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

[PATCH 00/13] ALSA: hda: Use LED classdev for mute controls consistently
by Takashi Iwai 18 Jun '20

18 Jun '20

Hi, this is a patch set inspired by the recent patch Kai-Heng posted about the HD-audio mic-mute LED control. Currently HD-audio driver deals with the mute and mic-mute LED in several different ways: primarily with the direct callback of vmaster hook and capture sync hook, while another with the LED class device binding. The latter has been used for binding with the platform device LEDs like Thinkpad, Dell, Huawei. And, yet, recently we added our own LED classdev for the mic-mute LED on some HP systems although they are controlled directly with the callback; it's exposed, however, for the DMIC that is governed by a different ASoC driver. This patch set is an attempt to sort out and make them consistent: namely, * All LEDs are now controlled via LED class device * The generic driver provides helper functions to easily build up the LED class dev and the relevant mixer controls * Conversion of the existing framework and clean ups The patches are lightly tested in my side with a couple of machines and also through hda-emu tests. Some devices receive new kcontrols for the mute LED behavior (that have been missing so far), but anything else look good though my tests. thanks, Takashi === Takashi Iwai (13): ALSA: hda: generic: Always call led-trigger for mic mute LED ALSA: hda: generic: Add a helper for mic-mute LED with LED classdev ALSA: hda/realtek: Convert to cdev-variant of mic-mute LED controls ALSA: hda/conexant: Convert to cdev-variant of mic-mute LED controls ALSA: hda/sigmatel: Convert to cdev-variant of mic-mute LED controls ALSA: hda: generic: Drop unused snd_hda_gen_fixup_micmute_led() ALSA: hda: generic: Drop the old mic-mute LED hook ALSA: hda: generic: Add vmaster mute LED helper ALSA: hda/realtek: Use the new vmaster mute LED helper ALSA: hda/conexant: Use the new vmaster mute LED helper ALSA: hda/sigmatel: Use the new vmaster mute LED helper ALSA: hda/realtek: Unify LED helper code ALSA: hda: Let LED cdev handling suspend/resume sound/pci/hda/Kconfig | 9 ++ sound/pci/hda/hda_generic.c | 134 ++++++++++++++++-------- sound/pci/hda/hda_generic.h | 13 ++- sound/pci/hda/patch_conexant.c | 47 ++++----- sound/pci/hda/patch_realtek.c | 220 +++++++++++++++++++--------------------- sound/pci/hda/patch_sigmatel.c | 22 ++-- sound/pci/hda/thinkpad_helper.c | 19 +--- 7 files changed, 248 insertions(+), 216 deletions(-) -- 2.16.4

1 13

[PATCH v3 1/2] ALSA: hda/realtek: Add COEF controlled micmute LED support
by Kai-Heng Feng 18 Jun '20

18 Jun '20

Currently, HDA codec LED class can only be used 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. 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> --- v3: - Wording. - Properly prefix exported symbol. v2: - Prevent platforms like Dell, Lenovoe and Huawei create double LED class devices. sound/pci/hda/hda_generic.c | 7 ++++--- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/patch_realtek.c | 29 ++++++++++++++++------------- 3 files changed, 21 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f4e9d9445e18..e4f534e9a88c 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 snd_hda_gen_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(snd_hda_gen_call_micmute_led_update); static void update_micmute_led(struct hda_codec *codec, struct snd_kcontrol *kcontrol, @@ -3945,7 +3946,7 @@ static void update_micmute_led(struct hda_codec *codec, spec->micmute_led.capture |= mask; else spec->micmute_led.capture &= ~mask; - call_micmute_led_update(codec); + snd_hda_gen_call_micmute_led_update(codec); } } @@ -3982,7 +3983,7 @@ static int micmute_led_mode_put(struct snd_kcontrol *kcontrol, if (mode == spec->micmute_led.led_mode) return 0; spec->micmute_led.led_mode = mode; - call_micmute_led_update(codec); + snd_hda_gen_call_micmute_led_update(codec); return 1; } diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index fb9f1a90238b..062be242339a 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 snd_hda_gen_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..0587d1e96b19 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; + snd_hda_gen_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

2 11

[RFC PATCH 0/2] TAS2563 DSP Firmware Loader
by Dan Murphy 18 Jun '20

18 Jun '20

Hello The TAS2563 amplifier has a DSP that can run programs and configurations to produce alternate audio experiences. The DSP firmware is not a typical firmware as the binary may contain various programs and configurations that are selectable during run time. These programs and configurations are selectable via files under the I2C dev node. There may be a better way to select this through ALSA controls but I was unable to find a good example of this. This is why this is an RFC patchset. Dan Dan Murphy (2): dt-bindings: tas2562: Add firmware support for tas2563 ASoc: tas2563: DSP Firmware loading support .../devicetree/bindings/sound/tas2562.yaml | 4 + sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/tas2562.c | 48 ++- sound/soc/codecs/tas2562.h | 25 ++ sound/soc/codecs/tas25xx_dsp_loader.c | 377 ++++++++++++++++++ sound/soc/codecs/tas25xx_dsp_loader.h | 93 +++++ 6 files changed, 530 insertions(+), 19 deletions(-) create mode 100644 sound/soc/codecs/tas25xx_dsp_loader.c create mode 100644 sound/soc/codecs/tas25xx_dsp_loader.h -- 2.26.2

