fireworks: change type of substream counter from atomic_t to unsigned int sound/firewire/fireworks/fireworks.h | 4 ++-- sound/firewire/fireworks/fireworks_midi.c | 16 +++++++++++---- sound/firewire/fireworks/fireworks_pcm.c | 28 +++++++++++++++++-------- sound/firewire/fireworks/fireworks_stream.c | 32 +++++++++++------------------ 4 files changed, 46 insertions(+), 34 deletions(-) -- 2.1.4

1 2

[alsa-devel] [PATCH 0/8] Add ASoC support for AMD APUs [v4]
by Alex Deucher 24 Oct '15

24 Oct '15

This patch set implements support for i2s audio and new AMD GPUs. The i2s codec is fed by a DMA engine on the GPU. To handle this we create mfd cells which we hang the i2s codec and DMA engine on. Because of this, this patch set covers two subsystems: drm and alsa. The drm patches add support for the ACP hw block which provides the DMA engine for the i2s codec. The alsa patches add the ASoC driver for the i2s codec. Since the alsa changes depend on the drm changes in this patch set as well as some other drm changes queued for 4.3, I'd like to take the alsa patches in via the drm tree. V2 changes: - Use the MFD subsystem rather than adding our own bus - Squash all sub-feature patches together - fix comments mentioned in previous review V3 changes: - Update the designware driver to handle slave mode, amd specific features - Use the designware driver directly for i2s - Move the DMA handling from the GPU driver into the AMD ASoC driver - Change the license on the ASoC driver to GPL v4 changes: - patch "ASoC : dwc : support dw i2s in slave mode" accepted - Add a _dai_fmt() operation that checks to make sure that the mode we're setting corresponds to what we read back from the hardware. - Split specific quirks into separate patches - Set the specific quirks that apply to AMD chips in the acp driver Patch 7 adds the register headers for the ACP block which is a pretty big patch so I've excluded it from email. The entire patch set can be viewed here: http://cgit.freedesktop.org/~agd5f/linux/log/?h=acp-upstream5 Thanks, Alex Maruthi Bayyavarapu (1): drm/amd: add ACP driver support Maruthi Srinivas Bayyavarapu (7): ASoC : dwc : add check for master/slave format ASoC : dwc : add different playback/capture base ASoC : dwc : add quirk for dwc controller instances ASoC : dwc : add quirk for different register offset ASoC : dwc : reconfigure dwc in 'resume' from 'suspend' ASoC : AMD : add ACP 2.2 register headers ASoC: AMD: add AMD ASoC ACP-I2S driver drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/acp/Kconfig | 10 + drivers/gpu/drm/amd/acp/Makefile | 9 + drivers/gpu/drm/amd/acp/acp_hw.c | 127 ++ drivers/gpu/drm/amd/acp/include/acp_gfx_if.h | 49 + drivers/gpu/drm/amd/amdgpu/Makefile | 13 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 12 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 275 +++ drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h | 41 + drivers/gpu/drm/amd/amdgpu/vi.c | 12 + drivers/gpu/drm/amd/include/amd_shared.h | 1 + include/linux/mfd/amd_acp.h | 43 + include/sound/designware_i2s.h | 6 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 4 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 518 ++++++ sound/soc/amd/acp.c | 736 +++++++++ sound/soc/amd/acp.h | 147 ++ sound/soc/amd/include/acp_2_2_d.h | 609 +++++++ sound/soc/amd/include/acp_2_2_enum.h | 1068 ++++++++++++ sound/soc/amd/include/acp_2_2_sh_mask.h | 2292 ++++++++++++++++++++++++++ sound/soc/dwc/designware_i2s.c | 206 ++- 24 files changed, 6121 insertions(+), 64 deletions(-) create mode 100644 drivers/gpu/drm/amd/acp/Kconfig create mode 100644 drivers/gpu/drm/amd/acp/Makefile create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.c create mode 100644 drivers/gpu/drm/amd/acp/include/acp_gfx_if.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h create mode 100644 include/linux/mfd/amd_acp.h create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c create mode 100644 sound/soc/amd/acp.c create mode 100644 sound/soc/amd/acp.h create mode 100644 sound/soc/amd/include/acp_2_2_d.h create mode 100644 sound/soc/amd/include/acp_2_2_enum.h create mode 100644 sound/soc/amd/include/acp_2_2_sh_mask.h -- 1.8.3.1

4 24

[alsa-devel] [RESEND PATCH 1/8] ASoC: dwc: add check for master/slave format
by Alex Deucher 23 Oct '15

23 Oct '15

From: Maruthi Srinivas Bayyavarapu <Maruthi.Bayyavarapu(a)amd.com> DW i2s controller's master/slave config can be read from a read-only register. Machine driver can try to set a master/slave format on cpu-dai using 'set_fmt' of dai ops. A check is added to verify codec is master when dwc is slave and vice-versa. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/soc/dwc/designware_i2s.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 3a52f82..f427325 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -341,12 +341,43 @@ static int dw_i2s_trigger(struct snd_pcm_substream *substream, return ret; } +static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + if (dev->capability & DW_I2S_SLAVE) + ret = 0; + else + ret = -EINVAL; + break; + case SND_SOC_DAIFMT_CBS_CFS: + if (dev->capability & DW_I2S_MASTER) + ret = 0; + else + ret = -EINVAL; + break; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + ret = -EINVAL; + break; + default: + dev_dbg(dev->dev, "dwc : Invalid master/slave format\n"); + ret = -EINVAL; + break; + } + return ret; +} + static struct snd_soc_dai_ops dw_i2s_dai_ops = { .startup = dw_i2s_startup, .shutdown = dw_i2s_shutdown, .hw_params = dw_i2s_hw_params, .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, + .set_fmt = dw_i2s_set_fmt, }; static const struct snd_soc_component_driver dw_i2s_component = { -- 1.8.3.1

1 0

[alsa-devel] [PATCH] ASoC: rt5645: Add the HWEQ for the speaker output
by Oder Chiou 23 Oct '15

23 Oct '15

The patch adds the HWEQ function for the speaker output. User can set the HWEQ parameters using the ALSA binary control byte-by-byte. We use the following struct array to store the HWEQ parameters for implementing as simple as possible. struct rt5645_eq_param_s { unsigned short reg; unsigned short val; }; It supports the variant length of the HWEQ parameters that are required. We add the validating function in the function "rt5645_hweq_put" of the ALSA binary control to avoid the user that puts the invalid parameters. In the HWEQ enable function of speaker event, we also add the validating function to prevent that the invalid parameters are applied to codec. The HWEQ parameters should be controlled by DAPM for a specific sequence, so the parameters will be applied to the codec in the speaker power up event of DAPM, and will be disabled in the speaker power down event of DAPM. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 110 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 110 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9d9797c..2813237 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -42,6 +42,8 @@ #define RT5645_PR_BASE (RT5645_PR_RANGE_BASE + (0 * RT5645_PR_SPACING)) +#define RT5645_HWEQ_NUM 57 + static const struct regmap_range_cfg rt5645_ranges[] = { { .name = "PR", @@ -224,6 +226,11 @@ static const struct reg_default rt5645_reg[] = { { 0xff, 0x6308 }, }; +struct rt5645_eq_param_s { + unsigned short reg; + unsigned short val; +}; + static const char *const rt5645_supply_names[] = { "avdd", "cpvdd", @@ -240,6 +247,7 @@ struct rt5645_priv { struct snd_soc_jack *btn_jack; struct delayed_work jack_detect_work; struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; + struct rt5645_eq_param_s *eq_param; int codec_type; int sysclk; @@ -477,6 +485,86 @@ static const DECLARE_TLV_DB_RANGE(spk_clsd_tlv, 7, 7, TLV_DB_SCALE_ITEM(228, 0, 0) ); +static int rt5645_hweq_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s); + + return 0; +} + +static int rt5645_hweq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + struct rt5645_eq_param_s *eq_param = + (struct rt5645_eq_param_s *)ucontrol->value.bytes.data; + int i; + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + eq_param[i].reg = cpu_to_be16(rt5645->eq_param[i].reg); + eq_param[i].val = cpu_to_be16(rt5645->eq_param[i].val); + } + + return 0; +} + +static bool rt5645_validate_hweq(unsigned short reg) +{ + if ((reg >= 0x1a4 && reg <= 0x1cd) | (reg >= 0x1e5 && reg <= 0x1f8) | + (reg == RT5645_EQ_CTRL2)) + return true; + + return false; +} + +static int rt5645_hweq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + struct rt5645_eq_param_s *eq_param = + (struct rt5645_eq_param_s *)ucontrol->value.bytes.data; + int i; + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + eq_param[i].reg = be16_to_cpu(eq_param[i].reg); + eq_param[i].val = be16_to_cpu(eq_param[i].val); + } + + /* The final setting of the table should be RT5645_EQ_CTRL2 */ + for (i = RT5645_HWEQ_NUM - 1; i >= 0; i--) { + if (eq_param[i].reg == 0) + continue; + else if (eq_param[i].reg != RT5645_EQ_CTRL2) + return 0; + else + break; + } + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + if (!rt5645_validate_hweq(eq_param[i].reg) && + eq_param[i].reg != 0) + return 0; + else if (eq_param[i].reg == 0) + break; + } + + memcpy(rt5645->eq_param, eq_param, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s)); + + return 0; +} + +#define RT5645_HWEQ(xname) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = rt5645_hweq_info, \ + .get = rt5645_hweq_get, \ + .put = rt5645_hweq_put \ +} + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -541,6 +629,7 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* I2S2 function select */ SOC_SINGLE("I2S2 Func Switch", RT5645_GPIO_CTRL1, RT5645_I2S2_SEL_SFT, 1, 1), + RT5645_HWEQ("Speaker HWEQ"), }; /** @@ -631,6 +720,22 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static int rt5645_enable_hweq(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + if (rt5645_validate_hweq(rt5645->eq_param[i].reg)) + regmap_write(rt5645->regmap, rt5645->eq_param[i].reg, + rt5645->eq_param[i].val); + else + break; + } + + return 0; +} + /** * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters * @codec: SoC audio codec device. @@ -1535,6 +1640,7 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: + rt5645_enable_hweq(codec); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, @@ -1543,6 +1649,7 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT5645_EQ_CTRL2, 0); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, 0); @@ -3060,6 +3167,9 @@ static int rt5645_probe(struct snd_soc_codec *codec) snd_soc_dapm_sync(dapm); } + rt5645->eq_param = devm_kzalloc(codec->dev, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s), GFP_KERNEL); + return 0; } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 1/2] BAT: Add missing locale.h header
by Caleb Crome 23 Oct '15

23 Oct '15

Missing include became apparent when -O3 was not specified. Added the header to remove error messages during debug compile. Signed-off-by: Caleb Crome <caleb(a)crome.org> --- bat/bat.c | 1 + 1 file changed, 1 insertion(+) diff --git a/bat/bat.c b/bat/bat.c index 56cfaf6..086b9fa 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -23,6 +23,7 @@ #include <getopt.h> #include <math.h> #include <limits.h> +#include <locale.h> #include "aconfig.h" #include "gettext.h" -- 2.6.1

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[alsa-devel] [PATCH v4 0/2] ASoC: nau8825: Add FLL config and device property debug
by Ben Zhang 23 Oct '15

23 Oct '15

This patch series adds the codec FLL configuration code to allow different input MCLK frequencies, and prints out the received device properties to make debugging easier with multiple versions of ACPI _DSD blob. Changes in v2: - Changed function/struct prefix from nau88xx_ to nau8825_ to match with the rest of the driver. Changes in v3: - Changed the implicit unsigned 64-bit divide to div_u64() to fix build errors on 32-bit arch. Changes in v4: - v2 and v3 updated only patch 1/2, not 2/2. The posted partial patch series makes code review confusing. v4 updates both patches in this series to the latest version. Ben Zhang (2): ASoC: nau8825: Add FLL configuration ASoC: nau8825: Show device properties sound/soc/codecs/nau8825.c | 193 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8825.h | 28 +++++-- 2 files changed, 216 insertions(+), 5 deletions(-) -- 2.6.0.rc2.230.g3dd15c0

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[alsa-devel] I/O errors, snd_pcm_hw_params_get_ functions don't deliver values and system freezes on drop/drain/close
by Bommes, Michael 23 Oct '15

23 Oct '15

Hello, I am trying to develop an audio capture application for an SoC (ASC8852A with a TW2866 Audio Codec). I have SSH access to the SoC and the operating system was set up by someone else. Libraries and drivers were precompiled by the manufacturer, as well as a demo system for a video capturing server. Unfortunately I'm not that familiar with operating systems, driver programming and linux in special (although I know some basics regularly used in high-level development). I'm not sure whether I installed ALSA libraries and drivers correctly. Currently I have: - loaded in kernel (output printed to terminal): Advanced Linux Sound Architecture Driver Version 1.0.18rc3. ASoC version 0.13.2 ALSA device list: No soundcards found. - loaded by /etc/modules and modprobe.conf TW2866_AUDIO CodecNum=2 i2c-gpio bus_num=2 scl0=6 sda0=7 scl1=12 sda1=13 - performed after system startup: snddevices script delivered in the SDK of the SoC. sh snddevices The script is placed in the same folder as the TW2866_AUDIO.ko - alsa.conf was placed in /etc/ALSA/alsa.conf because that place is used in the reference OS and some install script, too. After that I have performed some cat and ls commands: /dev/snd/ is created and holds entries (in the demo system there is another timer entry which is missing in my OS): controlC0 pcmC0D0c pcmC0D0p pcmC0D1c pcmC0D2c pcmC0D3c pcmC0D4c cat /proc/asound/devices 0: [ 0] : control 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 25: [ 0- 1]: digital audio capture 33: : timer and cat /proc/asound/cards 0 [VPL_AUDIO ]: VPL AUDIO - VPL Audio TW2866 Driver VPL Audio Codec Driver, TW2866. Now I have no idea whether the drivers and stuff were installed correctly. I tried to program a little sample program, but once I get pcm device to state SND_PCM_STATE_RUNNING the system freezes if I try to drain, drop or close the application, including all other ssh connections to the SoC so that a hard reset is necessary. Errors on readi occur before (10 seconds between each readi call and the error message), but none of the errors mentioned in the documentation (-EPIPE etc) but a -EIO error (Input/Output). But I'm not 100% sure whether the number of periods and buffer sizes were chosen correctly, so it might be possible that it's another user error (wrong buffer sizes delivered to driver could lead to freezes?). So I've reduced the code to setting hardware parameters and found out that snd_pcm_hw_params_get_ calls don't return errors but some of them always return 0 or don't change the output parameter. Here's sample source code and terminal output: ---------------------------------------------------------------------------------------------------------------- /* Use the newer ALSA API */ #define ALSA_PCM_NEW_HW_PARAMS_API #include <alsa/asoundlib.h> void testError(const char* task, int rc) { if (rc < 0) { fprintf(stderr,"ERROR: %s: %s\n", task, snd_strerror(rc)); exit(1); } } int main() { long loops; int rc; int size; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; snd_pcm_uframes_t frames; char *buffer; /* Open PCM device for recording (capture). */ rc = snd_pcm_open(&handle, "hw:0,0", SND_PCM_STREAM_CAPTURE, 0); testError("snd_pcm_open",rc); /* Allocate a hardware parameters object. */ // snd_pcm_hw_params_alloca(params); snd_pcm_hw_params_malloc(&params); /* Fill it in with default values. */ rc = snd_pcm_hw_params_any(handle, params); testError("snd_pcm_hw_params_any", rc); /* Set the desired hardware parameters. */ /* Interleaved mode */ rc = snd_pcm_hw_params_set_access(handle, params,SND_PCM_ACCESS_RW_INTERLEAVED); testError("snd_pcm_hw_params_set_access", rc); // test whether _set_access was applied snd_pcm_access_t access = SND_PCM_ACCESS_RW_NONINTERLEAVED; snd_pcm_hw_params_get_access (params, &access); if(access != SND_PCM_ACCESS_RW_INTERLEAVED) printf("ERROR! not SND_PCM_ACCESS_RW_INTERLEAVED\n"); /* Signed 16-bit little-endian format */ rc = snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_S16_LE); // non-16 bit formats fail to be set testError("snd_pcm_hw_params_set_format", rc); snd_pcm_format_t gottenFormat = SND_PCM_FORMAT_S8; snd_pcm_hw_params_get_format (params, &gottenFormat); if(gottenFormat != SND_PCM_FORMAT_S16_LE) { printf ("ERROR! not SND_PCM_FORMAT_S16_LE ... %i instead\n", gottenFormat); switch(gottenFormat) { case SND_PCM_FORMAT_S16_LE: printf("SND_PCM_FORMAT_S16_LE\n"); break; case SND_PCM_FORMAT_UNKNOWN: printf("SND_PCM_FORMAT_UNKNOWN\n"); break; default: break; //printf("%ld\n",gottenFormat); } } /* Two channels (stereo) */ rc = snd_pcm_hw_params_set_channels(handle, params, 2); testError("snd_pcm_hw_params_set_channels", rc); /* 44100 bits/second sampling rate (CD quality) */ unsigned int rate = 8000; //snd_pcm_hw_params_set_rate_near(handle, params,&val, &dir); rc = snd_pcm_hw_params_set_rate_near(handle, params,&rate, &dir); testError("snd_pcm_hw_params_set_rate_near", rc); printf("applied rate (return value): %i\n", (int)rate); unsigned int usedRate; rc = snd_pcm_hw_params_get_rate (params, &usedRate, &dir); testError("snd_pcm_hw_params_get_rate", rc); printf("used rate (getter value): %i\n", usedRate); unsigned int usedPeriods; snd_pcm_hw_params_get_periods(params, &usedPeriods, &dir); testError("", rc); printf("default periods: %i\n", usedPeriods); unsigned int periods = 8; // Set number of periods. Periods used to be called fragments. rc = snd_pcm_hw_params_set_periods(handle, params, periods, 0); testError("snd_pcm_hw_params_set_periods", rc); printf("Chosen number of periods (return value): %i\n", periods); rc = snd_pcm_hw_params_get_periods(params, &usedPeriods, &dir); testError("snd_pcm_hw_params_get_periods", rc); printf("used periods (getter value): %i\n", usedPeriods); /* Set period size to 32 frames. */ frames = 32; // snd_pcm_hw_params_set_period_size_near(handle,params, &frames, &dir); snd_pcm_hw_params_set_period_size_near(handle,params, &frames, &dir); testError("snd_pcm_hw_params_set_period_size_near", rc); printf("frames per period tried to set (return value): %ld\n", frames); printf("write to driver\n"); /* Write the parameters to the driver */ rc = snd_pcm_hw_params(handle, params); testError("snd_pcm_hw_params", rc); printf("written to driver\n"); snd_pcm_uframes_t maxSize; snd_pcm_hw_params_get_period_size_max(params, &maxSize, &dir); printf("max period size: %i\n", (int)maxSize); /* Use a buffer large enough to hold one period */ rc = snd_pcm_hw_params_get_period_size(params, &frames, &dir); testError("snd_pcm_hw_params_get_period_size", rc); size = frames * 4; /* 2 bytes/sample, 2 channels */ buffer = (char *) malloc(size); /* We want to loop for 5 seconds */ rc = snd_pcm_hw_params_get_period_time(params, &val, &dir); testError("snd_pcm_hw_params_get_period_time", rc); loops = 5000000 / val; printf("frames per period: %ld\n", frames); // ERROR: here: size will be 0, so loops will be performed but nothing is copied... while (loops > 0) { loops--; rc = snd_pcm_readi(handle, buffer, frames); if (rc == -EPIPE) { /* EPIPE means overrun */ fprintf(stderr, "overrun occurred\n"); snd_pcm_prepare(handle); } else if (rc < 0) { fprintf(stderr, "error from read: %s\n", snd_strerror(rc)); } else if (rc != (int)frames) { fprintf(stderr, "short read, read %d frames\n", rc); } rc = write(1, buffer, size); if (rc != size) fprintf(stderr, "short write: wrote %d bytes\n", rc); } snd_pcm_drain(handle); snd_pcm_close(handle); free(buffer); printf("finished\n"); return 0; } ------------------------------------------------------------------------------------------------------------------ with output: ERROR! not SND_PCM_FORMAT_S16_LE ... 0 instead applied rate (return value): 8000 used rate (getter value): 8000 default periods: 0 Chosen number of periods (return value): 8 (comment: output parameter meant here) used periods (getter value): 0 frames per period tried to set (return value): 32 (comment: output parameter meant here) write to driver hello, alsa~. written to driver max period size: 0 frames per period: 0 finished This code doesn't freeze but getter functions "return" (where I mean the output parameter) 0 for periods and framesPerPeriod and format-getter doesn't return the expected, while sample-rate getter works. I'm a but confused about the "hello, alsa~." output, is that normal behaviour for the driver or might it be some kind of debug output in library or driver? System freeze looks like driver issue, or might this only be a wrong usage of the driver? Is there a way to check which parts of the ALSA installation (library, driver, codecs) do or don't work? How can I be certain that snddevices script and .conf files are sufficient? Thank you very much in advance, Michael

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[alsa-devel] [PATCH] BAT: fix missing header
by Caleb Crome 23 Oct '15

23 Oct '15

There was no warning when compiling with -O3, but gave a warning without it. commit a19fa1f7dff40e56dc4327c5e5fdde9e15bd8a55 Author: Caleb Crome <ccrome(a)gmail.com> Date: Tue Oct 20 20:59:50 2015 +0000 Add missing locale.h header. diff --git a/bat/bat.c b/bat/bat.c index 57c9db8..14666b2 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -23,6 +23,7 @@ #include <getopt.h> #include <math.h> #include <limits.h> +#include <locale.h> #include "aconfig.h" #include "gettext.h"

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[alsa-devel] [PATCH] ASoC: fsl_sai: fix no frame clk in master mode
by Zidan Wang 23 Oct '15

23 Oct '15

After several open/close sai test with ctrl+c, there will be I/O error. The SAI can't work anymore, can't recover. There will be no frame clock. With adding the software reset in trigger stop, the issue can be fixed. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5c737f1..6a947dc 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -504,6 +504,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); + + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); } break; default: -- 1.9.1

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[alsa-devel] [GIT PULL] ASoC updates for v4.3
by Mark Brown 23 Oct '15

23 Oct '15

The following changes since commit 7379047d5585187d1288486d4627873170d0005a: Linux 4.3-rc6 (2015-10-18 16:08:42 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.3-rc6 for you to fetch changes up to f69eccc4b248ce53b1b830034babd0a795d5763c: Merge remote-tracking branches 'asoc/fix/rt298', 'asoc/fix/sx', 'asoc/fix/wm8904' and 'asoc/fix/wm8962' into asoc-linus (2015-10-23 08:44:14 +0900) ---------------------------------------------------------------- ASoC: Fixes for v4.3 A bunch of driver fixes plus one core fix which fixes problems with misreporting values from _SX controls following a recent refactoring. This had gone unnoticed as such controls are quite rare. ---------------------------------------------------------------- Bard Liao (2): ASoC: rt298: correct index default value ASoC: rt298: fix wrong setting of gpio2_en Charles Keepax (2): ASoC: Add info callback for SX_TLV controls ASoC: wm8904: Correct number of EQ registers Jiada Wang (1): ASoC: wm8962: mark cache_dirty flag after software reset in pm_resume Mark Brown (3): Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-fix-rt5645 Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt298', 'asoc/fix/sx', 'asoc/fix/wm8904' and 'asoc/fix/wm8962' into asoc-linus include/sound/soc.h | 6 ++++-- include/sound/wm8904.h | 2 +- sound/soc/codecs/rt298.c | 26 +++++++++++++------------- sound/soc/codecs/wm8962.c | 2 ++ sound/soc/soc-ops.c | 28 ++++++++++++++++++++++++++++ 5 files changed, 48 insertions(+), 16 deletions(-)

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[alsa-devel] [PATCH 1/3] ASoC: h1940_uda1380: Constify rate constraints
by Lars-Peter Clausen 23 Oct '15

23 Oct '15

The rate constraints in this board driver are not modified at runtime, so make them const. While we are at it also remove the unnecessary 0 initializer for the mask field. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/h1940_uda1380.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index c72e9fb..5f5825f 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -26,16 +26,15 @@ #include <mach/gpio-samsung.h> #include "s3c24xx-i2s.h" -static unsigned int rates[] = { +static const unsigned int rates[] = { 11025, 22050, 44100, }; -static struct snd_pcm_hw_constraint_list hw_rates = { +static const struct snd_pcm_hw_constraint_list hw_rates = { .count = ARRAY_SIZE(rates), .list = rates, - .mask = 0, }; static struct snd_soc_jack hp_jack; -- 2.1.4

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[alsa-devel] [PATCH 0/9] ASoC: rsnd: tidyup patch set
by Kuninori Morimoto 23 Oct '15

23 Oct '15

Hi Mark Renesas R-Car sound driver will have mod base common method on SSIU/CMD/AudioDMAC/AudioDMACpp/SSIparent soon. These are not directly related it, but cleanup/fixup/tidyup patch set. Kuninori Morimoto (9): ASoC: rsnd: fixup print debug message after read ASoC: rsnd: fixup struct rsnd_gen::res array size ASoC: rsnd: fixup devm_request_irq() option on ssi.c ASoC: rsnd: fixup rsnd_mod_call() behavior for debug ASoC: rsnd: fixup rsnd_dai_call() behavior for unimplemented method ASoC: rsnd: fixup rsnd_dai_call() behavior for .stop/.quit ASoC: rsnd: rename rsnd_mod_hw_start/stop to rsnd_mod_power_on/off ASoC: rsnd: remove unused rsnd_dma_to_ssi() macro ASoC: rsnd: remove duplicate parameter from rsnd_ssi_xxx() sound/soc/sh/rcar/adg.c | 3 +-- sound/soc/sh/rcar/core.c | 35 ++++++++++------------------------- sound/soc/sh/rcar/ctu.c | 4 ++-- sound/soc/sh/rcar/dma.c | 2 +- sound/soc/sh/rcar/dvc.c | 4 ++-- sound/soc/sh/rcar/gen.c | 14 +++++++++----- sound/soc/sh/rcar/mix.c | 4 ++-- sound/soc/sh/rcar/rsnd.h | 11 +++++++---- sound/soc/sh/rcar/src.c | 6 +++--- sound/soc/sh/rcar/ssi.c | 18 +++++++++--------- 10 files changed, 46 insertions(+), 55 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 1/2] ASoC: rt5640: Revise the input pin name of IN1 and IN2 in document of the devicetree
by Oder Chiou 23 Oct '15

23 Oct '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5640.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index bac4d9a..5d062a5 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -24,9 +24,9 @@ Pins on the device (for linking into audio routes) for RT5639/RT5640: * DMIC2 * MICBIAS1 * IN1P - * IN1R + * IN1N * IN2P - * IN2R + * IN2N * HPOL * HPOR * LOUTL -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [patch] ASoC: wm8904: harmless underflow in wm8904_put_deemph()
by Dan Carpenter 23 Oct '15

23 Oct '15

We try to check that "deemph" is zero or one, but because of an underflow bug, it can also be negative. It's fine, negative values are handled as non-zero. But it's messy and static checkers complain so let's fix it. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index b783743..2aa23f1 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -534,7 +534,7 @@ static int wm8904_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL;

3 2

[alsa-devel] [patch] ASoC: wm8731: harmless underflow in wm8731_put_deemph()
by Dan Carpenter 23 Oct '15

23 Oct '15

The code tries to verify that "deemph" is zero or one, but it fails to account for that it can be negative. It's harmless because negatives are treated the same as one, but we should fix it just to silence the static checker warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index ace8645..07cf1bd 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -132,7 +132,7 @@ static int wm8731_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; int ret = 0; if (deemph > 1)

3 2

[alsa-devel] [patch] ASoC: wm8960: harmless underflow in wm8960_put_deemph()
by Dan Carpenter 23 Oct '15

23 Oct '15

We should only accept "deemph" values of zero and one, but by mistake we accept negatives as well. It's harmless but let's clean it up anyway. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e3b7d0c..dc47c2f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -201,7 +201,7 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL;

3 2

[alsa-devel] [patch] ASoC: wm8903: harmless underflow in wm8903_put_deemph()
by Dan Carpenter 23 Oct '15

23 Oct '15

Valid values for "deemph" are zero and one, but we accidentally allow negative values as well. It's harmless, but static checkers complain and we may as well clean it up. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b011253..e4cc41e 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -452,7 +452,7 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; int ret = 0; if (deemph > 1)

3 2

[alsa-devel] [patch] ASoC: wm8955: harmless underflow in wm8955_put_deemph()
by Dan Carpenter 23 Oct '15

23 Oct '15

Valid values for "deemph" are zero and one, but we mistakenly allow negative values as well. It's harmless but we may as well clean it up. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 12e4435..9db00d5 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -402,7 +402,7 @@ static int wm8955_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL;

3 2

[alsa-devel] [patch] ASoC: wm2000: a couple harmless underflows
by Dan Carpenter 23 Oct '15

23 Oct '15

We want these to be zero or one, but by mistake we also accept negative values. It's harmless but we should still clean it up. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 786abd0..a67ea10 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -620,7 +620,7 @@ static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - int anc_active = ucontrol->value.integer.value[0]; + unsigned int anc_active = ucontrol->value.integer.value[0]; int ret; if (anc_active > 1) @@ -653,7 +653,7 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - int val = ucontrol->value.integer.value[0]; + unsigned int val = ucontrol->value.integer.value[0]; int ret; if (val > 1)

3 2

[alsa-devel] [PATCH v2] ASoC: Intel: sst: only use sst-firmware when DW DMAC is available
by Jie Yang 23 Oct '15

23 Oct '15

Currentlly, we use Synopsys DesignWare DMA Controller for baytrail/haswell/broadwell ADSP firmware loading, but for skylake, we don't use it, compiling sst-firmware.c may introduce error when CONFIG_DW_DMAC_CORE is not enabled: sound/built-in.o: In function `sst_dma_new': (.text+0xd7b38): undefined reference to `dw_dma_probe' sound/built-in.o: In function `sst_dma_free': (.text+0xd7c0a): undefined reference to `dw_dma_remove' Here we only compile sst-firmware when CONFIG_DW_DMAC_CORE is selected, to fix the linking error issue. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/common/Makefile | 6 +++++- sound/soc/intel/common/sst-dsp.c | 2 ++ sound/soc/intel/common/sst-dsp.h | 2 ++ 3 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index f24154c..d910558 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -1,7 +1,11 @@ -snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o +snd-soc-sst-dsp-objs := sst-dsp.o snd-soc-sst-acpi-objs := sst-acpi.o snd-soc-sst-ipc-objs := sst-ipc.o +ifneq ($(CONFIG_DW_DMAC_CORE),) +snd-soc-sst-dsp-objs += sst-firmware.o +endif + obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index a627236..c9452e0 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -420,6 +420,7 @@ void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes) } EXPORT_SYMBOL_GPL(sst_dsp_inbox_read); +#if IS_ENABLED(CONFIG_DW_DMAC_CORE) struct sst_dsp *sst_dsp_new(struct device *dev, struct sst_dsp_device *sst_dev, struct sst_pdata *pdata) { @@ -484,6 +485,7 @@ void sst_dsp_free(struct sst_dsp *sst) sst_dma_free(sst->dma); } EXPORT_SYMBOL_GPL(sst_dsp_free); +#endif /* Module information */ MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/intel/common/sst-dsp.h b/sound/soc/intel/common/sst-dsp.h index 1f45f18..859f0de 100644 --- a/sound/soc/intel/common/sst-dsp.h +++ b/sound/soc/intel/common/sst-dsp.h @@ -216,10 +216,12 @@ struct sst_pdata { void *dsp; }; +#if IS_ENABLED(CONFIG_DW_DMAC_CORE) /* Initialization */ struct sst_dsp *sst_dsp_new(struct device *dev, struct sst_dsp_device *sst_dev, struct sst_pdata *pdata); void sst_dsp_free(struct sst_dsp *sst); +#endif /* SHIM Read / Write */ void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value); -- 2.1.4

2 1

[alsa-devel] Applied "ASoC: Intel: switch from ioremap_cache to memremap" to the asoc tree
by Mark Brown 23 Oct '15

23 Oct '15

The patch ASoC: Intel: switch from ioremap_cache to memremap has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ba40a854ea4f9d81368dc023bd25ac4073058039 Mon Sep 17 00:00:00 2001 From: Dan Williams <dan.j.williams(a)intel.com> Date: Fri, 9 Oct 2015 18:16:36 -0400 Subject: [PATCH] ASoC: Intel: switch from ioremap_cache to memremap In preparation for deprecating ioremap_cache() convert its usage in skl-nhlt to memremap. Signed-off-by: Dan Williams <dan.j.williams(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-nhlt.c | 10 +++++----- sound/soc/intel/skylake/skl.h | 6 +++--- 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 13036b1..b0c7bd1 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -25,7 +25,7 @@ static u8 OSC_UUID[16] = {0x6E, 0x88, 0x9F, 0xA6, 0xEB, 0x6C, 0x94, 0x45, #define DSDT_NHLT_PATH "\\_SB.PCI0.HDAS" -void __iomem *skl_nhlt_init(struct device *dev) +void *skl_nhlt_init(struct device *dev) { acpi_handle handle; union acpi_object *obj; @@ -40,17 +40,17 @@ void __iomem *skl_nhlt_init(struct device *dev) if (obj && obj->type == ACPI_TYPE_BUFFER) { nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer; - return ioremap_cache(nhlt_ptr->min_addr, nhlt_ptr->length); + return memremap(nhlt_ptr->min_addr, nhlt_ptr->length, + MEMREMAP_WB); } dev_err(dev, "device specific method to extract NHLT blob failed\n"); return NULL; } -void skl_nhlt_free(void __iomem *addr) +void skl_nhlt_free(void *addr) { - iounmap(addr); - addr = NULL; + memunmap(addr); } static struct nhlt_specific_cfg *skl_get_specific_cfg( diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index e980d78..dd2e79a 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -62,7 +62,7 @@ struct skl { unsigned int init_failed:1; /* delayed init failed */ struct platform_device *dmic_dev; - void __iomem *nhlt; /* nhlt ptr */ + void *nhlt; /* nhlt ptr */ struct skl_sst *skl_sst; /* sst skl ctx */ struct skl_dsp_resource resource; @@ -83,8 +83,8 @@ struct skl_dma_params { int skl_platform_unregister(struct device *dev); int skl_platform_register(struct device *dev); -void __iomem *skl_nhlt_init(struct device *dev); -void skl_nhlt_free(void __iomem *addr); +void *skl_nhlt_init(struct device *dev); +void skl_nhlt_free(void *addr); struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance, u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn); -- 2.6.1

1 0

[alsa-devel] [PATCH v2 0/3] ASoC: dapm: add PGA kcontrol, dai_link enhancement &
by Vinod Koul 23 Oct '15

