October 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH 1/2] ASoC: dt-bindings: max98520: Add initial bindings
by George Song 13 Oct '21

13 Oct '21

From: George Song <george.song(a)analog.com> Add initial bindings for max98520 audio amplifier Signed-off-by: George Song <george.song(a)analog.com> --- .../bindings/sound/maxim,max98520.yaml | 35 +++++++++++++++++++ 1 file changed, 35 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98520.yaml diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml new file mode 100644 index 000000000000..d1f2b5ee775a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98520 Speaker Amplifier Driver + +maintainers: + - George Song <George.song(a)analog.com> + +properties: + compatible: + const: maxim,max98520 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98520: amplifier@38 { + compatible = "maxim,max98520"; + reg = <0x38>; + }; + }; -- 2.25.1

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[PATCH v2] component: do not leave master devres group open after bind
by Kai Vehmanen 13 Oct '21

13 Oct '21

In current code, the devres group for aggregate master is left open after call to component_master_add_*(). This leads to problems when the master does further managed allocations on its own. When any participating driver calls component_del(), this leads to immediate release of resources. This came up when investigating a page fault occurring with i915 DRM driver unbind with 5.15-rc1 kernel. The following sequence occurs: i915_pci_remove() -> intel_display_driver_unregister() -> i915_audio_component_cleanup() -> component_del() -> component.c:take_down_master() -> hdac_component_master_unbind() [via master->ops->unbind()] -> devres_release_group(master->parent, NULL) With older kernels this has not caused issues, but with audio driver moving to use managed interfaces for more of its allocations, this no longer works. Devres log shows following to occur: component_master_add_with_match() [ 126.886032] snd_hda_intel 0000:00:1f.3: DEVRES ADD 00000000323ccdc5 devm_component_match_release (24 bytes) [ 126.886045] snd_hda_intel 0000:00:1f.3: DEVRES ADD 00000000865cdb29 grp< (0 bytes) [ 126.886049] snd_hda_intel 0000:00:1f.3: DEVRES ADD 000000001b480725 grp< (0 bytes) audio driver completes its PCI probe() [ 126.892238] snd_hda_intel 0000:00:1f.3: DEVRES ADD 000000001b480725 pcim_iomap_release (48 bytes) component_del() called() at DRM/i915 unbind() [ 137.579422] i915 0000:00:02.0: DEVRES REL 00000000ef44c293 grp< (0 bytes) [ 137.579445] snd_hda_intel 0000:00:1f.3: DEVRES REL 00000000865cdb29 grp< (0 bytes) [ 137.579458] snd_hda_intel 0000:00:1f.3: DEVRES REL 000000001b480725 pcim_iomap_release (48 bytes) So the "devres_release_group(master->parent, NULL)" ends up freeing the pcim_iomap allocation. Upon next runtime resume, the audio driver will cause a page fault as the iomap alloc was released without the driver knowing about it. Fix this issue by using the "struct master" pointer as identifier for the devres group, and by closing the devres group after the master->ops->bind() call is done. This allows devres allocations done by the driver acting as master to be isolated from the binding state of the aggregate driver. This modifies the logic originally introduced in commit 9e1ccb4a7700 ("drivers/base: fix devres handling for master device") BugLink: https://gitlab.freedesktop.org/drm/intel/-/issues/4136 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Acked-by: Imre Deak <imre.deak(a)intel.com> Acked-by: Russell King (Oracle) <rmk+kernel(a)armlinux.org.uk> --- drivers/base/component.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) V2 changes: - after review form Imre and Russell, removing RFC tag - rebased on top of 5.15-rc2 (V1 was on drm-tip) - CI test results for V1 show that this patch fixes multiple failures in i915 unbind and module reload tests: https://patchwork.freedesktop.org/series/94889/ diff --git a/drivers/base/component.c b/drivers/base/component.c index 5e79299f6c3f..870485cbbb87 100644 --- a/drivers/base/component.c +++ b/drivers/base/component.c @@ -246,7 +246,7 @@ static int try_to_bring_up_master(struct master *master, return 0; } - if (!devres_open_group(master->parent, NULL, GFP_KERNEL)) + if (!devres_open_group(master->parent, master, GFP_KERNEL)) return -ENOMEM; /* Found all components */ @@ -258,6 +258,7 @@ static int try_to_bring_up_master(struct master *master, return ret; } + devres_close_group(master->parent, NULL); master->bound = true; return 1; } @@ -282,7 +283,7 @@ static void take_down_master(struct master *master) { if (master->bound) { master->ops->unbind(master->parent); - devres_release_group(master->parent, NULL); + devres_release_group(master->parent, master); master->bound = false; } } base-commit: e4e737bb5c170df6135a127739a9e6148ee3da82 -- 2.32.0

