June 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/3] ALSA: pcm:
by Takashi Sakamoto 09 Jun '17

09 Jun '17

Hi, This patchset is a revised version of my former RFC. [alsa-devel] [PATCH RFC 00/21] ALSA: pcm: add tracepoints for PCM params operation http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120548.html In ALSA PCM interface, applications can get hardware capability as 'struct snd_pcm_hw_params' type of data returned by a call of ioctl(2) with SNDRV_PCM_IOCTL_HW_REFINE/SNDRV_PCM_IOCTL_HW_PARAMS' commands. Results of these commands should be the same in a certain conditions but actually it's not. This is a part of layout of the structure. struct snd_pcm_hw_params { ... unsigned int info; unsigned int msbits; unsigned int rate_num; unsigned int rate_den; snd_pcm_uframes_t fifo_size; ... }; For example, although the 'msbits', 'rate_num' and 'rate_den' fields are filled in a result of HW_REFINE command conditionally, they're never filled in a result of HW_PARAMS command. This seems a bug. This patchset attempts to fix the bug. This affects userspace applications in cotents of the result, but it's acceptrable with a merit for the applications to get enough parameters. Takashi Sakamoto (3): ALSA: pcm: use helper functions to refer parameters as constants ALSA: pcm: calculate non-mask/non-interval parameters always when possible ALSA: pcm: move fixup of info flag after selecting single parameters sound/core/pcm_native.c | 88 +++++++++++++++++++++++++++++-------------------- 1 file changed, 53 insertions(+), 35 deletions(-) -- 2.11.0

2 5

[alsa-devel] [PATCH] ASoC: Intel: sst: Remove unused function sst_restore_shim64()
by Matthias Kaehlcke 09 Jun '17

09 Jun '17

Looks like the function has never been used since it was added by commit b0d94acd634a ("ASoC: Intel: mrfld - add shim save restore"). Removing it fixes the following warning when building with clang: sound/soc/intel/atom/sst/sst.c:360:20: error: unused function 'sst_restore_shim64' [-Werror,-Wunused-function] Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- sound/soc/intel/atom/sst/sst.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index f9ba71315e33..fb608d3a90c8 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -397,22 +397,6 @@ static inline void sst_save_shim64(struct intel_sst_drv *ctx, spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags); } -static inline void sst_restore_shim64(struct intel_sst_drv *ctx, - void __iomem *shim, - struct sst_shim_regs64 *shim_regs) -{ - unsigned long irq_flags; - - /* - * we only need to restore IMRX for this case, rest will be - * initialize by FW or driver when firmware is loaded - */ - spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags); - sst_shim_write64(shim, SST_IMRX, shim_regs->imrx); - sst_shim_write64(shim, SST_CSR, shim_regs->csr); - spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags); -} - void sst_configure_runtime_pm(struct intel_sst_drv *ctx) { pm_runtime_set_autosuspend_delay(ctx->dev, SST_SUSPEND_DELAY); -- 2.13.0.303.g4ebf302169-goog

4 6

[alsa-devel] [PATCH v2 0/8] [PATCH 0/8] ALSA: pcm: code refactoring for snd_pcm_hw_refine()
by Takashi Sakamoto 08 Jun '17