23 Oct '15

This adds few small patches in DAPM which we require for Intel DSPs First adds support for having kcontrol on PGAs. This is required for modules which are modelled as PGA but can be user configurable. Next is new callback on snd_soc_dai_link_events(). Lastly a macro to return widget associated with a kcontrol Jeeja KP (2): ASoC: dapm: Add kcontrol support for PGAs ASoC: dapm: Add startup & shutdown for dai_links Mythri P K (1): ASoC: dapm: Add snd_soc_dapm_kcontrol_widget() include/sound/soc-dapm.h | 3 +++ sound/soc/soc-dapm.c | 59 +++++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 56 insertions(+), 6 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH 0/3] ASoC: dapm: add PGA kcontrol, dai_link enhancement &
by Vinod Koul 23 Oct '15

23 Oct '15

This adds few small patches in DAPM which we require for Intel DSPs Firsts adds support for having kcontrol on PGAs. This is required for modules which are modelled as PGA but cna be user configurable. Next is new callback on snd_soc_dai_link_events(). Lastly a macro to return widget associated with a kcontrol Jeeja KP (2): ASoC: dapm: Add kcontrol support for PGAs ASoC: dapm: Add startup & shutdown for dai_links Mythri P K (1): ASoC: dapm: Add snd_soc_dapm_kcontrol_widget() include/sound/soc-dapm.h | 3 +++ sound/soc/soc-dapm.c | 64 ++++++++++++++++++++++++++++++++++++++++++------ 2 files changed, 60 insertions(+), 7 deletions(-) -- 1.9.1

3 7

[alsa-devel] [PATCH 1/2] ASoC: nau8825: Add FLL configuration
by Ben Zhang 23 Oct '15

23 Oct '15

snd_soc_codec_driver.set_pll is implemented to configure the FLL. The codec internal SYSCLK can be from either the MCLK pin directly, or the FLL. This is configured by snd_soc_codec_driver.set_pll. Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 162 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8825.h | 28 ++++++-- 2 files changed, 185 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 5c1badf..010f50b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -29,6 +29,58 @@ #include "nau8825.h" +#define NAU_FREF_MAX 13500000 +#define NAU_FVCO_MAX 100000000 +#define NAU_FVCO_MIN 90000000 + +struct nau88xx_fll { + int mclk_src; + int ratio; + int fll_frac; + int fll_int; + int clk_ref_div; +}; + +struct nau88xx_fll_attr { + unsigned int param; + unsigned int val; +}; + +/* scaling for mclk from sysclk_src output */ +static const struct nau88xx_fll_attr mclk_src_scaling[] = { + { 1, 0x0 }, + { 2, 0x2 }, + { 4, 0x3 }, + { 8, 0x4 }, + { 16, 0x5 }, + { 32, 0x6 }, + { 3, 0x7 }, + { 6, 0xa }, + { 12, 0xb }, + { 24, 0xc }, + { 48, 0xd }, + { 96, 0xe }, + { 5, 0xf }, +}; + +/* ratio for input clk freq */ +static const struct nau88xx_fll_attr fll_ratio[] = { + { 512000, 0x01 }, + { 256000, 0x02 }, + { 128000, 0x04 }, + { 64000, 0x08 }, + { 32000, 0x10 }, + { 8000, 0x20 }, + { 4000, 0x40 }, +}; + +static const struct nau88xx_fll_attr fll_pre_scalar[] = { + { 1, 0x0 }, + { 2, 0x1 }, + { 4, 0x2 }, + { 8, 0x3 }, +}; + static const struct reg_default nau8825_reg_defaults[] = { { NAU8825_REG_ENA_CTRL, 0x00ff }, { NAU8825_REG_CLK_DIVIDER, 0x0050 }, @@ -808,6 +860,115 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) return 0; } +/** + * nau88xx_calc_fll_param - Calculate FLL parameters. + * @fll_in: external clock provided to codec. + * @fs: sampling rate. + * @fll_param: Pointer to structure of FLL parameters. + * + * Calculate FLL parameters to configure codec. + * + * Returns 0 for success or negative error code. + */ +static int nau88xx_calc_fll_param(unsigned int fll_in, unsigned int fs, + struct nau88xx_fll *fll_param) +{ + u64 fref, fvco, fll_input; + int i; + + /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing + * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. + * FREF = freq_in / NAU8825_FLL_REF_DIV_MASK + */ + for (i = 0; i < ARRAY_SIZE(fll_pre_scalar); i++) { + fref = fll_in / fll_pre_scalar[i].param; + if (fref <= NAU_FREF_MAX) + break; + } + if (i == ARRAY_SIZE(fll_pre_scalar)) + return -EINVAL; + fll_param->clk_ref_div = fll_pre_scalar[i].val; + + /* Choose the FLL ratio based on FREF */ + for (i = 0; i < ARRAY_SIZE(fll_ratio); i++) { + if (fref >= fll_ratio[i].param) + break; + } + if (i == ARRAY_SIZE(fll_ratio)) + return -EINVAL; + fll_param->ratio = fll_ratio[i].val; + + /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. + * FDCO must be within the 90MHz - 100MHz or the FFL cannot be + * guaranteed across the full range of operation. + * FDCO = freq_out * 2 * mclk_src_scaling + */ + for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { + fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + if (NAU_FVCO_MIN < fvco && fvco < NAU_FVCO_MAX) + break; + } + if (i == ARRAY_SIZE(mclk_src_scaling)) + return -EINVAL; + fll_param->mclk_src = mclk_src_scaling[i].val; + + /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional + * input based on FDCO, FREF and FLL ratio. + */ + fll_input = (fvco << 16) / (fref * fll_param->ratio); + fll_param->fll_int = fll_input >> 16; + fll_param->fll_frac = fll_input & 0xFFFF; + return 0; +} + +static void nau88xx_fll_apply(struct nau8825 *nau8825, + struct nau88xx_fll *fll_param) +{ + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src); + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1, + NAU8825_FLL_RATIO_MASK, fll_param->ratio); + /* FLL 16-bit fractional input */ + regmap_write(nau8825->regmap, NAU8825_REG_FLL2, fll_param->fll_frac); + /* FLL 10-bit integer input */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL3, + NAU8825_FLL_INTEGER_MASK, fll_param->fll_int); + /* FLL pre-scaler */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL4, + NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); + /* select divided VCO input */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_FILTER_SW_MASK, 0x0000); + /* FLL sigma delta modulator enable */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); +} + +/* freq_out must be 256*Fs in order to achieve the best performance */ +static int nau8825_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct nau88xx_fll fll_param; + int ret, fs; + + fs = freq_out / 256; + ret = nau88xx_calc_fll_param(freq_in, fs, &fll_param); + if (ret < 0) { + dev_err(codec->dev, "Unsupported input clock %d\n", freq_in); + return ret; + } + dev_dbg(codec->dev, "mclk_src=%x ratio=%x fll_frac=%x fll_int=%x clk_ref_div=%x\n", + fll_param.mclk_src, fll_param.ratio, fll_param.fll_frac, + fll_param.fll_int, fll_param.clk_ref_div); + + nau88xx_fll_apply(nau8825, &fll_param); + mdelay(2); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + return 0; +} + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq) { @@ -920,6 +1081,7 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, static struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, .set_sysclk = nau8825_set_sysclk, + .set_pll = nau8825_set_pll, .set_bias_level = nau8825_set_bias_level, .suspend_bias_off = true, diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 8774923..dff8edb 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -101,13 +101,31 @@ #define NAU8825_ENABLE_SAR_SFT 1 /* CLK_DIVIDER (0x3) */ -#define NAU8825_CLK_SRC_SFT 15 -#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) -#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) -#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_SFT 15 +#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_MCLK_SRC_MASK (0xf << 0) + +/* FLL1 (0x04) */ +#define NAU8825_FLL_RATIO_MASK (0x7f << 0) + +/* FLL3 (0x06) */ +#define NAU8825_FLL_INTEGER_MASK (0x3ff << 0) + +/* FLL4 (0x07) */ +#define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) + +/* FLL5 (0x08) */ +#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) /* FLL6 (0x9) */ -#define NAU8825_DCO_EN (1 << 15) +#define NAU8825_DCO_EN_MASK (0x1 << 15) +#define NAU8825_DCO_EN (0x1 << 15) +#define NAU8825_DCO_DIS (0x0 << 15) +#define NAU8825_SDM_EN_MASK (0x1 << 14) +#define NAU8825_SDM_EN (0x1 << 14) +#define NAU8825_SDM_DIS (0x0 << 14) /* HSD_CTRL (0xc) */ #define NAU8825_HSD_AUTO_MODE (1 << 6) -- 2.6.0.rc2.230.g3dd15c0

3 7

[alsa-devel] [PATCH] ASoC: Document DAI signal polarity
by Anatol Pomozov 23 Oct '15

23 Oct '15

Currently there is no clear definition of what FSYNC polarity is. Different drivers use its own definition of what is "normal" and what is "inverted" fsync. This leads to compatibility problems between drivers. For example TegraX1 driver assumes that DSP-A format with frames starting at rising FSYNC edge has "inverted" polarity, while RT5677 assumes it is "normal" polarity. Explicitly specify meaning of BCLK/FSYNC polarity to avoid future compatibility problems. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- include/sound/soc-dai.h | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1..91e2e61 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -48,10 +48,25 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ /* - * DAI hardware signal inversions. + * DAI hardware signal polarity. * * Specifies whether the DAI can also support inverted clocks for the specified * format. + * + * BCLK: + * - "normal" polarity means signal is available at rising edge of BCLK + * - "inverted" polarity means signal is available at falling edge of BCLK + * + * FSYNC "normal" polarity depends on the frame format: + * - I2S: frame consists of left then right channel data. Left channel starts + * with falling FSYNC edge, right channel starts with rising FSYNC edge. + * - Left/Right Justified: frame consists of left then right channel data. + * Left channel starts with rising FSYNC edge, right channel starts with + * falling FSYNC edge. + * - DSP A/B: Frame starts with rising FSYNC edge. + * - AC97: Frame starts with rising FSYNC edge. + * + * "Negative" FSYNC polarity is the one opposite of "normal" polarity. */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ -- 2.6.0.rc2.230.g3dd15c0

3 2

[alsa-devel] Applied "ASoC: fsl-asoc-card: add AC'97 support" to the asoc tree
by Mark Brown 23 Oct '15

23 Oct '15

The patch ASoC: fsl-asoc-card: add AC'97 support has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 50760cad9de969fe85b24465afe6396b8bbc6a3f Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" <mail(a)maciej.szmigiero.name> Date: Sat, 19 Sep 2015 02:00:25 +0200 Subject: [PATCH] ASoC: fsl-asoc-card: add AC'97 support Add AC'97 support to fsl-asoc-card using generic ASoC AC'97 CODEC. The SSI controller will silently enable any TX AC'97 slots that have their bits set in SLOTREQ received from CODEC and then will redirect some of playback samples there. That's why it is important to make sure that any of CODEC playback slots that can pull samples are set to slots 3/4 (standard PCM playback slots). Currently, this applies to S/PDIF slots as they were seen to pull samples sometimes even with S/PDIF output being disabled. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/fsl-asoc-card.txt | 10 +- sound/soc/fsl/fsl-asoc-card.c | 140 ++++++++++++++++----- 2 files changed, 116 insertions(+), 34 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..ce55c0a 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit from the simplification of a new card support and the capability of the wide sample rates support through ASRC. -Note: The card is initially designed for those sound cards who use I2S and - PCM DAI formats. However, it'll be also possible to support those non - I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long - as the driver has been properly upgraded. +Note: The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. The compatible list for this generic sound card currently: + "fsl,imx-audio-ac97" + "fsl,imx-audio-cs42888" "fsl,imx-audio-wm8962" diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5aeb6ed..33628a0 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -14,6 +14,9 @@ #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_platform.h> +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) +#include <sound/ac97_codec.h> +#endif #include <sound/pcm_params.h> #include <sound/soc.h> @@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; +static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) +{ + return priv->dai_fmt == SND_SOC_DAIFMT_AC97; +} + static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, * set_bias_level(), bypass the remaining settings in hw_params(). * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. */ - if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) + if ((priv->card.set_bias_level && + priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || + fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, ext_port--; /* - * Use asynchronous mode (6 wires) for all cases. + * Use asynchronous mode (6 wires) for all cases except AC97. * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ @@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_TCLKDIR; break; default: - return -EINVAL; + if (!fsl_asoc_card_is_ac97(priv)) + return -EINVAL; + } + + if (fsl_asoc_card_is_ac97(priv)) { + int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR; } /* Asynchronous mode can not be set along with RCLKDIR */ - ret = imx_audmux_v2_configure_port(int_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(dev, "audmux internal port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); + + ret = imx_audmux_v2_configure_port(int_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(int_port, int_ptcr, @@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, return ret; } - ret = imx_audmux_v2_configure_port(ext_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(dev, "audmux external port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); + + ret = imx_audmux_v2_configure_port(ext_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, @@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + if (fsl_asoc_card_is_ac97(priv)) { +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + /* + * Use slots 3/4 for S/PDIF so SSI won't try to enable + * other slots and send some samples there + * due to SLOTREQ bits for S/PDIF received from codec + */ + snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, + AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); +#endif + + return 0; + } + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -407,7 +454,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct i2c_client *codec_dev; - struct clk *codec_clk; const char *codec_dai_name; u32 width; int ret; @@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Give a chance to old DT binding */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); - codec_np = of_parse_phandle(np, "audio-codec", 0); - if (!cpu_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); + if (!cpu_np) { + dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } @@ -434,22 +479,24 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto fail; } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; - } + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) + codec_dev = of_find_i2c_device_by_node(codec_np); + else + codec_dev = NULL; asrc_np = of_parse_phandle(np, "audio-asrc", 0); if (asrc_np) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ - codec_clk = clk_get(&codec_dev->dev, NULL); - if (!IS_ERR(codec_clk)) { - priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); - clk_put(codec_clk); + if (codec_dev) { + struct clk *codec_clk = clk_get(&codec_dev->dev, NULL); + + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } } /* Default sample rate and format, will be updated in hw_params() */ @@ -486,11 +533,21 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { + codec_dai_name = "ac97-hifi"; + priv->card.set_bias_level = NULL; + priv->dai_fmt = SND_SOC_DAIFMT_AC97; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; } + if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { + dev_err(&pdev->dev, "failed to find codec device\n"); + ret = -EINVAL; + goto asrc_fail; + } + /* Common settings for corresponding Freescale CPU DAI driver */ if (strstr(cpu_np->name, "ssi")) { /* Only SSI needs to configure AUDMUX */ @@ -507,7 +564,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } - sprintf(priv->name, "%s-audio", codec_dev->name); + snprintf(priv->name, sizeof(priv->name), "%s-audio", + fsl_asoc_card_is_ac97(priv) ? "ac97" : + codec_dev->name); /* Initialize sound card */ priv->pdev = pdev; @@ -531,8 +590,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; - priv->dai_link[0].codec_of_node = codec_np; priv->dai_link[0].codec_dai_name = codec_dai_name; + + if (!fsl_asoc_card_is_ac97(priv)) + priv->dai_link[0].codec_of_node = codec_np; + else { + u32 idx; + + ret = of_property_read_u32(cpu_np, "cell-index", &idx); + if (ret) { + dev_err(&pdev->dev, + "cannot get CPU index property\n"); + goto asrc_fail; + } + + priv->dai_link[0].codec_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, + "ac97-codec.%u", + (unsigned int)idx); + } + priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -543,6 +620,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_link[1].platform_of_node = asrc_np; priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].codec_name = + priv->dai_link[0].codec_name; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; @@ -578,14 +657,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); -fail: of_node_put(codec_np); +fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, -- 2.6.1

1 0

[alsa-devel] [PATCH] ASoC: rt5645: Add dmi "Google Reks" for chrome
by John Lin 23 Oct '15

23 Oct '15

Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index b0d96b6..84c5261 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3212,6 +3212,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"), }, }, + { + .ident = "Google Reks", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Reks"), + }, + }, { } }; -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [RESEND PATCH v3 0/2] ASoC: atmel-classd: add driver for Atmel CLASSD
by Songjun Wu 23 Oct '15

23 Oct '15

The Audio Class D Amplifier driver includes two parts. 1) Driver code to implement the Audio Class D Amplifier function. 2) Device tree binding document, it describes how to add the Audio Class D Amplifier in device tree. Changes in v3: - Remove '.owner' field in variable 'atmel_classd_driver', since the core will do it. - Correct 'IS_ERR(dd->aclk)' to 'IS_ERR(dd->gclk)', when geting the clock 'dd->gclk'. Changes in v2: - Change the "Mono", "Swap" and "Deemphasis" controls to "Mono Switch", "Swap Switch" and "Deemphasis Switch". - Merge "EQ Bass", "EQ Medium" and "EQ Treble" controls into one "EQ" control. - Change the "single-ended" and "differential" to "Single ended", "Differential". - Merge separate left and right controls into one single stereo control. - Add a warning if the user trys to specify an invalid value in function "atmel_classd_codec_probe". - Use the "dev_get_regmap()" in function "atmel_classd_codec_get_remap". - Check for errors when invoking the "clk_prepare_enable". - Remove the "dev_info" in function "atmel_classd_probe". - Add some code to create a sound card in function "atmel_classd_probe". - Remove the DT node "Sound" and the related code. - Delete the device node "Sound" in dts. Songjun Wu (2): ASoC: atmel-classd: add the Audio Class D Amplifier ASoC: atmel-classd: DT binding for Class D audio amplifier driver .../devicetree/bindings/sound/atmel-classd.txt | 52 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 679 ++++++++++++++++++++ sound/soc/atmel/atmel-classd.h | 120 ++++ 5 files changed, 862 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt create mode 100644 sound/soc/atmel/atmel-classd.c create mode 100644 sound/soc/atmel/atmel-classd.h -- 1.7.9.5

2 2

[alsa-devel] Applied "ASoC: atmel-classd: DT binding for Class D audio amplifier driver" to the asoc tree
by Mark Brown 23 Oct '15

23 Oct '15

The patch ASoC: atmel-classd: DT binding for Class D audio amplifier driver has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 391ac3ef50b9ec575c4d5c6f055efe5096ac1957 Mon Sep 17 00:00:00 2001 From: Songjun Wu <songjun.wu(a)atmel.com> Date: Thu, 8 Oct 2015 18:13:32 +0800 Subject: [PATCH] ASoC: atmel-classd: DT binding for Class D audio amplifier driver DT binding documentation for this new ASoC driver. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/atmel-classd.txt | 52 ++++++++++++++++++++++ 1 file changed, 52 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt diff --git a/Documentation/devicetree/bindings/sound/atmel-classd.txt b/Documentation/devicetree/bindings/sound/atmel-classd.txt new file mode 100644 index 0000000..0018451 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel-classd.txt @@ -0,0 +1,52 @@ +* Atmel ClassD driver under ALSA SoC architecture + +Required properties: +- compatible + Should be "atmel,sama5d2-classd". +- reg + Should contain ClassD registers location and length. +- interrupts + Should contain the IRQ line for the ClassD. +- dmas + One DMA specifiers as described in atmel-dma.txt and dma.txt files. +- dma-names + Must be "tx". +- clock-names + Tuple listing input clock names. + Required elements: "pclk", "gclk" and "aclk". +- clocks + Please refer to clock-bindings.txt. + +Optional properties: +- pinctrl-names, pinctrl-0 + Please refer to pinctrl-bindings.txt. +- atmel,model + The user-visible name of this sound complex. + The default value is "CLASSD". +- atmel,pwm-type + PWM modulation type, "single" or "diff". + The default value is "single". +- atmel,non-overlap-time + Set non-overlapping time, the unit is nanosecond(ns). + There are four values, + <5>, <10>, <15>, <20>, the default value is <10>. + Non-overlapping will be disabled if not specified. + +Example: +classd: classd@fc048000 { + compatible = "atmel,sama5d2-classd"; + reg = <0xfc048000 0x100>; + interrupts = <59 IRQ_TYPE_LEVEL_HIGH 7>; + dmas = <&dma0 + (AT91_XDMAC_DT_MEM_IF(0) | AT91_XDMAC_DT_PER_IF(1) + | AT91_XDMAC_DT_PERID(47))>; + dma-names = "tx"; + clocks = <&classd_clk>, <&classd_gclk>, <&audio_pll_pmc>; + clock-names = "pclk", "gclk", "aclk"; + + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_classd_default>; + atmel,model = "classd @ SAMA5D2-Xplained"; + atmel,pwm-type = "diff"; + atmel,non-overlap-time = <10>; +}; -- 2.6.1

1 0

[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: Convert to devm_snd_soc_register_card
by Axel Lin 23 Oct '15

23 Oct '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/intel/boards/skl_rt286.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 488c98e..a73a431 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -240,18 +240,11 @@ static int skylake_audio_probe(struct platform_device *pdev) { skylake_rt286.dev = &pdev->dev; - return snd_soc_register_card(&skylake_rt286); -} - -static int skylake_audio_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&skylake_rt286); - return 0; + return devm_snd_soc_register_card(&pdev->dev, &skylake_rt286); } static struct platform_driver skylake_audio = { .probe = skylake_audio_probe, - .remove = skylake_audio_remove, .driver = { .name = "skl_alc286s_i2s", }, -- 2.1.4

2 1

[alsa-devel] NULL pointer deref when reloading snd_soc_simple_card
by Felipe Balbi 22 Oct '15

22 Oct '15

Hi, I just triggered a NULL point deref with the following commands running on AM437x SK board. This is with v4.3-rc6: modprobe -r snd_soc_simple_card sleep 1 modprobe snd_soc_simple_card sleep 1 details below: [ 228.020921] Unable to handle kernel NULL pointer dereference at virtual address 000000f8 [ 228.029546] pgd = ed4bc000 [ 228.032375] [000000f8] *pgd=00000000 [ 228.036154] Internal error: Oops: 5 [#1] SMP ARM [ 228.040968] Modules linked in: snd_soc_simple_card(+) matrix_keypad matrix_keymap pwm_bl xhci_plat_hcd xhci_hcd usbcore joydev m25p80 spi_nor lis3lv02d_i2c lis3lv02d input_polldev cpufreq_dt thermal_sys hwmon dwc3_omap extcon tps65218_pwrbutton omap_wdt spi_ti_qspi evdev rtc_omap leds_gpio led_class dwc3 udc_core usb_common omapfb cfbfillrect cfbimgblt cfbcopyarea panel_dpi snd_soc_tlv320aic3x snd_soc_davinci_mcasp snd_soc_edma snd_soc_omap snd_soc_core omapdss snd_compress snd_pcm_dmaengine snd_pcm pwm_tiecap snd_timer snd soundcore phy_omap_usb2 autofs4 [last unloaded: snd_soc_simple_card] [ 228.096008] CPU: 0 PID: 710 Comm: modprobe Not tainted 4.3.0-rc6-00001-gada6475ae6e4 #97 [ 228.104436] Hardware name: Generic AM43 (Flattened Device Tree) [ 228.110608] task: ed4b9140 ti: ed52e000 task.ti: ed52e000 [ 228.116370] PC is at dapm_wcache_lookup+0x50/0x7c [snd_soc_core] [ 228.122664] LR is at dapm_wcache_lookup+0x38/0x7c [snd_soc_core] [ 228.128922] pc : [<bf16dd2c>] lr : [<bf16dd14>] psr: a0070013 [ 228.128922] sp : ed52fba8 ip : 00000005 fp : 00000000 [ 228.140883] r10: bf17d238 r9 : bf1f138c r8 : bf1f616c [ 228.146327] r7 : bf1f138c r6 : bf1f616c r5 : ee6f5158 r4 : 000000f4 [ 228.153126] r3 : 00000100 r2 : 00000052 r1 : ed1057c1 r0 : ffffffff [ 228.159925] Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none [ 228.167354] Control: 10c5387d Table: ad4bc059 DAC: 00000051 [ 228.173339] Process modprobe (pid: 710, stack limit = 0xed52e218) [ 228.179689] Stack: (0xed52fba8 to 0xed530000) [ 228.184233] fba0: 00000000 ee7c58c0 bf1f6050 bf16f244 00000000 ed52fc28 [ 228.192765] fbc0: ee577800 c1014a44 00000000 c06430d8 00000001 00000000 ee6c6e90 ee6f50a0 [ 228.201294] fbe0: 00000000 00000000 00000000 c09c3354 60070013 c00943bc ed4b9700 00000004 [ 228.209835] fc00: 00000004 ed4b9140 00000006 bf1f138c bf1ef834 c0091884 c063f37c ed4b9140 [ 228.218365] fc20: 00000001 c118a28c ed4b9140 c00919e8 ee6f506c 60070013 ee6f5070 c063f37c [ 228.226899] fc40: 00000001 00000000 bf16f430 00000003 00000004 ed4b9140 00000006 00000000 [ 228.235417] fc60: bf1ef834 c0091884 c0641060 bf1f138c 00000000 ee7c58c0 00000004 0000001b [ 228.243938] fc80: bf1f138c bf17d238 00000000 bf16f468 00000000 00000012 ee7c58c0 ee7c58c0 [ 228.252463] fca0: ee6c6e90 ee6c6e90 00000004 ee6c6ec0 00000000 bf1ef834 00000000 bf1efcec [ 228.260989] fcc0: ee7c5828 00000000 ee7c5810 ee6f5010 ee7c58c0 ee7c5858 ed496010 bf169ef4 [ 228.269506] fce0: ee6f5180 00000002 00000634 ee6f5010 00000000 00000000 00000000 00000000 [ 228.278029] fd00: ed440e0c bf16cb2c 00000000 ee6f5020 ee6f5180 bf17fd60 00000001 ee6f5028 [ 228.286547] fd20: ee6f5168 60070013 00000000 00000000 eeee3304 00000000 ee178410 ee6f5010 [ 228.295065] fd40: ed105c90 ee6f5010 ee178410 ee178410 ee178400 00000001 12f9f228 bf17968c [ 228.303583] fd60: ee6f5010 fffffdfb 00000001 eeee87d8 ee178410 bf0faa38 00000000 ee178410 [ 228.312108] fd80: ee178410 ee178410 ee178410 bf0fb28c fffffdfb 0000004e ed060e00 c03dff28 [ 228.320626] fda0: ee178410 c11be018 bf0fb28c 00000000 0000004e c03de5dc ee178410 bf0fb28c [ 228.329143] fdc0: ee178444 c098df20 00000000 c03de76c 00000000 bf0fb28c c03de6d8 c03dca40 [ 228.337666] fde0: ee0362a4 ee179f10 bf0fb28c ed462ec0 00000000 c03ddba4 bf0fb080 c09123a0 [ 228.346188] fe00: ed060d40 bf0fb28c c09123a0 ed060d40 bf0fd000 c03defb0 c09123a0 c09123a0 [ 228.354713] fe20: ed060d40 c0009804 60070093 00000000 00000000 00000000 0000000f 00000000 [ 228.363240] fe40: ef7c4464 40000000 0000002e c0091ccc ed060e40 000000d0 000000d0 c0162850 [ 228.371760] fe60: ed52ff58 c0091ccc c090e108 ee0000c0 a0070013 bf0fb300 bf0fb300 c09c34a8 [ 228.380285] fe80: ed060e40 bf0fb300 bf0fb348 00000001 12f9f228 c011b55c ed060e08 bf0fb300 [ 228.388815] fea0: ed52ff58 c09c34a8 ed060e08 c00cc6cc bf0fb30c 00007fff 00000000 c00c9e60 [ 228.397340] fec0: c119bfa4 bf0fb458 c090e990 bf0fb51c f07fa7bc bf0fb30c 00000000 c064c2c0 [ 228.405863] fee0: f07cd000 0002d80c 02e60649 00000000 0000000f 00000000 00000000 00000000 [ 228.414388] ff00: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 228.422911] ff20: 00000000 00000000 00000000 00000000 00000170 00000000 00000003 7f606ddc [ 228.431432] ff40: 0000017b c000f8e4 ed52e000 00000000 7f61a2e8 c00ccf24 f07cd000 0002d80c [ 228.439952] ff60: f07fa0dc f07eebe5 f07ef600 00001690 000019d0 00000000 00000000 00000000 [ 228.448475] ff80: 0000002c 0000002d 00000014 00000018 0000000f 00000000 7f607a28 00000000 [ 228.457000] ffa0: 1c7d6500 c000f740 7f607a28 00000000 00000003 7f606ddc 00000000 7f607d10 [ 228.465523] ffc0: 7f607a28 00000000 1c7d6500 0000017b 00040000 00000000 00000000 7f61a2e8 [ 228.474050] ffe0: bed179b0 bed179a0 7f5fd4bb b6f18852 80070030 00000003 185b0001 05000135 [ 228.482659] [<bf16dd2c>] (dapm_wcache_lookup [snd_soc_core]) from [<bf16f244>] (snd_soc_dapm_add_route+0x74/0x23c [snd_soc_core]) [ 228.494875] [<bf16f244>] (snd_soc_dapm_add_route [snd_soc_core]) from [<bf16f468>] (snd_soc_dapm_add_routes+0x5c/0xbc [snd_soc_core]) [ 228.507428] [<bf16f468>] (snd_soc_dapm_add_routes [snd_soc_core]) from [<bf1efcec>] (aic3x_probe+0x41c/0x57c [snd_soc_tlv320aic3x]) [ 228.519797] [<bf1efcec>] (aic3x_probe [snd_soc_tlv320aic3x]) from [<bf169ef4>] (soc_probe_component+0x208/0x348 [snd_soc_core]) [ 228.531812] [<bf169ef4>] (soc_probe_component [snd_soc_core]) from [<bf16cb2c>] (snd_soc_register_card+0x8d8/0x10f4 [snd_soc_core]) [ 228.544204] [<bf16cb2c>] (snd_soc_register_card [snd_soc_core]) from [<bf17968c>] (devm_snd_soc_register_card+0x2c/0x68 [snd_soc_core]) [ 228.556948] [<bf17968c>] (devm_snd_soc_register_card [snd_soc_core]) from [<bf0faa38>] (asoc_simple_card_probe+0x1f4/0x434 [snd_soc_simple_card]) [ 228.570587] [<bf0faa38>] (asoc_simple_card_probe [snd_soc_simple_card]) from [<c03dff28>] (platform_drv_probe+0x44/0xac) [ 228.581921] [<c03dff28>] (platform_drv_probe) from [<c03de5dc>] (driver_probe_device+0x1f4/0x2f0) [ 228.591170] [<c03de5dc>] (driver_probe_device) from [<c03de76c>] (__driver_attach+0x94/0x98) [ 228.599962] [<c03de76c>] (__driver_attach) from [<c03dca40>] (bus_for_each_dev+0x6c/0xa0) [ 228.608485] [<c03dca40>] (bus_for_each_dev) from [<c03ddba4>] (bus_add_driver+0x18c/0x214) [ 228.617100] [<c03ddba4>] (bus_add_driver) from [<c03defb0>] (driver_register+0x78/0xf8) [ 228.625451] [<c03defb0>] (driver_register) from [<c0009804>] (do_one_initcall+0x80/0x1dc) [ 228.633985] [<c0009804>] (do_one_initcall) from [<c011b55c>] (do_init_module+0x5c/0x1d0) [ 228.642421] [<c011b55c>] (do_init_module) from [<c00cc6cc>] (load_module+0x1a4c/0x20c0) [ 228.650769] [<c00cc6cc>] (load_module) from [<c00ccf24>] (SyS_finit_module+0x7c/0x90) [ 228.658936] [<c00ccf24>] (SyS_finit_module) from [<c000f740>] (ret_fast_syscall+0x0/0x1c) [ 228.667457] Code: e594300c e1550003 e243400c 0a000006 (e5131008) [ 228.673935] ---[ end trace d70ffb1b3028bdb3 ]--- -- balbi

1 0

[alsa-devel] ALSA DMA ping-pong
by Ran Shalit 22 Oct '15

22 Oct '15

Hello, I hope someone can help me understand the concept of ping-pong dma with ALSA. I am using davinci soc: /sound/soc/davinci/davinci_pcm.c But I am not sure if it using ping-pong dma or not. In the header of "davinci_pcm_enqueue_dma" function it states "Not used with ping/pong", but inside I see that it use period parameter in order to set the offset inside the dma buffer which is used. So, does it mean that ping-pong is used or not ? /* * Not used with ping/pong */ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) ..... period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; .... acnt = prtd->params->acnt; edma_set_src(link, src, INCR, W8BIT); edma_set_dest(link, dst, INCR, W8BIT); ...... } Regards, Ran

1 0

[alsa-devel] [PATCH v2 0/3] ASoC: rockchip: i2s: add 8 channels capture support and share lrck
by Sugar Zhang 22 Oct '15

22 Oct '15

support 8 channels capture support. share lrck tx/rx when symmetric_rates enabled. Sugar Zhang (3): ASoC: rockchip: i2s: add 8 channels capture support Documentation: DT bindings: rockchip-i2s: add capture property ASoC: rockchip: i2s: share tx/rx lrck when symmetric_rates enabled .../devicetree/bindings/sound/rockchip-i2s.txt | 2 + sound/soc/rockchip/rockchip_i2s.c | 48 +++++++++++++++++++++- sound/soc/rockchip/rockchip_i2s.h | 16 ++++++++ 3 files changed, 64 insertions(+), 2 deletions(-) -- 2.3.6

3 9

[alsa-devel] [PATCH] ALSA: Remove transfer_ack_{begin, end} callbacks from struct snd_pcm_runtime
by Lars-Peter Clausen 22 Oct '15

22 Oct '15

While there is nothing wrong with the transfer_ack_begin and transfer_ack_end callbacks per-se, the last documented user was part of the alsa-driver 0.5.12a package, which was released 14 years ago and even predates the upstream integration of the ALSA core and has subsequently been superseded by newer alsa-driver releases. This seems to indicate that there is no need for having these callbacks and they are just cruft that can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Documentation/DocBook/writing-an-alsa-driver.tmpl | 19 ++----------------- include/sound/pcm.h | 4 ---- sound/core/pcm_lib.c | 5 ----- 3 files changed, 2 insertions(+), 26 deletions(-) diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl index 84ef6a9..a27ab9f5 100644 --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -2181,10 +2181,6 @@ struct _snd_pcm_runtime { struct snd_pcm_hardware hw; struct snd_pcm_hw_constraints hw_constraints; - /* -- interrupt callbacks -- */ - void (*transfer_ack_begin)(struct snd_pcm_substream *substream); - void (*transfer_ack_end)(struct snd_pcm_substream *substream); - /* -- timer -- */ unsigned int timer_resolution; /* timer resolution */ @@ -2209,9 +2205,8 @@ struct _snd_pcm_runtime { For the operators (callbacks) of each sound driver, most of these records are supposed to be read-only. Only the PCM middle-layer changes / updates them. The exceptions are - the hardware description (hw), interrupt callbacks - (transfer_ack_xxx), DMA buffer information, and the private - data. Besides, if you use the standard buffer allocation + the hardware description (hw) DMA buffer information and the + private data. Besides, if you use the standard buffer allocation method via <function>snd_pcm_lib_malloc_pages()</function>, you don't need to set the DMA buffer information by yourself. </para> @@ -2538,16 +2533,6 @@ struct _snd_pcm_runtime { </para> </section> - <section id="pcm-interface-runtime-intr"> - <title>Interrupt Callbacks</title> - <para> - The field <structfield>transfer_ack_begin</structfield> and - <structfield>transfer_ack_end</structfield> are called at - the beginning and at the end of - <function>snd_pcm_period_elapsed()</function>, respectively. - </para> - </section> - </section> <section id="pcm-interface-operators"> diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2882ddd..3e0ffd2 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -402,10 +402,6 @@ struct snd_pcm_runtime { struct snd_pcm_hardware hw; struct snd_pcm_hw_constraints hw_constraints; - /* -- interrupt callbacks -- */ - void (*transfer_ack_begin)(struct snd_pcm_substream *substream); - void (*transfer_ack_end)(struct snd_pcm_substream *substream); - /* -- timer -- */ unsigned int timer_resolution; /* timer resolution */ int tstamp_type; /* timestamp type */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6dc4277..05a3ca9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1875,9 +1875,6 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) return; runtime = substream->runtime; - if (runtime->transfer_ack_begin) - runtime->transfer_ack_begin(substream); - snd_pcm_stream_lock_irqsave(substream, flags); if (!snd_pcm_running(substream) || snd_pcm_update_hw_ptr0(substream, 1) < 0) @@ -1889,8 +1886,6 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) #endif _end: snd_pcm_stream_unlock_irqrestore(substream, flags); - if (runtime->transfer_ack_end) - runtime->transfer_ack_end(substream); kill_fasync(&runtime->fasync, SIGIO, POLL_IN); } -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: rt5645: Add the HWEQ for the customized speaker output
by Oder Chiou 22 Oct '15