3 5

[PATCH v3] ASoC: rt1011: Fix 'I2S Reference' enum control caused error
by Peter Ujfalusi 13 Oct '21

13 Oct '21

Access to 'I2S Reference' enum causes alsamixer to fail to load: $ alsamixer cannot load mixer controls: Invalid argument cml_rt1011_rt5682 cml_rt1011_rt5682: control 2:0:0:TL I2S Reference:0: access overflow The reason is that the original patch adding the code was using ucontrol->value.integer.value[0] instead the correct ucontrol->value.enumerated.item[0] for an ENUM control. Fixes: 87f40af26c262 ("ASoC: rt1011: add i2s reference control for rt1011") Reported-by: Seppo Ingalsuo <seppo.ingalsuo(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi, Changes since v2: - Fix typo in commit message s/Is@/I2S Changes since v1: - Correct the ENUM declaration as well Regards, Peter sound/soc/codecs/rt1011.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index 508597866dff..bdfcbb81fa19 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1311,12 +1311,11 @@ static int rt1011_r0_load_info(struct snd_kcontrol *kcontrol, .put = rt1011_r0_load_mode_put \ } -static const char * const rt1011_i2s_ref[] = { +static const char * const rt1011_i2s_ref_texts[] = { "None", "Left Channel", "Right Channel" }; -static SOC_ENUM_SINGLE_DECL(rt1011_i2s_ref_enum, 0, 0, - rt1011_i2s_ref); +static SOC_ENUM_SINGLE_EXT_DECL(rt1011_i2s_ref_enum, rt1011_i2s_ref_texts); static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1325,7 +1324,7 @@ static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, snd_soc_kcontrol_component(kcontrol); struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - int i2s_ref_ch = ucontrol->value.integer.value[0]; + int i2s_ref_ch = ucontrol->value.enumerated.item[0]; switch (i2s_ref_ch) { case RT1011_I2S_REF_LEFT_CH: @@ -1344,7 +1343,7 @@ static int rt1011_i2s_ref_put(struct snd_kcontrol *kcontrol, dev_info(component->dev, "I2S Reference: Do nothing\n"); } - rt1011->i2s_ref = ucontrol->value.integer.value[0]; + rt1011->i2s_ref = ucontrol->value.enumerated.item[0]; return 0; } @@ -1357,7 +1356,7 @@ static int rt1011_i2s_ref_get(struct snd_kcontrol *kcontrol, struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - ucontrol->value.integer.value[0] = rt1011->i2s_ref; + ucontrol->value.enumerated.item[0] = rt1011->i2s_ref; return 0; } -- 2.33.0

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[PATCH] alsaloop: fixed parsing value 'auto' of option 'sync'
by Pavel Hofman 13 Oct '21

13 Oct '21

The value 'auto' was not checked which resulted in using SYNC_TYPE_NONE instead of SYNC_TYPE_AUTO. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- alsaloop/alsaloop.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/alsaloop/alsaloop.c b/alsaloop/alsaloop.c index 4192712..f5f2e37 100644 --- a/alsaloop/alsaloop.c +++ b/alsaloop/alsaloop.c @@ -528,6 +528,8 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, arg_sync = SYNC_TYPE_PLAYRATESHIFT; else if (optarg[0] == 'r') arg_sync = SYNC_TYPE_SAMPLERATE; + else if (optarg[0] == 'a') + arg_sync = SYNC_TYPE_AUTO; else arg_sync = atoi(optarg); if (arg_sync < 0 || arg_sync > SYNC_TYPE_LAST) -- 2.25.1

2 1

Re: [v2 2/2] ASoC : max98520 : Added AMplifier Driver Signed-off-by: George Song <george.song@maximintegrated.com>
by Mark Brown 13 Oct '21

13 Oct '21

On Wed, Oct 13, 2021 at 04:16:16PM +0900, George Song wrote: > From: "george.song" <george.song(a)analog.com> > > --- > sound/soc/codecs/Kconfig | 5 + You've not provided a Signed-off-by for this so I can't do anything with it, please see Documentation/process/submitting-patches.rst for details on what this is and why it's important. You've also sent two different versions of patch 2 but only one version of patch 1, please don't do this - if you need to send a new version please resend the whole thread otherwise it gets very hard to follow what's going on.

