June 2020 - Alsa-devel - mailman.alsa-project.org

pcm: direct: correctly apply existing interval settings
by GitHub pull_request - edited 17 Jun '20

17 Jun '20

alsa-project/alsa-lib pull request #62 was edited from aditpape: Feature 'variable periodsize' allows to extend user period size up to buffer_size/2 independent of slave period. On enlargement of the settings for period_time.max and period_size.max the setting for openmax was not updated. This lead to the effect, that if the slave period itself had openmax set it was still set on the extended size. Configuration of a period matching half buffer size was thus rejected. Example for failure: period size of 384 (half buffer size) is requested which is rejected and rounded down to 352: <pre> root@oracle-virtualbox:~# arecord -DDSNOOP_32 -r48000 -fS16_LE -c2 --period-size=382 -vv --dump-hw-params > /dev/null Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo HW Params of device "DSNOOP_32": -------------------- ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: (666 7334) PERIOD_SIZE: [32 352] PERIOD_BYTES: [128 1408] PERIODS: [2 24] BUFFER_TIME: (1333 16000] BUFFER_SIZE: [64 768] BUFFER_BYTES: [256 3072] TICK_TIME: ALL -------------------- Direct Snoop PCM Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 704 period_size : 352 period_time : 7333 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 352 period_event : 0 start_threshold : 1 stop_threshold : 704 silence_threshold: 0 silence_size : 0 boundary : 6341068275337658368 Hardware PCM card 0 'Intel 82801AA-ICH' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 768 period_size : 32 period_time : 666 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 32 period_event : 0 start_threshold : 1 stop_threshold : 6917529027641081856 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 appl_ptr : 0 hw_ptr : 224 When correctly applying the openmax setting: root@oracle-virtualbox:~# arecord -DDSNOOP_32 -r48000 -fS16_LE -c2 --period-size=382 -vv --dump-hw-params > /dev/null Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo HW Params of device "DSNOOP_32": -------------------- ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: (666 8000] PERIOD_SIZE: [32 384] PERIOD_BYTES: [128 1536] PERIODS: [2 24] BUFFER_TIME: (1333 16000] BUFFER_SIZE: [64 768] BUFFER_BYTES: [256 3072] TICK_TIME: ALL -------------------- Direct Snoop PCM Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 768 period_size : 384 period_time : 8000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 384 period_event : 0 start_threshold : 1 stop_threshold : 768 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 Hardware PCM card 0 'Intel 82801AA-ICH' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 768 period_size : 32 period_time : 666 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 32 period_event : 0 start_threshold : 1 stop_threshold : 6917529027641081856 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 appl_ptr : 0 hw_ptr : 224 </pre> Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/62 Patch URL : https://github.com/alsa-project/alsa-lib/pull/62.patch Repository URL: https://github.com/alsa-project/alsa-lib

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pcm: direct: correctly apply existing interval settings
by GitHub pull_request - opened 17 Jun '20