08 Jun '17

Hi, This patchset is revised version of my previous one. [alsa-devel] [PATCH 0/8] ALSA: pcm: code refactoring for snd_pcm_hw_refine() http://mailman.alsa-project.org/pipermail/alsa-devel/2017-June/121443.html Changes from v1: - get rid of __may_unused keyword - remove subtle changes Here, I repeat a description on my former message. In ALSA PCM interface, applications can get hardware capability as 'struct snd_pcm_hw_params' type of data returned by a call of ioctl(2) with SNDRV_PCM_IOCTL_HW_REFINE/SNDRV_PCM_IOCTL_HW_PARAMS' commands. In kernel land, the data is prepared in 'snd_pcm_hw_refine()' in 'sound/core/pcm_native.c'. This patchset is a code refactoring to this function. When applications execute open(2) to any PCM character device, runtime of corresponding PCM substream is created in kernel land. Then, the runtime gets information to configure the parameter data. When applications execute ioctl(2) with the above commands, the given parameter data is actually changed according to the information. The parameter data structure includes several types of information. Major part of the parameter data is represented by two kinds of shape; mask ('struct snd_mask') and interval ('struct snd_interval'). 15 types of data is pre-defined as 'SNDRV_PCM_HW_PARAM_XXX' macros with the two types. The runtime has several constraints, which is a kind of information to configure the parameters. There're two types of constraints; independent and dependent. The independent constrain is for each of single parameters. On the other hand, the dependent constrain is for each of combination between parameters, thus has specific structure, 'snd_pcm_hw_rule'. This structure describes relationships between several parameters. In 'sound/core/pcm_native.c', the above constraints and rules are applied to parameter data for userspace. This processing is done in 'snd_pcm_hw_refine()'. In this patchset, I attempt to clean up the processing so that developers easily understand it. This is a kind of refactoring, thus behaviour of the processing is not changed. Tracepoints added in my previous patchset is helpful for your test to check it. I wrote an application as low level program and it shall satisfies your interests about the above mechanism[1]. [1] refine-pcm-params.c https://github.com/takaswie/alsa-ioctl-test/ Takashi Sakamoto (8): ALSA: pcm: add a helper function to constrain mask-type parameters ALSA: pcm: add a helper function to constrain interval-type parameters ALSA: pcm: add a helper function to apply parameter rules ALSA: pcm: use goto statement instead of while statement to reduce indentation ALSA: pcm: remove function local variable with alternative evaluation ALSA: pcm: adaption of code formatting ALSA: pcm: use helper functions to check whether parameters are determined ALSA: pcm: add comment about application of rule to PCM parameters sound/core/pcm_native.c | 241 +++++++++++++++++++++++++++++++++--------------- 1 file changed, 169 insertions(+), 72 deletions(-) -- 2.11.0