1 0

[PATCH v3] alsaloop: added option prateshift for PLAYSHIFT ctl elem used in PLAYSHIFT
by Pavel Hofman 13 Oct '21

13 Oct '21

If snd-aloop device is on playback side, the required sync mode is PLAYSHIFT. That means Loopback ctl elem "PCM Rate Shift 100000" of the corresponding capture side of the Loopback pipe must be controlled (by a reciprocal). ASCII name of the playback rate shift ctl elem is specified with newly added option -x/--prateshift, e.g.: -P hw:Loopback,0 -S playshift \ -x iface=PCM,name='PCM Rate Shift 100000',device=1 Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- Notes: * using snd_ctl_elem_id_alloca instead of snd_ctl_elem_id_malloc * removed corresponding snd_ctl_elem_id_free calls alsaloop/alsaloop.1 | 11 +++++-- alsaloop/alsaloop.c | 11 ++++++- alsaloop/alsaloop.h | 1 + alsaloop/pcmjob.c | 77 +++++++++++++++++++++++++++++++++------------ 4 files changed, 77 insertions(+), 23 deletions(-) diff --git a/alsaloop/alsaloop.1 b/alsaloop/alsaloop.1 index 6bacfd5..006494d 100644 --- a/alsaloop/alsaloop.1 +++ b/alsaloop/alsaloop.1 @@ -57,6 +57,11 @@ Use given CTL device for playback. Use given CTL device for capture. +.TP +\fI\-x <ctl_ascii_name>\fP | \fI\-\-prateshift=<ctl_ascii_name>\fP + +Specify ctl ascii name for playshift sync mode (see the Examples section). + .TP \fI\-l <latency>\fP | \fI\-\-latency=<frames>\fP @@ -195,10 +200,12 @@ Verbose xrun profiling. Set process wake timeout. .SH EXAMPLES - -.TP +.nf \fBalsaloop \-C hw:0,0 \-P hw:1,0 \-t 50000\fR +\fBalsaloop \-C hw:soundcard,0 \-P hw:Loopback,0 \-t 50000 \-S playshift \\ +\-x iface=PCM,name='PCM Rate Shift 100000',device=1\fR +.ne .SH BUGS None known. .SH AUTHOR diff --git a/alsaloop/alsaloop.c b/alsaloop/alsaloop.c index 06ffadf..4192712 100644 --- a/alsaloop/alsaloop.c +++ b/alsaloop/alsaloop.c @@ -175,6 +175,7 @@ void help(void) "-C,--cdevice capture device\n" "-X,--pctl playback ctl device\n" "-Y,--cctl capture ctl device\n" +"-x,--prateshift playback 'PCM Rate Shift 100000' ascii ctl name\n" "-l,--latency requested latency in frames\n" "-t,--tlatency requested latency in usec (1/1000000sec)\n" "-f,--format sample format\n" @@ -362,6 +363,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, {"cdevice", 1, NULL, 'C'}, {"pctl", 1, NULL, 'X'}, {"cctl", 1, NULL, 'Y'}, + {"prateshift", 1, NULL, 'x'}, {"latency", 1, NULL, 'l'}, {"tlatency", 1, NULL, 't'}, {"format", 1, NULL, 'f'}, @@ -391,6 +393,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, char *arg_cdevice = NULL; char *arg_pctl = NULL; char *arg_cctl = NULL; + char *arg_prateshift = NULL; unsigned int arg_latency_req = 0; unsigned int arg_latency_reqtime = 10000; snd_pcm_format_t arg_format = SND_PCM_FORMAT_S16_LE; @@ -420,7 +423,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, while (1) { int c; if ((c = getopt_long(argc, argv, - "hdg:P:C:X:Y:l:t:F:f:c:r:s:benvA:S:a:m:T:O:w:UW:z", + "hdg:P:C:X:Y:x:l:t:F:f:c:r:s:benvA:S:a:m:T:O:w:UW:z", long_option, NULL)) < 0) break; switch (c) { @@ -446,6 +449,9 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, case 'Y': arg_cctl = strdup(optarg); break; + case 'x': + arg_prateshift = strdup(optarg); + break; case 'l': err = atoi(optarg); arg_latency_req = err >= 4 ? err : 4; @@ -627,6 +633,9 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, logit(LOG_CRIT, "Unable to add ossmixer controls.