22 Oct '15

The patch adds the HWEQ function for the customized speaker output that is in order to the customer's requirement. The customer can enable the HWEQ in the platform data, and set the parameters of the HWEQ in the ALSA binary control byte-by-byte. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5645.h | 2 + sound/soc/codecs/rt5645.c | 109 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 111 insertions(+) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index a5cf615..efa9f08 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -23,6 +23,8 @@ struct rt5645_platform_data { unsigned int jd_mode; /* Invert JD when jack insert */ bool jd_invert; + /* Speaker HWEQ */ + bool hweq; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4d54999..f26f08b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -42,6 +42,8 @@ #define RT5645_PR_BASE (RT5645_PR_RANGE_BASE + (0 * RT5645_PR_SPACING)) +#define RT5645_HWEQ_NUM 57 + static const struct regmap_range_cfg rt5645_ranges[] = { { .name = "PR", @@ -224,6 +226,11 @@ static const struct reg_default rt5645_reg[] = { { 0xff, 0x6308 }, }; +struct rt5645_eq_param_s { + unsigned short reg; + unsigned short val; +}; + static const char *const rt5645_supply_names[] = { "avdd", "cpvdd", @@ -240,6 +247,7 @@ struct rt5645_priv { struct snd_soc_jack *btn_jack; struct delayed_work jack_detect_work; struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; + struct rt5645_eq_param_s *eq_param; int codec_type; int sysclk; @@ -477,6 +485,80 @@ static const DECLARE_TLV_DB_RANGE(spk_clsd_tlv, 7, 7, TLV_DB_SCALE_ITEM(228, 0, 0) ); +static int rt5645_hweq_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s); + + return 0; +} + +static int rt5645_hweq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + struct rt5645_eq_param_s *eq_param = + (struct rt5645_eq_param_s *)ucontrol->value.bytes.data; + + if (rt5645->pdata.hweq) + memcpy(eq_param, rt5645->eq_param, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s)); + + return 0; +} + +static bool rt5645_validate_hweq(unsigned short reg) +{ + if ((reg >= 0x1a4 && reg <= 0x1cd) | (reg >= 0x1e5 && reg <= 0x1f8) | + (reg == RT5645_EQ_CTRL2)) + return true; + + return false; +} + +static int rt5645_hweq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + struct rt5645_eq_param_s *eq_param = + (struct rt5645_eq_param_s *)ucontrol->value.bytes.data; + int i; + + if (rt5645->pdata.hweq) { + /* The final setting of the table should be RT5645_EQ_CTRL2 */ + for (i = RT5645_HWEQ_NUM - 1; i >= 0; i--) { + if (eq_param[i].reg == 0) + continue; + else if (eq_param[i].reg != RT5645_EQ_CTRL2) + return 0; + else + break; + } + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + if (!rt5645_validate_hweq(eq_param[i].reg) && + eq_param[i].reg != 0) + return 0; + else if (eq_param[i].reg == 0) + break; + } + + memcpy(rt5645->eq_param, eq_param, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s)); + } + + return 0; +} + +#define RT5645_HWEQ(xname) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = rt5645_hweq_info, \ + .get = rt5645_hweq_get,\ + .put = rt5645_hweq_put } + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -541,6 +623,7 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* I2S2 function select */ SOC_SINGLE("I2S2 Func Switch", RT5645_GPIO_CTRL1, RT5645_I2S2_SEL_SFT, 1, 1), + RT5645_HWEQ("Speaker HWEQ"), }; /** @@ -631,6 +714,22 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static int rt5645_enable_hweq(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + if (rt5645_validate_hweq(rt5645->eq_param[i].reg)) + regmap_write(rt5645->regmap, rt5645->eq_param[i].reg, + rt5645->eq_param[i].val); + else + break; + } + + return 0; +} + /** * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters * @codec: SoC audio codec device. @@ -1532,9 +1631,12 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: + if (rt5645->pdata.hweq) + rt5645_enable_hweq(codec); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, @@ -1543,6 +1645,8 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + if (rt5645->pdata.hweq) + snd_soc_write(codec, RT5645_EQ_CTRL2, 0); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, 0); @@ -3060,6 +3164,11 @@ static int rt5645_probe(struct snd_soc_codec *codec) snd_soc_dapm_sync(dapm); } + if (rt5645->pdata.hweq) + rt5645->eq_param = devm_kzalloc(codec->dev, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s), + GFP_KERNEL); + return 0; } -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [patch] ASoC: es8328: harmless underflow in es8328_put_deemph()
by Dan Carpenter 22 Oct '15

22 Oct '15

Valid values for "deemph" are zero and one but we accidentally allow negative values as well. It's harmless but it causes static checker warnings and we may as well clean it up. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 6a09101..969e337 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -129,7 +129,7 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; int ret; if (deemph > 1)

2 2

[alsa-devel] [patch] ASoC: wm8960: harmless underflow in wm8960_put_deemph()
by Dan Carpenter 22 Oct '15

22 Oct '15

Valid values of "deemph" are zero and one, but we accidentally allow negative values as well. It's harmless but we may as well fix it. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e3b7d0c..dc47c2f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -201,7 +201,7 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL;

2 1

[alsa-devel] [PATCH v4] ASoC: soc-compress: add config item for soc-compress to make it compiled only when needed
by Jie Yang 22 Oct '15

22 Oct '15

We don't always need soc-compress in soc, here add a config item SND_SOC_COMPRESS, when nobody select it, the soc-compress will not be compiled. Here also change Kconfig to 'select SND_SOC_COMPRESS' for drivers that needed soc-compress. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/soc-dai.h | 2 +- include/sound/soc.h | 4 +++- sound/soc/Kconfig | 5 ++++- sound/soc/Makefile | 3 ++- sound/soc/intel/Kconfig | 1 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/soc-compress.c | 12 ++++++++++-- sound/soc/soc-core.c | 4 ++-- 8 files changed, 24 insertions(+), 9 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1..238200f 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -214,7 +214,7 @@ struct snd_soc_dai_driver { int (*suspend)(struct snd_soc_dai *dai); int (*resume)(struct snd_soc_dai *dai); /* compress dai */ - bool compress_dai; + int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); /* DAI is also used for the control bus */ bool bus_control; diff --git a/include/sound/soc.h b/include/sound/soc.h index 98e080b..10b7802 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -459,7 +459,9 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); -int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#ifdef CONFIG_SND_SOC_COMPRESS +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#endif struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 7de792b..7ff7d88 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -9,7 +9,6 @@ menuconfig SND_SOC select SND_JACK if INPUT=y || INPUT=SND select REGMAP_I2C if I2C select REGMAP_SPI if SPI_MASTER - select SND_COMPRESS_OFFLOAD ---help--- If you want ASoC support, you should say Y here and also to the @@ -30,6 +29,10 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_COMPRESS + bool + select SND_COMPRESS_OFFLOAD + config SND_SOC_TOPOLOGY bool diff --git a/sound/soc/Makefile b/sound/soc/Makefile index af0a571..8eb06db 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,6 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o ifneq ($(CONFIG_SND_SOC_TOPOLOGY),) snd-soc-core-objs += soc-topology.o diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 664df1f..aaafb13 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -12,6 +12,7 @@ config SND_MFLD_MACHINE config SND_SST_MFLD_PLATFORM tristate + select SND_SOC_COMPRESS config SND_SST_IPC tristate diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 5e9c316..0487cfa 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -512,7 +512,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { }, { .name = "compress-cpu-dai", - .compress_dai = 1, + .compress_new = snd_soc_new_compress, .ops = &sst_compr_dai_ops, .playback = { .stream_name = "Compress Playback", diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 025c38f..12a9820 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -612,8 +612,15 @@ static struct snd_compr_ops soc_compr_dyn_ops = { .get_codec_caps = soc_compr_get_codec_caps }; -/* create a new compress */ -int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +/** + * snd_soc_new_compress - create a new compress. + * + * @rtd: The runtime for which we will create compress + * @num: the device index number (zero based - shared with normal PCMs) + * + * Return: 0 for success, else error. + */ +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; @@ -703,3 +710,4 @@ compr_err: kfree(compr); return ret; } +EXPORT_SYMBOL_GPL(snd_soc_new_compress); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3b471f9..24b0960 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1370,9 +1370,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) soc_dpcm_debugfs_add(rtd); #endif - if (cpu_dai->driver->compress_dai) { + if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = soc_new_compress(rtd, num); + ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Let snd_soc_limit_volume() take a snd_soc_card
by Lars-Peter Clausen 22 Oct '15

22 Oct '15

snd_soc_limit_volume() operates on a card and the CODEC that is passed in is only used to look up the card. Let it directly take the card instead. This makes it possible to use it when no snd_soc_codec is available. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 2 +- sound/soc/omap/rx51.c | 2 +- sound/soc/soc-ops.c | 8 ++++---- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index bd85564..8b06678 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -612,7 +612,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_soc_limit_volume(struct snd_soc_codec *codec, +int snd_soc_limit_volume(struct snd_soc_card *card, const char *name, int max); int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 3bebfb1..9953890 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -297,7 +297,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) dev_err(card->dev, "Failed to add TPA6130A2 controls\n"); return err; } - snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); + snd_soc_limit_volume(card, "TPA6130A2 Headphone Playback Volume", 42); err = omap_mcbsp_st_add_controls(rtd, 2); if (err < 0) { diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 05977ae..ecd38e5 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -588,16 +588,16 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); /** * snd_soc_limit_volume - Set new limit to an existing volume control. * - * @codec: where to look for the control + * @card: where to look for the control * @name: Name of the control * @max: new maximum limit * * Return 0 for success, else error. */ -int snd_soc_limit_volume(struct snd_soc_codec *codec, +int snd_soc_limit_volume(struct snd_soc_card *card, const char *name, int max) { - struct snd_card *card = codec->component.card->snd_card; + struct snd_card *snd_card = card->snd_card; struct snd_kcontrol *kctl; struct soc_mixer_control *mc; int found = 0; @@ -607,7 +607,7 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec, if (unlikely(!name || max <= 0)) return -EINVAL; - list_for_each_entry(kctl, &card->controls, list) { + list_for_each_entry(kctl, &snd_card->controls, list) { if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { found = 1; break; -- 2.1.4

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[alsa-devel] [PATCH v2 0/2] *** wm8962 regmap related fix ***
by Jiada Wang 22 Oct '15

22 Oct '15

This patch set aims to fix issues in wm8962 codec driver related to regmap, currently any attempt to read from ALC Coefficient register will fail when wm8962 is in suspend mode. As ALC2 register is volatile register, it can't be read when cache_only flag is set. Another issue is, if wm8962's regulator is set to 'regulator-always-on' mode, then after wm8962 is resumed from suspend, wm8962 codec is reset, but cache_dirty flag isn't set, this cause difference between actual wm8962 HW and regmap cache. Changeset: -------------- v1 -> v2 * removed comment before regcache_mark_dirty Jiada Wang (2): ASoC: WM8962: mark cache_dirty flag after software reset in pm_resume ASoC: Codec: wm8962: declare ALC Coefficients as 4 separate registers sound/soc/codecs/wm8962.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) -- 2.4.5

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[alsa-devel] Fwd: Re: Gmail thinks most alsa-devel mail is spam
by A. C. Censi 22 Oct '15

22 Oct '15

-- A C Censi accensi @ gmail . com accensi @ montreal . com . br ---------- Forwarded message ---------- From: "A. C. Censi" <accensi(a)gmail.com> Date: Oct 21, 2015 22:50 Subject: Re: [alsa-devel] Gmail thinks most alsa-devel mail is spam To: "Timur Tabi" <timur(a)tabi.org> Cc: I have the same problem and not only with gmail, but my company server does the same check Probably the misclassification is due to: - Keep valid reverse DNS records for the IP address(es) from which you send mail, pointing to your domain. >From the source one email misclassified: Received: by 10.107.154.1 with SMTP id c1csp256822ioe; Wed, 21 Oct 2015 13:55:10 -0700 (PDT) X-Received: by 10.194.216.228 with SMTP id ot4mr13139641wjc.156.1445460910180; Wed, 21 Oct 2015 13:55:10 -0700 (PDT) Return-Path: <alsa-devel-bounces(a)alsa-project.org> Received: from alsa0.perex.cz ([2002:4d30:e0f1:1:5054:ff:fe8b:a8f0]) by mx.google.com with ESMTP id kb1si14026248wjc.16.2015.10.21.13.55.09; Wed, 21 Oct 2015 13:55:10 -0700 (PDT) Received-SPF: neutral (google.com: 2002:4d30:e0f1:1:5054:ff:fe8b:a8f0 is neither permitted nor denied by best guess record for domain of alsa-devel-bounces(a)alsa-project.org) client-ip=2002:4d30:e0f1:1:5054:ff:fe8b:a8f0; Authentication-Results: mx.google.com; spf=neutral (google.com: 2002:4d30:e0f1:1:5054:ff:fe8b:a8f0 is neither permitted nor denied by best guess record for domain of alsa-devel-bounces(a)alsa-project.org) smtp.mailfrom=alsa-devel-bounces(a)alsa-project.org Received: by alsa0.perex.cz (Postfix, from userid 1000) id 782D52667C3; Wed, 21 Oct 2015 22:55:09 +0200 (CEST) An correct email fom linux-fs-devel list: Received: by 10.107.154.1 with SMTP id c1csp276010ioe; Wed, 21 Oct 2015 14:43:58 -0700 (PDT) X-Received: by 10.66.147.74 with SMTP id ti10mr13215193pab.88.1445463838506; Wed, 21 Oct 2015 14:43:58 -0700 (PDT) Return-Path: <linux-fsdevel-owner(a)vger.kernel.org> Received: from vger.kernel.org (vger.kernel.org. [209.132.180.67]) by mx.google.com with ESMTP id qp7si1236731pbc.93.2015.10.21.14.43.44; Wed, 21 Oct 2015 14:43:58 -0700 (PDT) Received-SPF: pass (google.com: best guess record for domain of linux-fsdevel-owner(a)vger.kernel.org designates 209.132.180.67 as permitted sender) client-ip=209.132.180.67; Authentication-Results: mx.google.com; spf=pass (google.com: best guess record for domain of linux-fsdevel-owner(a)vger.kernel.org designates 209.132.180.67 as permitted sender) smtp.mailfrom=linux-fsdevel-owner(a)vger.kernel.org To solve adjust the DNS for the reverse entry and PTR record. ACC On Wed, Oct 21, 2015 at 8:53 PM, Timur Tabi <timur(a)tabi.org> wrote: > On Wed, Oct 21, 2015 at 2:32 PM, Caleb Crome <caleb(a)crome.org> wrote: > > > In gmail go to > > > > settings->Filters and blocked Addresses->Create New Filter > > > > Subject = [alsa-devel] > > > > Then, "Create Filter With This Search" > > > > Then "Apply The Label" = alsa > > and tick the box that says, "Never send it to Spam" > > All that does is disable the spam filter completely for alsa-devel, > which means I'll get more spam. Then I will have to scan my > alsa-devel folder to remove actual spam, which is more difficult than > fishing out legitimate mail from the spam folder. > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > -- A. C. Censi accensi [em] gmail [ponto] com accensi [em] montreal [ponto] com [ponto] br

2 1

[alsa-devel] HDMI codec, way forward?
by Jyri Sarha 22 Oct '15

22 Oct '15

After reading the ELCE Audio mini conf minutes [1] I gather that HDMI audio was not discussed after all. My conclusion from the Lars-Peter's latest mail [2] related to the subject is that the wind is currently blowing slightly in favour of my version of the hdmi codec [3], or at least not in favour of Arnaud's version [4]. So how should we proceed? Are we still waiting for some feedback from DRM/video side? There was not too many clear suggestions to my latest patch series [3], so I do not see too much point in sending yet another version of that series. Arnaud, if I'd try to make a patch series that would implement sti HDMI audio with my HDMI codec, would you be willing try that out? Best regards, Jyri [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/098847.ht… [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/098564.ht… [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-September/097667.… [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/098275.ht…

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[alsa-devel] Gmail thinks most alsa-devel mail is spam
by Timur Tabi 22 Oct '15

22 Oct '15

Hi, I use gmail to subscribe to Linux mailing lists, and most of the mail to alsa-devel ends up in the spam folder. According to gmail, the reason is: "Why is this message in Spam? It is in violation of Google's recommended email sender guidelines." and then it tell me to look here: https://support.google.com/mail/answer/81126?hl=en#authentication Is this something that can be fixed? It's really annoying have to check my spam folder every day to fish out messages from alsa-devel.

3 4

Re: [alsa-devel] ASoC: Intel: Skylake: Add SKL DSP initialization
by Dan Carpenter 21 Oct '15

21 Oct '15

Hello Subhransu S. Prusty, The patch a750ba5f5a56: "ASoC: Intel: Skylake: Add SKL DSP initialization" from Jul 10, 2015, leads to the following static checker warning: sound/soc/intel/skylake/skl-sst.c:270 skl_sst_dsp_init() info: ignoring unreachable code. sound/soc/intel/skylake/skl-sst.c 252 sst->dsp_ops = dsp_ops; 253 sst->fw_ops = skl_fw_ops; 254 255 ret = skl_ipc_init(dev, skl); 256 if (ret) 257 return ret; 258 259 ret = sst->fw_ops.load_fw(sst); 260 if (ret < 0) { 261 dev_err(dev, "Load base fw failed : %d", ret); 262 return ret; Presumably this should be a "goto free_ipc;" here? 263 } 264 265 if (dsp) 266 *dsp = skl; 267 268 return 0; 269 270 skl_ipc_free(&skl->ipc); ^^^^^^^^^^^^^^^^^^^^^^^ Unreachable. 271 return ret; 272 } 273 EXPORT_SYMBOL_GPL(skl_sst_dsp_init); regards, dan carpenter

1 0

Re: [alsa-devel] ASoC: Intel: Add Skylake HDA platform driver
by Dan Carpenter 21 Oct '15

21 Oct '15

Hello Jeeja KP, This is a semi-automatic email about new static checker warnings. The patch a40e693c7f5e: "ASoC: Intel: Add Skylake HDA platform driver" from Jul 9, 2015, leads to the following Smatch complaint: sound/soc/intel/skylake/skl-pcm.c:802 skl_get_position() warn: variable dereferenced before check 'substream->runtime' (see line 792) sound/soc/intel/skylake/skl-pcm.c 791 struct snd_pcm_substream *substream = hstr->substream; 792 struct hdac_ext_bus *ebus = get_bus_ctx(substream); ^^^^^^^^^^^^^^^^^^^^^^ Dereference inside function. 793 unsigned int pos; 794 int delay; 795 796 /* use the position buffer as default */ 797 pos = snd_hdac_stream_get_pos_posbuf(hdac_stream(hstream)); 798 799 if (pos >= hdac_stream(hstream)->bufsize) 800 pos = 0; 801 802 if (substream->runtime) { ^^^^^^^^^^^^^^^^^^ Checked too late. 803 delay = skl_get_delay_from_lpib(ebus, hstream, pos) 804 + codec_delay; regards, dan carpenter

1 0

[alsa-devel] No sound after suspend-to-ram
by George Gugulea 21 Oct '15

21 Oct '15

After a fresh reboot, the sound works ok but if I put the computer to sleep (pm-suspend), after it wakes up there is no sound anymore. I did try 'alsa force-reload' after suspend but still no sound. After 'pm-hibernate' sound is back again. I found an old thread here: https://bugzilla.kernel.org/show_bug.cgi?id=49421 where Michael seems to have the same problem which is still unsolved; maybe I can help with more info. I attached the 'alsa-info.sh' output before and after suspend: 'alsa-info.txt.before' and 'alsa-info.txt.after' The PC is here: http://www.dell.com/us/p/xps-27-2720-aio/pd?ref=PD_OC, but the 2y older version with no touch If I boot into windows (8.1) everything works great, including sound (even after suspend) -- gg

1 0

[alsa-devel] Hot to properly output DSD silence pattern before/after playback?
by Martin Pietryka 21 Oct '15

21 Oct '15

Hello Guys, I have already asked the question last week, but the thing got a bit more complicated. So we need to achieve the following when playing back DSD audio: 1.) Start outputting the DSD silence pattern (0x69) for ~40ms 2.) After ~10ms unset a mute GPIO 3.) Start outputting DSD data after silence is over and after DSD playback is finished a similar procedure: 1.) Start outputting the DSD silence pattern (0x69) for ~40ms 2.) After ~10ms set a mute GPIO 3.) (let silence run out) In addition to that, we also have the need to toggle the mute GPIO from an appropriate place at an appropriate time, the timing does not have to be exact but we need to be sure that the actual hardware already started outputting the silence samples. We have figured out three possibilities where to add the silence: 1.) Inside our application: We are using GStreamer and for various reasons we currently can't do it here without a significant effort. 2.) Inside alsa-lib: The idea here is to inject a snd_pcm_writei() call on the first call of snd_pcm_writei() and inside snd_pcm_drain(). This is really ugly, but doable, I already have a simple proof of concept working. But here I think the GPIO stuff will get really complicated. 3.) Inside the kernel: Modify the soc driver to output the silence pattern before and after playback. So what do you guys think would be the best approach to this problem, are there any other possibilities? I would imagine someone has had to do something similar before, but I couldn't find anything with Google, so if you know something regarding this, please let me know. Thanks, Martin

1 0

[alsa-devel] [PATCH] ASoC: dt-bindings: Drop unneeded properties rockchip i2s/spdif bindings
by Sjoerd Simons 21 Oct '15

21 Oct '15

Neither the rockchip i2s nor the rockchip spdif binding support child devices so #address-cells and #size-cells properties aren't required. Remove these from the bindings. Signed-off-by: Sjoerd Simons <sjoerd.simons(a)collabora.co.uk> --- This patch is on top of the ASoC tree Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 4 ---- Documentation/devicetree/bindings/sound/rockchip-spdif.txt | 4 ---- 2 files changed, 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index 9b82c20..085e0bc 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -12,8 +12,6 @@ Required properties: - reg: physical base address of the controller and length of memory mapped region. - interrupts: should contain the I2S interrupt. -- #address-cells: should be 1. -- #size-cells: should be 0. - dmas: DMA specifiers for tx and rx dma. See the DMA client binding, Documentation/devicetree/bindings/dma/dma.txt - dma-names: should include "tx" and "rx". @@ -28,8 +26,6 @@ i2s@ff890000 { compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s"; reg = <0xff890000 0x10000>; interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; - #address-cells = <1>; - #size-cells = <0>; dmas = <&pdma1 0>, <&pdma1 1>; dma-names = "tx", "rx"; clock-names = "i2s_hclk", "i2s_clk"; diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt index 33dd82c..e64dbde 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt @@ -12,8 +12,6 @@ Required properties: - reg: physical base address of the controller and length of memory mapped region. - interrupts: should contain the SPDIF interrupt. -- #address-cells: should be 1. -- #size-cells: should be 0. - dmas: DMA specifiers for tx dma. See the DMA client binding, Documentation/devicetree/bindings/dma/dma.txt - dma-names: should be "tx" @@ -33,8 +31,6 @@ spdif: spdif@0x1011e000 { compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; reg = <0x1011e000 0x2000>; interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; - #address-cells = <1>; - #size-cells = <0>; dmas = <&dmac1_s 8>; dma-names = "tx"; clock-names = "hclk", "mclk"; -- 2.6.1

2 2

[alsa-devel] [PATCH RFT] ASoC: da7219: Fix da7219->alc_en state when enabling ALC
by Axel Lin 21 Oct '15

21 Oct '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/da7219.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 86c528a..e36a7b7 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -402,7 +402,7 @@ static int da7219_alc_sw_put(struct snd_kcontrol *kcontrol, /* Force ALC offset calibration if enabling ALC */ if ((ucontrol->value.integer.value[0]) && (!da7219->alc_en)) { da7219_alc_calib(codec); - da7219->alc_en = false; + da7219->alc_en = true; } else { da7219->alc_en = false; } -- 2.1.4

3 3

[alsa-devel] [PATCH] ASoC: Intel: sst: only use sst-firmware when DW DMAC is available
by Jie Yang 21 Oct '15

21 Oct '15

Currentlly, we use Synopsys DesignWare DMA Controller for baytrail/haswell/broadwell ADSP firmware loading, but for skylake, we don't use it, compiling sst-firmware.c may introduce error when CONFIG_DW_DMAC_CORE is not enabled: sound/built-in.o: In function `sst_dma_new': (.text+0xd7b38): undefined reference to `dw_dma_probe' sound/built-in.o: In function `sst_dma_free': (.text+0xd7c0a): undefined reference to `dw_dma_remove' Here we only compile sst-firmware when CONFIG_DW_DMAC_CORE is selected, to fix the linking error issue. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/common/Makefile | 3 ++- sound/soc/intel/common/sst-dsp.c | 2 ++ sound/soc/intel/common/sst-dsp.h | 2 ++ 3 files changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index f24154c..55342a1 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -1,4 +1,5 @@ -snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o +snd-soc-sst-dsp-objs := sst-dsp.o +snd-soc-sst-dsp-$(CONFIG_DW_DMAC_CORE) += sst-firmware.o snd-soc-sst-acpi-objs := sst-acpi.o snd-soc-sst-ipc-objs := sst-ipc.o diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index a627236..b5bbdf4 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -420,6 +420,7 @@ void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes) } EXPORT_SYMBOL_GPL(sst_dsp_inbox_read); +#ifdef CONFIG_DW_DMAC_CORE struct sst_dsp *sst_dsp_new(struct device *dev, struct sst_dsp_device *sst_dev, struct sst_pdata *pdata) { @@ -484,6 +485,7 @@ void sst_dsp_free(struct sst_dsp *sst) sst_dma_free(sst->dma); } EXPORT_SYMBOL_GPL(sst_dsp_free); +#endif /* Module information */ MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/intel/common/sst-dsp.h b/sound/soc/intel/common/sst-dsp.h index 1f45f18..0b84c71 100644 --- a/sound/soc/intel/common/sst-dsp.h +++ b/sound/soc/intel/common/sst-dsp.h @@ -216,10 +216,12 @@ struct sst_pdata { void *dsp; }; +#ifdef CONFIG_DW_DMAC_CORE /* Initialization */ struct sst_dsp *sst_dsp_new(struct device *dev, struct sst_dsp_device *sst_dev, struct sst_pdata *pdata); void sst_dsp_free(struct sst_dsp *sst); +#endif /* SHIM Read / Write */ void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value); -- 2.1.4

1 0

Re: [alsa-devel] linux-next: Tree for Oct 20 (sound/soc/intel)
by Randy Dunlap 21 Oct '15

21 Oct '15

On 10/19/15 22:26, Stephen Rothwell wrote: > Hi all, > > Changes since 20151019: > on x86_64: when CONFIG_DW_DMAC is not enabled. sound/built-in.o: In function `sst_dma_new': (.text+0xd7b38): undefined reference to `dw_dma_probe' sound/built-in.o: In function `sst_dma_free': (.text+0xd7c0a): undefined reference to `dw_dma_remove' Full randconfig file is attached. -- ~Randy

2 1

[alsa-devel] [PATCH 00/18] ALSA: Add helper function to add single value constraint
by Lars-Peter Clausen 21 Oct '15

21 Oct '15

Apologies if you got this series twice, forgot to Cc the mailinglist on the first try. This series introduces a new helper function which can be used to constrain a configuration parameter to a single. So far the recommended and most efficient way to do this was to set a min-max constraint with the same value for minimum and maximum. It is not necessarily immediately obvious though that this is the right way to do things and some driver have come up with different ways to achieve the same effect, e.g. by installing a list constraint with a single item. This is less efficient since list constraints are dynamic constraints which needed to be re-evaluated multiple times at runtime while a min-max constraint is a static constraint. The function introduced in this patch series is called snd_pcm_hw_constraint_single() which hopefully is semantically more expressive and makes it clear that this is the function to use when setting a single value constraint. All existing drivers that want to set a single value constraint are update to use the new helper function. - Lars Lars-Peter Clausen (18): ALSA: Add helper function to add single value constraint ALSA: korg1212: Use snd_pcm_hw_constraint_single() ALAA: lx6464es: Use snd_pcm_hw_constraint_single() ALSA: rme32: Use snd_pcm_hw_constraint_single() ALSA: rme96: Use snd_pcm_hw_constraint_single() ALSA: rme9652: Use snd_pcm_hw_constraint_single() ASoC: adav80x: Use snd_pcm_hw_constraint_single() ASoC: twl4030: Use snd_pcm_hw_constraint_single() ASoC: uda134x: Use snd_pcm_hw_constraint_single() ASoC: wl1273: Use snd_pcm_hw_constraint_single() ASoC: n810: Use snd_pcm_hw_constraint_single() ASoC: rx51: Use snd_pcm_hw_constraint_single() ASoC: pcm: Use snd_pcm_hw_constraint_single() ASoC: ux500: Use snd_pcm_hw_constraint_single() ASoC: bytcr_rt5640: Use snd_pcm_hw_constraint_single() ASoC: cht_bsw_max98090: Use snd_pcm_hw_constraint_single() ASoC: cht_bsw_rt5645: Use snd_pcm_hw_constraint_single() ASoC: cht_bsw_rt5672: Use snd_pcm_hw_constraint_single() include/sound/pcm.h | 16 ++++++++++++++++ sound/pci/korg1212/korg1212.c | 8 +++++--- sound/pci/lx6464es/lx6464es.c | 4 ++-- sound/pci/rme32.c | 4 ++-- sound/pci/rme96.c | 8 ++++---- sound/pci/rme9652/hdspm.c | 9 ++++----- sound/soc/codecs/adav80x.c | 4 ++-- sound/soc/codecs/twl4030.c | 13 +++++-------- sound/soc/codecs/uda134x.c | 6 ++---- sound/soc/codecs/wl1273.c | 9 ++++----- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++------------ sound/soc/intel/boards/cht_bsw_max98090_ti.c | 14 ++------------ sound/soc/intel/boards/cht_bsw_rt5645.c | 14 ++------------ sound/soc/intel/boards/cht_bsw_rt5672.c | 14 ++------------ sound/soc/omap/n810.c | 3 +-- sound/soc/omap/rx51.c | 3 +-- sound/soc/soc-pcm.c | 10 ++++------ sound/soc/ux500/ux500_msp_dai.c | 4 ++-- 18 files changed, 62 insertions(+), 95 deletions(-) -- 2.1.4

5 24

[alsa-devel] [PATCH v4] ASoC: rt5645: Add the HW EQ for the customized speaker output of Google Celes
by Oder Chiou 21 Oct '15

21 Oct '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5645.h | 2 + sound/soc/codecs/rt5645.c | 103 +++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 104 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index a5cf615..efa9f08 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -23,6 +23,8 @@ struct rt5645_platform_data { unsigned int jd_mode; /* Invert JD when jack insert */ bool jd_invert; + /* Speaker HWEQ */ + bool hweq; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4d54999..6809d1c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -42,6 +42,8 @@ #define RT5645_PR_BASE (RT5645_PR_RANGE_BASE + (0 * RT5645_PR_SPACING)) +#define RT5645_HWEQ_NUM 57 + static const struct regmap_range_cfg rt5645_ranges[] = { { .name = "PR", @@ -224,6 +226,11 @@ static const struct reg_default rt5645_reg[] = { { 0xff, 0x6308 }, }; +struct rt5645_eq_param_s { + unsigned short reg; + unsigned short val; +}; + static const char *const rt5645_supply_names[] = { "avdd", "cpvdd", @@ -240,6 +247,7 @@ struct rt5645_priv { struct snd_soc_jack *btn_jack; struct delayed_work jack_detect_work; struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; + struct rt5645_eq_param_s *eq_param; int codec_type; int sysclk; @@ -477,6 +485,49 @@ static const DECLARE_TLV_DB_RANGE(spk_clsd_tlv, 7, 7, TLV_DB_SCALE_ITEM(228, 0, 0) ); +static int rt5645_hweq_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s); + + return 0; +} + +static int rt5645_hweq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + char *p = ucontrol->value.bytes.data; + + if (rt5645->pdata.hweq) + memcpy(p, rt5645->eq_param, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s)); + + return 0; +} + +static int rt5645_hweq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + char *p = ucontrol->value.bytes.data; + + if (rt5645->pdata.hweq) + memcpy(rt5645->eq_param, p, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s)); + + return 0; +} + +#define RT5645_HWEQ(xname) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = rt5645_hweq_info, \ + .get = rt5645_hweq_get,\ + .put = rt5645_hweq_put } + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -541,6 +592,7 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* I2S2 function select */ SOC_SINGLE("I2S2 Func Switch", RT5645_GPIO_CTRL1, RT5645_I2S2_SEL_SFT, 1, 1), + RT5645_HWEQ("Speaker HWEQ"), }; /** @@ -631,6 +683,31 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static bool rt5645_validate_hweq(unsigned short reg) +{ + if ((reg >= 0x1a4 && reg <= 0x1cd) | (reg >= 0x1e5 && reg <= 0x1f8) | + (reg == RT5645_EQ_CTRL2)) + return true; + + return false; +} + +static int rt5645_enable_hweq(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + if (rt5645_validate_hweq(rt5645->eq_param[i].reg)) + regmap_write(rt5645->regmap, rt5645->eq_param[i].reg, + rt5645->eq_param[i].val); + else + break; + } + + return 0; +} + /** * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters * @codec: SoC audio codec device. @@ -1532,9 +1609,12 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: + if (rt5645->pdata.hweq) + rt5645_enable_hweq(codec); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, @@ -1543,6 +1623,8 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + if (rt5645->pdata.hweq) + snd_soc_write(codec, RT5645_EQ_CTRL2, 0); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, 0); @@ -3060,6 +3142,11 @@ static int rt5645_probe(struct snd_soc_codec *codec) snd_soc_dapm_sync(dapm); } + if (rt5645->pdata.hweq) + rt5645->eq_param = devm_kzalloc(codec->dev, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s), + GFP_KERNEL); + return 0; } @@ -3209,6 +3296,20 @@ static int strago_quirk_cb(const struct dmi_system_id *id) return 1; } +static struct rt5645_platform_data celes_platform_data = { + .dmic1_data_pin = RT5645_DMIC1_DISABLE, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, + .hweq = true, +}; + +static int celes_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &celes_platform_data; + + return 1; +} + static const struct dmi_system_id dmi_platform_intel_braswell[] = { { .ident = "Intel Strago", @@ -3219,7 +3320,7 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { }, { .ident = "Google Celes", - .callback = strago_quirk_cb, + .callback = celes_quirk_cb, .matches = { DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), }, -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] Headphone Mic and PA
by Takashi Iwai 21 Oct '15