2 9

[alsa-devel] [PATCH 00/31] Constify hw_constrains
by Takashi Iwai 08 Jun '17

08 Jun '17

Hi, this is a very trivial patchset for adding const prefix to some static PCM hw constraints. Just for hardening. The similar patchset for ASoC will follow later. thanks, Takashi === Takashi Iwai (31): ALSA: cs423x: Constify hw_constraints ALSA: es1688: Constify hw_constraints ALSA: es18xx: Constify hw_constraints ALSA: gus: Constify hw_constraints ALSA: sb: Constify hw_constraints ALSA: wss: Constify hw_constraints ALSA: parisc: Constify hw_constraints ALSA: ali5451: Constify hw_constraints ALSA: atiixp: Constify hw_constraints ALSA: au88x0: Constify hw_constraints ALSA: azt3328: Constify hw_constraints ALSA: bt87x: Constify hw_constraints ALSA: cmipci: Constify hw_constraints ALSA: cs46xx: Constify hw_constraints ALSA: emu10k1: Constify hw_constraints ALSA: fm801: Constify hw_constraints ALSA: ens137x: Constify hw_constraints ALSA: es1938: Constify hw_constraints ALSA: hda: Constify hw_constraints ALSA: ice1712: Constify hw_constraints ALSA: ice1724: Constify hw_constraints ALSA: intel8x0: Constify hw_constraints ALSA: nm256: Constify hw_constraints ALSA: rme32: Constify hw_constraints ALSA: rme96: Constify hw_constraints ALSA: hdsp: Constify hw_constraints ALSA: hdspm: Constify hw_constraints ALSA: rme9652: Constify hw_constraints ALSA: sonicvibes: Constify hw_constraints ALSA: via82xx: Constify hw_constraints ALSA: sparc: Constify hw_constraints sound/isa/cs423x/cs4236_lib.c | 4 ++-- sound/isa/es1688/es1688_lib.c | 4 ++-- sound/isa/es18xx.c | 8 ++++---- sound/isa/gus/gus_pcm.c | 4 ++-- sound/isa/sb/sb8_main.c | 6 +++--- sound/isa/wss/wss_lib.c | 4 ++-- sound/parisc/harmony.c | 6 +++--- sound/pci/ali5451/ali5451.c | 4 ++-- sound/pci/atiixp_modem.c | 4 ++-- sound/pci/au88x0/au88x0_pcm.c | 4 ++-- sound/pci/azt3328.c | 4 ++-- sound/pci/bt87x.c | 4 ++-- sound/pci/cmipci.c | 12 ++++++------ sound/pci/cs46xx/cs46xx_lib.c | 4 ++-- sound/pci/emu10k1/emupcm.c | 8 ++++---- sound/pci/ens1370.c | 16 ++++++++-------- sound/pci/es1938.c | 4 ++-- sound/pci/fm801.c | 8 ++++---- sound/pci/hda/patch_hdmi.c | 10 +++++----- sound/pci/hda/patch_si3054.c | 4 ++-- sound/pci/ice1712/ice1712.c | 4 ++-- sound/pci/ice1712/ice1712.h | 2 +- sound/pci/ice1712/ice1724.c | 8 ++++---- sound/pci/ice1712/juli.c | 6 +++--- sound/pci/ice1712/maya44.c | 4 ++-- sound/pci/ice1712/quartet.c | 6 +++--- sound/pci/intel8x0.c | 12 ++++++------ sound/pci/intel8x0m.c | 4 ++-- sound/pci/nm256/nm256.c | 4 ++-- sound/pci/rme32.c | 5 ++--- sound/pci/rme96.c | 4 ++-- sound/pci/rme9652/hdsp.c | 8 ++++---- sound/pci/rme9652/hdspm.c | 4 ++-- sound/pci/rme9652/rme9652.c | 4 ++-- sound/pci/sonicvibes.c | 4 ++-- sound/pci/via82xx.c | 4 ++-- sound/pci/via82xx_modem.c | 4 ++-- sound/sparc/cs4231.c | 4 ++-- 38 files changed, 106 insertions(+), 107 deletions(-) -- 2.13.0

1 31

[alsa-devel] [PATCH] ASoC: Back out post commit widget creation changes
by Mark Brown 08 Jun '17

08 Jun '17

Due to build errors revert commit c8597af855f3 (ASoC: topology: Allow bespoke configuration post widget creation) until they can be fixed. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-topology.c | 24 ++---------------------- 1 file changed, 2 insertions(+), 22 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 7006cf3007b5..f4ec236a418e 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -344,24 +344,12 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, return 0; } -/* optionally pass new dynamic widget to component driver. This is mainly for - * external widgets where we can assign private data/ops */ -static int soc_tplg_widget_ready(struct soc_tplg *tplg, - struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) -{ - if (tplg->comp && tplg->ops && tplg->ops->widget_ready) - return tplg->ops->widget_ready(tplg->comp, w, tplg_w); - - return 0; -} - /* pass DAI configurations to component driver for extra initialization */ static int soc_tplg_dai_load(struct soc_tplg *tplg, - struct snd_soc_dai_driver *dai_drv, - struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) + struct snd_soc_dai_driver *dai_drv) { if (tplg->comp && tplg->ops && tplg->ops->dai_load) - return tplg->ops->dai_load(tplg->comp, dai_drv, pcm, dai); + return tplg->ops->dai_load(tplg->comp, dai_drv); return 0; } @@ -1591,16 +1579,8 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, widget->dobj.ops = tplg->ops; widget->dobj.index = tplg->index; list_add(&widget->dobj.list, &tplg->comp->dobj_list); - - ret = soc_tplg_widget_ready(tplg, widget, w); - if (ret < 0) - goto ready_err; - return 0; -ready_err: - snd_soc_tplg_widget_remove(widget); - snd_soc_dapm_free_widget(widget); hdr_err: kfree(template.sname); err: -- 2.11.0