\n"); exit(EXIT_FAILURE); } + if (arg_prateshift) + play->prateshift_name = arg_prateshift; + #ifdef USE_SAMPLERATE loop->src_enable = arg_samplerate > 0; if (loop->src_enable) diff --git a/alsaloop/alsaloop.h b/alsaloop/alsaloop.h index 1dbcefe..7a98ef3 100644 --- a/alsaloop/alsaloop.h +++ b/alsaloop/alsaloop.h @@ -127,6 +127,7 @@ struct loopback_handle { snd_ctl_elem_value_t *ctl_format; snd_ctl_elem_value_t *ctl_rate; snd_ctl_elem_value_t *ctl_channels; + char *prateshift_name; /* ascii name for the playback rate shift ctl elem */ }; struct loopback { diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 845ab82..13fa595 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1101,7 +1101,8 @@ void update_pitch(struct loopback *loop) #endif } else if (loop->sync == SYNC_TYPE_PLAYRATESHIFT) { - set_rate_shift(loop->play, pitch); + // pitch is capture-based, playback side requires reciprocal + set_rate_shift(loop->play, 1 / pitch); #ifdef USE_SAMPLERATE if (loop->use_samplerate) { loop->src_data.src_ratio = @@ -1172,44 +1173,80 @@ static int get_channels(struct loopback_handle *lhandle) return snd_ctl_elem_value_get_integer(lhandle->ctl_channels, 0); } -static void openctl_elem(struct loopback_handle *lhandle, - int device, int subdevice, - const char *name, - snd_ctl_elem_value_t **elem) +static int openctl_elem_id(struct loopback_handle *lhandle, snd_ctl_elem_id_t *id, + snd_ctl_elem_value_t **elem) { int err; if (snd_ctl_elem_value_malloc(elem) < 0) { *elem = NULL; + return -ENOMEM; + } + snd_ctl_elem_value_set_id(*elem, id); + snd_ctl_elem_value_set_interface(*elem, SND_CTL_ELEM_IFACE_PCM); + err = snd_ctl_elem_read(lhandle->ctl, *elem); + if (err < 0) { + snd_ctl_elem_value_free(*elem); + *elem = NULL; + return err; } else { - snd_ctl_elem_value_set_interface(*elem, - SND_CTL_ELEM_IFACE_PCM); - snd_ctl_elem_value_set_device(*elem, device); - snd_ctl_elem_value_set_subdevice(*elem, subdevice); - snd_ctl_elem_value_set_name(*elem, name); - err = snd_ctl_elem_read(lhandle->ctl, *elem); - if (err < 0) { - snd_ctl_elem_value_free(*elem); - *elem = NULL; - } else { - if (verbose) - snd_output_printf(lhandle->loopback->output, - "Opened PCM element %s of %s, device %d, subdevice %d\n", - name, snd_ctl_name(lhandle->ctl), device, subdevice); - } + snd_output_printf(lhandle->loopback->output, + "Opened PCM element %s of %s, device %d, subdevice %d\n", + snd_ctl_elem_id_get_name(id), snd_ctl_name(lhandle->ctl), + snd_ctl_elem_id_get_device(id), + snd_ctl_elem_id_get_subdevice(id)); + return 0; } } +static int openctl_elem(struct loopback_handle *lhandle, + int device, int subdevice, + const char *name, + snd_ctl_elem_value_t **elem) +{ + snd_ctl_elem_id_t *id; + + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_id_set_device(id, device); + snd_ctl_elem_id_set_subdevice(id, subdevice); + snd_ctl_elem_id_set_name(id, name); + return openctl_elem_id(lhandle, id, elem); +} + +static int openctl_elem_ascii(struct loopback_handle *lhandle, char *ascii_name, + snd_ctl_elem_value_t **elem) +{ + snd_ctl_elem_id_t *id; + + snd_ctl_elem_id_alloca(&id); + if (snd_ctl_ascii_elem_id_parse(id, ascii_name)) { + fprintf(stderr, "Wrong control identifier: %s\n", ascii_name); + return -EINVAL; + } + return openctl_elem_id(lhandle, id, elem); +} + static int openctl(struct loopback_handle *lhandle, int device, int subdevice) { int err; lhandle->ctl_rate_shift = NULL; if (lhandle->loopback->play == lhandle) { + // play only + if (lhandle->prateshift_name) { + err = openctl_elem_ascii(lhandle, lhandle->prateshift_name, + &lhandle->ctl_rate_shift); + if (err < 0) { + logit(LOG_CRIT, "Unable to open playback PCM Rate Shift elem '%s'.\n", + lhandle->prateshift_name); + exit(EXIT_FAILURE); + } + } if (lhandle->loopback->controls) goto __events; return 0; } + // capture only openctl_elem(lhandle, device, subdevice, "PCM Notify", &lhandle->ctl_notify); openctl_elem(lhandle, device, subdevice, "PCM Rate Shift 100000", -- 2.25.1