17 Jun '20

alsa-project/alsa-lib pull request #62 was opened from aditpape: Feature 'variable periodsize' allows to extend user period size up to buffer_size/2 independent of slave period. On enlargement of the settings for period_time.max and period_size.max the setting for openmax was not updated. This lead to the effect, that if the slave period itself had openmax set it was still set on the extended size. Configuration of a period matching half buffer size was thus rejected. Example for failure: period size of 384 (half buffer size) is requested which is rejected and rounded down to 352: root@oracle-virtualbox:~# arecord -DDSNOOP_32 -r48000 -fS16_LE -c2 --period-size=382 -vv --dump-hw-params > /dev/null Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo HW Params of device "DSNOOP_32": -------------------- ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: (666 7334) PERIOD_SIZE: [32 352] PERIOD_BYTES: [128 1408] PERIODS: [2 24] BUFFER_TIME: (1333 16000] BUFFER_SIZE: [64 768] BUFFER_BYTES: [256 3072] TICK_TIME: ALL -------------------- Direct Snoop PCM Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 704 period_size : 352 period_time : 7333 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 352 period_event : 0 start_threshold : 1 stop_threshold : 704 silence_threshold: 0 silence_size : 0 boundary : 6341068275337658368 Hardware PCM card 0 'Intel 82801AA-ICH' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 768 period_size : 32 period_time : 666 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 32 period_event : 0 start_threshold : 1 stop_threshold : 6917529027641081856 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 appl_ptr : 0 hw_ptr : 224 When correctly applying the openmax setting: root@oracle-virtualbox:~# arecord -DDSNOOP_32 -r48000 -fS16_LE -c2 --period-size=382 -vv --dump-hw-params > /dev/null Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo HW Params of device "DSNOOP_32": -------------------- ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: (666 8000] PERIOD_SIZE: [32 384] PERIOD_BYTES: [128 1536] PERIODS: [2 24] BUFFER_TIME: (1333 16000] BUFFER_SIZE: [64 768] BUFFER_BYTES: [256 3072] TICK_TIME: ALL -------------------- Direct Snoop PCM Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 768 period_size : 384 period_time : 8000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 384 period_event : 0 start_threshold : 1 stop_threshold : 768 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 Hardware PCM card 0 'Intel 82801AA-ICH' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 768 period_size : 32 period_time : 666 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 32 period_event : 0 start_threshold : 1 stop_threshold : 6917529027641081856 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 appl_ptr : 0 hw_ptr : 224 Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/62 Patch URL : https://github.com/alsa-project/alsa-lib/pull/62.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2 1/2] ALSA: hda/realtek: Add COEF controlled micmute LED support
by Kai-Heng Feng 17 Jun '20

17 Jun '20

Currently, HDA codec LED class can only used by by GPIO controlled LED. However, there are some new systems that control LED via COEF instead of GPIO. In order to support those systems, create a new helper that can be facilitated by both COEF controlled and GPIO controlled LED, and use generic call_micmute_led_update() to control both case. In addition to that, add LED_CORE_SUSPENDRESUME flag since some systems don't restore the LED properly after suspend. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v2: - Prevent platforms like Dell, Lenovoe and Huawei create double LED class devices. sound/pci/hda/hda_generic.c | 3 ++- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/patch_realtek.c | 29 ++++++++++++++++------------- 3 files changed, 19 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f4e9d9445e18..9c3f1990c621 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3897,7 +3897,7 @@ enum { MICMUTE_LED_FOLLOW_MUTE, }; -static void call_micmute_led_update(struct hda_codec *codec) +void call_micmute_led_update(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; unsigned int val; @@ -3924,6 +3924,7 @@ static void call_micmute_led_update(struct hda_codec *codec) if (spec->micmute_led.update) spec->micmute_led.update(codec); } +EXPORT_SYMBOL_GPL(call_micmute_led_update); static void update_micmute_led(struct hda_codec *codec, struct snd_kcontrol *kcontrol, diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index fb9f1a90238b..9f08a9ee9112 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -353,6 +353,7 @@ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on); int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin); +void call_micmute_led_update(struct hda_codec *codec); int snd_hda_gen_add_micmute_led(struct hda_codec *codec, void (*hook)(struct hda_codec *)); void snd_hda_gen_fixup_micmute_led(struct hda_codec *codec, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d73f8beadb6..9691af8241cf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4114,10 +4114,10 @@ static int micmute_led_set(struct led_classdev *led_cdev, enum led_brightness brightness) { struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent); - struct alc_spec *spec = codec->spec; + struct hda_gen_spec *spec = codec->spec; - alc_update_gpio_led(codec, spec->gpio_mic_led_mask, - spec->micmute_led_polarity, !!brightness); + spec->micmute_led.led_mode = !brightness; + call_micmute_led_update(codec); return 0; } @@ -4126,7 +4126,17 @@ static struct led_classdev micmute_led_cdev = { .max_brightness = 1, .brightness_set_blocking = micmute_led_set, .default_trigger = "audio-micmute", + .flags = LED_CORE_SUSPENDRESUME, }; + +static void alc_register_micmute_led(struct hda_codec *codec) +{ + micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + if (devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev)) + codec_warn(codec, "failed to register micmute LED\n"); +} +#else +static inline void alc_register_micmute_led(struct hda_codec *codec) {} #endif /* setup mute and mic-mute GPIO bits, add hooks appropriately */ @@ -4136,9 +4146,6 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, unsigned int micmute_mask) { struct alc_spec *spec = codec->spec; -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) - int err; -#endif alc_fixup_gpio(codec, action, mute_mask | micmute_mask); @@ -4151,13 +4158,7 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, if (micmute_mask) { spec->gpio_mic_led_mask = micmute_mask; snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); - -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) - micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); - err = devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev); - if (err) - codec_warn(codec, "failed to register micmute LED\n"); -#endif + alc_register_micmute_led(codec); } } @@ -4305,6 +4306,7 @@ static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec, spec->mic_led_coefbit_on = 1<<13; spec->mic_led_coefbit_off = 0; snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update); + alc_register_micmute_led(codec); } } @@ -4319,6 +4321,7 @@ static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec, spec->mic_led_coefbit_on = 2<<2; spec->mic_led_coefbit_off = 1<<2; snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update); + alc_register_micmute_led(codec); } } -- 2.17.1