21 Oct '15

Hi David, I'm currently checking the bug report regarding PA, and this looks like an issue with "Headphone Mic" jack control. The reported problem is that PA doesn't react properly when both HP and dock line-out jacks are plugged and line-out is unplugged. The machine in question is Dell E7250, and I see that it has only "Headphone Mic Jack" kctl without "Headphone Jack" kctl. Is this the intentional behavior? I see the code in hda_jack.c: int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { const hda_nid_t *p; int i, err; for (i = 0; i < cfg->num_inputs; i++) { /* If we have headphone mics; make sure they get the right name before grabbed by output pins */ if (cfg->inputs[i].is_headphone_mic) { if (auto_cfg_hp_outs(cfg) == 1) err = add_jack_kctl(codec, auto_cfg_hp_pins(cfg)[0], cfg, "Headphone Mic"); else err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, "Headphone Mic"); } else err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, NULL); So, the driver seems doing what it was written. Meanwhile, the headphone mic comes from the quirk: [ALC293_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x18, 0x01a1913d }, /* use as headphone mic, without its own jack detect */ { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ { } }, .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, and this appears as if it were supposed to be a phantom jack. The alsa-info output is found in the bugzilla https://bugzilla.suse.com/show_bug.cgi?id=948979 thanks, Takashi

3 5

[alsa-devel] [PATCH 1/1] BAT: Add man page
by han.lu＠intel.com 21 Oct '15

21 Oct '15

From: "Lu, Han" <han.lu(a)intel.com> Add more information about BAT utility, including design intent, principles, use mode, hardware connection, options, example and return value. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/bat/Makefile.am b/bat/Makefile.am index 0849d0d..3a76d7f 100644 --- a/bat/Makefile.am +++ b/bat/Makefile.am @@ -1,4 +1,7 @@ bin_PROGRAMS = bat +man_MANS = bat.1 + +EXTRA_DIST = bat.1 bat_SOURCES = \ bat.c \ diff --git a/bat/bat.1 b/bat/bat.1 new file mode 100644 index 0000000..e00fc27 --- /dev/null +++ b/bat/bat.1 @@ -0,0 +1,158 @@ +.TH BAT 1 "20th October 2015" +.SH NAME +bat \- command\-line sound tester for ALSA sound card driver + +.SH SYNOPSIS +\fBbat\fP [\fIflags\fP] + +.SH DESCRIPTION +\fBBAT(Basic Audio Tester)\fP is a simple command\-line utility intended +to help automate audio driver and sound server testing with little human +interaction. BAT can be used to test audio quality, stress test features +and test audio before and after PM state changes. + +BAT's design is relatively simple. BAT plays an audio stream and captures +the same stream in either a digital or analog loop back. It then compares +the captured stream using a FFT to the original to determine if the test +case passes or fails. + +BAT can either run wholly on the target machine being tested (standalone +mode) or can run as a client/server mode where by bat client runs on the +target and runs as a server on a separate tester machine. The client/server +mode still requires some manual interaction for synchronization, but this +is actively being developed for future releases. + +The hardware testing configuration may require the use of an analog cable +connecting target to tester machines or a cable to create an analog +loopback if no loopback mode is not available on the sound hardware that +is being tested. +An analog loopback cable can be used to connect the "line in" to "line out" +jacks to create a loopback. If only headphone and mic jacks (or combo jack) +are available then the following simple circuit can be used to create an +analog loopback :- + +https://source.android.com/devices/audio/loopback.html + +.SH OPTIONS +.TP +\fI\-h, \-\-help\fP +Help: show syntax. +.TP +\fI\-D\fP +Select sound card to be tested by name. +.TP +\fI\-P\fP +Select the playback PCM device. +.TP +\fI\-C\fP +Select the capture PCM device. +.TP +\fI\-f\fP +Sample format +.br +Recognized sample formats are: U8 S16_LE S24_3LE S32_LE +.br +Some of these may not be available on selected hardware +.br +The available format shortcuts are: +.nf +\-f cd (16 bit little endian, 44100, stereo) [\-f S16_LE \-c2 \-r44100] +\-f dat (16 bit little endian, 48000, stereo) [\-f S16_LE \-c2 \-r48000] +.fi +If no format is given S16_LE is used. +.TP +\fI\-c\fP +The number of channels. The default is one channel. +Valid values at the moment are 1 or 2. +.TP +\fI\-r\fP +Sampling rate in Hertz. The default rate is 44100 Hertz. +Valid values depends on hardware support. +.TP +\fI\-n\fP +Duration of generated signal. +The value could be either of the two forms: +.br +1. Decimal integer, means number of frames; +.br +2. Floating point with suffix 's', means number of seconds. +.br +The default is 2 seconds. +.TP +\fI\-k\fP +Sigma k value for analysis. +.br +The analysis function reads data from WAV file, run FFT against the data +to get magnitude of frequency vectors, and then calculates the average +value and standard deviation of frequency vectors. After that, we define +a threshold: +.br +threshold = k * standard_deviation + mean_value +.br +Frequencies with amplitude larger than threshold will be recognized as a +peak, and the frequency with largest peak value will be recognized as a +detected frequency. +.br +BAT then compares the detected frequency to target frequency, to decide +if the detecting passes or fails. +.br +The default value is 3.0. +.TP +\fI\-F\fP +Target frequency for signal generation and analysis, in Hertz. +The default is 997.0 Hertz. +Valid range is (DC_THRESHOLD, 40% * Sampling rate). +.TP +\fI\-p\fP +Total number of periods to play or capture. +.TP +\fI\-\-log=#\fP +Write stderr and stdout output to this log file. +.TP +\fI\-\-file=#\fP +Input WAV file for playback. +.TP +\fI\-\-saveplay=#\fP +Target WAV file to save capture test content. +.TP +\fI\-\-local\fP +Internal loopback mode. +Playback, capture and analysis internal to BAT only. This is intended for +developers to test new BAT features as no audio is routed outside of BAT. + +.SH EXAMPLES + +.TP +\fBbat \-P plughw:0,0 \-C plughw:0,0 \-c 2 \-f S32_LE \-F 250\fR +Generate and play a sine wave of 250 Hertz with 2 channel and S32_LE format, +and then capture and analyze. + +.TP +\fBbat \-P plughw:0,0 \-C plughw:0,0 \-\-file 500Hz.wav\fR +Play the RIFF WAV file "500Hz.wav" which contains 500 Hertz waveform LPCM +data, and then capture and analyze. + +.SH RETURN VALUE +.br +On success, returns 0. +.br +If no peak be detected, returns -1001; +.br +If only DC be detected, returns -1002; +.br +If peak frequency does not match with the target frequency, returns -1003. + +.SH SEE ALSO +\fB +aplay(1) +\fP + +.SH BUGS +Currently only support RIFF WAV format with PCM data. Please report any bugs to +the alsa-devel mailing list. + +.SH AUTHOR +\fBbat\fP is by Liam Girdwood <liam.r.girdwood(a)linux.intel.com>, Bernard Gautier +<bernard.gautier(a)intel.com> and Han Lu <han.lu(a)intel.com>. +This document is by Liam Girdwood <liam.r.girdwood(a)linux.intel.com> and Han Lu +<han.lu(a)intel.com>. -- 1.9.1

2 1

[alsa-devel] [PATCH 0/5 v2] ALSA: firewire-tascam: fix sending the same MIDI bytes in two transactions
by Takashi Sakamoto 21 Oct '15

21 Oct '15

Hi, This patchset is revised version of my previous post: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099123.ht… The main purpose of this patchset is to fix an issue that a transaction can include MIDI bytes in the next message. The last patch solves this issue, and the first 4 patches are for better shape of code including a small fixes. Changes: * add buffer alias * refine local variables Takashi Sakamoto (5): ALSA: firewire-tascam: remove buffer initialization in driver side ALSA: firewire-tascam: change type of valiables according to function prototype ALSA: firewire-tascam: use better name for local variables to describe their intension ALSA: firewire-tascam: fix loop condition with some readable variables ALSA: firewire-tascam: clear extra MIDI byte in an asynchronous transaction sound/firewire/tascam/tascam-transaction.c | 73 +++++++++++++++++------------- 1 file changed, 41 insertions(+), 32 deletions(-) -- 2.1.4

2 7

[alsa-devel] [PATCH] ASoC: wm8904: Correct number of EQ registers
by Charles Keepax 20 Oct '15

20 Oct '15

There are 24 EQ registers not 25, I suspect this bug came about because the registers start at EQ1 not zero. The bug is relatively harmless as the extra register written is an unused one. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/wm8904.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h index 898be3a..6d8f8fb 100644 --- a/include/sound/wm8904.h +++ b/include/sound/wm8904.h @@ -119,7 +119,7 @@ #define WM8904_MIC_REGS 2 #define WM8904_GPIO_REGS 4 #define WM8904_DRC_REGS 4 -#define WM8904_EQ_REGS 25 +#define WM8904_EQ_REGS 24 /** * DRC configurations are specified with a label and a set of register -- 1.7.2.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: ssm2518: Drop .volatile_reg implementation
by Axel Lin 20 Oct '15

20 Oct '15

The implementation of ssm2518_register_volatile always returns false, this behavior is the same as no .volatile_reg callback implementation when cache_type != REGCACHE_NONE. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/ssm2518.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index ddb0203..86b81a60 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -723,17 +723,11 @@ static struct snd_soc_codec_driver ssm2518_codec_driver = { .num_dapm_routes = ARRAY_SIZE(ssm2518_routes), }; -static bool ssm2518_register_volatile(struct device *dev, unsigned int reg) -{ - return false; -} - static const struct regmap_config ssm2518_regmap_config = { .val_bits = 8, .reg_bits = 8, .max_register = SSM2518_REG_DRC_9, - .volatile_reg = ssm2518_register_volatile, .cache_type = REGCACHE_RBTREE, .reg_defaults = ssm2518_reg_defaults, -- 2.1.4

3 5

[alsa-devel] BAT: why MAX_NUMBER_OF_CHANNELS and CHANNEL_MAX separate?
by Caleb Crome 20 Oct '15

20 Oct '15

Hi, I'm starting to bring up BAT on my board to see if it's going to work for me. I'm starting from https://github.com/hanlui/bat, is that the correct place to start from? I'm bringing up a 16 channel board, so need BAT to support more channels (currently it supports only 2). bat/include/common.h has the following defines: #define MAX_NUMBER_OF_CHANNELS 2 #define CHANNEL_MAX 2 #define CHANNEL_MIN 1 Is there any reason we need both MAX_NUMBER_OF_CHANNELS and CHANNEL_MAX? They sure seem like they can be merged to me. Once I get the program up and running, I'll add the ramp test, and start thinking about the other tests we talked about. Thanks, -Caleb

3 5

[alsa-devel] Getting ICS43432 MEMS mic working with TMS320DM365 processor
by Ankhit Vivekananda 20 Oct '15

20 Oct '15

Hello, I have configured the TM320DM365 McBSP with clock and frame synchronization signals. Also see the ICS43432 MIC serial data on the oscilloscope, but do not get the right output in my application. I have mapped the DaVinci McBSP with the CQ0093VC audio codec driver. Is this mapping correct or is it necessary to map the McBSP with ICS-43432 codec driver? Since, I have not applied the patch, I have used CQ0093VC codec driver. Please do let me know if the ICS-43432 patch really necessary for the MIC to work. Thank you, Ankhit

2 1

[alsa-devel] [PATCH] ASoC: rt5640: Fill up the IN3's support
by Oder Chiou 20 Oct '15

20 Oct '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5640.txt | 9 ++++++--- include/sound/rt5640.h | 3 ++- sound/soc/codecs/rt5640.c | 22 +++++++++++++++++++++- 3 files changed, 29 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index bac4d9a..9e62f6e 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -14,7 +14,8 @@ Optional properties: - realtek,in1-differential - realtek,in2-differential - Boolean. Indicate MIC1/2 input are differential, rather than single-ended. +- realtek,in3-differential + Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended. - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. @@ -24,9 +25,11 @@ Pins on the device (for linking into audio routes) for RT5639/RT5640: * DMIC2 * MICBIAS1 * IN1P - * IN1R + * IN1N * IN2P - * IN2R + * IN2N + * IN3P + * IN3N * HPOL * HPOR * LOUTL diff --git a/include/sound/rt5640.h b/include/sound/rt5640.h index 59d26dd..e3c84b9 100644 --- a/include/sound/rt5640.h +++ b/include/sound/rt5640.h @@ -12,9 +12,10 @@ #define __LINUX_SND_RT5640_H struct rt5640_platform_data { - /* IN1 & IN2 can optionally be differential */ + /* IN1 & IN2 & IN3 can optionally be differential */ bool in1_diff; bool in2_diff; + bool in3_diff; bool dmic_en; bool dmic1_data_pin; /* 0 = IN1P; 1 = GPIO3 */ diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e1ceeb8..f2beb1a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -405,11 +405,14 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), - /* IN1/IN2 Control */ + /* IN1/IN2/IN3 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, RT5640_BST_SFT1, 8, 0, bst_tlv), SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4, RT5640_BST_SFT2, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN3 Boost", RT5640_IN1_IN2, + RT5640_BST_SFT2, 8, 0, bst_tlv), + /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5640_INL_INR_VOL, RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT, @@ -598,6 +601,8 @@ static const struct snd_kcontrol_new rt5640_rec_l_mix[] = { RT5640_M_HP_L_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("INL Switch", RT5640_REC_L2_MIXER, RT5640_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_L2_MIXER, + RT5640_M_BST2_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_L2_MIXER, RT5640_M_BST4_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_L2_MIXER, @@ -611,6 +616,8 @@ static const struct snd_kcontrol_new rt5640_rec_r_mix[] = { RT5640_M_HP_R_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("INR Switch", RT5640_REC_R2_MIXER, RT5640_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_R2_MIXER, + RT5640_M_BST2_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_R2_MIXER, RT5640_M_BST4_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_R2_MIXER, @@ -1065,6 +1072,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1N"), SND_SOC_DAPM_INPUT("IN2P"), SND_SOC_DAPM_INPUT("IN2N"), + SND_SOC_DAPM_INPUT("IN3P"), + SND_SOC_DAPM_INPUT("IN3N"), SND_SOC_DAPM_PGA("DMIC L1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC R1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC L2", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1081,6 +1090,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { RT5640_PWR_BST1_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("BST2", RT5640_PWR_ANLG2, RT5640_PWR_BST4_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST3", RT5640_PWR_ANLG2, + RT5640_PWR_BST2_BIT, 0, NULL, 0), /* Input Volume */ SND_SOC_DAPM_PGA("INL VOL", RT5640_PWR_VOL, RT5640_PWR_IN_L_BIT, 0, NULL, 0), @@ -1310,6 +1321,7 @@ static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, + {"IN3P", NULL, "LDO2"}, {"DMIC L1", NULL, "DMIC1"}, {"DMIC R1", NULL, "DMIC1"}, @@ -1320,18 +1332,22 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"BST1", NULL, "IN1N"}, {"BST2", NULL, "IN2P"}, {"BST2", NULL, "IN2N"}, + {"BST3", NULL, "IN3P"}, + {"BST3", NULL, "IN3N"}, {"INL VOL", NULL, "IN2P"}, {"INR VOL", NULL, "IN2N"}, {"RECMIXL", "HPOL Switch", "HPOL"}, {"RECMIXL", "INL Switch", "INL VOL"}, + {"RECMIXL", "BST3 Switch", "BST3"}, {"RECMIXL", "BST2 Switch", "BST2"}, {"RECMIXL", "BST1 Switch", "BST1"}, {"RECMIXL", "OUT MIXL Switch", "OUT MIXL"}, {"RECMIXR", "HPOR Switch", "HPOR"}, {"RECMIXR", "INR Switch", "INR VOL"}, + {"RECMIXR", "BST3 Switch", "BST3"}, {"RECMIXR", "BST2 Switch", "BST2"}, {"RECMIXR", "BST1 Switch", "BST1"}, {"RECMIXR", "OUT MIXR Switch", "OUT MIXR"}, @@ -2260,6 +2276,10 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); + if (rt5640->pdata.in3_diff) + regmap_update_bits(rt5640->regmap, RT5640_IN1_IN2, + RT5640_IN_DF2, RT5640_IN_DF2); + rt5640->hp_mute = 1; return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix deadlock at error in building PCM
by Takashi Iwai 20 Oct '15

20 Oct '15

The HDA codec driver issues snd_hda_codec_reset() at the error path of PCM build. This was needed in the earlier code base, but the recent rewrite to use the standard bus binding made this a deadlock: modprobe D 0000000000000005 0 720 716 0x00000080 Call Trace: [<ffffffff816a5dbe>] schedule+0x3e/0x90 [<ffffffff816a61a5>] schedule_preempt_disabled+0x15/0x20 [<ffffffff816a7ae5>] __mutex_lock_slowpath+0xb5/0x120 [<ffffffff816a7b6b>] mutex_lock+0x1b/0x30 [<ffffffff8148656b>] device_release_driver+0x1b/0x30 [<ffffffff81485c15>] bus_remove_device+0x105/0x180 [<ffffffff814822b9>] device_del+0x139/0x260 [<ffffffffa05e0ec5>] snd_hdac_device_unregister+0x25/0x30 [snd_hda_core] [<ffffffffa074fa6a>] snd_hda_codec_reset+0x2a/0x70 [snd_hda_codec] [<ffffffffa075007b>] snd_hda_codec_build_pcms+0x18b/0x1b0 [snd_hda_codec] [<ffffffffa074a44e>] hda_codec_driver_probe+0xbe/0x140 [snd_hda_codec] [<ffffffff81486ac4>] driver_probe_device+0x1f4/0x460 [<ffffffff81486dc0>] __driver_attach+0x90/0xa0 [<ffffffff81484844>] bus_for_each_dev+0x64/0xa0 [<ffffffff814862de>] driver_attach+0x1e/0x20 [<ffffffff81485e7b>] bus_add_driver+0x1eb/0x280 [<ffffffff81487680>] driver_register+0x60/0xe0 [<ffffffffa074a0da>] __hda_codec_driver_register+0x5a/0x60 [snd_hda_codec] [<ffffffffa070a01e>] realtek_driver_init+0x1e/0x1000 [snd_hda_codec_realtek] [<ffffffff810002f3>] do_one_initcall+0xb3/0x200 [<ffffffff816a1fc5>] do_init_module+0x60/0x1f8 [<ffffffff810ee5c3>] load_module+0x1653/0x1bd0 [<ffffffff810eed48>] SYSC_finit_module+0x98/0xc0 [<ffffffff810eed8e>] SyS_finit_module+0xe/0x10 [<ffffffff816aa032>] entry_SYSCALL_64_fastpath+0x16/0x75 The simple fix is just to remove this call, since we don't need to think about unbinding at there any longer. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=948758 Cc: <stable(a)vger.kernel.org> # v4.1+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 37f43a1b34ef..a249d5486889 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3367,10 +3367,8 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) int dev, err; err = snd_hda_codec_parse_pcms(codec); - if (err < 0) { - snd_hda_codec_reset(codec); + if (err < 0) return err; - } /* attach a new PCM streams */ list_for_each_entry(cpcm, &codec->pcm_list_head, list) { -- 2.6.1

1 0

[alsa-devel] [PATCH V4 0/4] Some fixes against previous comments
by han.lu＠intel.com 20 Oct '15

20 Oct '15

From: "Lu, Han" <han.lu(a)intel.com> Some fixes and adjustments against previous comments of Takashi Lu, Han (4): BAT: Remove redundant message strings BAT: Use colon instead of comma for separation BAT: Change comments and interface of usage() BAT: Use dynamic temp file bat/bat.c | 106 +++++++++++++++++++++++++++++++---------------------------- bat/common.h | 2 +- 2 files changed, 57 insertions(+), 51 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH 0/6] HD-audio modalias support patchset
by Takashi Iwai 20 Oct '15

20 Oct '15

Hi, this is a patchset to add the support of automatic modalias creation for HD-audio codec drivers like other standard drivers. Currently, the HDA codec driver has put module alias manually due to ugly historical reasons. This patch finally moves to the saner way. Also it adds the udev modalias support so that it can be autoloaded in udev. The patchset was initiated by Intel people for SKL ASoC drivers. Now I brushed up and made it working for legacy drivers. The current patchset is found in topic/hda-modalias branch of sound git tree, too. thanks, Takashi === Subhransu S. Prusty (2): ALSA: hda - Add hduadio support to DEVTABLE ALSA: hda - Add hdaudio bus modalias support Takashi Iwai (3): ALSA: hda - Add api_version to hda_device_id struct ALSA: hda - Add a common helper to give the codec modalias string ALSA: hda - convert to hda_device_id Thierry Reding (1): ALSA: hda - Advertise MODALIAS in uevent include/linux/mod_devicetable.h | 8 ++ include/sound/hdaudio.h | 12 +-- scripts/mod/devicetable-offsets.c | 5 + scripts/mod/file2alias.c | 17 +++ sound/hda/hda_bus_type.c | 13 +++ sound/hda/hdac_device.c | 15 +++ sound/hda/hdac_sysfs.c | 8 ++ sound/pci/hda/hda_bind.c | 32 +++--- sound/pci/hda/hda_codec.h | 19 ++-- sound/pci/hda/hda_generic.c | 7 +- sound/pci/hda/patch_analog.c | 37 ++++--- sound/pci/hda/patch_ca0110.c | 15 ++- sound/pci/hda/patch_ca0132.c | 9 +- sound/pci/hda/patch_cirrus.c | 21 ++-- sound/pci/hda/patch_cmedia.c | 15 ++- sound/pci/hda/patch_conexant.c | 118 ++++++--------------- sound/pci/hda/patch_hdmi.c | 201 +++++++++++++---------------------- sound/pci/hda/patch_realtek.c | 122 ++++++++++------------ sound/pci/hda/patch_si3054.c | 39 +++---- sound/pci/hda/patch_sigmatel.c | 214 +++++++++++++++++++------------------- sound/pci/hda/patch_via.c | 153 ++++++++++----------------- 21 files changed, 475 insertions(+), 605 deletions(-) -- 2.6.1

2 9

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Convert to devm_snd_soc_register_card
by Axel Lin 20 Oct '15

20 Oct '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/intel/boards/skl_rt286.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 488c98e..3b6f48c 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -240,18 +240,11 @@ static int skylake_audio_probe(struct platform_device *pdev) { skylake_rt286.dev = &pdev->dev; - return snd_soc_register_card(&skylake_rt286); -} - -static int skylake_audio_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&skylake_rt286); - return 0; + return devm_snd_soc_register_card(&skylake_rt286); } static struct platform_driver skylake_audio = { .probe = skylake_audio_probe, - .remove = skylake_audio_remove, .driver = { .name = "skl_alc286s_i2s", }, -- 2.1.4

1 0

[alsa-devel] Implementing sub-stream multiplexing in DAI
by Ughreja, Rakesh A 19 Oct '15

19 Oct '15

Hi Mark, In order to implement sub-stream multiplexing as a part of single BE DAI I have following observations. 1. All BE DAI ops gets called only once even if two different FE DAIs are connected to a BE DAI. 2. BE DAI ops are always called with sub-stream 0. are these expected ? What is the best way to allow calling of BE DAI ops with different sub-stream ID so that I can do link configuration based on sub-stream parameters ? can you give some pointers ? Regards, Rakesh

2 1

[alsa-devel] [PATCH] ASoC: rockchip: spdif: Convert to use devm_snd_dmaengine_pcm_register
by Axel Lin 19 Oct '15

19 Oct '15

Use resource managed API then we can remove snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() calls in .probe error path and .remove. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/rockchip/rockchip_spdif.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 9d5c470..a38a302 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -355,16 +355,14 @@ static int rk_spdif_probe(struct platform_device *pdev) goto err_pm_runtime; } - ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - goto err_pcm_register; + goto err_pm_runtime; } return 0; -err_pcm_register: - snd_dmaengine_pcm_unregister(&pdev->dev); err_pm_runtime: pm_runtime_disable(&pdev->dev); @@ -381,8 +379,6 @@ static int rk_spdif_remove(struct platform_device *pdev) clk_disable_unprepare(spdif->mclk); clk_disable_unprepare(spdif->hclk); - snd_dmaengine_pcm_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- 2.1.4

2 1

[alsa-devel] [PATCH v3] ASoC: Intel: Skylake - Add Skylake RT286 I2S machine driver
by Vinod Koul 19 Oct '15

19 Oct '15

From: Omair M Abdullah <omair.m.abdullah(a)intel.com> Add the SKL I2S machine driver using Realtek ALC286S codec in I2S mode. Signed-off-by: Omair M Abdullah <omair.m.abdullah(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- v3: fixed sparse warns v2: added depends on X86 and ACPI based on LKP reports sound/soc/intel/Kconfig | 13 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_rt286.c | 266 +++++++++++++++++++++++++++++++++++++ 3 files changed, 281 insertions(+) create mode 100644 sound/soc/intel/boards/skl_rt286.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 664df1f44530..6d67357f2d74 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -140,3 +140,16 @@ config SND_SOC_INTEL_SKYLAKE select SND_HDA_EXT_CORE select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST + +config SND_SOC_INTEL_SKL_RT286_MACH + tristate "ASoC Audio driver for SKL with RT286 I2S mode" + depends on X86 && ACPI + select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT286 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Skylake platforms + with RT286 I2S audio codec. + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index cb94895c9edb..371c4565cad8 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o +snd-soc-skl_rt286-objs := skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -15,3 +16,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o +obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c new file mode 100644 index 000000000000..488c98ee1ded --- /dev/null +++ b/sound/soc/intel/boards/skl_rt286.c @@ -0,0 +1,266 @@ +/* + * Intel Skylake I2S Machine Driver + * + * Copyright (C) 2014-2015, Intel Corporation. All rights reserved. + * + * Modified from: + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * 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 published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include "../../codecs/rt286.h" + +static struct snd_soc_jack skylake_headset; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin skylake_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new skylake_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget skylake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route skylake_rt286_map[] = { + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphone Jack", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC2"}, + {"DMIC AIF", NULL, "SoC DMIC"}, + + /* CODEC BE connections */ + { "AIF1 Playback", NULL, "ssp0 Tx"}, + { "ssp0 Tx", NULL, "codec0_out"}, + { "ssp0 Tx", NULL, "codec1_out"}, + + { "codec0_in", NULL, "ssp0 Rx" }, + { "codec1_in", NULL, "ssp0 Rx" }, + { "ssp0 Rx", NULL, "AIF1 Capture" }, + + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "Capture" }, + + { "hif1", NULL, "iDisp Tx"}, + { "iDisp Tx", NULL, "iDisp_out"}, + +}; + +static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &skylake_headset, + skylake_headset_pins, ARRAY_SIZE(skylake_headset_pins)); + + if (ret) + return ret; + + rt286_mic_detect(codec, &skylake_headset); + + return 0; +} + + +static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The output is 48KHz, stereo, 16bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static int skylake_rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "set codec sysclk failed: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops skylake_rt286_ops = { + .hw_params = skylake_rt286_hw_params, +}; + +/* skylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link skylake_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "Skl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_playback = 1, + }, + { + .name = "Skl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_capture = 1, + }, + { + .name = "Skl Audio Reference cap", + .stream_name = "refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .ignore_suspend = 1, + .nonatomic = 1, + .dynamic = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .be_id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .init = skylake_rt286_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = skylake_ssp0_fixup, + .ops = &skylake_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .be_id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, +}; + +/* skylake audio machine driver for SPT + RT286S */ +static struct snd_soc_card skylake_rt286 = { + .name = "skylake-rt286", + .owner = THIS_MODULE, + .dai_link = skylake_rt286_dais, + .num_links = ARRAY_SIZE(skylake_rt286_dais), + .controls = skylake_controls, + .num_controls = ARRAY_SIZE(skylake_controls), + .dapm_widgets = skylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(skylake_widgets), + .dapm_routes = skylake_rt286_map, + .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map), + .fully_routed = true, +}; + +static int skylake_audio_probe(struct platform_device *pdev) +{ + skylake_rt286.dev = &pdev->dev; + + return snd_soc_register_card(&skylake_rt286); +} + +static int skylake_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&skylake_rt286); + return 0; +} + +static struct platform_driver skylake_audio = { + .probe = skylake_audio_probe, + .remove = skylake_audio_remove, + .driver = { + .name = "skl_alc286s_i2s", + }, +}; + +module_platform_driver(skylake_audio) + +/* Module information */ +MODULE_AUTHOR("Omair Mohammed Abdullah <omair.m.abdullah(a)intel.com>"); +MODULE_DESCRIPTION("Intel SST Audio for Skylake"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:skl_alc286s_i2s"); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: ad193x-spi: Add adau1328 to ad193x_spi_id table
by Axel Lin 19 Oct '15

19 Oct '15

This driver also supports adau1328, thus add adau1328 to ad193x_spi_id. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/ad193x-spi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c index b55055a..431f95d 100644 --- a/sound/soc/codecs/ad193x-spi.c +++ b/sound/soc/codecs/ad193x-spi.c @@ -41,6 +41,7 @@ static const struct spi_device_id ad193x_spi_id[] = { { "ad1934", AD1934 }, { "ad1938", AD193X }, { "ad1939", AD193X }, + { "adau1328", AD193X }, { } }; MODULE_DEVICE_TABLE(spi, ad193x_spi_id); -- 2.1.4

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[alsa-devel] [PATCH 0/7] ALSA: USB-audio: Support Zoom R16 playback
by Ricard Wanderlof 19 Oct '15

19 Oct '15

The Zoom R16 has a nonstandard playback format where each isochronous packet contains a length descriptor in the first four bytes. Curiously, capture data does not contain this and requires no quirk. The quirk involves adding the extra length descriptor whenever outgoing isochronous packets are generated, both in pcm.c (outgoing audio) and endpoint.c (silent data). The calculation for the max packet size in endpoint.c is also affected. This patch series refactors some code in pcm.c and endpoint.c before the actual quirk implementation. A commit then cleans up the entry in the quirks table. Tested with a Zoom R16, both separately using arecord and aplay (8 channel capture and 2 channel playback, respectively), as well as capture and playback together together using the Ardour digitial audio workstation application. The R24 reportedly is compatible with the R16, but I don't have access to one so I can't test it, and consequently the patch series only mentions the R16. Also tested using an Edirol UA-5 in both class compliant (16-bit) and "advanced" (24 bit, forces the use of quirks) modes in order to do some form of regression testing. Ricard Wanderlof (7): Break out copying to urb from prepare_playback_urb() Also move out hwptr_done wrap from prepare_playback_urb() Break out creation of silent urbs from prepare_outbound_urb() Add offset parameter to copy_to_urb() Add quirk for Zoom R16 playback Adjust max packet size calculation for tx_length_quirk Remove mixer entry from Zoom R16/24 quirk sound/usb/card.h | 1 + sound/usb/endpoint.c | 73 +++++++++++++++++++++++++++++++++-------------- sound/usb/pcm.c | 74 +++++++++++++++++++++++++++++++++++++----------- sound/usb/quirks-table.h | 14 +++------ sound/usb/quirks.c | 3 ++ sound/usb/stream.c | 1 + sound/usb/usbaudio.h | 1 + 7 files changed, 119 insertions(+), 48 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH 0/3] Add support for Cirrus Logic CS47L24 and WM1831 codecs
by Richard Fitzgerald 19 Oct '15

19 Oct '15

mfd patch based on Lee Jones's for-next gpio patch based on Linus Walleij's for-next ASoC patch based on Mark Brown's for-next Lee/Mark - can one of you take all three patches (with maintainer acks) so that the patch set is kept together? Richard Fitzgerald (3): mfd: arizona: Support Cirrus Logic CS47L24 and WM1831 gpio: arizona: Support Cirrus Logic CS47L24 and WM1831 ASoC: cs47l24: Add driver for Cirrus Logic CS47L24 and WM1831 codecs Documentation/devicetree/bindings/mfd/arizona.txt | 4 +- drivers/gpio/gpio-arizona.c | 4 + drivers/mfd/Kconfig | 19 +- drivers/mfd/Makefile | 3 + drivers/mfd/arizona-core.c | 72 +- drivers/mfd/arizona-irq.c | 40 +- drivers/mfd/arizona-spi.c | 7 + drivers/mfd/arizona.h | 4 + drivers/mfd/cs47l24-tables.c | 1629 +++++++++++++++++++++ include/linux/mfd/arizona/core.h | 3 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs47l24.c | 1147 +++++++++++++++ sound/soc/codecs/cs47l24.h | 23 + 14 files changed, 2940 insertions(+), 25 deletions(-) create mode 100644 drivers/mfd/cs47l24-tables.c create mode 100644 sound/soc/codecs/cs47l24.c create mode 100644 sound/soc/codecs/cs47l24.h -- 1.9.1

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[alsa-devel] [patch] ALSA: firewire-tascam: off by one in identify_model()
by Dan Carpenter 19 Oct '15

19 Oct '15

In the original code, we potentially put a NUL character in model[8] and it caused a static checker warning. We can put the NUL in model[7] instead. Fixes: 53b3ffee7885 ('ALSA: firewire-tascam: change device probing processing') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index c6747a4..6ef2bcc 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -51,7 +51,7 @@ static int identify_model(struct snd_tscm *tscm) } /* Pick up model name from certain addresses. */ - for (i = 0; i < 8; i++) { + for (i = 0; i < 7; i++) { c = config_rom[28 + i / 4] >> (24 - 8 * (i % 4)); if (c == '\0') break;

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[alsa-devel] [PATCH 0/3] ALSA: oxfw: add support for TASCAM FireOne
by Takashi Sakamoto 19 Oct '15