2 1

The default jack detection type overridden by sof_sdw_quirk
by Chris Chiu 13 Oct '21

13 Oct '21

Hi, I have an Intel AlderLake machine which uses the sof-soundwire driver for audio functions. When the headset is plugged in, it will incorrectly trigger the event KEY_PLAYPAUSE and never stops. To fix the problem, I used to apply the quirk RT711_JD2 proposed in sound/soc/intel/boards/sof_sdw.c and the jack detect function will work as expected. However, when I look into the code of sound/soc/codecs/rt711.c and rt711-sdca.c, the default value for jd_src (https://github.com/torvalds/linux/blob/master/sound/soc/codecs/rt711.c#L1209) is already RT711_JD2, which means it is overridden by other drivers. After digging deeper, the jd_src value is overridden by rt711_add_codec_device_props() with the value RT711_JD1 comes from sof_sdw_quirk. Is there any reason why sof_sdw_quirk is assigned with RT711_JD1 by default? Can I simply fix it by assigning the sof_sdw_quirk to RT711_JD_NULL as follows? -unsigned long sof_sdw_quirk = RT711_JD1; +unsigned long sof_sdw_quirk = RT711_JD_NULL; Please let me know if there's any potential problem I didn't notice and suggest if there's any better solution. Thanks Chris

1 0

[PATCH] ASoC: atmel: add COMPILE_TEST for SND_ATMEL_SOC_PDC
by Randy Dunlap 13 Oct '21

13 Oct '21

Geert pointed out that since sound/soc has the soc_dummy_driver for NO_DMA platforms, it is possible (desirable) to have drivers that depend on HAS_DMA to alternately depend on COMPILE_TEST. This means that SND_ATMEL_SOC_PDC can depend on HAS_DMA || COMPIE_TEST. Fixes: 6c5c659dfe3f ("ASoC: atmel: ATMEL drivers don't need HAS_DMA") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Suggested-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/atmel/Kconfig | 1 + 1 file changed, 1 insertion(+) --- linux-next-20211001.orig/sound/soc/atmel/Kconfig +++ linux-next-20211001/sound/soc/atmel/Kconfig @@ -11,6 +11,7 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC bool + depends on HAS_DMA || COMPILE_TEST config SND_ATMEL_SOC_DMA bool

3 4

[PATCH 1/2] ASoC: dt-bindings: max98520: Add initial bindings
by George Song 13 Oct '21

13 Oct '21

Signed-off-by: George Song <george.song(a)analog.com> --- .../bindings/sound/maxim,max98520.yaml | 35 +++++++++++++++++++ 1 file changed, 35 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98520.yaml diff --git a/Documentation/devicetree/bindings/sound/maxim,max98520.yaml b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml new file mode 100644 index 000000000000..d1f2b5ee775a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98520.yaml @@ -0,0 +1,35 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98520.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98520 Speaker Amplifier Driver + +maintainers: + - George Song <George.song(a)analog.com> + +properties: + compatible: + const: maxim,max98520 + + reg: + maxItems: 1 + description: I2C address of the device. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98520: amplifier@38 { + compatible = "maxim,max98520"; + reg = <0x38>; + }; + }; -- 2.25.1

1 0

[PATCH v2] alsaloop: added option prateshift for PLAYSHIFT ctl elem used in PLAYSHIFT
by Pavel Hofman 13 Oct '21

