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

[PATCH v2 0/8] Add RZ/G2L Sound support
by Biju Das 02 Aug '21

02 Aug '21

This patch series aims to add ASoC support on RZ/G2L SoC's. It is based on the work done by Chris Brandt for RZ/A ASoC driver. Biju Das (8): ASoC: dt-bindings: Document RZ/G2L bindings sound: soc: sh: Add RZ/G2L SSIF-2 driver arm64: dts: renesas: r9a07g044: Add external audio clock nodes arm64: dts: renesas: r9a07g044: Add SSI support arm64: defconfig: Enable ASoC sound support for RZ/G2L SoC ASoC: dt-bindings: sound: renesas,rz-ssi: Document DMA support sound: sh: rz-ssi: Add SSI DMAC support arm64: dts: renesas: r9a07g044: Add SSI DMA support .../bindings/sound/renesas,rz-ssi.yaml | 100 ++ arch/arm64/boot/dts/renesas/r9a07g044.dtsi | 105 ++ arch/arm64/configs/defconfig | 1 + sound/soc/sh/Kconfig | 10 + sound/soc/sh/Makefile | 4 + sound/soc/sh/rz-ssi.c | 1078 +++++++++++++++++ 6 files changed, 1298 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml create mode 100644 sound/soc/sh/rz-ssi.c -- 2.17.1

6 15

[PATCH] ASoC: intel: skylake: Drop superfluous mmap callback
by Takashi Iwai 02 Aug '21

02 Aug '21

skl_platform_soc_mmap() just calls the standard mmap helper, hence it's superfluous. Let's drop it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/skylake/skl-pcm.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index b1ca64d2f7ea..c604200de80e 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1214,13 +1214,6 @@ static snd_pcm_uframes_t skl_platform_soc_pointer( return bytes_to_frames(substream->runtime, pos); } -static int skl_platform_soc_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *area) -{ - return snd_pcm_lib_default_mmap(substream, area); -} - static u64 skl_adjust_codec_delay(struct snd_pcm_substream *substream, u64 nsec) { @@ -1460,7 +1453,6 @@ static const struct snd_soc_component_driver skl_component = { .trigger = skl_platform_soc_trigger, .pointer = skl_platform_soc_pointer, .get_time_info = skl_platform_soc_get_time_info, - .mmap = skl_platform_soc_mmap, .pcm_construct = skl_platform_soc_new, .module_get_upon_open = 1, /* increment refcount when a pcm is opened */ }; -- 2.26.2

3 5

[PATCH 0/5] ASoC: Fix reference to PCM buffer address
by Takashi Iwai 02 Aug '21

02 Aug '21

Hi, while updating the memory allocation API for WC pages, I casually stumbled on quite a few drivers dealing with the PCM buffer address wrongly via substream->dma_buffer.addr. This patch series corrects those usages. Takashi === Takashi Iwai (5): ASoC: amd: Fix reference to PCM buffer address ASoC: intel: atom: Fix reference to PCM buffer address ASoC: xilinx: Fix reference to PCM buffer address ASoC: uniphier: Fix reference to PCM buffer address ASoC: kirkwood: Fix reference to PCM buffer address sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 3 +-- sound/soc/kirkwood/kirkwood-dma.c | 26 ++++++++++++++------ sound/soc/uniphier/aio-dma.c | 2 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 4 +-- 7 files changed, 25 insertions(+), 16 deletions(-) -- 2.26.2

3 9

[PATCH 00/10] soundwire/ASoC: add mockup codec support
by Bard Liao 02 Aug '21

02 Aug '21

Adding mockup SoundWire codec is useful to debug driver/topology changes without having any actual device connected. Bard Liao (2): soundwire: stream: don't abort bank switch on Command_Ignored/-ENODATA soundwire: stream: don't program mockup device ports Pierre-Louis Bossart (8): ASoC: codecs: add SoundWire mockup device support ASoC: soc-acpi: cnl: add table for SoundWire mockup devices ASoC: soc-acpi: tgl: add table for SoundWire mockup devices ASoC: Intel: boards: sof_sdw: add SoundWire mockup codecs for tests soundwire: add flag to ignore all command/control for mockup devices soundwire: bus: squelch error returned by mockup devices soundwire: cadence: add debugfs interface for PDI loopbacks soundwire: cadence: override PDI configurations to create loopback drivers/soundwire/bus.c | 10 +- drivers/soundwire/cadence_master.c | 174 ++++++++-- drivers/soundwire/cadence_master.h | 3 + drivers/soundwire/stream.c | 5 +- include/linux/soundwire/sdw.h | 3 + sound/soc/codecs/Kconfig | 18 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sdw-mockup.c | 312 ++++++++++++++++++ sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_sdw.c | 41 +++ sound/soc/intel/common/Makefile | 3 +- .../intel/common/soc-acpi-intel-cnl-match.c | 15 + .../common/soc-acpi-intel-sdw-mockup-match.c | 166 ++++++++++ .../common/soc-acpi-intel-sdw-mockup-match.h | 17 + .../intel/common/soc-acpi-intel-tgl-match.c | 23 ++ 15 files changed, 754 insertions(+), 39 deletions(-) create mode 100644 sound/soc/codecs/sdw-mockup.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-sdw-mockup-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-sdw-mockup-match.h -- 2.17.1