19 Oct '15

Hi, This parchset adds support for TASCAM FireOne to ALSA OXFW driver. PCM and MIDI functionalities become available including physical controls. The first and second patches are preparation for it. This model supports some extra commands. So as the other ALSA drivers for sound and music units on IEEE 1394 bus, they should be implemented in userspace applications. For your information, I note them: (Jog/Shuttle display mode) ./firewire-request /dev/fw1 fcp 0x01ff0000022e46493110ffff response: 000: 0c ff 00 00 02 2e 46 49 31 10 06 * The last byte is: * 0x00: always off * 0x01: always on * 0x02: breathe * 0x03: metronome * 0x04: MIDI clock rotate * 0x05: MIDI clock flash * 0x06: jog slow rotate * 0x07: Jog track (Control surface mode) ./firewire-request /dev/fw1 fcp 0x01ff0000022e46493111ffff response: 000: 0c ff 00 00 02 2e 46 49 31 11 00 * The last bytes is: * 0x00: FireOne native * 0x01: Mackie Human User Interface emulation (Input monitor monaural) ./firewire-request /dev/fw1 fcp 0x01ff0000022e46493112ffff response: 000: 0c ff 00 00 02 2e 46 49 31 12 00 * The last byte is: * 0x00: off * 0x01: on (Firmware version, read-only) ./firewire-request /dev/fw1 fcp 0x01ff0000022e46493113ffff response: 000: 0c ff 00 00 02 2e 46 49 31 13 02 * The last byte is: * 0x02: version 2 I can see additional commands from Windows driver. They seem to be used to retrieve the name of data channels. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff0000000040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 00 00 00 12 0c ......HST`...... response: 010: 41 75 64 69 6f 20 4f 75 74 20 4c 00 Audio Out L. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff0000010040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 00 00 01 12 0c ......HST`...... response: 010: 41 75 64 69 6f 20 4f 75 74 20 52 00 Audio Out R. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff0000020040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 00 00 02 12 09 ......HST`...... response: 010: 4d 69 64 69 20 4f 75 74 20 Midi Out ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff0001000040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 00 01 00 12 0b ......HST`...... response: 010: 41 75 64 69 6f 20 49 6e 20 41 00 Audio In A. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff0001010040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 00 01 01 12 0b ......HST`...... response: 010: 41 75 64 69 6f 20 49 6e 20 42 00 Audio In B. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff0001020040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 00 01 02 12 08 ......HST`...... response: 010: 4d 69 64 69 20 49 6e 20 Midi In ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535470ff0000000040 response: 000: 0c ff 00 00 02 2e 48 53 54 70 ff 00 00 00 12 06 ......HSTp...... response: 010: 41 75 64 69 6f 20 Audio ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535470ff0000010040 response: 000: 0c ff 00 00 02 2e 48 53 54 70 ff 00 00 01 12 00 ......HSTp...... ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535470ff0000020040 response: 000: 0a ff 00 00 02 2e 48 53 54 70 ff 00 00 02 00 40 ......HSTp.....@ ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535470ff0001000040 response: 000: 0c ff 00 00 02 2e 48 53 54 70 ff 00 01 00 12 06 ......HSTp...... response: 010: 41 75 64 69 6f 20 Audio ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535470ff0001010040 response: 000: 0c ff 00 00 02 2e 48 53 54 70 ff 00 01 01 12 00 ......HSTp...... ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535470ff0001020040 response: 000: 0a ff 00 00 02 2e 48 53 54 70 ff 00 01 02 00 40 ......HSTp.....@ ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff8100000040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 81 00 00 12 0c ......HST`...... response: 010: 4d 69 64 69 20 49 6e 20 4f 6e 65 00 Midi In One. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff8101000040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 81 01 00 12 0d ......HST`...... response: 010: 4d 69 64 69 20 4f 75 74 20 4f 6e 65 00 Midi Out One. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff8200000040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 82 00 00 12 0c ......HST`...... response: 010: 4d 69 64 69 20 49 6e 20 54 77 6f 00 Midi In Two. ./firewire-request /dev/fw1 fcp 0x01ff0000022e48535460ff8201000040 response: 000: 0c ff 00 00 02 2e 48 53 54 60 ff 82 01 00 12 0d ......HST`...... response: 010: 4d 69 64 69 20 4f 75 74 20 54 77 6f 00 Midi Out Two. Takashi Sakamoto (3): ALSA: oxfw: calculating MIDI ports in stream discover ALSA: oxfw: support more MIDI ports ALSA: oxfw: add an entry for TASCAM FireOne sound/firewire/oxfw/oxfw-midi.c | 24 +++--------------------- sound/firewire/oxfw/oxfw-stream.c | 37 +++++++++++++++++++++++++++++++++++-- sound/firewire/oxfw/oxfw.c | 17 +++++++++++++++++ 3 files changed, 55 insertions(+), 23 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH v3] drm/i915: Improve kernel-doc for i915_audio_component struct
by David Henningsson 19 Oct '15

19 Oct '15

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Now rebased against drm-intel git master. include/drm/i915_component.h | 69 ++++++++++++++++++++++++++++++++------------ 1 file changed, 51 insertions(+), 18 deletions(-) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index 30d89e0..fab1385 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -31,47 +31,80 @@ #define MAX_PORTS 5 /** - * struct i915_audio_component_ops - callbacks defined in gfx driver - * @owner: the module owner - * @get_power: get the POWER_DOMAIN_AUDIO power well - * @put_power: put the POWER_DOMAIN_AUDIO power well - * @codec_wake_override: Enable/Disable generating the codec wake signal - * @get_cdclk_freq: get the Core Display Clock in KHz - * @sync_audio_rate: set n/cts based on the sample rate + * struct i915_audio_component_ops - Ops implemented by i915 driver, called by hda driver */ struct i915_audio_component_ops { + /** + * @owner: i915 module + */ struct module *owner; + /** + * @get_power: get the POWER_DOMAIN_AUDIO power well + * + * Request the power well to be turned on. + */ void (*get_power)(struct device *); + /** + * @put_power: put the POWER_DOMAIN_AUDIO power well + * + * Allow the power well to be turned off. + */ void (*put_power)(struct device *); + /** + * @codec_wake_override: Enable/disable codec wake signal + */ void (*codec_wake_override)(struct device *, bool enable); + /** + * @get_cdclk_freq: Get the Core Display Clock in kHz + */ int (*get_cdclk_freq)(struct device *); + /** + * @sync_audio_rate: set n/cts based on the sample rate + * + * Called from audio driver. After audio driver sets the + * sample rate, it will call this function to set n/cts + */ int (*sync_audio_rate)(struct device *, int port, int rate); }; +/** + * struct i915_audio_component_audio_ops - Ops implemented by hda driver, called by i915 driver + */ struct i915_audio_component_audio_ops { + /** + * @audio_ptr: Pointer to be used in call to pin_eld_notify + */ void *audio_ptr; /** - * Call from i915 driver, notifying the HDA driver that - * pin sense and/or ELD information has changed. - * @audio_ptr: HDA driver object - * @port: Which port has changed (PORTA / PORTB / PORTC etc) + * @pin_eld_notify: Notify the HDA driver that pin sense and/or ELD information has changed + * + * Called when the i915 driver has set up audio pipeline or has just + * begun to tear it down. This allows the HDA driver to update its + * status accordingly (even when the HDA controller is in power save + * mode). */ void (*pin_eld_notify)(void *audio_ptr, int port); }; /** - * struct i915_audio_component - used for audio video interaction - * @dev: the device from gfx driver - * @aud_sample_rate: the array of audio sample rate per port - * @ops: callback for audio driver calling - * @audio_ops: Call from i915 driver + * struct i915_audio_component - Used for direct communication between i915 and hda drivers */ struct i915_audio_component { + /** + * @dev: i915 device, used as parameter for ops + */ struct device *dev; + /** + * @aud_sample_rate: the array of audio sample rate per port + */ int aud_sample_rate[MAX_PORTS]; - + /** + * @ops: Ops implemented by i915 driver, called by hda driver + */ const struct i915_audio_component_ops *ops; - + /** + * @audio_ops: Ops implemented by hda driver, called by i915 driver + */ const struct i915_audio_component_audio_ops *audio_ops; }; -- 1.9.1

2 1

[alsa-devel] [PATCH 0/5] ALSA: firewire: suppress sparse warnings due to endianness handling
by Takashi Sakamoto 19 Oct '15

19 Oct '15

Hi, This patchset fixes some issues related to sparse warnings about endianness handling for ALSA firewire stack. For writing this patchset, I use 'C=1 CF=-D__CHECK_ENDIAN__' options to 'make' command. If they're not enough, please inform to me. Takashi Sakamoto (5): ALSA: dice: correct variable types for __be32 data ALSA: dice: assign converted data to the same type of variable ALSA: fireworks: use u32 type for be32_to_cpup() macro ALSA: bebob: use correct type for __be32 data ALSA: firewire-digi00x/firewire-tascam: remove wrong conversion for Config ROM sound/firewire/bebob/bebob_maudio.c | 2 +- sound/firewire/dice/dice-stream.c | 12 ++++++------ sound/firewire/dice/dice.c | 3 ++- sound/firewire/digi00x/digi00x.c | 3 +-- sound/firewire/fireworks/fireworks_command.c | 2 +- sound/firewire/tascam/tascam.c | 3 +-- 6 files changed, 12 insertions(+), 13 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH] ALSA: firewire-lib: avoid NULL pointer dereference after closing MIDI port
by Takashi Sakamoto 19 Oct '15

19 Oct '15

When asynchronous MIDI port is closed before callbacked, the callback function causes NULL pointer dereference to missing MIDI substream. This commit fixes this bug. Fixes: e8a40d9bcb23('ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages') Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/lib.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c index edf1c8b..f80aafa 100644 --- a/sound/firewire/lib.c +++ b/sound/firewire/lib.c @@ -74,7 +74,11 @@ static void async_midi_port_callback(struct fw_card *card, int rcode, struct snd_fw_async_midi_port *port = callback_data; struct snd_rawmidi_substream *substream = ACCESS_ONCE(port->substream); - if (rcode == RCODE_COMPLETE && substream != NULL) + /* This port is closed. */ + if (substream == NULL) + return; + + if (rcode == RCODE_COMPLETE) snd_rawmidi_transmit_ack(substream, port->consume_bytes); else if (!rcode_is_permanent_error(rcode)) /* To start next transaction immediately for recovery. */ -- 2.1.4

2 1

[alsa-devel] [PATCH v6 1/4] drm/i915: Add audio sync_audio_rate callback
by libin.yang＠intel.com 19 Oct '15

19 Oct '15

From: Libin Yang <libin.yang(a)intel.com> Add the sync_audio_rate callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS or N/M based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..8ad6f1b 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,13 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + /** + * @sync_audio_rate: set n/cts based on the sample rate + * + * Called from audio driver. After audio driver sets the + * sample rate, it will call this function to set n/cts + */ + int (*sync_audio_rate)(struct device *, int port, int rate); } *ops; }; -- 1.9.1

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[alsa-devel] audio-over-SPI driver for ASoC
by Brian Clarkson 18 Oct '15

18 Oct '15

Hi I'm about to embark on the development of an ASoC capture driver that uses SPI to stream the samples to the processor. I'm still spending a lot of my time going through the ALSA kernel code but as I do that may I have some suggestions as to which examples would be closest to my case? My development hardware is AM335x --> McSPI --> ADS8868. So far I think I have achieved a solid understanding of the "simple-card" case where you have dai-link that connects a processor peripheral driver (e.g. McASP ) with a codec driver and forms a snd_soc_card that can be registered. Is my best option to: - write a new CPU DAI (snd_soc_dai_driver) based on the existing SPI driver - write a new CODEC DAI (snd_soc_dai_driver) which will be mostly empty - write a new platform driver that populates and registers a new snd_soc_card struct with the above in a dai_link? Any advice would be much appreciated. Cheers, Brian -- Orthogonal Devices Tokyo, Japan www.orthogonaldevices.com +81 80 8700 2340

2 1

[alsa-devel] [PATCH 0/3] fix sending the same MIDI bytes in two transactions
by Takashi Sakamoto 18 Oct '15

18 Oct '15

Hi, In former commits, MIDI functionality is added to TASCAM driver, while it includes a bug to send the same MIDI bytes in two continuous transactions. This patchset is to fix this bug, including some code improvements. Takashi Sakamoto (3): ALSA: tascam: remove buffer initialization in driver side ALSA: tascam: improve loop condition to filling MIDI bytes in transaction ALSA: tascam: clear extra MIDI byte of asynchronous transaction sound/firewire/tascam/tascam-transaction.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) -- 2.1.4

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[alsa-devel] Sildes of topology status for ELEC 2015
by Lin, Mengdong 18 Oct '15

18 Oct '15

Hi, The slides of topology status is attached. And Libin will update DP MST audio slides based on recent alignment. Thanks Mengdong

4 6

[alsa-devel] [PATCH] ALSA: USB-audio: Add support for Novation Nocturn MIDIcontrol surface
by Ricard Wanderlof 18 Oct '15

18 Oct '15

The Nocturn needs the MIDI_RAW_BYTES quirk, like other Novation devices. Tested that the Nocturn shows up in aconnect, and that it can be used as a control surface (using the xtor synthesizer patch editor). Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- I've noticed that checkpatch complains about the space between the & and (, but in the interest of consistency I've not changed this. Is there some reason for the space, or would it be worth while cleaning up the file? sound/usb/quirks-table.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e475665..99de061 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2664,6 +2664,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE(0x1235, 0x000a), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Novation", */ + /* .product_name = "Nocturn", */ + .ifnum = 0, + .type = QUIRK_MIDI_RAW_BYTES + } +}, +{ USB_DEVICE(0x1235, 0x000e), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "Novation", */ -- 2.1.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 4

[alsa-devel] [PATCH] ALSA: fireworks/bebob/oxfw/dice: enable to make as built-in
by Takashi Sakamoto 18 Oct '15

18 Oct '15

When committed to upstream, these four modules had wrong entries for Makefile. This forces them to be loadable modules even if they're set as built-in. This commit fixes this bug. Fixes: b5b04336015e('ALSA: fireworks: Add skelton for Fireworks based devices') Fixes: fd6f4b0dc167('ALSA: bebob: Add skelton for BeBoB based devices') Fixes: 1a4e39c2e5ca('ALSA: oxfw: Move to its own directory') Fixes: 14ff6a094815('ALSA: dice: Move file to its own directory') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/Makefile | 2 +- sound/firewire/dice/Makefile | 2 +- sound/firewire/fireworks/Makefile | 2 +- sound/firewire/oxfw/Makefile | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/bebob/Makefile b/sound/firewire/bebob/Makefile index 6cf470c..af7ed66 100644 --- a/sound/firewire/bebob/Makefile +++ b/sound/firewire/bebob/Makefile @@ -1,4 +1,4 @@ snd-bebob-objs := bebob_command.o bebob_stream.o bebob_proc.o bebob_midi.o \ bebob_pcm.o bebob_hwdep.o bebob_terratec.o bebob_yamaha.o \ bebob_focusrite.o bebob_maudio.o bebob.o -obj-m += snd-bebob.o +obj-$(CONFIG_SND_BEBOB) += snd-bebob.o diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 9ef228e..55b4be9 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,3 +1,3 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o -obj-m += snd-dice.o +obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/fireworks/Makefile b/sound/firewire/fireworks/Makefile index 0c74408..15ef7f7 100644 --- a/sound/firewire/fireworks/Makefile +++ b/sound/firewire/fireworks/Makefile @@ -1,4 +1,4 @@ snd-fireworks-objs := fireworks_transaction.o fireworks_command.o \ fireworks_stream.o fireworks_proc.o fireworks_midi.o \ fireworks_pcm.o fireworks_hwdep.o fireworks.o -obj-m += snd-fireworks.o +obj-$(CONFIG_SND_FIREWORKS) += snd-fireworks.o diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile index a926850..06ff50f 100644 --- a/sound/firewire/oxfw/Makefile +++ b/sound/firewire/oxfw/Makefile @@ -1,3 +1,3 @@ snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-control.o oxfw-pcm.o \ oxfw-proc.o oxfw-midi.o oxfw-hwdep.o oxfw.o -obj-m += snd-oxfw.o +obj-$(CONFIG_SND_OXFW) += snd-oxfw.o -- 2.1.4

2 3

[alsa-devel] [PATCH 2/3] ALSA: hda - Remove leftover snd_hda_bus() prototype
by Takashi Iwai 17 Oct '15

17 Oct '15

It was forgotten to be removed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5bcfe7407d83..373fcad840ea 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -301,10 +301,6 @@ struct hda_codec { /* * constructors */ -int snd_hda_bus_new(struct snd_card *card, - const struct hdac_bus_ops *ops, - const struct hdac_io_ops *io_ops, - struct hda_bus **busp); int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, unsigned int codec_addr, struct hda_codec **codecp); int snd_hda_codec_configure(struct hda_codec *codec); -- 2.6.0

1 1

[alsa-devel] [PATCH 1/3] ALSA: hda - Fix bogus codec address check for mixer name assignment
by Takashi Iwai 17 Oct '15

17 Oct '15

The recent commit [7fbe824a0f0e: ALSA: hda - Update mixer name for the lower codec address] tried to improve the mixer chip name assignment in the order of codec address. However, this fix was utterly bogus; it checks the field set in each codec, thus this value is reset at each codec creation, of course. For really handling this priority, the assignment has to be remembered in the common place, namely in hda_bus, instead of hda_codec. Fixes: 7fbe824a0f0e ('ALSA: hda - Update mixer name for the lower codec address') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_bind.c | 4 ++-- sound/pci/hda/hda_codec.c | 1 - sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_controller.c | 1 + 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 152acdaa0a45..70671ad65d24 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -63,11 +63,11 @@ int snd_hda_codec_set_name(struct hda_codec *codec, const char *name) /* update the mixer name */ if (!*codec->card->mixername || - codec->mixer_assigned >= codec->core.addr) { + codec->bus->mixer_assigned >= codec->core.addr) { snprintf(codec->card->mixername, sizeof(codec->card->mixername), "%s %s", codec->core.vendor_name, codec->core.chip_name); - codec->mixer_assigned = codec->core.addr; + codec->bus->mixer_assigned = codec->core.addr; } return 0; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0e55c6a6cc7e..2eeaf5ea20f9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -851,7 +851,6 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); codec->depop_delay = -1; codec->fixup_id = HDA_FIXUP_ID_NOT_SET; - codec->mixer_assigned = -1; #ifdef CONFIG_PM codec->power_jiffies = jiffies; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 055a80522282..5bcfe7407d83 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -70,6 +70,7 @@ struct hda_bus { unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ int primary_dig_out_type; /* primary digital out PCM type */ + unsigned int mixer_assigned; /* codec addr for mixer name */ }; /* from hdac_bus to hda_bus */ @@ -260,7 +261,6 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; #endif - unsigned int mixer_assigned; /* filter the requested power state per nid */ unsigned int (*power_filter)(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 944455997fdc..d6b93a20361b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1045,6 +1045,7 @@ int azx_bus_init(struct azx *chip, const char *model, mutex_init(&bus->prepare_mutex); bus->pci = chip->pci; bus->modelname = model; + bus->mixer_assigned = -1; bus->core.snoop = azx_snoop(chip); if (chip->get_position[0] != azx_get_pos_lpib || chip->get_position[1] != azx_get_pos_lpib) -- 2.6.0

1 0

[alsa-devel] [patch 2/2] ALSA: firewire-tascam: off by one in handle_midi_tx()
by Dan Carpenter 17 Oct '15

17 Oct '15

My static checker complains because tscm->spec->midi_capture_ports is either 2 or 4 but the tscm->tx_midi_substreams[] array has 4 elements so this is possibly off by one. I have looked at the code and I think it should be >= instead of > as well. Fixes: 107cc0129a68 ('ALSA: firewire-tascam: add support for incoming MIDI messages by asynchronous transaction') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c index 45c3ce3..d4f64ae 100644 --- a/sound/firewire/tascam/tascam-transaction.c +++ b/sound/firewire/tascam/tascam-transaction.c @@ -158,7 +158,7 @@ static void handle_midi_tx(struct fw_card *card, struct fw_request *request, port = b[0] >> 4; /* TODO: support virtual MIDI ports. */ - if (port > tscm->spec->midi_capture_ports) + if (port >= tscm->spec->midi_capture_ports) goto end; /* Assume the message length. */

2 1

[alsa-devel] [patch 1/2] ALSA: firewire-tascam: fix an LED bug
by Dan Carpenter 17 Oct '15

17 Oct '15

We recently tried to add some new code to support turning the LED on and off but the code in snd_tscm_transaction_reregister() is unreachable. Fixes: e65e2cb99e44 ('ALSA: firewire-tascam: Turn on/off FireWire LED') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c index 1c9a88b..45c3ce3 100644 --- a/sound/firewire/tascam/tascam-transaction.c +++ b/sound/firewire/tascam/tascam-transaction.c @@ -249,7 +249,7 @@ int snd_tscm_transaction_reregister(struct snd_tscm *tscm) /* Turn on messaging. */ reg = cpu_to_be32(0x00000001); - return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, + err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST, TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ON, &reg, sizeof(reg), 0); if (err < 0)

2 1

[alsa-devel] [PATCH] ASoC: rt5645: Recheck the jack detect status after resuming from S3
by Oder Chiou 16 Oct '15

16 Oct '15

The patch rechecks the jack detect status after resuming from S3. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 89ba4ff..4d54999 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2745,6 +2745,10 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_FV1 | RT5645_PWR_FV2, RT5645_PWR_FV1 | RT5645_PWR_FV2); + if (rt5645->en_button_func && + snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) + queue_delayed_work(system_power_efficient_wq, + &rt5645->jack_detect_work, msecs_to_jiffies(0)); break; case SND_SOC_BIAS_OFF: -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH RESEND] ASoC: rt298: correct index default value
by Bard Liao 16 Oct '15

16 Oct '15

Some of the default value on rt298_index_def are incorrect. Change them to the correct value. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- This patch is base on topic/rt298 branch. --- sound/soc/codecs/rt298.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index ff126a7..8c5cb7e 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -50,24 +50,24 @@ struct rt298_priv { }; static struct reg_default rt298_index_def[] = { - { 0x01, 0xaaaa }, - { 0x02, 0x8aaa }, + { 0x01, 0xa5a8 }, + { 0x02, 0x8e95 }, { 0x03, 0x0002 }, - { 0x04, 0xaf01 }, - { 0x08, 0x000d }, - { 0x09, 0xd810 }, - { 0x0a, 0x0120 }, + { 0x04, 0xaf67 }, + { 0x08, 0x200f }, + { 0x09, 0xd010 }, + { 0x0a, 0x0100 }, { 0x0b, 0x0000 }, { 0x0d, 0x2800 }, - { 0x0f, 0x0000 }, - { 0x19, 0x0a17 }, + { 0x0f, 0x0022 }, + { 0x19, 0x0217 }, { 0x20, 0x0020 }, { 0x33, 0x0208 }, { 0x46, 0x0300 }, - { 0x49, 0x0004 }, - { 0x4f, 0x50e9 }, - { 0x50, 0x2000 }, - { 0x63, 0x2902 }, + { 0x49, 0x4004 }, + { 0x4f, 0x50c9 }, + { 0x50, 0x3000 }, + { 0x63, 0x1b02 }, { 0x67, 0x1111 }, { 0x68, 0x1016 }, { 0x69, 0x273f }, -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] Remove Renesas sound SRU for Gen1 support
by Kuninori Morimoto 16 Oct '15

16 Oct '15

Hi ALSA, and SH ML I would like to remove Renesas sound SRU support of Gen1 in the future. It was created for preparation of Gen2 SRC support, and it converts sampling rate. But no-one is using this feature in Gen1. board-bockw had used SRU before, but it was pass through mode. This means it is same as SSI. 2nd reason is that board-bockw was removed from upstream code. BockW is now supported by DT, but it doesn't use SRU. More detail, r8a7778.dtsi has rcar_sound,src entry, but no-one is using this feature. SRU proving has no relation to this removing feature. This means there is no effect for DT compatibility. Gen2 SRC was created from Gen1 SRU, these are similar but not same IP. Keeping Gen1 SRU in current driver is a little bit difficult, and no-one is using it today. Gen1 sound is still supported via SSI. If there is no objection, I would like to remove it from upstream step1. add warning message in v4.x step2. remove it in v4.x+1 Best regards --- Kuninori Morimoto

2 2

[alsa-devel] [PATCH] ASoC: rsnd: Gen1 probe is not error
by Kuninori Morimoto 16 Oct '15

16 Oct '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Probing from Gen1 is not error. This patch fixup it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ctu.c | 6 ++---- sound/soc/sh/rcar/dvc.c | 6 ++---- sound/soc/sh/rcar/mix.c | 6 ++---- 3 files changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 6e34909c..27d56a0 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -125,10 +125,8 @@ int rsnd_ctu_probe(struct platform_device *pdev, int i, nr, ret; /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) { - dev_warn(dev, "CTU is not supported on Gen1\n"); - return -EINVAL; - } + if (rsnd_is_gen1(priv)) + return 0; rsnd_of_parse_ctu(pdev, of_data, priv); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 08e58d0..63e8e52 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -338,10 +338,8 @@ int rsnd_dvc_probe(struct platform_device *pdev, int i, nr, ret; /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) { - dev_warn(dev, "CMD is not supported on Gen1\n"); - return -EINVAL; - } + if (rsnd_is_gen1(priv)) + return 0; rsnd_of_parse_dvc(pdev, of_data, priv); diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 2311d49..a5485b2 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -158,10 +158,8 @@ int rsnd_mix_probe(struct platform_device *pdev, int i, nr, ret; /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) { - dev_warn(dev, "MIX is not supported on Gen1\n"); - return -EINVAL; - } + if (rsnd_is_gen1(priv)) + return 0; rsnd_of_parse_mix(pdev, of_data, priv); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Add info callback for SX_TLV controls
by Charles Keepax 16 Oct '15

16 Oct '15

SX_TLV controls are intended for situations where the register behind the control has some non-zero value indicating the minimum gain and then gains increasing from there and eventually overflowing through zero. Currently every CODEC implementing these controls specifies the minimum as the non-zero value for the minimum and the maximum as the number of gain settings available. This means when the info callback subtracts the minimum value from the maximum value to calculate the number of gain levels available it is actually under reporting the available levels. This patch fixes this issue by adding a new snd_soc_info_volsw_sx callback that does not subtract the minimum value. Fixes: 1d99f2436d0d ("ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV") Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 6 ++++-- sound/soc/soc-ops.c | 28 ++++++++++++++++++++++++++++ 2 files changed, 32 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 98e080b..bd85564 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -86,7 +86,7 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array),\ - .info = snd_soc_info_volsw, \ + .info = snd_soc_info_volsw_sx, \ .get = snd_soc_get_volsw_sx,\ .put = snd_soc_put_volsw_sx, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ @@ -156,7 +156,7 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw, \ + .info = snd_soc_info_volsw_sx, \ .get = snd_soc_get_volsw_sx, \ .put = snd_soc_put_volsw_sx, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ @@ -593,6 +593,8 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +int snd_soc_info_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); #define snd_soc_info_bool_ext snd_ctl_boolean_mono_info int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 100d92b..05977ae 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -207,6 +207,34 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_info_volsw); /** + * snd_soc_info_volsw_sx - Mixer info callback for SX TLV controls + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a single mixer control, or a double + * mixer control that spans 2 registers of the SX TLV type. SX TLV controls + * have a range that represents both positive and negative values either side + * of zero but without a sign bit. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + snd_soc_info_volsw(kcontrol, uinfo); + /* Max represents the number of levels in an SX control not the + * maximum value, so add the minimum value back on + */ + uinfo->value.integer.max += mc->min; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_sx); + +/** * snd_soc_get_volsw - single mixer get callback * @kcontrol: mixer control * @ucontrol: control element information -- 1.7.2.5

3 3

[alsa-devel] race condition between dmaengine_pcm_dma_complete and snd_pcm_release
by Matt Campbell 16 Oct '15

16 Oct '15

Hi All, I've been trying to track down a kernel crash I've been getting when closing ALSA devices. There is a race condition between the bottom half interrupt handler for the DMA system (dmaengine_pcm_dma_complete in pcm_dmaengine.c) and the releasing of the sub-stream resources it receives as an argument (when snd_pcm_release is called). An older thread from 2013 discussed this to a good extent so I wont go into the details here. I've been unable to track down either a patch to fix this or even a good solution that I could implement. I've spent a couple days trying to think of an elegant solution and come up with nothing so far. Any input would be appreciated. Link to original thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-October/067111.ht… It's worth nothing that the original thread points out how this can arise on multi-core systems. In my case I'm on a single core system, but with the real-time patch enabled and the userspace ALSA thread running at a higher priority than the kernel's tasklet thread which can reproduce this almost 100% of the time. ~Matt

2 5

[alsa-devel] [PATCH] ALSA: timer: add config item to export PCM timer disabling for expert
by Jie Yang 16 Oct '15

16 Oct '15

PCM timer is not always used. For embedded device, we need an interface to disable it when it is not needed, to shrink the kernel size and memory footprint, here add CONFIG_SND_PCM_TIMER for it. When both CONFIG_SND_PCM_TIMER and CONFIG_SND_TIMER is unselected, about 25KB saving bonus we can get. Please be noted that when disabled, those stubs who using pcm timer (e.g. dmix, dsnoop & co) may work incorrectlly. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/pcm.h | 8 +++++++- sound/core/Kconfig | 13 ++++++++++++- sound/core/Makefile | 3 ++- sound/core/pcm_lib.c | 2 ++ sound/core/pcm_native.c | 36 ++++++++++++++++-------------------- 5 files changed, 39 insertions(+), 23 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 691e7ee..79cf943 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1117,10 +1117,16 @@ static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substrea * Timer interface */ +#ifdef CONFIG_SND_PCM_TIMER void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream); void snd_pcm_timer_init(struct snd_pcm_substream *substream); void snd_pcm_timer_done(struct snd_pcm_substream *substream); - +#else +static inline void +snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream) {} +static inline void snd_pcm_timer_init(struct snd_pcm_substream *substream) {} +static inline void snd_pcm_timer_done(struct snd_pcm_substream *substream) {} +#endif /** * snd_pcm_gettime - Fill the timespec depending on the timestamp mode * @runtime: PCM runtime instance diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 6c96fee..e3e9491 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -4,7 +4,7 @@ config SND_TIMER config SND_PCM tristate - select SND_TIMER + select SND_TIMER if SND_PCM_TIMER config SND_PCM_ELD bool @@ -93,6 +93,17 @@ config SND_PCM_OSS_PLUGINS support conversion of channels, formats and rates. It will behave like most of new OSS/Free drivers in 2.4/2.6 kernels. +config SND_PCM_TIMER + bool "PCM timer interface" if EXPERT + default y + help + If you disable this option, pcm timer will be inavailable, so + those stubs used pcm timer (e.g. dmix, dsnoop & co) may work + incorrectlly. + + For some embedded device, we may disable it to reduce memory + footprint, about 20KB on x86_64 platform. + config SND_SEQUENCER_OSS bool "OSS Sequencer API" depends on SND_SEQUENCER diff --git a/sound/core/Makefile b/sound/core/Makefile index 3354f91..48ab4b8 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -13,8 +13,9 @@ snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o -snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ +snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \ pcm_memory.o memalloc.o +snd-pcm-$(CONFIG_SND_PCM_TIMER) += pcm_timer.o snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7d45645..6dc4277 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1883,8 +1883,10 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) snd_pcm_update_hw_ptr0(substream, 1) < 0) goto _end; +#ifdef CONFIG_SND_PCM_TIMER if (substream->timer_running) snd_timer_interrupt(substream->timer, 1); +#endif _end: snd_pcm_stream_unlock_irqrestore(substream, flags); if (runtime->transfer_ack_end) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 75888dd..d5672ed 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -486,6 +486,16 @@ static void snd_pcm_set_state(struct snd_pcm_substream *substream, int state) snd_pcm_stream_unlock_irq(substream); } +static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream, + int event) +{ +#ifdef CONFIG_SND_PCM_TIMER + if (substream->timer) + snd_timer_notify(substream->timer, event, + &substream->runtime->trigger_tstamp); +#endif +} + static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -1042,9 +1052,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTART, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); } static struct action_ops snd_pcm_action_start = { @@ -1092,9 +1100,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) if (runtime->status->state != state) { snd_pcm_trigger_tstamp(substream); runtime->status->state = state; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTOP); } wake_up(&runtime->sleep); wake_up(&runtime->tsleep); @@ -1208,18 +1214,12 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push) snd_pcm_trigger_tstamp(substream); if (push) { runtime->status->state = SNDRV_PCM_STATE_PAUSED; - if (substream->timer) - snd_timer_notify(substream->timer, - SNDRV_TIMER_EVENT_MPAUSE, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MPAUSE); wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } else { runtime->status->state = SNDRV_PCM_STATE_RUNNING; - if (substream->timer) - snd_timer_notify(substream->timer, - SNDRV_TIMER_EVENT_MCONTINUE, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MCONTINUE); } } @@ -1267,9 +1267,7 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) snd_pcm_trigger_tstamp(substream); runtime->status->suspended_state = runtime->status->state; runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSUSPEND, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSUSPEND); wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } @@ -1373,9 +1371,7 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state) struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); runtime->status->state = runtime->status->suspended_state; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MRESUME); } static struct action_ops snd_pcm_action_resume = { -- 2.1.4

2 1

[alsa-devel] [PATCH V3 0/4] BAT: some fixes
by han.lu＠intel.com 16 Oct '15

16 Oct '15

From: "Lu, Han" <han.lu(a)intel.com> some fixes and adjusts. Lu, Han (4): BAT: Merge message strings BAT: Use colon instead of comma for separation BAT: Adjust parameters and strings BAT: Use dynamic temp file bat/bat.c | 77 +++++++++++++++++++++++++++++++++--------------------------- bat/common.h | 2 +- 2 files changed, 43 insertions(+), 36 deletions(-) -- 1.9.1

3 6

[alsa-devel] [RFC PATCH] ASoC: Modify check condition of multiple bindings of components
by Koro Chen 16 Oct '15

16 Oct '15

The patch "ASoC: Prevent components from being bound to multiple cards" adds check to prevent multiple cards from using the same component. However, snd_soc_register_platform() or snd_soc_register_codec() will also create components, and sharing the same platform by multiple cards is then refused. This happens with a platform having multiple independent DAIs that share the same DMA controller. Relax the condition by checking component->registered_as_component, which is only true in case of snd_soc_register_component() and will be false for components created by snd_soc_register_platform() or snd_soc_register_codec(). Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 42575b0..eca169a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1106,7 +1106,8 @@ static int soc_probe_component(struct snd_soc_card *card, return 0; if (component->card) { - if (component->card != card) { + if (component->card != card && + component->registered_as_component) { dev_err(component->dev, "Trying to bind component to card \"%s\" but is already bound to card \"%s\"\n", card->name, component->card->name); -- 1.7.9.5

3 9

[alsa-devel] [PATCH MC Next Gen 0/2] Add mixer and control interface media entities
by Shuah Khan 15 Oct '15

15 Oct '15

This patch series adds the following: Add support for creating MEDIA_ENT_F_AUDIO_MIXER entity for each mixer and a MEDIA_INTF_T_ALSA_CONTROL control interface entity that links to mixer entities. MEDIA_INTF_T_ALSA_CONTROL entity corresponds to the control device for the card. Change au0828_create_media_graph() to create pad link between MEDIA_ENT_F_AUDIO_MIXER entity and decoder's AU8522_PAD_AUDIO_OUT. This patch series is dependent on an earlier patch series: Update ALSA, and au0828 drivers to use Managed Media Controller API: https://www.mail-archive.com/linux-media@vger.kernel.org/msg92752.html Please find the media graph with mixer and control interface nodes: https://drive.google.com/folderview?id=0B0NIL0BQg-Alb3JFb2diMXRoQlU&usp=sha… New file: graph_mixer.png Shuah Khan (2): sound/usb: Create media mixer function and control interface entities media: au0828 create link between ALSA Mixer and decoder drivers/media/usb/au0828/au0828-core.c | 12 +++++ drivers/media/usb/au0828/au0828.h | 1 + sound/usb/card.c | 5 ++ sound/usb/media.c | 85 ++++++++++++++++++++++++++++++++++ sound/usb/media.h | 4 ++ sound/usb/mixer.h | 1 + sound/usb/usbaudio.h | 1 + 7 files changed, 109 insertions(+) -- 2.1.4

1 2

[alsa-devel] DSD: Playing silence before and after actual audio data
by Martin Pietryka 15 Oct '15

15 Oct '15

Hi guys, I'm trying to figure out the best way to insert ~40ms worth of silence pattern before and after playing back DSD data. There seems to be support inside ALSA for a DSD silence pattern, but as far as I have seen it's only used when an underrun might occur (silence_threshold, silence_size). Is there some possibility inside ALSA to basically insert silence before and after playback, or how hard would it be to implement something like this? I have looked a bit for myself and for me it seems that the modifications needed inside ALSA to achieve this feature are substantial, but of course I also might have missed something. Martin

2 1

[alsa-devel] Audio Mini Summit 2015, 8th October, Dublin
by Mark Brown 15 Oct '15

15 Oct '15

[This message is BCCed to everyone currently on the signup list] As previously announce the audio mini-summit for this year will be this Thursday (8th August) at CCD in Dublin. Topics we currently have on the list for discussion are: - Power saving and general system design decisions (is what we're doing lining up with modern hardware with things like multiple output streams?). - What to do with HDMI and its integration with graphics. - Test tools (Intel BAT and others). - BATCH flag. - ACPI simple-card and general generic card support. - Topology & media controller integration. - Kernel tinification. - Android <-> alsa-lib configuration translation. - HDA restructuring. - PCM core simplification - the code is currently quite complex! - Soundwire driver support plans. If you're planning to attend and haven't done so already please sign up at: https://docs.google.com/spreadsheets/d/1bi-8GHsqlzt41FR20WiMuWILe_cZowFireV… Once I have details on the room & so on I will follow up to this mail with both. We'll also try and sort out some dinner on Thursday night, if there are any special requirements for that please say (I'm assuming that we will have at least some vegetarians).