13 Oct '21

If snd-aloop device is on playback side, the required sync mode is PLAYSHIFT. That means Loopback ctl elem "PCM Rate Shift 100000" of the corresponding capture side of the Loopback pipe must be controlled (by pitch reciprocal). ASCII name of the playback rate shift ctl elem is specified with newly added option -x/--prateshift, e.g.: -P hw:Loopback,0 -S playshift \ -x iface=PCM,name='PCM Rate Shift 100000',device=1 Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- alsaloop/alsaloop.1 | 11 ++++-- alsaloop/alsaloop.c | 11 +++++- alsaloop/alsaloop.h | 1 + alsaloop/pcmjob.c | 83 ++++++++++++++++++++++++++++++++++----------- 4 files changed, 83 insertions(+), 23 deletions(-) diff --git a/alsaloop/alsaloop.1 b/alsaloop/alsaloop.1 index 6bacfd5..006494d 100644 --- a/alsaloop/alsaloop.1 +++ b/alsaloop/alsaloop.1 @@ -57,6 +57,11 @@ Use given CTL device for playback. Use given CTL device for capture. +.TP +\fI\-x <ctl_ascii_name>\fP | \fI\-\-prateshift=<ctl_ascii_name>\fP + +Specify ctl ascii name for playshift sync mode (see the Examples section). + .TP \fI\-l <latency>\fP | \fI\-\-latency=<frames>\fP @@ -195,10 +200,12 @@ Verbose xrun profiling. Set process wake timeout. .SH EXAMPLES - -.TP +.nf \fBalsaloop \-C hw:0,0 \-P hw:1,0 \-t 50000\fR +\fBalsaloop \-C hw:soundcard,0 \-P hw:Loopback,0 \-t 50000 \-S playshift \\ +\-x iface=PCM,name='PCM Rate Shift 100000',device=1\fR +.ne .SH BUGS None known. .SH AUTHOR diff --git a/alsaloop/alsaloop.c b/alsaloop/alsaloop.c index 06ffadf..4192712 100644 --- a/alsaloop/alsaloop.c +++ b/alsaloop/alsaloop.c @@ -175,6 +175,7 @@ void help(void) "-C,--cdevice capture device\n" "-X,--pctl playback ctl device\n" "-Y,--cctl capture ctl device\n" +"-x,--prateshift playback 'PCM Rate Shift 100000' ascii ctl name\n" "-l,--latency requested latency in frames\n" "-t,--tlatency requested latency in usec (1/1000000sec)\n" "-f,--format sample format\n" @@ -362,6 +363,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, {"cdevice", 1, NULL, 'C'}, {"pctl", 1, NULL, 'X'}, {"cctl", 1, NULL, 'Y'}, + {"prateshift", 1, NULL, 'x'}, {"latency", 1, NULL, 'l'}, {"tlatency", 1, NULL, 't'}, {"format", 1, NULL, 'f'}, @@ -391,6 +393,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, char *arg_cdevice = NULL; char *arg_pctl = NULL; char *arg_cctl = NULL; + char *arg_prateshift = NULL; unsigned int arg_latency_req = 0; unsigned int arg_latency_reqtime = 10000; snd_pcm_format_t arg_format = SND_PCM_FORMAT_S16_LE; @@ -420,7 +423,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, while (1) { int c; if ((c = getopt_long(argc, argv, - "hdg:P:C:X:Y:l:t:F:f:c:r:s:benvA:S:a:m:T:O:w:UW:z", + "hdg:P:C:X:Y:x:l:t:F:f:c:r:s:benvA:S:a:m:T:O:w:UW:z", long_option, NULL)) < 0) break; switch (c) { @@ -446,6 +449,9 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, case 'Y': arg_cctl = strdup(optarg); break; + case 'x': + arg_prateshift = strdup(optarg); + break; case 'l': err = atoi(optarg); arg_latency_req = err >= 4 ? err : 4; @@ -627,6 +633,9 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, logit(LOG_CRIT, "Unable to add ossmixer controls.