3 19

File Plugin seems to drop data when writing to FIFO
by GitHub issues - opened 18 Jun '20

18 Jun '20

alsa-project/alsa-lib issue #63 was opened from riggiding: Hi, we're using the file plugin to pipe data for use in a vu-meter application. This worked well but since updating to 1.1.4.1 the data received is incomplete (tested using sine test tones and Audacity to playback / inspect the data that went through the fifo). **When writing to a normal file all is fine.** Also tried using tee, same result. Any hints or ideas would be great! **The fifo is created by:** #! /bin/sh if [ ! -p "$ALSA_FIFO" ]; then # create a fifo for alsa -> vumeter mkfifo "$ALSA_FIFO" chmod 666 "$ALSA_FIFO" fi cat <>"$ALSA_FIFO" **asound.conf:** pcm.hiface { type hw card 0 device 0 } pcm.vumeter { type rate slave { pcm "alsaFifo" rate 44100 format "S16_LE" } } pcm.alsaFifo { type file slave.pcm "null" file "/tmp/alsa-fifo" } pcm.alsaFifoTee { type empty slave.pcm "tee:null,'/tmp/alsa-fifo',raw" } pcm.direct { type route slave.pcm "split_direct" ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 } pcm.split_direct { type multi slaves.a.pcm "hiface" slaves.a.channels 2 slaves.b.pcm "vumeter" slaves.b.channels 2 bindings.0.slave a bindings.0.channel 0 bindings.1.slave a bindings.1.channel 1 bindings.2.slave b bindings.2.channel 0 bindings.3.slave b bindings.3.channel 1 } Issue URL : https://github.com/alsa-project/alsa-lib/issues/63 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: hda: Intel: add missing PCI IDs for ICL-H, TGL-H and EKL
by Pierre-Louis Bossart 18 Jun '20

18 Jun '20

Mirror PCI ids used for SOF. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d20aedd103c6..3565e2ab0965 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2470,6 +2470,9 @@ static const struct pci_device_id azx_ids[] = { /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Icelake-H */ + { PCI_DEVICE(0x8086, 0x3dc8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Jasperlake */ { PCI_DEVICE(0x8086, 0x38c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, @@ -2478,9 +2481,14 @@ static const struct pci_device_id azx_ids[] = { /* Tigerlake */ { PCI_DEVICE(0x8086, 0xa0c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Tigerlake-H */ + { PCI_DEVICE(0x8086, 0x43c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4b58), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, -- 2.20.1

2 1

[PATCH 1/2] ASoC: amd: Removing unnecessary instance initialization
by Ravulapati Vishnu vardhan rao 18 Jun '20

18 Jun '20

In trigger we already get the selected instance details from runtime->private_data.So, removing the local initialization which may corrupt the instance selected details and this leads to corrupt data. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-i2s.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index a532e01a2622..14607563abd2 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -151,20 +151,12 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, struct i2s_stream_instance *rtd; struct snd_soc_pcm_runtime *prtd; struct snd_soc_card *card; - struct acp3x_platform_info *pinfo; u32 ret, val, period_bytes, reg_val, ier_val, water_val; u32 buf_size, buf_reg; prtd = substream->private_data; rtd = substream->runtime->private_data; card = prtd->card; - pinfo = snd_soc_card_get_drvdata(card); - if (pinfo) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - rtd->i2s_instance = pinfo->play_i2s_instance; - else - rtd->i2s_instance = pinfo->cap_i2s_instance; - } period_bytes = frames_to_bytes(substream->runtime, substream->runtime->period_size); buf_size = frames_to_bytes(substream->runtime, -- 2.17.1

1 0

[PATCH AUTOSEL 4.4 52/60] ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failed
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit 36124fb19f1ae68a500cd76a76d40c6e81bee346 ] fsl_asrc_dma_hw_params() invokes dma_request_channel() or fsl_asrc_get_dma_channel(), which returns a reference of the specified dma_chan object to "pair->dma_chan[dir]" with increased refcnt. The reference counting issue happens in one exception handling path of fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End, the function forgets to decrease the refcnt increased by dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when config DMA channel failed. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Link: https://lore.kernel.org/r/1590415966-52416-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_asrc_dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index ffc000bc1f15..56a873ba08e4 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -243,6 +243,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); if (ret) { dev_err(dev, "failed to config DMA channel for Back-End\n"); + dma_release_channel(pair->dma_chan[dir]); return ret; } -- 2.25.1

1 0

[PATCH AUTOSEL 4.4 24/60] ALSA: usb-audio: Improve frames size computation
by Sasha Levin 18 Jun '20