9 25

[alsa-devel] [PATCH] ASoC: bcm2835-i2s: Fix module autoload for OF platform drivers
by Luis de Bethencourt 15 Oct '15

15 Oct '15

These platform drivers have a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/bcm/bcm2835-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 03fa1cb..8c435be 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -862,6 +862,8 @@ static const struct of_device_id bcm2835_i2s_of_match[] = { {}, }; +MODULE_DEVICE_TABLE(of, bcm2835_i2s_of_match); + static struct platform_driver bcm2835_i2s_driver = { .probe = bcm2835_i2s_probe, .driver = { -- 2.4.6

2 2

Re: [alsa-devel] Sound not working after new kernel installed
by James Cameron 15 Oct '15

15 Oct '15

On Thu, Oct 15, 2015 at 01:17:15PM +0200, Simon Wolk wrote: > The Kernel changed from 3.2.0 to 3.19.0 for me. By installing the > 3.19. the sound stopped working. Yes, Ubuntu kernels change like that, but you can easily run more recent kernel versions, see Ubuntu Wiki page: https://wiki.ubuntu.com/Kernel/MainlineBuilds While you wait for help, you can install latest v4.2.3 built 3rd October 2015. You can also test kernels between 3.2 and 3.19, if you don't get any other help. -- James Cameron http://quozl.linux.org.au/

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - consolidate chip rename functions
by Takashi Iwai 15 Oct '15

15 Oct '15

A few multiple codec drivers do renaming the chip_name string but all these are open-coded and some of them have even no error check. Let's make common helpers to do it properly. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 1 + sound/hda/hdac_device.c | 22 ++++++++++++++++++++++ sound/pci/hda/hda_bind.c | 35 +++++++++++++++++++++++------------ sound/pci/hda/hda_codec.h | 2 ++ sound/pci/hda/hda_sysfs.c | 3 +-- sound/pci/hda/patch_realtek.c | 12 +----------- sound/pci/hda/patch_via.c | 18 ++++-------------- 7 files changed, 54 insertions(+), 39 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 26e956f4b7c6..49df61c7afdc 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -117,6 +117,7 @@ int snd_hdac_device_init(struct hdac_device *dev, struct hdac_bus *bus, void snd_hdac_device_exit(struct hdac_device *dev); int snd_hdac_device_register(struct hdac_device *codec); void snd_hdac_device_unregister(struct hdac_device *codec); +int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name); int snd_hdac_refresh_widgets(struct hdac_device *codec); int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b3b0ad289df1..4b06b26cee06 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -164,6 +164,28 @@ void snd_hdac_device_unregister(struct hdac_device *codec) EXPORT_SYMBOL_GPL(snd_hdac_device_unregister); /** + * snd_hdac_device_set_chip_name - set/update the codec name + * @codec: the HDAC device + * @name: name string to set + * + * Returns 0 if the name is set or updated, or a negative error code. + */ +int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name) +{ + char *newname; + + if (!name) + return 0; + newname = kstrdup(name, GFP_KERNEL); + if (!newname) + return -ENOMEM; + kfree(codec->chip_name); + codec->chip_name = newname; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name); + +/** * snd_hdac_make_cmd - compose a 32bit command word to be sent to the * HD-audio controller * @codec: the codec object diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index d5ac25cc7fee..ef6b8f836a87 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -45,15 +45,31 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) codec->patch_ops.unsol_event(codec, ev); } -/* reset the codec name from the preset */ -static int codec_refresh_name(struct hda_codec *codec, const char *name) +/** + * snd_hda_codec_set_name - set the codec name + * @codec: the HDA codec + * @name: name string to set + */ +int snd_hda_codec_set_name(struct hda_codec *codec, const char *name) { - if (name) { - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup(name, GFP_KERNEL); + int err; + + if (!name) + return 0; + err = snd_hdac_device_set_chip_name(&codec->core, name); + if (err < 0) + return err; + + /* update the mixer name */ + if (!*codec->card->mixername) { + snprintf(codec->card->mixername, + sizeof(codec->card->mixername), "%s %s", + codec->core.vendor_name, codec->core.chip_name); } - return codec->core.chip_name ? 0 : -ENOMEM; + + return 0; } +EXPORT_SYMBOL_GPL(snd_hda_codec_set_name); static int hda_codec_driver_probe(struct device *dev) { @@ -64,7 +80,7 @@ static int hda_codec_driver_probe(struct device *dev) if (WARN_ON(!codec->preset)) return -EINVAL; - err = codec_refresh_name(codec, codec->preset->name); + err = snd_hda_codec_set_name(codec, codec->preset->name); if (err < 0) goto error; err = snd_hdac_regmap_init(&codec->core); @@ -251,11 +267,6 @@ int snd_hda_codec_configure(struct hda_codec *codec) } } - /* audio codec should override the mixer name */ - if (codec->core.afg || !*codec->card->mixername) - snprintf(codec->card->mixername, - sizeof(codec->card->mixername), "%s %s", - codec->core.vendor_name, codec->core.chip_name); return 0; error: diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 95991e463abb..b6d937784afa 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -463,6 +463,8 @@ void snd_hda_unlock_devices(struct hda_bus *bus); void snd_hda_bus_reset(struct hda_bus *bus); void snd_hda_bus_reset_codecs(struct hda_bus *bus); +int snd_hda_codec_set_name(struct hda_codec *codec, const char *name); + /* * power management */ diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index a6e3d9b511ab..64e0d1d81ca5 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -595,8 +595,7 @@ static void parse_model_mode(char *buf, struct hda_bus *bus, static void parse_chip_name_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - kfree((*codecp)->core.chip_name); - (*codecp)->core.chip_name = kstrdup(buf, GFP_KERNEL); + snd_hda_codec_set_name(*codecp, buf); } #define DEFINE_PARSE_ID_MODE(name) \ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 16b8dcba5c12..e1ffb0202ebc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -822,17 +822,7 @@ static const struct hda_codec_ops alc_patch_ops = { }; -/* replace the codec chip_name with the given string */ -static int alc_codec_rename(struct hda_codec *codec, const char *name) -{ - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup(name, GFP_KERNEL); - if (!codec->core.chip_name) { - alc_free(codec); - return -ENOMEM; - } - return 0; -} +#define alc_codec_rename(codec, name) snd_hda_codec_set_name(codec, name) /* * Rename codecs appropriately from COEF value or subvendor id diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index da5366405eda..d714a57e9460 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -785,21 +785,11 @@ static int patch_vt1708S(struct hda_codec *codec) override_mic_boost(codec, 0x1e, 0, 3, 40); /* correct names for VT1708BCE */ - if (get_codec_type(codec) == VT1708BCE) { - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup("VT1708BCE", GFP_KERNEL); - snprintf(codec->card->mixername, - sizeof(codec->card->mixername), - "%s %s", codec->core.vendor_name, codec->core.chip_name); - } + if (get_codec_type(codec) == VT1708BCE) + snd_hda_codec_set_name(codec, "VT1708BCE"); /* correct names for VT1705 */ - if (codec->core.vendor_id == 0x11064397) { - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup("VT1705", GFP_KERNEL); - snprintf(codec->card->mixername, - sizeof(codec->card->mixername), - "%s %s", codec->core.vendor_name, codec->core.chip_name); - } + if (codec->core.vendor_id == 0x11064397) + snd_hda_codec_set_name(codec, "VT1705"); /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); -- 2.6.1

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[alsa-devel] [PATCH v6 0/10] Fix broken DMAFLUSHP on Rockchips platform
by Shawn Lin 15 Oct '15

15 Oct '15

The purpose of the DMAFLUSHP instruction: - Tell the peripheral to clear its status and control registers. - Send a message to the peripheral to resend its level status. There are 3 timings described in PL330 Technical Reference Manual: - Timing 1: Burst request, can work well without DMAFLUSHP. - Timing 2: Single and burst request, DMAC will ignore the single transfer request. This timing happens if there are single and burst request. - Timing 3: Single transfers for a burst request, DMAC should signals datype to request the peripheral to flush the contents of any control registers. This timing happens if there is not enough MFIFO to places the burst data. A peripheral may signal a DMA request during the execution of DMAFLUSHP instruction, that cause DMA request being ignored by DMAC. But DMAC and all peripherals on RK3X SoCs DO NOT support DMAFLUSHP. It can't send a message to the peripheral to resend DMA request, and the peripheral can't acknowledge a flush request from DMAC. So all DMA requests should NOT be ignored by DMAC, and DMAC will not notify the peripheral to flush. To fix this problem, we need: - Do NOT execute DMAFLUSHP instruction. - Timing 2 and timing 3 should not happen. Because on RK3X SoCs, there are 6 or below channels and 32 MFIFO depth for DMAC_BUS, and 8 channels and 64 MFIFO depth for DMAC_PERI, it is impossible to hit the timing 3 if burst length is equal or less than 4. Since the request type signal by the peripheral can only be set by software. We can set Rockchip Soc's GRF_PERIDMAC_CON0[2:1] to select single or burst request, if it is set b01, all of the peripharals will signal a brust request. So the timing 2 will not happen, too. So DMAC on RK3X can support single or burst transfer, but can't support mixed transfer. Because burst transfer is more efficient than single transfer, this is confirmed by our ASIC team, who strongly suggest to use burst transfer. And this is confirmed by Addy's test on RK3288-Pink2 board, the speed of spi flash burst transfer will increase about two times than single transfer. Also, I have tested dw_mmc with pl330 on RK3188 platform to double confirm the result. That means burst transfer is reansonable. So we need a quirk not to execute DMAFLUSHP instruction and to use burst transfer. Note: - The Rockchip Soc default value of GRF_PERIDMAC_CON0[2:1] is b01. To support brust transfer, these bits should not be changed in bootloader. Changes in v6: - remove expose quirk and add dma max_burst caps for clients - remove expose quirks and add max_burst for dmaengine - remove get quirks and add get slave caps - remove quirks and get dma caps in order to limit burst Changes in v5: - add Mark's tag for spi changes - remove unnecessary whitespace change - use switch statement for i2s quirk - use switch statement for dma_quirk's manipulation Changes in v4: - remove spi & i2s dts changes and query quirk from dmaengine API suggeseted by Mark. - fix typo - Add dmaengine_get_quirk hook and implement it for pl330 Changes in v3: - add Sunny's tag - add more rockchip drivers' changes in this patchset - add Reviewed-by: Sonny Rao <sonnyrao(a)chromium.org> Changes in v2: - amend the author - reorder the patches suggested by Doug - add Reviewed-by: Doug Anderson <dianders(a)chromium.org> for rk3288.dtsi patch and arm-pl330.txt patch - fix Olof's mail address Changes in v1: - add From original author. - remove Sunny's tag - rename broken-no-flushp to "arm,pl330-broken-no-flushp" suggested by Krzysztof. Addy Ke (3): DMA: pl330: add quirk for broken no flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3288 platform spi: rockchip: modify DMA max burst to 1 Boojin Kim (1): DMA: pl330: support burst mode for dev-to-mem and mem-to-dev transmit Shawn Lin (5): Documentation: arm-pl330: add description of arm,pl330-broken-no-flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3xxx platform DMA: dmaengine: expose max burst capability to clients DMA: pl330: add max burst for dmaengine snd: dmaengine-pcm: add snd_dmaengine_pcm_get_caps interface Yiwei Cai (1): ASoC: rockchip_i2s: modify DMA max burst to 1 .../devicetree/bindings/dma/arm-pl330.txt | 1 + arch/arm/boot/dts/rk3288.dtsi | 3 + arch/arm/boot/dts/rk3xxx.dtsi | 3 + drivers/dma/dmaengine.c | 1 + drivers/dma/pl330.c | 106 +++++++++++++++------ drivers/spi/spi-rockchip.c | 12 ++- include/linux/dmaengine.h | 4 + sound/soc/rockchip/rockchip_i2s.c | 18 +++- sound/soc/soc-generic-dmaengine-pcm.c | 24 +++++ 9 files changed, 138 insertions(+), 34 deletions(-) -- 2.3.7

4 16

[alsa-devel] [PATCH] ALSA: hda - Enable widget power saving for Cirrus codecs
by Takashi Iwai 15 Oct '15

15 Oct '15

Cirrus codecs have also fine power controls on each widget, thus it gets benefit from the recent widget power-saving feature. As we haven't seen any obvious regressions with tests on some MacBooks, let's try to enable it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 85813de26da8..01c464a468f6 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -570,6 +570,7 @@ static struct cs_spec *cs_alloc_spec(struct hda_codec *codec, int vendor_nid) return NULL; codec->spec = spec; spec->vendor_nid = vendor_nid; + codec->power_save_node = 1; snd_hda_gen_spec_init(&spec->gen); return spec; -- 2.6.1

1 0

[alsa-devel] Sound not working after new kernel installed
by Simon Wolk 15 Oct '15

15 Oct '15

Hello everybody, I don't know if my question is right here, but either way I am asking: I am using an old Benq notebook (Joybook S73) mit Xubuntu 15.04. When I installed a new kernel (then 3.19.0-16, but now I am using 3.19.0.28) the sound on the speakers stoped working and there were no more audible sounds. With headphones plugged in and a the highest possible volume I could hear very silent the music played on the PC. Up to Kernel 3.2.0-70 the sound worked fine. System update and upgrade unfortunately did not help. Do you have any idea what may be the problem? I did the following to show the chip used: simon@simon-Joybook-S73:~$ lspci -nnk | grep -i audio -A2 00:1b.0 Audio device [0403]: Intel Corporation NM10/ICH7 Family High Definition Audio Controller [8086:27d8] (rev 02) Subsystem: Benq Corporation Device [17ff:0600] Kernel driver in use: snd_hda_intel simon@simon-Joybook-S73:~$ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x6c400000 irq 29 Thank you for your help! Simon

3 2

Re: [alsa-devel] ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages
by Dan Carpenter 15 Oct '15

15 Oct '15

Hello Takashi Sakamoto, This is a semi-automatic email about new static checker warnings. The patch e8a40d9bcb23: "ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages" from Oct 9, 2015, leads to the following Smatch complaint: sound/firewire/lib.c:88 async_midi_port_callback() error: we previously assumed 'substream' could be null (see line 77) sound/firewire/lib.c 76 77 if (rcode == RCODE_COMPLETE && substream != NULL) ^^^^^^^^^^^^^^^^^ Existing code assumes substream can be NULL. 78 snd_rawmidi_transmit_ack(substream, port->consume_bytes); 79 else if (!rcode_is_permanent_error(rcode)) 80 /* To start next transaction immediately for recovery. */ 81 port->next_ktime = ktime_set(0, 0); 82 else 83 /* Don't continue processing. */ 84 port->error = true; 85 86 port->idling = true; 87 88 if (!snd_rawmidi_transmit_empty(substream)) ^^^^^^^^^ Patch adds an unchecked dereference (inside the function call). 89 schedule_work(&port->work); 90 } regards, dan carpenter

2 1

[alsa-devel] [PATCH v4 28/79] hdspm.h: use __u8, __u32 and __u64 from linux/types.h instead of stdint.h
by Mikko Rapeli 15 Oct '15

15 Oct '15

Kernel headers should use linux/types.h based definitions. Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/hdspm.h | 40 ++++++++++++++++++---------------------- 1 file changed, 18 insertions(+), 22 deletions(-) diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index 5737332..c4db6f5 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -20,11 +20,7 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ -#ifdef __KERNEL__ #include <linux/types.h> -#else -#include <stdint.h> -#endif /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 @@ -46,15 +42,15 @@ enum hdspm_speed { /* -------------------- IOCTL Peak/RMS Meters -------------------- */ struct hdspm_peak_rms { - uint32_t input_peaks[64]; - uint32_t playback_peaks[64]; - uint32_t output_peaks[64]; + __u32 input_peaks[64]; + __u32 playback_peaks[64]; + __u32 output_peaks[64]; - uint64_t input_rms[64]; - uint64_t playback_rms[64]; - uint64_t output_rms[64]; + __u64 input_rms[64]; + __u64 playback_rms[64]; + __u64 output_rms[64]; - uint8_t speed; /* enum {ss, ds, qs} */ + __u8 speed; /* enum {ss, ds, qs} */ int status2; }; @@ -155,21 +151,21 @@ enum hdspm_syncsource { }; struct hdspm_status { - uint8_t card_type; /* enum hdspm_io_type */ + __u8 card_type; /* enum hdspm_io_type */ enum hdspm_syncsource autosync_source; - uint64_t card_clock; - uint32_t master_period; + __u64 card_clock; + __u32 master_period; union { struct { - uint8_t sync_wc; /* enum hdspm_sync */ - uint8_t sync_madi; /* enum hdspm_sync */ - uint8_t sync_tco; /* enum hdspm_sync */ - uint8_t sync_in; /* enum hdspm_sync */ - uint8_t madi_input; /* enum hdspm_madi_input */ - uint8_t channel_format; /* enum hdspm_madi_channel_format */ - uint8_t frame_format; /* enum hdspm_madi_frame_format */ + __u8 sync_wc; /* enum hdspm_sync */ + __u8 sync_madi; /* enum hdspm_sync */ + __u8 sync_tco; /* enum hdspm_sync */ + __u8 sync_in; /* enum hdspm_sync */ + __u8 madi_input; /* enum hdspm_madi_input */ + __u8 channel_format; /* enum hdspm_madi_channel_format */ + __u8 frame_format; /* enum hdspm_madi_frame_format */ } madi; } card_specific; }; @@ -184,7 +180,7 @@ struct hdspm_status { #define HDSPM_ADDON_TCO 1 struct hdspm_version { - uint8_t card_type; /* enum hdspm_io_type */ + __u8 card_type; /* enum hdspm_io_type */ char cardname[20]; unsigned int serial; unsigned short firmware_rev; -- 2.5.0

2 1

[alsa-devel] [PATCH v4 17/79] include/uapi/sound/emu10k1.h: added EMU10K1 version of DECLARE_BITMAP macro
by Mikko Rapeli 15 Oct '15

15 Oct '15

Fixes userspace compilation error: error: expected specifier-qualifier-list before ‘DECLARE_BITMAP’ DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ DECLARE_BITMAP macro is not meant for userspace headers and thus added here as private copy for emu10k.h. Fix was suggested by Arnd Bergmann <arnd(a)arndb.de> in message <2168807.4Yxh5gl11Q@wuerfel> and Takashi Iwai <tiwai(a)suse.de> in message <s5h1thx88tk.wl-tiwai(a)suse.de> on lkml. Signed-off-by: Mikko Rapeli <mikko.rapeli(a)iki.fi> --- include/uapi/sound/emu10k1.h | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index ec1535b..5175e16 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -34,6 +34,14 @@ #define EMU10K1_FX8010_PCM_COUNT 8 +/* + * Following definition is copied from linux/types.h to support compiling + * this header file in userspace since they are not generally available for + * uapi headers. + */ +#define __EMU10K1_DECLARE_BITMAP(name,bits) \ + unsigned long name[(bits) / (sizeof(unsigned long) * 8)] + /* instruction set */ #define iMAC0 0x00 /* R = A + (X * Y >> 31) ; saturation */ #define iMAC1 0x01 /* R = A + (-X * Y >> 31) ; saturation */ @@ -300,7 +308,7 @@ struct snd_emu10k1_fx8010_control_old_gpr { struct snd_emu10k1_fx8010_code { char name[128]; - DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ + __EMU10K1_DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ __u32 __user *gpr_map; /* initializers */ unsigned int gpr_add_control_count; /* count of GPR controls to add/replace */ @@ -313,11 +321,11 @@ struct snd_emu10k1_fx8010_code { unsigned int gpr_list_control_total; /* total count of GPR controls */ struct snd_emu10k1_fx8010_control_gpr __user *gpr_list_controls; /* listed GPR controls */ - DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */ + __EMU10K1_DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */ __u32 __user *tram_data_map; /* data initializers */ __u32 __user *tram_addr_map; /* map initializers */ - DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */ + __EMU10K1_DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */ __u32 __user *code; /* one instruction - 64 bits */ }; -- 2.5.0

2 1

[alsa-devel] [patch] ALSA: oss: underflow in snd_mixer_oss_proc_write()
by Dan Carpenter 15 Oct '15

15 Oct '15

We cap the upper bound of "idx" but not the negative side. Let's make it unsigned to fix this. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index a99f720..7a8c79d 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1177,7 +1177,8 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, struct snd_mixer_oss *mixer = entry->private_data; char line[128], str[32], idxstr[16]; const char *cptr; - int ch, idx; + unsigned int idx; + int ch; struct snd_mixer_oss_assign_table *tbl; struct slot *slot;

2 1

[alsa-devel] [PATCH v5 0/10] Fix broken DMAFLUSHP on Rockchips platform
by Shawn Lin 14 Oct '15

14 Oct '15

The purpose of the DMAFLUSHP instruction: - Tell the peripheral to clear its status and control registers. - Send a message to the peripheral to resend its level status. There are 3 timings described in PL330 Technical Reference Manual: - Timing 1: Burst request, can work well without DMAFLUSHP. - Timing 2: Single and burst request, DMAC will ignore the single transfer request. This timing happens if there are single and burst request. - Timing 3: Single transfers for a burst request, DMAC should signals datype to request the peripheral to flush the contents of any control registers. This timing happens if there is not enough MFIFO to places the burst data. A peripheral may signal a DMA request during the execution of DMAFLUSHP instruction, that cause DMA request being ignored by DMAC. But DMAC and all peripherals on RK3X SoCs DO NOT support DMAFLUSHP. It can't send a message to the peripheral to resend DMA request, and the peripheral can't acknowledge a flush request from DMAC. So all DMA requests should NOT be ignored by DMAC, and DMAC will not notify the peripheral to flush. To fix this problem, we need: - Do NOT execute DMAFLUSHP instruction. - Timing 2 and timing 3 should not happen. Because on RK3X SoCs, there are 6 or below channels and 32 MFIFO depth for DMAC_BUS, and 8 channels and 64 MFIFO depth for DMAC_PERI, it is impossible to hit the timing 3 if burst length is equal or less than 4. Since the request type signal by the peripheral can only be set by software. We can set Rockchip Soc's GRF_PERIDMAC_CON0[2:1] to select single or burst request, if it is set b01, all of the peripharals will signal a brust request. So the timing 2 will not happen, too. So DMAC on RK3X can support single or burst transfer, but can't support mixed transfer. Because burst transfer is more efficient than single transfer, this is confirmed by our ASIC team, who strongly suggest to use burst transfer. And this is confirmed by Addy's test on RK3288-Pink2 board, the speed of spi flash burst transfer will increase about two times than single transfer. Also, I have tested dw_mmc with pl330 on RK3188 platform to double confirm the result. That means burst transfer is reansonable. So we need a quirk not to execute DMAFLUSHP instruction and to use burst transfer. Note: - The Rockchip Soc default value of GRF_PERIDMAC_CON0[2:1] is b01. To support brust transfer, these bits should not be changed in bootloader. Changes in v5: - add Mark's tag for spi changes - remove unnecessary whitespace change - use switch statement for I2S dma_quirk's manipulation Changes in v4: - remove spi & i2s dts changes and query quirk from dmaengine API suggeseted by Mark. - fix typo - Add dmaengine_get_quirk hook and implement it for pl330 Changes in v3: - add Sunny's tag - add more rockchip drivers' changes in this patchset - add Reviewed-by: Sonny Rao <sonnyrao(a)chromium.org> Changes in v2: - amend the author - reorder the patches suggested by Doug - add Reviewed-by: Doug Anderson <dianders(a)chromium.org> for rk3288.dtsi patch and arm-pl330.txt patch - amend Olof's mail address Changes in v1: - rename broken-no-flushp to "arm,pl330-broken-no-flushp" suggested by Krzysztof. - add From original author. - remove Sunny's tag Addy Ke (3): DMA: pl330: add quirk for broken no flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3288 platform spi: rockchip: modify DMA max burst to 1 Boojin Kim (1): DMA: pl330: support burst mode for dev-to-mem and mem-to-dev transmit Shawn Lin (5): Documentation: arm-pl330: add description of arm,pl330-broken-no-flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3xxx platform dmaengine: add API for getting dma controller's quirk DMA: pl330: implement dmaengine_get_quirks hook snd: dmaengine-pcm: add snd_dmaengine_pcm_get_quirks interface Yiwei Cai (1): ASoC: rockchip_i2s: modify DMA max burst to 1 .../devicetree/bindings/dma/arm-pl330.txt | 1 + arch/arm/boot/dts/rk3288.dtsi | 3 + arch/arm/boot/dts/rk3xxx.dtsi | 3 + drivers/dma/pl330.c | 110 +++++++++++++++------ drivers/spi/spi-rockchip.c | 13 ++- include/linux/dmaengine.h | 9 ++ sound/soc/rockchip/rockchip_i2s.c | 21 ++++ sound/soc/soc-generic-dmaengine-pcm.c | 24 +++++ 8 files changed, 153 insertions(+), 31 deletions(-) -- 2.3.7

4 21

[alsa-devel] snd_pcm_hw_params_set_buffer_size_near yields EINVAL for the value 1681
by Andrew Kelley 14 Oct '15

14 Oct '15

When I call snd_pcm_hw_params_set_buffer_size_near with the value 1681, I get an EINVAL error. These other values all work fine: 1 1679 1680 1682 1683 1684 99999 What in the world is going on here?

2 1

[alsa-devel] [PATCH v4 0/8] Add SPDIF support for rockchip
by Sjoerd Simons 13 Oct '15

13 Oct '15

This patchset adds support for the Rockchip SPDIF transceiver as present on RK3066, RK3188 and RK3288 boards and enables it on a Radxa rock pro. Tested on a Radxa Rock Pro board and Radxa Rock 2 Square board. Quite a few changes since v3 (mostly based on review comment). One important addition is proper RK3288 SoC support; While according to the documentation it has 2 SPDIF "solutions" with the default being the old one, this variation doesn't actually work.. So support has been added to switch to the new SPDIF solution on RK3288. Changes in v4: - Require rockchip,grf on RK3288 as the 8 channel solution has to be selected on that SoC - Make the compatibility string one of a known list rather then requiring a precise list of options. - Change the clock names to hclk and mclk instead of spdif_hclk and spdif_mclk to better match the implementation and data sheets. - Use hclk and mclk as the clock names matching the update devicetree documentation - Drop something useless to_info function - Open-code tx control in rk_spdif_trigge - Use devm_regmap_init_mmio_clk to allow the regmap subsystem to manage the hclk as needed - Use a more conventionale pm setup (turn everything on, mark the device as active, enable pm and request the device to go idle) - re-add the platfrorm:rockchip-spdif module alias - Also manage the hclk in the PM callbacks - Check the return code of clk_set_rate - Switch to the 8 channel spdif solutions on RK3288 - Add support for 192khz audio rate - Adjust clock names to match the new names (hclk/mclk) Changes in v3: - Fix typos in commit message - Resent patches to a more complete list of maintainers. Changes in v2: - Remove platform: module alias as it was unused - Call MODULE_DEVICE_TABLE(of, ) right after the of match table - use rk_spdif as a prefix consistenly throughout the driver - Check return code of regmap_update and bubble it up - Sort the spdif node properties - Drop the 0x prefix from the node name - Rename the spdif@ node to sound@ Sjoerd Simons (8): ASoC: dt-bindings: add rockchip tranceiver bindings ASoC: rockchip: Add rockchip SPDIF transceiver driver ARM: dts: rockchip: Add SPDIF transceiver for RK3188 ARM: dts: rockchip: Add SPDIF optical out on Radxa Rock clk: rockchip: Allow the RK3288 SPDIF clocks to change their parent ARM: dts: rockchip: Add SPDIF transceiver for RK3288 ARM: dts: rockchip: Add SPDIF optical out on Radxa Rock2 Square ARM: multi_v7_defconfig: Add rockchip audio support .../devicetree/bindings/sound/rockchip-spdif.txt | 44 +++ arch/arm/boot/dts/rk3188-radxarock.dts | 19 + arch/arm/boot/dts/rk3188.dtsi | 22 ++ arch/arm/boot/dts/rk3288-rock2-square.dts | 18 + arch/arm/boot/dts/rk3288.dtsi | 23 ++ arch/arm/configs/multi_v7_defconfig | 5 + drivers/clk/rockchip/clk-rk3288.c | 16 +- sound/soc/rockchip/Kconfig | 8 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_spdif.c | 409 +++++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.h | 63 ++++ 11 files changed, 621 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 sound/soc/rockchip/rockchip_spdif.c create mode 100644 sound/soc/rockchip/rockchip_spdif.h -- 2.6.1

7 17

[alsa-devel] [PATCH v3 0/2] make it possible to not compile compress APIs
by Jie Yang 13 Oct '15

13 Oct '15

Compress APIs are not always needed, but they are always built currentlly, in both ALSA core and ASoC core. This series make it possible to not compile it when unneeded. Hi, Takashi & all, I am not sure you will like the patch 2/2, it split soc-compress to a module, but introduced several extra 'EXPORT_SYMBOL_GPL' there. If don't like it, you can just apply the first one(1/2) and skip 2/2. thanks, ~Keyon Changes in this version: 1. change to use a callback for compress_new, and remove compress_dai bool. 2. remove the empty module_init/exit for soc-compress module. Jie Yang (2): ASoC: soc-compress: add a config item for soc-compress ASoC: soc-compress: split soc-compress to a module include/sound/soc-dai.h | 2 +- include/sound/soc.h | 2 ++ sound/soc/Kconfig | 6 +++++- sound/soc/Makefile | 5 ++++- sound/soc/intel/Kconfig | 1 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/soc-compress.c | 10 ++++++++++ sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 1 + sound/soc/soc-pcm.c | 13 +++++++++++++ 10 files changed, 40 insertions(+), 6 deletions(-) -- 1.9.1

5 12

[alsa-devel] [PATCH v2] drm/i915: Improve kernel-doc for i915_audio_component struct
by David Henningsson 13 Oct '15

13 Oct '15

To make kernel-doc happy, the i915_audio_component_audio_ops struct cannot be nested. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/DocBook/drm.tmpl | 1 + include/drm/i915_component.h | 92 ++++++++++++++++++++++++++++++------------ 2 files changed, 67 insertions(+), 26 deletions(-) diff --git a/Documentation/DocBook/drm.tmpl b/Documentation/DocBook/drm.tmpl index 9ddf8c6..f16e4e8 100644 --- a/Documentation/DocBook/drm.tmpl +++ b/Documentation/DocBook/drm.tmpl @@ -4051,6 +4051,7 @@ int num_ioctls;</synopsis> <title>High Definition Audio</title> !Pdrivers/gpu/drm/i915/intel_audio.c High Definition Audio over HDMI and Display Port !Idrivers/gpu/drm/i915/intel_audio.c +!Iinclude/drm/i915_component.h </sect2> <sect2> <title>Panel Self Refresh PSR (PSR/SRD)</title> diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index 89dc7d6..76c10c8 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -30,38 +30,78 @@ */ #define MAX_PORTS 5 +/** + * struct i915_audio_component_ops - Ops implemented by i915 driver, called by hda driver + */ +struct i915_audio_component_ops { + /** + * @owner: i915 module + */ + struct module *owner; + /** + * @get_power: Request that power well is to be turned on + */ + void (*get_power)(struct device *); + /** + * @put_power: Allow the power well to be turned off + */ + void (*put_power)(struct device *); + /** + * @codec_wake_override: Force the audio codec to stay awake + */ + void (*codec_wake_override)(struct device *, bool enable); + /** + * @get_cdclk_freq: Query the i915 driver about the current cdclk frequency + */ + int (*get_cdclk_freq)(struct device *); + /** + * @sync_audio_rate: set n/cts based on the sample rate + * + * Called from audio driver. After audio driver sets the + * sample rate, it will call this function to set n/cts + */ + int (*sync_audio_rate)(struct device *, int port, int rate); +}; + +/** + * struct i915_audio_component_audio_ops - Ops implemented by hda driver, called by i915 driver + */ +struct i915_audio_component_audio_ops { + /** + * @audio_ptr: Pointer to be used in call to pin_eld_notify + */ + void *audio_ptr; + /** + * @pin_eld_notify: Notify the HDA driver that pin sense and/or ELD information has changed + * + * Called when the i915 driver has set up audio pipeline or has just + * begun to tear it down. This allows the HDA driver to update its + * status accordingly (even when the HDA controller is in power save + * mode). + */ + void (*pin_eld_notify)(void *audio_ptr, int port); +}; + +/** + * struct i915_audio_component - Used for direct communication between i915 and hda drivers + */ struct i915_audio_component { + /** + * @dev: i915 device, used as parameter for ops + */ struct device *dev; /** * @aud_sample_rate: the array of audio sample rate per port */ int aud_sample_rate[MAX_PORTS]; - - const struct i915_audio_component_ops { - struct module *owner; - void (*get_power)(struct device *); - void (*put_power)(struct device *); - void (*codec_wake_override)(struct device *, bool enable); - int (*get_cdclk_freq)(struct device *); - /** - * @sync_audio_rate: set n/cts based on the sample rate - * - * Called from audio driver. After audio driver sets the - * sample rate, it will call this function to set n/cts - */ - int (*sync_audio_rate)(struct device *, int port, int rate); - } *ops; - - const struct i915_audio_component_audio_ops { - void *audio_ptr; - /** - * Call from i915 driver, notifying the HDA driver that - * pin sense and/or ELD information has changed. - * @audio_ptr: HDA driver object - * @port: Which port has changed (PORTA / PORTB / PORTC etc) - */ - void (*pin_eld_notify)(void *audio_ptr, int port); - } *audio_ops; + /** + * @ops: Ops implemented by i915 driver, called by hda driver + */ + const struct i915_audio_component_ops *ops; + /** + * @audio_ops: Ops implemented by hda driver, called by i915 driver + */ + const struct i915_audio_component_audio_ops *audio_ops; }; #endif /* _I915_COMPONENT_H_ */ -- 1.9.1