\n"); exit(EXIT_FAILURE); } + if (arg_prateshift) + play->prateshift_name = arg_prateshift; + #ifdef USE_SAMPLERATE loop->src_enable = arg_samplerate > 0; if (loop->src_enable) diff --git a/alsaloop/alsaloop.h b/alsaloop/alsaloop.h index 1dbcefe..7a98ef3 100644 --- a/alsaloop/alsaloop.h +++ b/alsaloop/alsaloop.h @@ -127,6 +127,7 @@ struct loopback_handle { snd_ctl_elem_value_t *ctl_format; snd_ctl_elem_value_t *ctl_rate; snd_ctl_elem_value_t *ctl_channels; + char *prateshift_name; /* ascii name for the playback rate shift ctl elem */ }; struct loopback { diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 845ab82..8658fa6 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1101,7 +1101,8 @@ void update_pitch(struct loopback *loop) #endif } else if (loop->sync == SYNC_TYPE_PLAYRATESHIFT) { - set_rate_shift(loop->play, pitch); + // pitch is capture-based, playback side requires reciprocal + set_rate_shift(loop->play, 1 / pitch); #ifdef USE_SAMPLERATE if (loop->use_samplerate) { loop->src_data.src_ratio = @@ -1172,44 +1173,86 @@ static int get_channels(struct loopback_handle *lhandle) return snd_ctl_elem_value_get_integer(lhandle->ctl_channels, 0); } -static void openctl_elem(struct loopback_handle *lhandle, - int device, int subdevice, - const char *name, - snd_ctl_elem_value_t **elem) +static int openctl_elem_id(struct loopback_handle *lhandle, snd_ctl_elem_id_t *id, + snd_ctl_elem_value_t **elem) { int err; if (snd_ctl_elem_value_malloc(elem) < 0) { *elem = NULL; + snd_ctl_elem_id_free(id); + return -ENOMEM; + } + snd_ctl_elem_value_set_id(*elem, id); + snd_ctl_elem_value_set_interface(*elem, SND_CTL_ELEM_IFACE_PCM); + err = snd_ctl_elem_read(lhandle->ctl, *elem); + if (err < 0) { + snd_ctl_elem_id_free(id); + snd_ctl_elem_value_free(*elem); + *elem = NULL; + return err; } else { - snd_ctl_elem_value_set_interface(*elem, - SND_CTL_ELEM_IFACE_PCM); - snd_ctl_elem_value_set_device(*elem, device); - snd_ctl_elem_value_set_subdevice(*elem, subdevice); - snd_ctl_elem_value_set_name(*elem, name); - err = snd_ctl_elem_read(lhandle->ctl, *elem); - if (err < 0) { - snd_ctl_elem_value_free(*elem); - *elem = NULL; - } else { - if (verbose) - snd_output_printf(lhandle->loopback->output, - "Opened PCM element %s of %s, device %d, subdevice %d\n", - name, snd_ctl_name(lhandle->ctl), device, subdevice); - } + snd_output_printf(lhandle->loopback->output, + "Opened PCM element %s of %s, device %d, subdevice %d\n", + snd_ctl_elem_id_get_name(id), snd_ctl_name(lhandle->ctl), + snd_ctl_elem_id_get_device(id), + snd_ctl_elem_id_get_subdevice(id)); + return 0; } } +static int openctl_elem(struct loopback_handle *lhandle, + int device, int subdevice, + const char *name, + snd_ctl_elem_value_t **elem) +{ + snd_ctl_elem_id_t *id; + + if (snd_ctl_elem_id_malloc(&id)) + return -ENOMEM; + snd_ctl_elem_id_set_device(id, device); + snd_ctl_elem_id_set_subdevice(id, subdevice); + snd_ctl_elem_id_set_name(id, name); + return openctl_elem_id(lhandle, id, elem); +} + +static int openctl_elem_ascii(struct loopback_handle *lhandle, char *ascii_name, + snd_ctl_elem_value_t **elem) +{ + snd_ctl_elem_id_t *id; + + if (snd_ctl_elem_id_malloc(&id)) { + return -ENOMEM; + } + if (snd_ctl_ascii_elem_id_parse(id, ascii_name)) { + snd_ctl_elem_id_free(id); + fprintf(stderr, "Wrong control identifier: %s\n", ascii_name); + return -EINVAL; + } + return openctl_elem_id(lhandle, id, elem); +} + static int openctl(struct loopback_handle *lhandle, int device, int subdevice) { int err; lhandle->ctl_rate_shift = NULL; if (lhandle->loopback->play == lhandle) { + // play only + if (lhandle->prateshift_name) { + err = openctl_elem_ascii(lhandle, lhandle->prateshift_name, + &lhandle->ctl_rate_shift); + if (err < 0) { + logit(LOG_CRIT, "Unable to open playback PCM Rate Shift elem '%s'.\n", + lhandle->prateshift_name); + exit(EXIT_FAILURE); + } + } if (lhandle->loopback->controls) goto __events; return 0; } + // capture only openctl_elem(lhandle, device, subdevice, "PCM Notify", &lhandle->ctl_notify); openctl_elem(lhandle, device, subdevice, "PCM Rate Shift 100000", -- 2.25.1

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