1 1

[alsa-devel] [asoc:topic/topology 1/4] sound/soc/soc-topology.c:352:42: error: 'struct snd_soc_tplg_ops' has no member named 'widget_ready'
by kbuild test robot 08 Jun '17

08 Jun '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/topology head: 294de6e372673229432dc8bcd80964223bc1589d commit: c8597af855f3e34aaebaff0e5c3dbd07611c87f1 [1/4] ASoC: topology: Allow bespoke configuration post widget creation config: x86_64-randconfig-g0-06082243 (attached as .config) compiler: gcc-5 (Debian 5.4.1-2) 5.4.1 20160904 reproduce: git checkout c8597af855f3e34aaebaff0e5c3dbd07611c87f1 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/soc/soc-topology.c: In function 'soc_tplg_widget_ready': >> sound/soc/soc-topology.c:352:42: error: 'struct snd_soc_tplg_ops' has no member named 'widget_ready' if (tplg->comp && tplg->ops && tplg->ops->widget_ready) ^ sound/soc/soc-topology.c:353:19: error: 'struct snd_soc_tplg_ops' has no member named 'widget_ready' return tplg->ops->widget_ready(tplg->comp, w, tplg_w); ^ sound/soc/soc-topology.c: In function 'soc_tplg_dai_load': sound/soc/soc-topology.c:364:10: error: too many arguments to function 'tplg->ops->dai_load' return tplg->ops->dai_load(tplg->comp, dai_drv, pcm, dai); ^ sound/soc/soc-topology.c: In function 'soc_tplg_dai_create': sound/soc/soc-topology.c:1723:8: error: too few arguments to function 'soc_tplg_dai_load' ret = soc_tplg_dai_load(tplg, dai_drv); ^ sound/soc/soc-topology.c:359:12: note: declared here static int soc_tplg_dai_load(struct soc_tplg *tplg, ^ sound/soc/soc-topology.c: In function 'soc_tplg_dai_config': sound/soc/soc-topology.c:2206:8: error: too few arguments to function 'soc_tplg_dai_load' ret = soc_tplg_dai_load(tplg, dai_drv); ^ sound/soc/soc-topology.c:359:12: note: declared here static int soc_tplg_dai_load(struct soc_tplg *tplg, ^ vim +352 sound/soc/soc-topology.c 346 347 /* optionally pass new dynamic widget to component driver. This is mainly for 348 * external widgets where we can assign private data/ops */ 349 static int soc_tplg_widget_ready(struct soc_tplg *tplg, 350 struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) 351 { > 352 if (tplg->comp && tplg->ops && tplg->ops->widget_ready) 353 return tplg->ops->widget_ready(tplg->comp, w, tplg_w); 354 355 return 0; --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [asoc:topic/topology 1/4] sound//soc/soc-topology.c:352:42: error: 'struct snd_soc_tplg_ops' has no member named 'widget_ready'; did you mean 'widget_load'?
by kbuild test robot 08 Jun '17