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[alsa-devel] Feasibility of adding alternative audio transport besides I2S/PWM/SPDIF, etc
by Caleb Crome 13 Oct '15

13 Oct '15

Hi All, I'm in a constant struggle to bring up many channel audio on each separate SoC. I can easily put a microcontroller in place that will collect and distrubute all the TDM channels to the codecs, and connect the hardware via an SPI interface to the SoC. So, instead of: CODECS <---TDM---> SoC It would be CODECS <---TDM---> uC <---SPI---> SoC So, my questions are: * I suspect the SPI interface could be used more universally than each individual I2S/TDM interface (like FSL SSI vs. Ti McBSP vs. Ti McASP, etc). and the SPI port would provide a very common API regardless of SoC. Is that true? * Is it feasible to integrate an SPI transport into the SoC/alsa layer? What are the difficulties involved with that? Cheers, -Caleb

4 7

[alsa-devel] [PATCH v2] drm/i915: set proper N/CTS in modeset
by libin.yang＠intel.com 13 Oct '15

13 Oct '15

From: Libin Yang <libin.yang(a)intel.com> When modeset occurs and the TMDS frequency is set to some speical values, the N/CTS need to be set manually if audio is playing. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- drivers/gpu/drm/i915/intel_audio.c | 57 ++++++++++++++++++++++++++++++++------ include/drm/i915_component.h | 10 +++++++ 2 files changed, 58 insertions(+), 9 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index e3c32ce..642dfbd 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -128,6 +128,20 @@ static int audio_config_get_n(const struct drm_display_mode *mode, int rate) return 0; } +static uint32_t audio_config_setup_n_reg(int n, uint32_t val) +{ + int n_low, n_up; + uint32_t tmp = val; + + n_low = n & 0xfff; + n_up = (n >> 12) & 0xff; + tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); + tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | + (n_low << AUD_CONFIG_LOWER_N_SHIFT) | + AUD_CONFIG_N_PROG_ENABLE); + return tmp; +} + /* check whether N/CTS/M need be set manually */ static bool audio_rate_need_prog(struct intel_crtc *crtc, struct drm_display_mode *mode) @@ -262,9 +276,14 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, struct drm_i915_private *dev_priv = connector->dev->dev_private; struct intel_crtc *intel_crtc = to_intel_crtc(encoder->base.crtc); enum pipe pipe = intel_crtc->pipe; + struct i915_audio_component *acomp = dev_priv->audio_component; const uint8_t *eld = connector->eld; + struct intel_digital_port *intel_dig_port = + enc_to_dig_port(&encoder->base); + enum port port = intel_dig_port->port; uint32_t tmp; int len, i; + int n, rate; DRM_DEBUG_KMS("Enable audio codec on pipe %c, %u bytes ELD\n", pipe_name(pipe), drm_eld_size(eld)); @@ -302,12 +321,29 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, /* Enable timestamps */ tmp = I915_READ(HSW_AUD_CFG(pipe)); tmp &= ~AUD_CONFIG_N_VALUE_INDEX; - tmp &= ~AUD_CONFIG_N_PROG_ENABLE; tmp &= ~AUD_CONFIG_PIXEL_CLOCK_HDMI_MASK; if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT)) tmp |= AUD_CONFIG_N_VALUE_INDEX; else tmp |= audio_config_hdmi_pixel_clock(mode); + + tmp &= ~AUD_CONFIG_N_PROG_ENABLE; + if (audio_rate_need_prog(intel_crtc, mode)) { + if (!acomp) + rate = 0; + else if (port >= PORT_A && port <= PORT_E) + rate = acomp->aud_sample_rate[port]; + else { + DRM_ERROR("invalid port: %d\n", port); + rate = 0; + } + n = audio_config_get_n(mode, rate); + if (n != 0) + tmp = audio_config_setup_n_reg(n, tmp); + else + DRM_DEBUG_KMS("no suitable N value is found\n"); + } + I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -594,9 +630,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, struct intel_digital_port *intel_dig_port; struct intel_crtc *crtc; struct drm_display_mode *mode; + struct i915_audio_component *acomp = dev_priv->audio_component; enum pipe pipe = -1; u32 tmp; - int n_low, n_up, n; + int n; /* HSW, BDW SKL need this fix */ if (!IS_SKYLAKE(dev_priv) && @@ -630,6 +667,9 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, pipe_name(pipe), port_name(port)); mode = &crtc->config->base.adjusted_mode; + /* port must be valid now, otherwise the pipe will be invalid */ + acomp->aud_sample_rate[port] = rate; + /* 2. check whether to set the N/CTS/M manually or not */ if (!audio_rate_need_prog(crtc, mode)) { tmp = I915_READ(HSW_AUD_CFG(pipe)); @@ -649,15 +689,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, mutex_unlock(&dev_priv->av_mutex); return 0; } - n_low = n & 0xfff; - n_up = (n >> 12) & 0xff; - /* 4. set the N/CTS/M */ + /* 3. set the N/CTS/M */ tmp = I915_READ(HSW_AUD_CFG(pipe)); - tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); - tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | - (n_low << AUD_CONFIG_LOWER_N_SHIFT) | - AUD_CONFIG_N_PROG_ENABLE); + tmp = audio_config_setup_n_reg(n, tmp); I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -678,6 +713,7 @@ static int i915_audio_component_bind(struct device *i915_dev, { struct i915_audio_component *acomp = data; struct drm_i915_private *dev_priv = dev_to_i915(i915_dev); + int i; if (WARN_ON(acomp->ops || acomp->dev)) return -EEXIST; @@ -685,6 +721,9 @@ static int i915_audio_component_bind(struct device *i915_dev, drm_modeset_lock_all(dev_priv->dev); acomp->ops = &i915_audio_component_ops; acomp->dev = i915_dev; + BUILD_BUG_ON(MAX_PORTS != I915_MAX_PORTS); + for (i = 0; i < ARRAY_SIZE(acomp->aud_sample_rate); i++) + acomp->aud_sample_rate[i] = 0; dev_priv->audio_component = acomp; drm_modeset_unlock_all(dev_priv->dev); diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index e6d35d7..89dc7d6 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,8 +24,18 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +/* MAX_PORT is the number of port + * It must be sync with I915_MAX_PORTS defined i915_drv.h + * 5 should be enough as only HSW, BDW, SKL need such fix. + */ +#define MAX_PORTS 5 + struct i915_audio_component { struct device *dev; + /** + * @aud_sample_rate: the array of audio sample rate per port + */ + int aud_sample_rate[MAX_PORTS]; const struct i915_audio_component_ops { struct module *owner; -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: Intel: Skylake - Add Skylake RT286 I2S machine driver
by Vinod Koul 13 Oct '15

13 Oct '15

From: Omair M Abdullah <omair.m.abdullah(a)intel.com> Add the SKL I2S machine driver using Realtek ALC286S codec in I2S mode. Signed-off-by: Omair M Abdullah <omair.m.abdullah(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- v2: added depends on X86 and ACPI based on LKP reports sound/soc/intel/Kconfig | 13 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_rt286.c | 268 +++++++++++++++++++++++++++++++++++++ 3 files changed, 283 insertions(+) create mode 100644 sound/soc/intel/boards/skl_rt286.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 664df1f44530..6d67357f2d74 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -140,3 +140,16 @@ config SND_SOC_INTEL_SKYLAKE select SND_HDA_EXT_CORE select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST + +config SND_SOC_INTEL_SKL_RT286_MACH + tristate "ASoC Audio driver for SKL with RT286 I2S mode" + depends on X86 && ACPI + select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT286 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Skylake platforms + with RT286 I2S audio codec. + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index cb94895c9edb..371c4565cad8 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o +snd-soc-skl_rt286-objs := skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -15,3 +16,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o +obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c new file mode 100644 index 000000000000..71907b2c1611 --- /dev/null +++ b/sound/soc/intel/boards/skl_rt286.c @@ -0,0 +1,268 @@ +/* + * Intel Skylake I2S Machine Driver + * + * Copyright (C) 2014-2015, Intel Corporation. All rights reserved. + * + * Modified from: + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * 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 published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include "../../codecs/rt286.h" + +static struct snd_soc_jack skylake_headset; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin skylake_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new skylake_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget skylake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route skylake_rt286_map[] = { + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphone Jack", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC2"}, + {"DMIC AIF", NULL, "SoC DMIC"}, + + /* CODEC BE connections */ + { "AIF1 Playback", NULL, "ssp0 Tx"}, + { "ssp0 Tx", NULL, "codec0_out"}, + { "ssp0 Tx", NULL, "codec1_out"}, + + { "codec0_in", NULL, "ssp0 Rx" }, + { "codec1_in", NULL, "ssp0 Rx" }, + { "ssp0 Rx", NULL, "AIF1 Capture" }, + + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "Capture" }, + + { "hif1", NULL, "iDisp Tx"}, + { "iDisp Tx", NULL, "iDisp_out"}, + +}; + +static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &skylake_headset, + skylake_headset_pins, ARRAY_SIZE(skylake_headset_pins)); + + if (ret) + return ret; + + rt286_mic_detect(codec, &skylake_headset); + + return 0; +} + + +static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The output is 48KHz, stereo, 16bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static int skylake_rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "set codec sysclk failed: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops skylake_rt286_ops = { + .hw_params = skylake_rt286_hw_params, +}; + +/* skylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link skylake_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "Skl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_playback = 1, + }, + { + .name = "Skl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_capture = 1, + }, + { + .name = "Skl Audio Reference cap", + .stream_name = "refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .ignore_suspend = 1, + .nonatomic = 1, + .dynamic = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .be_id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .init = skylake_rt286_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = skylake_ssp0_fixup, + .ops = &skylake_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .be_id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, +}; + +/* skylake audio machine driver for SPT + RT286S */ +static struct snd_soc_card skylake_rt286 = { + .name = "skylake-rt286", + .owner = THIS_MODULE, + .dai_link = skylake_rt286_dais, + .num_links = ARRAY_SIZE(skylake_rt286_dais), + .controls = skylake_controls, + .num_controls = ARRAY_SIZE(skylake_controls), + .dapm_widgets = skylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(skylake_widgets), + .dapm_routes = skylake_rt286_map, + .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map), + .fully_routed = true, +}; + +static int skylake_audio_probe(struct platform_device *pdev) +{ + skylake_rt286.dev = &pdev->dev; + + return snd_soc_register_card(&skylake_rt286); +} + +static int skylake_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&skylake_rt286); + return 0; +} + +static struct platform_driver skylake_audio = { + .probe = skylake_audio_probe, + .remove = skylake_audio_remove, + .driver = { + .name = "skl_alc286s_i2s", + }, +}; + +module_platform_driver(skylake_audio) + +/* Module information */ +MODULE_AUTHOR("Omair Mohammed Abdullah <omair.m.abdullah(a)intel.com>"); +MODULE_DESCRIPTION("Intel SST Audio for Skylake"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:skl_alc286s_i2s"); -- 2.4.3

2 1

[alsa-devel] [patch] ALSA: pcm: harmless underflow issue in snd_pcm_sw_params()
by Dan Carpenter 13 Oct '15

13 Oct '15

"params->tstamp_mode" is a user supplied int. We cap the maximum but allow negative values by mistake. It's harmless but static checkers complain. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 75888dd..4863af5 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -650,7 +650,8 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, } snd_pcm_stream_unlock_irq(substream); - if (params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST) + if (params->tstamp_mode < 0 || + params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST) return -EINVAL; if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12) && params->tstamp_type > SNDRV_PCM_TSTAMP_TYPE_LAST)

2 2

[alsa-devel] [PATCH V2 0/7] BAT: Add Basic Audio Tester command line tool
by han.lu＠intel.com 13 Oct '15

13 Oct '15

From: "Lu, Han" <han.lu(a)intel.com> BAT (Basic Audio Tester) is a simple command line utility intended to automate audio driver and sound server QA testing with no human interaction. BAT works by playing an audio stream and capturing the same stream in either a digital or analog loop back. It then compares the captured stream to the original to determine if the test case passes or fails. The main idea of frequency detecting is: The analysis function reads data from WAV file, run FFT against the data to get magnitude of frequency vectors, and then calculates the average value and standard deviation of frequency vectors. After that, we define a threshold: threshold = 3 * standard_deviation + average_value Frequencies with amplitude larger than threshold will be recognized as a peak, and the frequency with largest peak value will be recognized as a detected frequency. BAT then compares the detected frequency to target frequency, to decide if the detecting passes or fails. BAT supports 4 working modes: 1. Single line playback; 2. Single line capture and analysis; 3. Playback and capture in loop, and analyze captured data; 4. Local analyze without actual playback or capture. BAT will check devices input by user to decide which mode to use. BAT will create threads for playback and record, and will run spectrum analysis for mode 2, 3 or 4. Changes on V2: 1. Adopt Caleb Crome's method to generate sine signal to avoids unwanted distortion on waveform. Lu, Han (7): BAT: Add initial functions BAT: Add common definitions and functions BAT: Add playback and record functions BAT: Add signal generator BAT: Add converting functions BAT: Add spectrum analysis functions BAT: Add Makefile and configures Makefile.am | 3 + bat/Makefile.am | 14 ++ bat/alsa.c | 594 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/alsa.h | 20 ++ bat/analyze.c | 314 +++++++++++++++++++++++++++++ bat/analyze.h | 16 ++ bat/bat.c | 603 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/common.c | 198 +++++++++++++++++++ bat/common.h | 176 +++++++++++++++++ bat/convert.c | 113 +++++++++++ bat/convert.h | 23 +++ bat/signal.c | 182 +++++++++++++++++ bat/signal.h | 30 +++ configure.ac | 19 +- 14 files changed, 2304 insertions(+), 1 deletion(-) create mode 100644 bat/Makefile.am create mode 100644 bat/alsa.c create mode 100644 bat/alsa.h create mode 100644 bat/analyze.c create mode 100644 bat/analyze.h create mode 100644 bat/bat.c create mode 100644 bat/common.c create mode 100644 bat/common.h create mode 100644 bat/convert.c create mode 100644 bat/convert.h create mode 100644 bat/signal.c create mode 100644 bat/signal.h -- 1.9.1

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[alsa-devel] [PATCH v4] ALSA: usb-audio: Fix max packet size calculation for USB audio
by Ricard Wanderlof 13 Oct '15

13 Oct '15

Rounding must take place before multiplication with the frame size, since each packet contains a whole number of frames. We must also properly consider the data interval, as a larger data interval will result in larger packets, which, depending on the sampling frequency, can result in packet sizes that are less than integral multiples of the packet size for a lower data interval. Detailed explanation and rationale: The code before this commit had the following expression on line 613 to calculate the maximum isochronous packet size: maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) >> (16 - ep->datainterval); Here, ep->freqmax is the maximum assumed sample frequency, calculated from the nominal sample frequency plus 25%. It is ultimately derived from ep->freqn, which is in the units of frames per packet, from get_usb_full_speed_rate() or usb_high_speed_rate(), as applicable, in Q16.16 format. The expression essentially adds the Q16.16 equivalent of 0.999... (i.e. the largest number less than one) to the sample rate, in order to get a rate whose integer part is rounded up from the fractional value. The multiplication with (frame_bits >> 3) yields the number of bytes in a packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back to an integer, taking into consideration the bDataInterval field of the endpoint descriptor (which describes how often isochronous packets are transmitted relative to the (micro)frame rate (125us or 1ms, for USB high speed and full speed, respectively)). For this discussion we will initially assume a bDataInterval of 0, so the second line of the expression just converts the Q16.16 value to an integer. In order to illustrate the problem, we will set frame_bits 64, which corresponds to a frame size of 8 bytes. The problem here is twofold. First, the rounding operation consists of the addition of 0x0.ffff and subsequent conversion to integer, but as the expression stands, the conversion to integer is done after multiplication with the frame size, rather than before. This results in the resulting maxsize becoming too large. Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is 0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000. The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 . However, if we do the number of bytes calculation in a less obscure way it's more apparent what the true corresponding packet size is: we get ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612, and the 8000 is the number of isochronous packets per second on a high speed USB connection (125 us microframe interval). This is fixed by performing the complete rounding operation prior to multiplication with the frame rate. The second problem is that when considering the ep->datainterval, this must be done before rounding, in order to take the advantage of the fact that if the number of bytes per packet is not an integer, the resulting rounded-up integer is not necessarily a factor of two when the data interval is increased by the same factor. For instance, assuming a freqency of 41 kHz, the resulting bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or 0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0), this means that 6 frames per packet are needed, whereas with a data interval of 2 we need 10.25, i.e. 11 frames needed. Rephrasing the maxsize expression to: maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * (frame_bits >> 3); for the above 96 kHz example we instead get ((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value. We can also do the calculation with a non-integer sample rate which is when rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn = 0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)): Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down) True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56 New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56 This is also corroborated by the wMaxPacketSize check on line 616. Assume that wMaxPacketSize = 104, with ep->maxpacksize then having the same value. As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to (104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111 (with decimals 111.99988). Clearly, we should get back the 104 here, which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 . (The error has not been a problem because it only results in maxsize being a bit too big which just wastes a couple of bytes, either as a result of the first maxsize calculation, or because the resulting calculation will hit the wMaxPacketSize value before the packet is too big, resulting in fixing the size to wMaxPacketSize even though the packet is actually not too long.) Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V2: Added tested device to commit message. No other changes. V3: Updated after input from Clemens. Since the expression is now getting rather convoluted, added a description in the comments in the code. New commit message to reflect the new version. Updated rationale below. V4: Moved most of the rationale (slightly rephrased) to the commit message as suggested by Takashi. Although the error is tolerable because it just wastes a couple of bytes in each packet, however, I'm working on a quirk for a device which stuffs an extra four (non sample) bytes into each isochronous packet, and that means that the maxsize calculation must take this into account. During this work it became obvious that there was something wrong with the maxsize calculation, as the resulting packet size when wMaxPacketSize was hit did not correspond to the packet size which triggered the if statement to start with. Since this seems to me an obvious (after looking at it for a bit...) bug I'm submitting this as a separate patch rather than part of an upcoming patch series. sound/usb/endpoint.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e6f7189..a77d9c8 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -610,8 +610,23 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, /* assume max. frequency is 25% higher than nominal */ ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - ep->datainterval); + /* Round up freqmax to nearest integer in order to calculate maximum + * packet size, which must represent a whole number of frames. + * This is accomplished by adding 0x0.ffff before converting the + * Q16.16 format into integer. + * In order to accurately calculate the maximum packet size when + * the data interval is more than 1 (i.e. ep->datainterval > 0), + * multiply by the data interval prior to rounding. For instance, + * a freqmax of 41 kHz will result in a max packet size of 6 (5.125) + * frames with a data interval of 1, but 11 (10.25) frames with a + * data interval of 2. + * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the + * maximum datainterval value of 3, at USB full speed, higher for + * USB high speed, noting that ep->freqmax is in units of + * frames per packet in Q16.16 format.) + */ + maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * + (frame_bits >> 3); /* but wMaxPacketSize might reduce this */ if (ep->maxpacksize && ep->maxpacksize < maxsize) { /* whatever fits into a max. size packet */ -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[alsa-devel] [PATCH] ALSA: hdac: Explicitly add io.h
by Vinod Koul 13 Oct '15

13 Oct '15

Compiling the hdac extended core on arm fails with below error: sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_writel': >> sound/hda/ext/hdac_ext_bus.c:29:2: error: implicit declaration of >> function +'writel' [-Werror=implicit-function-declaration] writel(value, addr); ^ sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_readl': >> sound/hda/ext/hdac_ext_bus.c:34:2: error: implicit declaration of >> function +'readl' [-Werror=implicit-function-declaration] return readl(addr); This is fixed by explicitly including io.h Fixes: 99463b3a3994 - ('ALSA: hda: provide default bus io ops extended hdac') Reported-by: kbuild test robot <lkp(a)intel.com> Suggested-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/hda/ext/hdac_ext_bus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 4449d1a99089..2433f7c81472 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/slab.h> +#include <linux/io.h> #include <sound/hdaudio_ext.h> MODULE_DESCRIPTION("HDA extended core"); -- 2.4.3

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix inverted internal mic on Lenovo G50-80
by David Henningsson 13 Oct '15

13 Oct '15

Add the appropriate quirk to indicate the Lenovo G50-80 has a stereo mic input where one channel has reverse polarity. Alsa-info available at: https://launchpadlibrarian.net/220846272/AlsaInfo.txt Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1504778 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ca03c40..2f0ec7c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -819,6 +819,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), + SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), -- 1.9.1

2 1

[alsa-devel] How is /proc/asound/cardX/eld#Y.Z generated?
by VDR User 12 Oct '15

12 Oct '15

Hi. I was wondering if someone can tell me how exactly the eld files are generated. Additionally, if they can be overridden somehow. The reason is if I start x while my tv is turned on, the eld reports only stereo audio capability for hdmi even though the tv is plugged into my surround receiver. I don't care what the tv reports, my surround receiver is 7.1 so I always want the eld to reflect my -actual- audio channel setup. Thanks.

2 2

[alsa-devel] [PATCH] ASoC: rt298: set register non-volatile by default
by Bard Liao 12 Oct '15

12 Oct '15

It is not necessary to set registers volatile. So, return false for default case of rt298_volatile_register. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 5a9aadb..56e5e1a 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -128,7 +128,7 @@ static bool rt298_volatile_register(struct device *dev, unsigned int reg) case VERB_CMD(AC_VERB_GET_EAPD_BTLENABLE, RT298_HP_OUT, 0): return true; default: - return true; + return false; } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: rt298: correct index default value
by Bard Liao 12 Oct '15

12 Oct '15

Some of the default value on rt298_index_def are incorrect. Change them to the correct value. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 83bbc62..5a9aadb 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -49,24 +49,24 @@ struct rt298_priv { }; static const struct reg_default rt298_index_def[] = { - { 0x01, 0xaaaa }, - { 0x02, 0x8aaa }, + { 0x01, 0xa5a8 }, + { 0x02, 0x8e95 }, { 0x03, 0x0002 }, - { 0x04, 0xaf01 }, - { 0x08, 0x000d }, - { 0x09, 0xd810 }, - { 0x0a, 0x0120 }, + { 0x04, 0xaf67 }, + { 0x08, 0x200f }, + { 0x09, 0xd010 }, + { 0x0a, 0x0100 }, { 0x0b, 0x0000 }, { 0x0d, 0x2800 }, - { 0x0f, 0x0000 }, - { 0x19, 0x0a17 }, + { 0x0f, 0x0022 }, + { 0x19, 0x0217 }, { 0x20, 0x0020 }, { 0x33, 0x0208 }, { 0x46, 0x0300 }, - { 0x49, 0x0004 }, - { 0x4f, 0x50e9 }, - { 0x50, 0x2000 }, - { 0x63, 0x2902 }, + { 0x49, 0x4004 }, + { 0x4f, 0x50c9 }, + { 0x50, 0x3000 }, + { 0x63, 0x1b02 }, { 0x67, 0x1111 }, { 0x68, 0x1016 }, { 0x69, 0x273f }, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v3] ASoC: rt5645: Add the HW EQ for the customized speaker output of Google Celes
by Oder Chiou 12 Oct '15

12 Oct '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5645.h | 1 + sound/soc/codecs/rt5645.c | 86 ++++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 86 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index a5cf615..42f0842 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -23,6 +23,7 @@ struct rt5645_platform_data { unsigned int jd_mode; /* Invert JD when jack insert */ bool jd_invert; + bool hweq; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c4fe6b..d9532bb 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -22,6 +22,7 @@ #include <linux/acpi.h> #include <linux/dmi.h> #include <linux/regulator/consumer.h> +#include <linux/firmware.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -224,6 +225,16 @@ static const struct reg_default rt5645_reg[] = { { 0xff, 0x6308 }, }; +struct rt5645_eq_param_s { + unsigned short reg; + unsigned short val; +}; + +struct rt5645_hweq_s { + unsigned short num; + struct rt5645_eq_param_s *param; +}; + static const char *const rt5645_supply_names[] = { "avdd", "cpvdd", @@ -240,6 +251,7 @@ struct rt5645_priv { struct snd_soc_jack *btn_jack; struct delayed_work jack_detect_work; struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; + struct rt5645_hweq_s hweq; int codec_type; int sysclk; @@ -631,6 +643,59 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static int rt5645_read_hweq_param(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + const struct firmware *fw; + int i; + + if (!request_firmware(&fw, "rt5645_hweq.bin", codec->dev)) { + rt5645->hweq.num = fw->size / sizeof(struct rt5645_eq_param_s); + rt5645->hweq.param = devm_kmemdup(codec->dev, fw->data, + fw->size, GFP_KERNEL); + if (rt5645->hweq.param) { + for (i = 0; i < rt5645->hweq.num; i++) { + rt5645->hweq.param[i].reg = + be16_to_cpu(rt5645->hweq.param[i].reg); + rt5645->hweq.param[i].val = + be16_to_cpu(rt5645->hweq.param[i].val); + } + } + release_firmware(fw); + } + + return 0; +} + +static bool rt5645_validate_hweq(unsigned short reg) +{ + if ((reg >= 0x1a4 && reg <= 0x1cd) | (reg >= 0x1e5 && reg <= 0x1f8) | + (reg == RT5645_EQ_CTRL2)) + return true; + + return false; +} + +static int rt5645_enable_hweq(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int i; + + if (!rt5645->hweq.param) + rt5645_read_hweq_param(codec); + + if (rt5645->hweq.param) { + for (i = 0; i < rt5645->hweq.num; i++) { + if (rt5645_validate_hweq(rt5645->hweq.param[i].reg)) + regmap_write(rt5645->regmap, + rt5645->hweq.param[i].reg, + rt5645->hweq.param[i].val); + } + } + + return 0; +} + /** * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters * @codec: SoC audio codec device. @@ -1532,9 +1597,12 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: + if (rt5645->pdata.hweq) + rt5645_enable_hweq(codec); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, @@ -1543,6 +1611,8 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + if (rt5645->pdata.hweq) + snd_soc_write(codec, RT5645_EQ_CTRL2, 0); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, 0); @@ -3205,6 +3275,20 @@ static int strago_quirk_cb(const struct dmi_system_id *id) return 1; } +static struct rt5645_platform_data celes_platform_data = { + .dmic1_data_pin = RT5645_DMIC1_DISABLE, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, + .hweq = true, +}; + +static int celes_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &celes_platform_data; + + return 1; +} + static const struct dmi_system_id dmi_platform_intel_braswell[] = { { .ident = "Intel Strago", @@ -3215,7 +3299,7 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { }, { .ident = "Google Celes", - .callback = strago_quirk_cb, + .callback = celes_quirk_cb, .matches = { DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), }, -- 1.8.1.1.439.g50a6b54

2 1

Re: [alsa-devel] [PATCH] ASoC: Intel: Skylake - Add Skylake RT286 I2S machine driver
by Koul, Vinod 12 Oct '15

12 Oct '15

On Sat, 2015-10-10 at 16:42 +0800, kbuild test robot wrote: > Hi Omair, > > [auto build test ERROR on next-20151009 -- if it's inappropriate base, please > ignore] > > config: arm-allmodconfig (attached as .config) > reproduce: That's nice, thanks. We should add a depends on X86 for this to not compile on non X86 code Will send updated patch -- ~Vinod

3 3

[alsa-devel] Applied "ASoc: sh: Fit typo in Kconfig" to the asoc tree
by Mark Brown 12 Oct '15

12 Oct '15

The patch ASoc: sh: Fit typo in Kconfig has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d795eb3f5c10d36ed51ca7cd5ce411677f6f7aab Mon Sep 17 00:00:00 2001 From: Masanari Iida <standby24x7(a)gmail.com> Date: Sat, 10 Oct 2015 14:17:58 +0900 Subject: [PATCH] ASoc: sh: Fit typo in Kconfig s/SUR/SRU/g Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 07114b0..de668b2 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -40,7 +40,7 @@ config SND_SOC_RCAR select SND_SIMPLE_CARD select REGMAP_MMIO help - This option enables R-Car SUR/SCU/SSIU/SSI sound support + This option enables R-Car SRU/SCU/SSIU/SSI sound support config SND_SOC_RSRC_CARD tristate "Renesas Sampling Rate Convert Sound Card" -- 2.5.0

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Fix max packet size calculation for USB audio
by Ricard Wanderlof 12 Oct '15

12 Oct '15

The addition of 0xffff ("almost 1" in Q16.16 format), resulting in the rounding up of the resulting value, was in the wrong place, as it is the resulting packet size that needs to be rounded up to the nearest integer (e.g. if the resulting packet size is calculated as 55.125 bytes, we need a packet of 56 bytes), rather than the rate value (ep->freqmax). Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- Since this clearly is a sensitive part of the code, here follows a lengthy rationale with examples to illustrate the problem. The code as it stands has the following expression on line 613 to calculate the maximum isochronous packet size: maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) >> (16 - ep->datainterval); Here, ep->freqmax is the maximum assumed sample frequency, calculated from the nominal sample frequency plus 25%. It is ultimately derived from ep->freqn, which is in the units of samples per packet, from get_usb_full_speed_rate() or usb_high_speed_rate(), as applicable, in Q16.16 format. The expression essentially adds the Q16.16 equivalent of 0.999... (i.e. the largest number less than one) to the sample rate, in order to get a rate whose integer part is rounded up from the fractional value. The multiplication with (frame_bits >> 3) yields the number of bytes in a packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back to an integer, taking into consideration the bDataInterval field of the endpoint descriptor, which describes how often isochronous packets are transmitted relative to the (micro)frame rate (125us or 1ms, for USB high speed and full speed, respectively). For this discussion we will assume a bDataInterval of 0, so the second line of the expression just converts the Q16.16 value to an integer. In order to illustrate the problem, we will set frame_bits 64, which corresponds to a frame size of 8 bytes. The problem here is that the addition of 0xffff is in the wrong place. We don't want the rounding up to take place on the frequency, but on the resulting packet size. Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is 786432 (see usb_high_speed_rate()). Add 25% (line 612) and we get 983040. The calculated maxsize is then ((983040 + 65535) * 8) >> 16 = 127 . However, if we do the number of bytes calculation in a less obscure way it's more apparent what the true corresponding packet size is: we get (96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612, and the 8000 is the number of isochronous packets per second on a high speed USB connection (125 us microframe interval). Rephrasing the maxsize expression to: maxsize = (ep->freqmax * (frame_bits >> 3) + 0xffff) >> (16 - ep->datainterval); we instead get (983040 * 8 + 65535) >> 16 = 120 which is the correct value. We can also do the calculation with a non-integer sample rate which is when rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn = 361267, and resulting ep->freqmax 361267 * 1.25 = 451583 (rounded down). Original maxsize = ((451583 + 65535) * 8) << 16 = 63 (63.124.. rounded down) True maxsize = (44100 * 1.25 / 8000) * 8 = 55.125, i.e. 56 bytes required New maxsize = (451583 * 8 + 65535) >> 16 = 56 This is also corroborated by the wMaxPacketSize check on line 616. Assume that wMaxPacketSize = 108, with ep->maxpacksize then having the same value. As 108 < 127, we get maxsize = 108. ep->freqmax is then recalculated to (108 / 8) << 16 = 884736 . Putting that rate into the original maxsize calculation yields a maxsize of ((884736 + 65535) * 8) >> 16 = 115 (with decimals 115.99988). Clearly, we should get back the 108 here, which we would with the new expression: (884736 * 8 + 65535) >> 16 = 108 . As it currently stands, the error is tolerable because it only results in maxsize being a bit too big which wastes a couple of bytes, either as a result of the first maxsize calculation, or because the resulting calculation will hit the wMaxPacketSize value before the packet is too big, resulting in fixing the size to wMaxPacketSize even though the packet is actually not too long. However, I'm working on a quirk for a device which stuffs an extra four (non sample) bytes into each isochronous packet, and that means that the maxsize calculation must take this into account. During this work it became obvious that there was something wrong with the maxsize calculation, as the resulting packet size when wMaxPacketSize was hit did not correspond to the packet size which triggered the if statement to start with. Since this seems to me an obvious (after looking at it for a bit...) bug I'm submitting this as a separate patch rather than part of the upcoming patch series. sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e6f7189..be8a972 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -610,7 +610,7 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, /* assume max. frequency is 25% higher than nominal */ ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) + maxsize = (ep->freqmax * (frame_bits >> 3) + 0xffff) >> (16 - ep->datainterval); /* but wMaxPacketSize might reduce this */ if (ep->maxpacksize && ep->maxpacksize < maxsize) { -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 2

[alsa-devel] [PATCH alsa-lib] fix _GNU_SOURCE handling & header inclusion
by Mike Frysinger 12 Oct '15