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[PATCH v2 1/2] ASoC: bindings: fsl_spdif: Add new compatible string for imx6sx
by Shengjiu Wang 17 Jun '20

17 Jun '20

Add new compatible string "fsl,imx6sx-spdif" in the binding document. And add compatible string "fsl,vf610-spdif" which was missed before. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.txt | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt index 8b324f82a782..e1365b0ee1e9 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt @@ -6,7 +6,11 @@ a fibre cable. Required properties: - - compatible : Compatible list, must contain "fsl,imx35-spdif". + - compatible : Compatible list, should contain one of the following + compatibles: + "fsl,imx35-spdif", + "fsl,vf610-spdif", + "fsl,imx6sx-spdif", - reg : Offset and length of the register set for the device. -- 2.21.0

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alsa-gobject: timer/seq: use const qualifier to self argument for getter API
by GitHub pull_request - opened 17 Jun '20

17 Jun '20

alsa-project/alsa-gobject pull request #40 was opened from takaswie: In ALSATimer and ALSASeq, some methods have prototype such that self argument don't have const qualifier even if it's getter method. This patchset adds const qualifier to the methods. Request URL : https://github.com/alsa-project/alsa-gobject/pull/40 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/40.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH 1/2] ASoC: bindings: fsl_spdif: Add new compatible string for imx6sx
by Shengjiu Wang 17 Jun '20

17 Jun '20

Add new compatible string "fsl,imx6sx-spdif" in the binding document. And add compatible string "fsl,vf610-spdif" which was missed before. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.txt | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt index 8b324f82a782..e1365b0ee1e9 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt @@ -6,7 +6,11 @@ a fibre cable. Required properties: - - compatible : Compatible list, must contain "fsl,imx35-spdif". + - compatible : Compatible list, should contain one of the following + compatibles: + "fsl,imx35-spdif", + "fsl,vf610-spdif", + "fsl,imx6sx-spdif", - reg : Offset and length of the register set for the device. -- 2.21.0

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[PATCH 1/2] ASoC: bindings: fsl-asoc-card: Add compatible string for MQS
by Shengjiu Wang 17 Jun '20

17 Jun '20

Add compatible string "fsl,imx-audio-mqs" for MQS Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl-asoc-card.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index c60a5732d29c..6417c5dffa9a 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -34,6 +34,8 @@ The compatible list for this generic sound card currently: "fsl,imx-audio-wm8960" + "fsl,imx-audio-mqs" + Required properties: - compatible : Contains one of entries in the compatible list. -- 2.21.0