08 Jun '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/topology head: 294de6e372673229432dc8bcd80964223bc1589d commit: c8597af855f3e34aaebaff0e5c3dbd07611c87f1 [1/4] ASoC: topology: Allow bespoke configuration post widget creation config: x86_64-randconfig-u0-06082216 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout c8597af855f3e34aaebaff0e5c3dbd07611c87f1 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound//soc/soc-topology.c: In function 'soc_tplg_widget_ready': >> sound//soc/soc-topology.c:352:42: error: 'struct snd_soc_tplg_ops' has no member named 'widget_ready'; did you mean 'widget_load'? if (tplg->comp && tplg->ops && tplg->ops->widget_ready) ^~ sound//soc/soc-topology.c:353:19: error: 'struct snd_soc_tplg_ops' has no member named 'widget_ready'; did you mean 'widget_load'? return tplg->ops->widget_ready(tplg->comp, w, tplg_w); ^~ sound//soc/soc-topology.c: In function 'soc_tplg_dai_load': sound//soc/soc-topology.c:364:10: error: too many arguments to function 'tplg->ops->dai_load' return tplg->ops->dai_load(tplg->comp, dai_drv, pcm, dai); ^~~~ sound//soc/soc-topology.c: In function 'soc_tplg_dai_create': >> sound//soc/soc-topology.c:1723:8: error: too few arguments to function 'soc_tplg_dai_load' ret = soc_tplg_dai_load(tplg, dai_drv); ^~~~~~~~~~~~~~~~~ sound//soc/soc-topology.c:359:12: note: declared here static int soc_tplg_dai_load(struct soc_tplg *tplg, ^~~~~~~~~~~~~~~~~ sound//soc/soc-topology.c: In function 'soc_tplg_dai_config': sound//soc/soc-topology.c:2206:8: error: too few arguments to function 'soc_tplg_dai_load' ret = soc_tplg_dai_load(tplg, dai_drv); ^~~~~~~~~~~~~~~~~ sound//soc/soc-topology.c:359:12: note: declared here static int soc_tplg_dai_load(struct soc_tplg *tplg, ^~~~~~~~~~~~~~~~~ vim +352 sound//soc/soc-topology.c 346 347 /* optionally pass new dynamic widget to component driver. This is mainly for 348 * external widgets where we can assign private data/ops */ 349 static int soc_tplg_widget_ready(struct soc_tplg *tplg, 350 struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) 351 { > 352 if (tplg->comp && tplg->ops && tplg->ops->widget_ready) > 353 return tplg->ops->widget_ready(tplg->comp, w, tplg_w); 354 355 return 0; 356 } --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v2] ASoC: sgtl5000: add avc enable control
by Richard Leitner 08 Jun '17

08 Jun '17

This control enables the automatic volume control (AVC) of SGTL5000 using its DAP_AVC_CTRL register. Signed-off-by: Richard Leitner <richard.leitner(a)skidata.com> --- CHANGES v2: - added trailing "Switch" as covered in ControlNames.txt --- sound/soc/codecs/sgtl5000.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 5a2702e..0477f2e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -434,6 +434,8 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 0x1f, 1, lineout_volume), SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1), + + SOC_SINGLE("AVC Enable Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0), }; /* mute the codec used by alsa core */ -- 2.1.4

2 1

[alsa-devel] [PATCH 0/8] ALSA: pcm: code refactoring for snd_pcm_hw_refine()
by Takashi Sakamoto 08 Jun '17

08 Jun '17

Hi, This patchset is an update of part of my RFC, and goes for upstream. [alsa-devel] [PATCH RFC 00/21] ALSA: pcm: add tracepoints for PCM params operation http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120548.html In ALSA PCM interface, applications can get hardware capability as 'struct snd_pcm_hw_params' type of data returned by a call of ioctl(2) with SNDRV_PCM_IOCTL_HW_REFINE/SNDRV_PCM_IOCTL_HW_PARAMS' commands. In kernel land, the data is prepared in 'snd_pcm_hw_refine()' in 'sound/core/pcm_native.c'. This patchset is a code refactoring to this function. When applications execute open(2) to any PCM character device, runtime of corresponding PCM substream is created in kernel land. Then, the runtime gets information to configure the parameter data. When applications execute ioctl(2) with the above commands, the given parameter data is actually changed according to the information. The parameter data structure includes several types of information. Major part of the parameter data is represented by two kinds of shape; mask ('struct snd_mask') and interval ('struct snd_interval'). 15 types of data is pre-defined as 'SNDRV_PCM_HW_PARAM_XXX' macros with the two types. The runtime has several constraints, which is a kind of information to configure the parameters. There're two types of constraints; independent and dependent. The independent constrain is for each of single parameters. On the other hand, the dependent constrain is for each of combination between parameters, thus has specific structure, 'snd_pcm_hw_rule'. This structure describes relationships between several parameters. In 'sound/core/pcm_native.c', the above constraints and rules are applied to parameter data for userspace. This processing is done in 'snd_pcm_hw_refine()'. In this patchset, I attempt to clean up the processing so that developers easily understand it. This is a kind of refactoring, thus behaviour of the processing is not changed. Tracepoints added in my previous patchset is helpful for your test to check it. I wrote an application as low level program and it shall satisfies your interests about the above mechanism[1]. [1] refine-pcm-params.c https://github.com/takaswie/alsa-ioctl-test/ Takashi Sakamoto (8): ALSA: pcm: add a helper function to constrain mask-type parameters ALSA: pcm: add a helper function to constrain interval-type parameters ALSA: pcm: add a helper function to apply parameter rules ALSA: pcm: use goto statement instead of while statement to reduce indentation ALSA: pcm: remove function local variable with alternative evaluation ALSA: pcm: adaption of code formatting ALSA: pcm: use helper functions to check whether parameters are determined ALSA: pcm: add comment about application of rule to PCM parameters sound/core/pcm_native.c | 253 ++++++++++++++++++++++++++++++++++-------------- 1 file changed, 179 insertions(+), 74 deletions(-) -- 2.11.0