12 Oct '15

From: Mike Frysinger <vapier(a)chromium.org> The configure script blindly adds -D_GNU_SOURCE to all build settings, even on non-GNU systems. This isn't too much of a big deal (even if it uses the wrong variable -- CFLAGS instead of CPPFLAGS), except that the alsa-lib source itself determines whether to use GNU features when this is defined (such as versionsort). So when we build on non-glibc systems, we get build failures like: src/ucm/parser.c:1268:18: error: 'versionsort' undeclared (first use in this function) #define SORTFUNC versionsort ^ src/ucm/parser.c:1272:54: note: in expansion of macro 'SORTFUNC' err = scandir(filename, &namelist, filename_filter, SORTFUNC); ^ The correct way to add these flags is to use the autoconf helper AC_USE_SYSTEM_EXTENSIONS. Unfortunately, that triggers some more bugs in the alsa build. This macro adds defines to config.h and not directly to CPPFLAGS, so it relies on files correctly including config.h before anything else. A number of alsa files do not do this leading to build failures. The fix there is to shuffle the includes around so that the local ones come first. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- configure.ac | 4 +--- src/conf.c | 2 +- src/timer/timer.c | 9 ++------- src/timer/timer_hw.c | 6 ------ src/timer/timer_local.h | 5 ++--- src/timer/timer_query.c | 6 ------ src/timer/timer_query_hw.c | 6 ------ src/ucm/ucm_local.h | 2 +- 8 files changed, 7 insertions(+), 33 deletions(-) diff --git a/configure.ac b/configure.ac index a14e52d..9cb8614 100644 --- a/configure.ac +++ b/configure.ac @@ -27,11 +27,9 @@ AC_PREFIX_DEFAULT(/usr) dnl Checks for programs. -CFLAGS="$CFLAGS -D_GNU_SOURCE" - - AC_PROG_CC AC_PROG_CPP +AC_USE_SYSTEM_EXTENSIONS AC_PROG_INSTALL AC_PROG_LN_S AC_DISABLE_STATIC diff --git a/src/conf.c b/src/conf.c index c6a83ee..91fca25 100644 --- a/src/conf.c +++ b/src/conf.c @@ -414,12 +414,12 @@ beginning:</P> */ +#include "local.h" #include <stdarg.h> #include <limits.h> #include <sys/stat.h> #include <dirent.h> #include <locale.h> -#include "local.h" #ifdef HAVE_LIBPTHREAD #include <pthread.h> #endif diff --git a/src/timer/timer.c b/src/timer/timer.c index b71a9f8..a25e4f7 100644 --- a/src/timer/timer.c +++ b/src/timer/timer.c @@ -67,15 +67,10 @@ This example shows opening a timer device and reading of timer events. * \anchor example_test_timer */ -#include <stdio.h> -#include <stdlib.h> -#include <unistd.h> -#include <string.h> -#include <fcntl.h> -#include <signal.h> -#include <sys/ioctl.h> #include "timer_local.h" +#include <signal.h> + static int snd_timer_open_conf(snd_timer_t **timer, const char *name, snd_config_t *timer_root, snd_config_t *timer_conf, int mode) diff --git a/src/timer/timer_hw.c b/src/timer/timer_hw.c index e833fc8..e61b994 100644 --- a/src/timer/timer_hw.c +++ b/src/timer/timer_hw.c @@ -19,12 +19,6 @@ * */ -#include <stdio.h> -#include <stdlib.h> -#include <unistd.h> -#include <string.h> -#include <fcntl.h> -#include <sys/ioctl.h> #include "timer_local.h" #ifndef PIC diff --git a/src/timer/timer_local.h b/src/timer/timer_local.h index 8040b05..eef3b06 100644 --- a/src/timer/timer_local.h +++ b/src/timer/timer_local.h @@ -19,10 +19,9 @@ * */ -#include <stdio.h> -#include <stdlib.h> -#include <limits.h> #include "local.h" +#include <limits.h> +#include <sys/ioctl.h> #ifndef DOC_HIDDEN typedef struct { diff --git a/src/timer/timer_query.c b/src/timer/timer_query.c index 50b098a..93d2455 100644 --- a/src/timer/timer_query.c +++ b/src/timer/timer_query.c @@ -26,12 +26,6 @@ * */ -#include <stdio.h> -#include <stdlib.h> -#include <unistd.h> -#include <string.h> -#include <fcntl.h> -#include <sys/ioctl.h> #include "timer_local.h" static int snd_timer_query_open_conf(snd_timer_query_t **timer, diff --git a/src/timer/timer_query_hw.c b/src/timer/timer_query_hw.c index 289ca52..d0c4391 100644 --- a/src/timer/timer_query_hw.c +++ b/src/timer/timer_query_hw.c @@ -19,12 +19,6 @@ * */ -#include <stdio.h> -#include <stdlib.h> -#include <unistd.h> -#include <string.h> -#include <fcntl.h> -#include <sys/ioctl.h> #include "timer_local.h" #ifndef PIC diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index c1655c7..3a5d2c2 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -36,8 +36,8 @@ #define UC_MGR_DEBUG #endif -#include <pthread.h> #include "local.h" +#include <pthread.h> #include "use-case.h" #define MAX_FILE 256 -- 2.5.2

2 1

[alsa-devel] [RFC] i2c: Revert back to old device naming for ACPI enumerated I2C slaves
by Jarkko Nikula 12 Oct '15

12 Oct '15

Commit 70762abb9f89 ("i2c: Use stable dev_name for ACPI enumerated I2C slaves") broke the lm-sensors which relies on I2C hwmon slave devices under /sys/bus/i2c/devices/ to be named as "x-00yz". However if those hwmon devices are ACPI 5 enumerated their name became "i2c-INTABCD:ij" and sysfs code in lm-sensors does not find them anymore: lib/sysfs.c:665: if ((!subsys || !strcmp(subsys, "i2c")) && sscanf(dev_name, "%hd-%x", &entry.chip.bus.nr, &entry.chip.addr) == 2) { This patch fixes this by reverting back the old device naming in i2c-core but at the same avoids regression to ALSA SoC drivers that depend on stable device binding. Reverted I2C slave device naming is handled in ALSA SoC core by using the ACPI name as a component name if device is ACPI probed. This keeps the component binding in ALSA SoC stable and requires only minimal changes to affected machine drivers. Fixes: 70762abb9f89 i2c: Use stable dev_name for ACPI enumerated I2C slaves Reported-by: Dustin Byford <dustin(a)cumulusnetworks.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- This is on top of 4.2.0-rc8. I didn't check exact kernel versions where this still applies but I know we would need a few different versions for older stable versions. Mainly because of added and moved drivers in sound/soc/intel/. This is for discussion so I didn't cc stable(a)vger.kernel.org yet. I was thinking would it work if we'd keep the stable name but have an another symlink in /sys/bus/i2c/devices/ that uses "x-00yz" name. However this feels ill-use of devices directory and probably causes more troubles elsewhere. I don't know how common ACPI enumerated I2C hwmon devices are but I guess they exists since Dustin notices this issue and wrote "With that change, /sys/bus/i2c/devices/i2c-0-004a, for example, became /sys/bus/i2c/devices/i2c-PNPXXXX:xx". It could be possible there is use for the new device naming. I found a comment from Documentation/hid/hid-sensor.txt introduced by commit b2eafd7282fd ("HID: sensor: Update document for custom sensor") that seems to indicate current i2c-INTABCD:xy naming may be used from userspace too but I didn't find examples of that. --- drivers/i2c/i2c-core.c | 21 ++++----------------- sound/soc/intel/boards/broadwell.c | 6 +++--- sound/soc/intel/boards/byt-max98090.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5645.c | 4 ++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +++--- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/soc-core.c | 10 ++++++++++ 10 files changed, 28 insertions(+), 31 deletions(-) diff --git a/drivers/i2c/i2c-core.c b/drivers/i2c/i2c-core.c index c83e4d13cfc5..e9c227b9dc92 100644 --- a/drivers/i2c/i2c-core.c +++ b/drivers/i2c/i2c-core.c @@ -913,22 +913,6 @@ void i2c_unlock_adapter(struct i2c_adapter *adapter) } EXPORT_SYMBOL_GPL(i2c_unlock_adapter); -static void i2c_dev_set_name(struct i2c_adapter *adap, - struct i2c_client *client) -{ - struct acpi_device *adev = ACPI_COMPANION(&client->dev); - - if (adev) { - dev_set_name(&client->dev, "i2c-%s", acpi_dev_name(adev)); - return; - } - - /* For 10-bit clients, add an arbitrary offset to avoid collisions */ - dev_set_name(&client->dev, "%d-%04x", i2c_adapter_id(adap), - client->addr | ((client->flags & I2C_CLIENT_TEN) - ? 0xa000 : 0)); -} - /** * i2c_new_device - instantiate an i2c device * @adap: the adapter managing the device @@ -987,7 +971,10 @@ i2c_new_device(struct i2c_adapter *adap, struct i2c_board_info const *info) client->dev.of_node = info->of_node; client->dev.fwnode = info->fwnode; - i2c_dev_set_name(adap, client); + /* For 10-bit clients, add an arbitrary offset to avoid collisions */ + dev_set_name(&client->dev, "%d-%04x", i2c_adapter_id(adap), + client->addr | ((client->flags & I2C_CLIENT_TEN) + ? 0xa000 : 0)); status = device_register(&client->dev); if (status) goto out_err; diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 8bafaf6ceab1..3abcf0d7682e 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -205,7 +205,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, - .codec_name = "i2c-INT343A:00", + .codec_name = "INT343A:00", .codec_dai_name = "rt286-aif1", .init = broadwell_rt286_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | @@ -223,7 +223,7 @@ static int broadwell_suspend(struct snd_soc_card *card){ struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + if (!strcmp(codec->component.name, "INT343A:00")) { dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); rt286_mic_detect(codec, NULL); break; @@ -236,7 +236,7 @@ static int broadwell_resume(struct snd_soc_card *card){ struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-INT343A:00")) { + if (!strcmp(codec->component.name, "INT343A:00")) { dev_dbg(codec->dev, "enabling jack detect for resume.\n"); rt286_mic_detect(codec, &broadwell_headset); break; diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c index 7ab8cc9fbfd5..ae336b8522a8 100644 --- a/sound/soc/intel/boards/byt-max98090.c +++ b/sound/soc/intel/boards/byt-max98090.c @@ -116,7 +116,7 @@ static struct snd_soc_dai_link byt_max98090_dais[] = { .stream_name = "Audio", .cpu_dai_name = "baytrail-pcm-audio", .codec_dai_name = "HiFi", - .codec_name = "i2c-193C9890:00", + .codec_name = "193C9890:00", .platform_name = "baytrail-pcm-audio", .init = byt_max98090_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | diff --git a/sound/soc/intel/boards/byt-rt5640.c b/sound/soc/intel/boards/byt-rt5640.c index ae89b9b966d9..275b056c495c 100644 --- a/sound/soc/intel/boards/byt-rt5640.c +++ b/sound/soc/intel/boards/byt-rt5640.c @@ -186,7 +186,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { .stream_name = "Audio", .cpu_dai_name = "baytrail-pcm-audio", .codec_dai_name = "rt5640-aif1", - .codec_name = "i2c-10EC5640:00", + .codec_name = "10EC5640:00", .platform_name = "baytrail-pcm-audio", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 7f55d59024a8..973f07548b08 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -171,7 +171,7 @@ static struct snd_soc_dai_link byt_dailink[] = { .platform_name = "sst-mfld-platform", .no_pcm = 1, .codec_dai_name = "rt5640-aif1", - .codec_name = "i2c-10EC5640:00", + .codec_name = "10EC5640:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .be_hw_params_fixup = byt_codec_fixup, diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 70f832114a5a..1a5348771e03 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -232,7 +232,7 @@ static struct snd_soc_ops cht_be_ssp2_ops = { static struct snd_soc_aux_dev cht_max98090_headset_dev = { .name = "Headset Chip", .init = cht_max98090_headset_init, - .codec_name = "i2c-104C227E:00", + .codec_name = "104C227E:00", }; static struct snd_soc_dai_link cht_dailink[] = { @@ -265,7 +265,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .platform_name = "sst-mfld-platform", .no_pcm = 1, .codec_dai_name = "HiFi", - .codec_name = "i2c-193C9890:00", + .codec_name = "193C9890:00", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index bdcaf467842a..c05841264dc7 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -290,7 +290,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .platform_name = "sst-mfld-platform", .no_pcm = 1, .codec_dai_name = "rt5645-aif1", - .codec_name = "i2c-10EC5645:00", + .codec_name = "10EC5645:00", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, @@ -365,7 +365,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) } } card->dev = &pdev->dev; - sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); + sprintf(codec_name, "%s:00", drv->acpi_card->codec_id); /* set correct codec name */ strcpy((char *)card->dai_link[2].codec_name, codec_name); snd_soc_card_set_drvdata(card, drv); diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2c9cc5be439e..632926d0c595 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -280,7 +280,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .no_pcm = 1, .nonatomic = true, .codec_dai_name = "rt5670-aif1", - .codec_name = "i2c-10EC5670:00", + .codec_name = "10EC5670:00", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, @@ -296,7 +296,7 @@ static int cht_suspend_pre(struct snd_soc_card *card) struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + if (!strcmp(codec->component.name, "10EC5670:00")) { dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); rt5670_jack_suspend(codec); break; @@ -310,7 +310,7 @@ static int cht_resume_post(struct snd_soc_card *card) struct snd_soc_codec *codec; list_for_each_entry(codec, &card->codec_dev_list, card_list) { - if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + if (!strcmp(codec->component.name, "10EC5670:00")) { dev_dbg(codec->dev, "enabling jack detect for resume.\n"); rt5670_jack_resume(codec); break; diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 22558572cb9c..6fd6491e368d 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -160,7 +160,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, - .codec_name = "i2c-INT33CA:00", + .codec_name = "INT33CA:00", .codec_dai_name = "rt5640-aif1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0e1e69c7abd5..811d63e31a7a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -34,6 +34,7 @@ #include <linux/ctype.h> #include <linux/slab.h> #include <linux/of.h> +#include <linux/acpi.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> @@ -2476,10 +2477,18 @@ static char *fmt_single_name(struct device *dev, int *id) { char *found, name[NAME_SIZE]; int id1, id2; + struct acpi_device *adev; if (dev_name(dev) == NULL) return NULL; + if (ACPI_HANDLE(dev) && + !acpi_bus_get_device(ACPI_HANDLE(dev), &adev)) { + /* use ACPI name as a component name for ACPI probed devices */ + strlcpy(name, acpi_dev_name(adev), NAME_SIZE); + goto out; + } + strlcpy(name, dev_name(dev), NAME_SIZE); /* are we a "%s.%d" name (platform and SPI components) */ @@ -2508,6 +2517,7 @@ static char *fmt_single_name(struct device *dev, int *id) *id = 0; } +out: return kstrdup(name, GFP_KERNEL); } -- 2.5.0

6 16

[alsa-devel] [PATCH 0/6 v3] ALSA: firewire-digi00x: support MIDI functionality
by Takashi Sakamoto 12 Oct '15

12 Oct '15

Hi, This commit updates my former post (patch 13, 15, 16): [alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096739.html Digi 00x driver may be newly available in Linux 4.4. This patchset adds ALSA MIDI ports to support physical MIDI ports and physical controls. I tested physical MIDI ports, while never tested physical controls because I'm not an owner of 'console' models. The functionality for physical controls is written according to my assumption. When I investigated Digi 002 rack, Windows driver registers a certain address to the device, and adds virtual MIDI ports to the system. These ports are named as 'control'. When MIDI application sends MIDI messages to the port, asynchronous transactions start on IEEE 1394 bus. On the other hand, I have no way to confirm asynchronous transaction in opposite direction because the rack model doesn't have the functionality. In this time, I apply the same format of outgoing asynchronous transaction to incoming asynchronous transaction. Takashi Sakamoto (6): firewire-digi00x: use in-kernel representation for the type of 8 bits firewire-digi00x: handle MIDI messages in isochronous packets firewire-digi00x: add support for MIDI ports corresponding to isochronous packet streaming firewire-digi00x: add support of asynchronous transaction for incoming MIDI messages from physical controls firewire-digi00x: add support of asynchronous transaction for outgoing MIDI messages to physical controls firewire-digi00x: add support for MIDI ports for physical controls sound/firewire/digi00x/Makefile | 2 +- sound/firewire/digi00x/amdtp-dot.c | 150 +++++++++++++++++++++---- sound/firewire/digi00x/digi00x-midi.c | 160 +++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-stream.c | 6 +- sound/firewire/digi00x/digi00x-transaction.c | 70 ++++++++++-- sound/firewire/digi00x/digi00x.c | 4 + sound/firewire/digi00x/digi00x.h | 15 ++- 7 files changed, 376 insertions(+), 31 deletions(-) create mode 100644 sound/firewire/digi00x/digi00x-midi.c -- 2.1.4

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[alsa-devel] [RESEND PATCH 1/1] Allow any MIDI endpoint to drive use of interrupt transfer on newer Roland devices
by maillist＠superlative.org 11 Oct '15

11 Oct '15

From: "Keith A. Milner" <maillist(a)superlative.org> This patch enables interrupt transfer mode for MIDI ports on newer Boss/Roland devices such as the GT-100/001 which support interrupt transfer on both IN and OUT MIDI endpoints. Previously this wasn't being enabled for these devices as the code was specifically looking for the scenario where the IN endpoint supported interrupt transfer and the OUT endpoint was bulk transfer. Newer devices support interrupt transfer for both endpoints. This has been tested on Boss devices GT-001, BR-80 and JS-8 and Roland VS-20. It would benefit from some regresison testing with other devices if possible. Signed-off-by: Keith A. Milner <maillist(a)superlative.org> diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b901f46..d50c649 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1860,9 +1860,14 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi) hostif = &intf->altsetting[1]; intfd = get_iface_desc(hostif); + /* If either or both of the endpoints support interrupt transfer, + * then use the alternate setting + */ if (intfd->bNumEndpoints != 2 || - (get_endpoint(hostif, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK || - (get_endpoint(hostif, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT) + !((get_endpoint(hostif, 0)->bmAttributes + & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT || + (get_endpoint(hostif, 1)->bmAttributes + & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT)) return; snd_printdd(KERN_INFO "switching to altsetting %d with int ep\n", -- 1.9.1

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[alsa-devel] [PATCH v3] ALSA: usb-audio: Fix max packet size calculation for USB audio
by Ricard Wanderlof 11 Oct '15

11 Oct '15

Rounding must take place before multiplication with the frame size, since each packet contains a whole number of frames. We must also properly consider the data interval, as a larger data interval will result in larger packets, which, depending on the sampling frequency, can result in packet sizes that are less than integral multiples of the packet size for a lower data interval. Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V2: Added tested device to commit message. No other changes. V3: Updated after input from Clemens. Since the expression is now getting rather convoluted, added a description in the comments in the code. New commit message to reflect the new version. Updated rationale below. The code as it stands has the following expression on line 613 to calculate the maximum isochronous packet size: maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) >> (16 - ep->datainterval); Here, ep->freqmax is the maximum assumed sample frequency, calculated from the nominal sample frequency plus 25%. It is ultimately derived from ep->freqn, which is in the units of frames per packet, from get_usb_full_speed_rate() or usb_high_speed_rate(), as applicable, in Q16.16 format. The expression essentially adds the Q16.16 equivalent of 0.999... (i.e. the largest number less than one) to the sample rate, in order to get a rate whose integer part is rounded up from the fractional value. The multiplication with (frame_bits >> 3) yields the number of bytes in a packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back to an integer, taking into consideration the bDataInterval field of the endpoint descriptor (which describes how often isochronous packets are transmitted relative to the (micro)frame rate (125us or 1ms, for USB high speed and full speed, respectively)). For this discussion we will initially assume a bDataInterval of 0, so the second line of the expression just converts the Q16.16 value to an integer. In order to illustrate the problem, we will set frame_bits 64, which corresponds to a frame size of 8 bytes. The problem here is twofold. First, the rounding operation consists of the addition of 0x0.ffff and subsequent conversion to integer, but as the expression stands, the conversion to integer is done after multiplication with the frame size, rather than before. This results in the resulting maxsize becoming too large. Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is 0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000. The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 . However, if we do the number of bytes calculation in a less obscure way it's more apparent what the true corresponding packet size is: we get ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612, and the 8000 is the number of isochronous packets per second on a high speed USB connection (125 us microframe interval). This is fixed by performing the complete rounding operation prior to multiplication with the frame rate. The second problem is that when considering the ep->datainterval, this must be done before rounding, in order to take the advantage of the fact that the number of bytes per packet is not an integer, the resulting rounded-up integer is not necessarily a factor of two when the data interval is increased by the same factor. For instance, assuming a freqency of 41 kHz, the resulting bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or 0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0), this 6 frames per packet is required, whereas with a data interval of 2 we need 10.25, i.e. 11 frames needed. Rephrasing the maxsize expression to: maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * (frame_bits >> 3); for the above 96 kHz example we instead get ((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value. We can also do the calculation with a non-integer sample rate which is when rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn = 0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)): Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down) True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56 New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56 This is also corroborated by the wMaxPacketSize check on line 616. Assume that wMaxPacketSize = 104, with ep->maxpacksize then having the same value. As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to (104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111 (with decimals 111.99988). Clearly, we should get back the 104 here, which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 . As it currently stands, the error is tolerable because it only results in maxsize being a bit too big which wastes a couple of bytes, either as a result of the first maxsize calculation, or because the resulting calculation will hit the wMaxPacketSize value before the packet is too big, resulting in fixing the size to wMaxPacketSize even though the packet is actually not too long. However, I'm working on a quirk for a device which stuffs an extra four (non sample) bytes into each isochronous packet, and that means that the maxsize calculation must take this into account. During this work it became obvious that there was something wrong with the maxsize calculation, as the resulting packet size when wMaxPacketSize was hit did not correspond to the packet size which triggered the if statement to start with. Since this seems to me an obvious (after looking at it for a bit...) bug I'm submitting this as a separate patch rather than part of an upcoming patch series. sound/usb/endpoint.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e6f7189..a77d9c8 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -610,8 +610,23 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, /* assume max. frequency is 25% higher than nominal */ ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - ep->datainterval); + /* Round up freqmax to nearest integer in order to calculate maximum + * packet size, which must represent a whole number of frames. + * This is accomplished by adding 0x0.ffff before converting the + * Q16.16 format into integer. + * In order to accurately calculate the maximum packet size when + * the data interval is more than 1 (i.e. ep->datainterval > 0), + * multiply by the data interval prior to rounding. For instance, + * a freqmax of 41 kHz will result in a max packet size of 6 (5.125) + * frames with a data interval of 1, but 11 (10.25) frames with a + * data interval of 2. + * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the + * maximum datainterval value of 3, at USB full speed, higher for + * USB high speed, noting that ep->freqmax is in units of + * frames per packet in Q16.16 format.) + */ + maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * + (frame_bits >> 3); /* but wMaxPacketSize might reduce this */ if (ep->maxpacksize && ep->maxpacksize < maxsize) { /* whatever fits into a max. size packet */ -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH] ALSA: firewire-lib: continue packet processing at detecting wrong CIP headers
by Takashi Sakamoto 11 Oct '15

11 Oct '15

In firewire-lib, isochronous packet streaming is stopped when detecting wrong value for FMT field of CIP headers. Although this is appropriate to IEC 61883-1 and 6, some BeBoB based devices with vendors' customization use invalid value to FMT field of CIP headers in the beginning of streaming. $ journalctl snd-bebob fw1.0: Detect unexpected protocol: 01000000 8000ffff I got this log with M-Audio FireWire 1814. In this line, the value of FMT field is 0x00, while it should be 0x10 in usual AMDTP. Except for the beginning, these devices continue to transfer packets with valid value for FMT field, except for the beginning. Therefore, in this case, firewire-lib should continue to process packets. The former implementation of firewire-lib performs it. This commit loosens the handling of wrong value, to continue packet processing in the case. Fixes: 414ba022a528('ALSA: firewire-lib: add support arbitrary value for fmt/fdf fields in CIP header') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index fa10b58..ed29026 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -468,10 +468,12 @@ static int handle_in_packet(struct amdtp_stream *s, /* Check valid protocol or not. */ fmt = (cip_header[1] & CIP_FMT_MASK) >> CIP_FMT_SHIFT; if (fmt != s->fmt) { - dev_err(&s->unit->device, - "Detect unexpected protocol: %08x %08x\n", - cip_header[0], cip_header[1]); - return -EIO; + dev_info_ratelimited(&s->unit->device, + "Detect unexpected protocol: %08x %08x\n", + cip_header[0], cip_header[1]); + *data_blocks = 0; + pcm_frames = 0; + goto end; } /* Calculate data blocks */ -- 2.1.4

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[alsa-devel] [PATCH] ALSA: bebob: constify snd_bebob_rate_spec structures
by Julia Lawall 11 Oct '15

11 Oct '15

The snd_bebob_rate_spec structure type contains only function pointers. In all instances of this structure type, the functions are provided statically and are never changed subsequently. Add a const annotation on all initializations and uses. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/firewire/bebob/bebob.c | 2 +- sound/firewire/bebob/bebob.h | 2 +- sound/firewire/bebob/bebob_focusrite.c | 4 ++-- sound/firewire/bebob/bebob_maudio.c | 4 ++-- sound/firewire/bebob/bebob_pcm.c | 2 +- sound/firewire/bebob/bebob_proc.c | 2 +- sound/firewire/bebob/bebob_stream.c | 2 +- sound/firewire/bebob/bebob_terratec.c | 2 +- sound/firewire/bebob/bebob_yamaha.c | 2 +- 9 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/firewire/bebob/bebob_yamaha.c b/sound/firewire/bebob/bebob_yamaha.c index 5810170..2a9ffdb 100644 --- a/sound/firewire/bebob/bebob_yamaha.c +++ b/sound/firewire/bebob/bebob_yamaha.c @@ -51,7 +51,7 @@ static struct snd_bebob_clock_spec clock_spec = { .types = clk_src_types, .get = &clk_src_get, }; -static struct snd_bebob_rate_spec rate_spec = { +static const struct snd_bebob_rate_spec rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate, }; diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c index a1a3949..e62923e 100644 --- a/sound/firewire/bebob/bebob_focusrite.c +++ b/sound/firewire/bebob/bebob_focusrite.c @@ -260,7 +260,7 @@ saffire_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size) return err; } -static struct snd_bebob_rate_spec saffirepro_both_rate_spec = { +static const struct snd_bebob_rate_spec saffirepro_both_rate_spec = { .get = &saffirepro_both_clk_freq_get, .set = &saffirepro_both_clk_freq_set, }; @@ -287,7 +287,7 @@ struct snd_bebob_spec saffirepro_10_spec = { .meter = NULL }; -static struct snd_bebob_rate_spec saffire_both_rate_spec = { +static const struct snd_bebob_rate_spec saffire_both_rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate, }; diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 0ed5e5f..091290d 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -335,7 +335,7 @@ static void bebob_remove(struct fw_unit *unit) snd_card_free_when_closed(bebob->card); } -static struct snd_bebob_rate_spec normal_rate_spec = { +static const struct snd_bebob_rate_spec normal_rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate }; diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index d3c9d8d..aff9da3 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -71,7 +71,7 @@ struct snd_bebob_meter_spec { }; struct snd_bebob_spec { struct snd_bebob_clock_spec *clock; - struct snd_bebob_rate_spec *rate; + const struct snd_bebob_rate_spec *rate; struct snd_bebob_meter_spec *meter; }; diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c index 9242e33..470d526 100644 --- a/sound/firewire/bebob/bebob_terratec.c +++ b/sound/firewire/bebob/bebob_terratec.c @@ -55,7 +55,7 @@ phase24_series_clk_src_get(struct snd_bebob *bebob, unsigned int *id) return 0; } -static struct snd_bebob_rate_spec phase_series_rate_spec = { +static const struct snd_bebob_rate_spec phase_series_rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate, }; diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index 2fdc1f1..ef224d6 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -155,7 +155,7 @@ static int pcm_open(struct snd_pcm_substream *substream) { struct snd_bebob *bebob = substream->private_data; - struct snd_bebob_rate_spec *spec = bebob->spec->rate; + const struct snd_bebob_rate_spec *spec = bebob->spec->rate; unsigned int sampling_rate; enum snd_bebob_clock_type src; int err; diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index a2baa47..fc7a66d 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -580,7 +580,7 @@ end: int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) { - struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; + const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; struct amdtp_stream *master, *slave; enum cip_flags sync_mode; unsigned int curr_rate; diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index 057495d..dc9cd0c 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -712,7 +712,7 @@ end: } /* for special customized devices */ -static struct snd_bebob_rate_spec special_rate_spec = { +static const struct snd_bebob_rate_spec special_rate_spec = { .get = &special_get_rate, .set = &special_set_rate, }; @@ -733,7 +733,7 @@ struct snd_bebob_spec maudio_special_spec = { }; /* Firewire 410 specification */ -static struct snd_bebob_rate_spec usual_rate_spec = { +static const struct snd_bebob_rate_spec usual_rate_spec = { .get = &snd_bebob_stream_get_rate, .set = &snd_bebob_stream_set_rate, }; diff --git a/sound/firewire/bebob/bebob_proc.c b/sound/firewire/bebob/bebob_proc.c index 301cc6a..0f3b2ca 100644 --- a/sound/firewire/bebob/bebob_proc.c +++ b/sound/firewire/bebob/bebob_proc.c @@ -138,7 +138,7 @@ proc_read_clock(struct snd_info_entry *entry, "SYT-Match", }; struct snd_bebob *bebob = entry->private_data; - struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; + const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock; enum snd_bebob_clock_type src; unsigned int rate;

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[alsa-devel] [PATCH 1/1] Allow any MIDI endpoint to drive use of interrupt transfer on Roland devices
by maillist＠superlative.org 11 Oct '15

11 Oct '15

From: "Keith A. Milner" <kamilner(a)superlative.org> Signed-off-by: Keith A. Milner <maillist(a)superlative.org> diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b901f46..52b8c72 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1860,9 +1860,14 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi) hostif = &intf->altsetting[1]; intfd = get_iface_desc(hostif); + /* If either or both of the endpoints support interrupt transfer, + * then use the alternate setting + */ if (intfd->bNumEndpoints != 2 || - (get_endpoint(hostif, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK || - (get_endpoint(hostif, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT) + !((get_endpoint(hostif, 0)->bmAttributes + & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT || + (get_endpoint(hostif, 1)->bmAttributes + & USB_ENDPOINT_XFERTYPE_MASK) == USB_ENDPOINT_XFER_INT)) return; snd_printdd(KERN_INFO "switching to altsetting %d with int ep\n", -- 1.9.1

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[alsa-devel] [PATCH v2] ALSA: usb-audio: Fix max packet size calculation for USB audio
by Ricard Wanderlof 10 Oct '15

10 Oct '15

The addition of 0xffff ("almost 1" in Q16.16 format), resulting in the rounding up of the resulting value, was in the wrong place, as it is the resulting packet size that needs to be rounded up to the nearest integer (e.g. if the resulting packet size is calculated as 55.125 bytes, we need a packet of 56 bytes), rather than the rate value (ep->freqmax). Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- V2: Added tested device to commit message. No other changes. Since this clearly is a sensitive part of the code, here follows a lengthy rationale with examples to illustrate the problem. The code as it stands has the following expression on line 613 to calculate the maximum isochronous packet size: maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) >> (16 - ep->datainterval); Here, ep->freqmax is the maximum assumed sample frequency, calculated from the nominal sample frequency plus 25%. It is ultimately derived from ep->freqn, which is in the units of samples per packet, from get_usb_full_speed_rate() or usb_high_speed_rate(), as applicable, in Q16.16 format. The expression essentially adds the Q16.16 equivalent of 0.999... (i.e. the largest number less than one) to the sample rate, in order to get a rate whose integer part is rounded up from the fractional value. The multiplication with (frame_bits >> 3) yields the number of bytes in a packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back to an integer, taking into consideration the bDataInterval field of the endpoint descriptor, which describes how often isochronous packets are transmitted relative to the (micro)frame rate (125us or 1ms, for USB high speed and full speed, respectively). For this discussion we will assume a bDataInterval of 0, so the second line of the expression just converts the Q16.16 value to an integer. In order to illustrate the problem, we will set frame_bits 64, which corresponds to a frame size of 8 bytes. The problem here is that the addition of 0xffff is in the wrong place. We don't want the rounding up to take place on the frequency, but on the resulting packet size. Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is 786432 (see usb_high_speed_rate()). Add 25% (line 612) and we get 983040. The calculated maxsize is then ((983040 + 65535) * 8) >> 16 = 127 . However, if we do the number of bytes calculation in a less obscure way it's more apparent what the true corresponding packet size is: we get (96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612, and the 8000 is the number of isochronous packets per second on a high speed USB connection (125 us microframe interval). Rephrasing the maxsize expression to: maxsize = (ep->freqmax * (frame_bits >> 3) + 0xffff) >> (16 - ep->datainterval); we instead get (983040 * 8 + 65535) >> 16 = 120 which is the correct value. We can also do the calculation with a non-integer sample rate which is when rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn = 361267, and resulting ep->freqmax 361267 * 1.25 = 451583 (rounded down). Original maxsize = ((451583 + 65535) * 8) << 16 = 63 (63.124.. rounded down) True maxsize = (44100 * 1.25 / 8000) * 8 = 55.125, i.e. 56 bytes required New maxsize = (451583 * 8 + 65535) >> 16 = 56 This is also corroborated by the wMaxPacketSize check on line 616. Assume that wMaxPacketSize = 108, with ep->maxpacksize then having the same value. As 108 < 127, we get maxsize = 108. ep->freqmax is then recalculated to (108 / 8) << 16 = 884736 . Putting that rate into the original maxsize calculation yields a maxsize of ((884736 + 65535) * 8) >> 16 = 115 (with decimals 115.99988). Clearly, we should get back the 108 here, which we would with the new expression: (884736 * 8 + 65535) >> 16 = 108 . As it currently stands, the error is tolerable because it only results in maxsize being a bit too big which wastes a couple of bytes, either as a result of the first maxsize calculation, or because the resulting calculation will hit the wMaxPacketSize value before the packet is too big, resulting in fixing the size to wMaxPacketSize even though the packet is actually not too long. However, I'm working on a quirk for a device which stuffs an extra four (non sample) bytes into each isochronous packet, and that means that the maxsize calculation must take this into account. During this work it became obvious that there was something wrong with the maxsize calculation, as the resulting packet size when wMaxPacketSize was hit did not correspond to the packet size which triggered the if statement to start with. Since this seems to me an obvious (after looking at it for a bit...) bug I'm submitting this as a separate patch rather than part of an upcoming patch series. sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index e6f7189..be8a972 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -610,7 +610,7 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, /* assume max. frequency is 25% higher than nominal */ ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) + maxsize = (ep->freqmax * (frame_bits >> 3) + 0xffff) >> (16 - ep->datainterval); /* but wMaxPacketSize might reduce this */ if (ep->maxpacksize && ep->maxpacksize < maxsize) { -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] How to download and install patch for ICS-43432 codec driver
by Ankhit Vivekananda 10 Oct '15

10 Oct '15

Hello, I am new to developing with ALSA. I have a situation where I need to use ICS-43432 MEMS microphone with a TMS320DM365 processor. I wanted to know if I could download the patch for ICS-43432 codec driver: http://permalink.gmane.org/gmane.linux.alsa.devel/143442. Appreciate any help. Thank you, Ankhit

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