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[PATCH alsa-lib] control: ctlparse - use type-specific bound on element count
by John Keeping 16 Jun '20

16 Jun '20

Using a fixed bound of 128 means that too many values may be set for an INTEGER64 type and that any elements past 128 are out of reach for BYTE type controls. Derive the maximum number of elements from the type so that the full range is parsed for all types. Signed-off-by: John Keeping <john(a)metanate.com> --- src/control/ctlparse.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index 1ac31bb9..ee1e0602 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -304,6 +304,25 @@ static int get_ctl_enum_item_index(snd_ctl_t *handle, return -1; } +static unsigned int get_ctl_type_max_elements(snd_ctl_elem_type_t type) +{ + struct snd_ctl_elem_value value; + + switch (type) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + case SND_CTL_ELEM_TYPE_INTEGER: + return ARRAY_SIZE(value.value.integer.value); + case SND_CTL_ELEM_TYPE_INTEGER64: + return ARRAY_SIZE(value.value.integer64.value); + case SND_CTL_ELEM_TYPE_ENUMERATED: + return ARRAY_SIZE(value.value.enumerated.item); + case SND_CTL_ELEM_TYPE_BYTES: + return ARRAY_SIZE(value.value.bytes.data); + default: + return 0; + } +} + /** * \brief parse ASCII string as CTL element value * \param handle CTL handle @@ -331,8 +350,11 @@ int snd_ctl_ascii_value_parse(snd_ctl_t *handle, type = snd_ctl_elem_info_get_type(info); count = snd_ctl_elem_info_get_count(info); snd_ctl_elem_value_set_id(dst, &myid); + + if (count > get_ctl_type_max_elements(type)) + count = get_ctl_type_max_elements(type); - for (idx = 0; idx < count && idx < 128 && ptr && *ptr; idx++) { + for (idx = 0; idx < count && ptr && *ptr; idx++) { if (*ptr == ',') goto skip; switch (type) { -- 2.27.0

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[RESENDx2][PATCH] sound: qcom: Kconfig: Tweak dependencies on SND_SOC_SDM845
by John Stultz 16 Jun '20

16 Jun '20

CROS_EC isn't strictly required for audio to work on other SDM845 platforms (like the Dragonboard 845c). So lets remove the dependency and select the related CROS_EC options if CROS_EC is already enabled. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Rohit kumar <rohitkr(a)codeaurora.org> Cc: Patrick Lai <plai(a)codeaurora.org> Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: John Stultz <john.stultz(a)linaro.org> --- sound/soc/qcom/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index f51b28d1b94d..0434f2c7d746 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -99,12 +99,12 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" - depends on QCOM_APR && CROS_EC && I2C && SOUNDWIRE + depends on QCOM_APR && I2C && SOUNDWIRE select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 select SND_SOC_MAX98927 - select SND_SOC_CROS_EC_CODEC + select SND_SOC_CROS_EC_CODEC if CROS_EC help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. -- 2.17.1

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pcm: dmic: assembly: add x32 support.
by GitHub pull_request - opened 16 Jun '20

16 Jun '20

alsa-project/alsa-lib pull request #61 was opened from hvdijk: x32 is the x86_64 ABI that uses 32-bit pointers, so requires loading addresses into edi/esi/ebx rather than rdi/rsi/rbx. Note that instructions such as movl %eax, (%rdi) do not require updating, as loading an address into %edi zeroes the high bits, causing the full %rdi register to hold a valid address. Signed-off-by: Harald van Dijk <harald(a)gigawatt.nl> Without this, even aplay segfaults when building with `-mx32`. Request URL : https://github.com/alsa-project/alsa-lib/pull/61 Patch URL : https://github.com/alsa-project/alsa-lib/pull/61.patch Repository URL: https://github.com/alsa-project/alsa-lib

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