18 Jun '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit f0bd62b64016508938df9babe47f65c2c727d25c ] For computation of the the next frame size current value of fs/fps and accumulated fractional parts of fs/fps are used, where values are stored in Q16.16 format. This is quite natural for computing frame size for asynchronous endpoints driven by explicit feedback, since in this case fs/fps is a value provided by the feedback endpoint and it's already in the Q format. If an error is accumulated over time, the device can adjust fs/fps value to prevent buffer overruns/underruns. But for synchronous endpoints the accuracy provided by these computations is not enough. Due to accumulated error the driver periodically produces frames with incorrect size (+/- 1 audio sample). This patch fixes this issue by implementing a different algorithm for frame size computation. It is based on accumulating of the remainders from division fs/fps and it doesn't accumulate errors over time. This new method is enabled for synchronous and adaptive playback endpoints. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 4 ++++ sound/usb/endpoint.c | 43 ++++++++++++++++++++++++++++++++++++++----- sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 2 ++ 4 files changed, 45 insertions(+), 5 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 71778ca4b26a..844c68863810 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -80,6 +80,10 @@ struct snd_usb_endpoint { dma_addr_t sync_dma; /* DMA address of syncbuf */ unsigned int pipe; /* the data i/o pipe */ + unsigned int framesize[2]; /* small/large frame sizes in samples */ + unsigned int sample_rem; /* remainder from division fs/fps */ + unsigned int sample_accum; /* sample accumulator */ + unsigned int fps; /* frames per second */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ int freqshift; /* how much to shift the feedback value to get Q16.16 */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 66648b4bdd28..666731317b33 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -137,12 +137,12 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep) /* * For streaming based on information derived from sync endpoints, - * prepare_outbound_urb_sizes() will call next_packet_size() to + * prepare_outbound_urb_sizes() will call slave_next_packet_size() to * determine the number of samples to be sent in the next packet. * - * For implicit feedback, next_packet_size() is unused. + * For implicit feedback, slave_next_packet_size() is unused. */ -int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -159,6 +159,29 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) return ret; } +/* + * For adaptive and synchronous endpoints, prepare_outbound_urb_sizes() + * will call next_packet_size() to determine the number of samples to be + * sent in the next packet. + */ +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +{ + int ret; + + if (ep->fill_max) + return ep->maxframesize; + + ep->sample_accum += ep->sample_rem; + if (ep->sample_accum >= ep->fps) { + ep->sample_accum -= ep->fps; + ret = ep->framesize[1]; + } else { + ret = ep->framesize[0]; + } + + return ret; +} + static void retire_outbound_urb(struct snd_usb_endpoint *ep, struct snd_urb_ctx *urb_ctx) { @@ -203,6 +226,8 @@ static void prepare_silent_urb(struct snd_usb_endpoint *ep, if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); @@ -879,10 +904,17 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->maxpacksize = fmt->maxpacksize; ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); - if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) { ep->freqn = get_usb_full_speed_rate(rate); - else + ep->fps = 1000; + } else { ep->freqn = get_usb_high_speed_rate(rate); + ep->fps = 8000; + } + + ep->sample_rem = rate % ep->fps; + ep->framesize[0] = rate / ep->fps; + ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps; /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; @@ -941,6 +973,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; + ep->sample_accum = 0; snd_usb_endpoint_start_quirk(ep); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 584f295d7c77..4aad49cbeb5f 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -27,6 +27,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct snd_usb_endpoint *ep); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f84c55ecd0fb..53d91cae86f9 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1473,6 +1473,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); -- 2.25.1

1 0

[PATCH AUTOSEL 4.4 05/60] ALSA: isa/wavefront: prevent out of bounds write in ioctl
by Sasha Levin 18 Jun '20

18 Jun '20

From: Dan Carpenter <dan.carpenter(a)oracle.com> [ Upstream commit 7f0d5053c5a9d23fe5c2d337495a9d79038d267b ] The "header->number" comes from the ioctl and it needs to be clamped to prevent out of bounds writes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/r/20200501094011.GA960082@mwanda Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/wavefront/wavefront_synth.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 718d5e3b7806..6c06d0645779 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1174,7 +1174,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) "alias for %d\n", header->number, header->hdr.a.OriginalSample); - + + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + munge_int32 (header->number, &alias_hdr[0], 2); munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2); munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset), @@ -1205,6 +1208,9 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) int num_samples; unsigned char *msample_hdr; + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL); if (! msample_hdr) return -ENOMEM; -- 2.25.1

1 0

[PATCH AUTOSEL 4.9 69/80] ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failed
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit 36124fb19f1ae68a500cd76a76d40c6e81bee346 ] fsl_asrc_dma_hw_params() invokes dma_request_channel() or fsl_asrc_get_dma_channel(), which returns a reference of the specified dma_chan object to "pair->dma_chan[dir]" with increased refcnt. The reference counting issue happens in one exception handling path of fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End, the function forgets to decrease the refcnt increased by dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when config DMA channel failed. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Link: https://lore.kernel.org/r/1590415966-52416-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_asrc_dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index dc30d780f874..3fcf174b99d3 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -243,6 +243,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); if (ret) { dev_err(dev, "failed to config DMA channel for Back-End\n"); + dma_release_channel(pair->dma_chan[dir]); return ret; } -- 2.25.1

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