2 14

[alsa-devel] [PATCH] ALSA: firewire: arrange common PCM info/constraints for AMDTP engine applications
by Takashi Sakamoto 08 Jun '17

08 Jun '17

In ALSA firewire stack, 8 drivers uses IEC 61883-1/6 engine for data transmission. They have common PCM info/constraints and duplicated codes. This commit unifies the codes into fireiwre-lib. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 19 +++++++++++++++++ sound/firewire/bebob/bebob_pcm.c | 21 ------------------- sound/firewire/dice/dice-pcm.c | 20 ------------------ sound/firewire/digi00x/digi00x-pcm.c | 33 ++++++++++-------------------- sound/firewire/fireface/ff-pcm.c | 21 ------------------- sound/firewire/fireworks/fireworks_pcm.c | 21 ------------------- sound/firewire/motu/motu-pcm.c | 20 ------------------ sound/firewire/oxfw/oxfw-pcm.c | 20 ------------------ sound/firewire/tascam/tascam-pcm.c | 35 ++++++-------------------------- 9 files changed, 36 insertions(+), 174 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 4316d9db404d..9678bc75dc5b 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -148,8 +148,27 @@ EXPORT_SYMBOL(amdtp_rate_table); int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, struct snd_pcm_runtime *runtime) { + struct snd_pcm_hardware *hw = &runtime->hw; int err; + hw->info = SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_JOINT_DUPLEX | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID; + + /* SNDRV_PCM_INFO_BATCH */ + hw->periods_min = 2; + hw->periods_max = UINT_MAX; + + /* bytes for a frame */ + hw->period_bytes_min = 4 * hw->channels_max; + + /* Just to prevent from allocating much pages. */ + hw->period_bytes_max = hw->period_bytes_min * 2048; + hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; + /* * Currently firewire-lib processes 16 packets in one software * interrupt callback. This equals to 2msec but actually the diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index 657e15a27e5c..e6adab3ef42e 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -92,19 +92,6 @@ limit_channels_and_rates(struct snd_pcm_hardware *hw, } } -static void -limit_period_and_buffer(struct snd_pcm_hardware *hw) -{ - hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ - hw->periods_max = UINT_MAX; - - hw->period_bytes_min = 4 * hw->channels_max; /* bytes for a frame */ - - /* Just to prevent from allocating much pages. */ - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; -} - static int pcm_init_hw_params(struct snd_bebob *bebob, struct snd_pcm_substream *substream) @@ -114,13 +101,6 @@ pcm_init_hw_params(struct snd_bebob *bebob, struct snd_bebob_stream_formation *formations; int err; - runtime->hw.info = SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS; s = &bebob->tx_stream; @@ -132,7 +112,6 @@ pcm_init_hw_params(struct snd_bebob *bebob, } limit_channels_and_rates(&runtime->hw, formations); - limit_period_and_buffer(&runtime->hw); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, formations, diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 2dda74695069..7cb9e9713ac3 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -51,18 +51,6 @@ static int limit_channels_and_rates(struct snd_dice *dice, return 0; } -static void limit_period_and_buffer(struct snd_pcm_hardware *hw) -{ - hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ - hw->periods_max = UINT_MAX; - - hw->period_bytes_min = 4 * hw->channels_max; /* byte for a frame */ - - /* Just to prevent from allocating much pages. */ - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; -} - static int init_hw_info(struct snd_dice *dice, struct snd_pcm_substream *substream) { @@ -74,13 +62,6 @@ static int init_hw_info(struct snd_dice *dice, unsigned int count, size; int err; - hw->info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_BLOCK_TRANSFER; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { hw->formats = AM824_IN_PCM_FORMAT_BITS; dir = AMDTP_IN_STREAM; @@ -107,7 +88,6 @@ static int init_hw_info(struct snd_dice *dice, substream->pcm->device, size); if (err < 0) return err; - limit_period_and_buffer(hw); return amdtp_am824_add_pcm_hw_constraints(stream, runtime); } diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index f76cf5e383db..796f4b4645f5 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -58,31 +58,11 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, static int pcm_init_hw_params(struct snd_dg00x *dg00x, struct snd_pcm_substream *substream) { - static const struct snd_pcm_hardware hardware = { - .info = SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000, - .rate_min = 44100, - .rate_max = 96000, - .channels_min = 10, - .channels_max = 18, - .period_bytes_min = 4 * 18, - .period_bytes_max = 4 * 18 * 2048, - .buffer_bytes_max = 4 * 18 * 2048 * 2, - .periods_min = 2, - .periods_max = UINT_MAX, - }; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hardware *hw = &runtime->hw; struct amdtp_stream *s; int err; - substream->runtime->hw = hardware; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; @@ -92,6 +72,15 @@ static int pcm_init_hw_params(struct snd_dg00x *dg00x, s = &dg00x->rx_stream; } + hw->channels_min = 10; + hw->channels_max = 18; + + hw->rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + snd_pcm_limit_hw_rates(runtime); + err = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, NULL, diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index ad974b5a2561..d12a0e3a4219 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -91,18 +91,6 @@ static void limit_channels_and_rates(struct snd_pcm_hardware *hw, } } -static void limit_period_and_buffer(struct snd_pcm_hardware *hw) -{ - hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ - hw->periods_max = UINT_MAX; - - hw->period_bytes_min = 4 * hw->channels_max; /* bytes for a frame */ - - /* Just to prevent from allocating much pages. */ - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; -} - static int pcm_init_hw_params(struct snd_ff *ff, struct snd_pcm_substream *substream) { @@ -111,13 +99,6 @@ static int pcm_init_hw_params(struct snd_ff *ff, const unsigned int *pcm_channels; int err; - runtime->hw.info = SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; s = &ff->tx_stream; @@ -128,9 +109,7 @@ static int pcm_init_hw_params(struct snd_ff *ff, pcm_channels = ff->spec->pcm_playback_channels; } - /* limit rates */ limit_channels_and_rates(&runtime->hw, pcm_channels); - limit_period_and_buffer(&runtime->hw); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, (void *)pcm_channels, diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index 346e2647ed1f..40faed5e6968 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -129,19 +129,6 @@ limit_channels(struct snd_pcm_hardware *hw, unsigned int *pcm_channels) } } -static void -limit_period_and_buffer(struct snd_pcm_hardware *hw) -{ - hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ - hw->periods_max = UINT_MAX; - - hw->period_bytes_min = 4 * hw->channels_max; /* bytes for a frame */ - - /* Just to prevent from allocating much pages. */ - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; -} - static int pcm_init_hw_params(struct snd_efw *efw, struct snd_pcm_substream *substream) @@ -151,13 +138,6 @@ pcm_init_hw_params(struct snd_efw *efw, unsigned int *pcm_channels; int err; - runtime->hw.info = SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS; s = &efw->tx_stream; @@ -173,7 +153,6 @@ pcm_init_hw_params(struct snd_efw *efw, snd_pcm_limit_hw_rates(runtime); limit_channels(&runtime->hw, pcm_channels); - limit_period_and_buffer(&runtime->hw); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, pcm_channels, diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index e3ef89cee565..a2b50df70874 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -96,18 +96,6 @@ static void limit_channels_and_rates(struct snd_motu *motu, snd_pcm_limit_hw_rates(runtime); } -static void limit_period_and_buffer(struct snd_pcm_hardware *hw) -{ - hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ - hw->periods_max = UINT_MAX; - - hw->period_bytes_min = 4 * hw->channels_max; /* byte for a frame */ - - /* Just to prevent from allocating much pages. */ - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; -} - static int init_hw_info(struct snd_motu *motu, struct snd_pcm_substream *substream) { @@ -117,13 +105,6 @@ static int init_hw_info(struct snd_motu *motu, struct snd_motu_packet_format *formats; int err; - hw->info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_BLOCK_TRANSFER; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { hw->formats = SNDRV_PCM_FMTBIT_S32; stream = &motu->tx_stream; @@ -135,7 +116,6 @@ static int init_hw_info(struct snd_motu *motu, } limit_channels_and_rates(motu, runtime, formats); - limit_period_and_buffer(hw); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, motu_rate_constraint, formats, diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index bc1a3a36ab06..3dd46285c0e2 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -106,18 +106,6 @@ static void limit_channels_and_rates(struct snd_pcm_hardware *hw, u8 **formats) } } -static void limit_period_and_buffer(struct snd_pcm_hardware *hw) -{ - hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ - hw->periods_max = UINT_MAX; - - hw->period_bytes_min = 4 * hw->channels_max; /* bytes for a frame */ - - /* Just to prevent from allocating much pages. */ - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; -} - static int init_hw_params(struct snd_oxfw *oxfw, struct snd_pcm_substream *substream) { @@ -126,13 +114,6 @@ static int init_hw_params(struct snd_oxfw *oxfw, struct amdtp_stream *stream; int err; - runtime->hw.info = SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS; stream = &oxfw->tx_stream; @@ -144,7 +125,6 @@ static int init_hw_params(struct snd_oxfw *oxfw, } limit_channels_and_rates(&runtime->hw, formats); - limit_period_and_buffer(&runtime->hw); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, formats, diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index 3c4482aa7231..6ec8ec634d4d 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -8,41 +8,14 @@ #include "tascam.h" -static void set_buffer_params(struct snd_pcm_hardware *hw) -{ - hw->period_bytes_min = 4 * hw->channels_min; - hw->period_bytes_max = hw->period_bytes_min * 2048; - hw->buffer_bytes_max = hw->period_bytes_max * 2; - - hw->periods_min = 2; - hw->periods_max = UINT_MAX; -} - static int pcm_init_hw_params(struct snd_tscm *tscm, struct snd_pcm_substream *substream) { - static const struct snd_pcm_hardware hardware = { - .info = SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_JOINT_DUPLEX | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000, - .rate_min = 44100, - .rate_max = 96000, - .channels_min = 10, - .channels_max = 18, - }; struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hardware *hw = &runtime->hw; struct amdtp_stream *stream; unsigned int pcm_channels; - runtime->hw = hardware; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; stream = &tscm->tx_stream; @@ -59,7 +32,11 @@ static int pcm_init_hw_params(struct snd_tscm *tscm, pcm_channels += 2; runtime->hw.channels_min = runtime->hw.channels_max = pcm_channels; - set_buffer_params(&runtime->hw); + hw->rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + snd_pcm_limit_hw_rates(runtime); return amdtp_tscm_add_pcm_hw_constraints(stream, runtime); } -- 2.11.0

2 1