4 15

[PATCH] soundwire: cadence: add paranoid check on self-clearing bits
by Bard Liao 02 Aug '21

02 Aug '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The Cadence IP exposes a small number of self-clearing bits in the MCP_CONTROL and MCP_CONFIG_UPDATE registers. We currently do not check that those bits are indeed cleared, e.g. during resume operations. That could lead to resuming peripheral devices too early. In addition, if we happen to read these registers, update one of the fields and write the register back, we may be writing stale data that might have been cleared in hardware. These sort of race conditions could lead to e.g. doing a hw_reset twice or stopping a clock that just restarted. There is no clear way of avoiding these potential race conditions other than making sure that these registers fields are cleared before any read-modify-write sequence. If we detect this sort of errors, we only log them since there is no clear recovery possible. The only way out is likely to restart the IP with a suspend/resume cycle. Note that the checks are performed before updating the registers, as well as after the Intel 'sync go' sequence in multi-link mode. That should cover both the start and end of suspend/resume hardware configurations. The Multi-Master mode gates the configuration updates until the 'sync go' signal is asserted, so we only check on init and after the end of the 'sync go' sequence. The duration of the usleep_range() was defined by the GSYNC frequency used in multi-master mode. With a 4kHz frequency, any configuration change might be deferred by up to 250us. Extending the range to 1000-1500us should guarantee that the configuration change is completed without any significant impact on the overall resume time. Suggested-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/cadence_master.c | 47 ++++++++++++++++++++++++++++++ drivers/soundwire/cadence_master.h | 4 +++ drivers/soundwire/intel.c | 14 +++++++++ 3 files changed, 65 insertions(+) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index c7372519452b..0b7f037e6cd0 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -979,6 +979,49 @@ static void cdns_update_slave_status_work(struct work_struct *work) } +/* paranoia check to make sure self-cleared bits are indeed cleared */ +void sdw_cdns_check_self_clearing_bits(struct sdw_cdns *cdns, const char *string, + bool initial_delay, int reset_iterations) +{ + u32 mcp_control; + u32 mcp_config_update; + int i; + + if (initial_delay) + usleep_range(1000, 1500); + + mcp_control = cdns_readl(cdns, CDNS_MCP_CONTROL); + + /* the following bits should be cleared immediately */ + if (mcp_control & CDNS_MCP_CONTROL_CMD_RST) + dev_err(cdns->dev, "%s failed: MCP_CONTROL_CMD_RST is not cleared\n", string); + if (mcp_control & CDNS_MCP_CONTROL_SOFT_RST) + dev_err(cdns->dev, "%s failed: MCP_CONTROL_SOFT_RST is not cleared\n", string); + if (mcp_control & CDNS_MCP_CONTROL_SW_RST) + dev_err(cdns->dev, "%s failed: MCP_CONTROL_SW_RST is not cleared\n", string); + if (mcp_control & CDNS_MCP_CONTROL_CLK_STOP_CLR) + dev_err(cdns->dev, "%s failed: MCP_CONTROL_CLK_STOP_CLR is not cleared\n", string); + mcp_config_update = cdns_readl(cdns, CDNS_MCP_CONFIG_UPDATE); + if (mcp_config_update & CDNS_MCP_CONFIG_UPDATE_BIT) + dev_err(cdns->dev, "%s failed: MCP_CONFIG_UPDATE_BIT is not cleared\n", string); + + i = 0; + while (mcp_control & CDNS_MCP_CONTROL_HW_RST) { + if (i == reset_iterations) { + dev_err(cdns->dev, "%s failed: MCP_CONTROL_HW_RST is not cleared\n", string); + break; + } + + dev_dbg(cdns->dev, "%s: MCP_CONTROL_HW_RST is not cleared at iteration %d\n", string, i); + i++; + + usleep_range(1000, 1500); + mcp_control = cdns_readl(cdns, CDNS_MCP_CONTROL); + } + +} +EXPORT_SYMBOL(sdw_cdns_check_self_clearing_bits); + /* * init routines */ @@ -1256,6 +1299,8 @@ int sdw_cdns_init(struct sdw_cdns *cdns) cdns_init_clock_ctrl(cdns); + sdw_cdns_check_self_clearing_bits(cdns, __func__, false, 0); + /* reset msg_count to default value of FIFOLEVEL */ cdns->msg_count = cdns_readl(cdns, CDNS_MCP_FIFOLEVEL); @@ -1500,6 +1545,8 @@ int sdw_cdns_clock_stop(struct sdw_cdns *cdns, bool block_wake) struct sdw_slave *slave; int ret; + sdw_cdns_check_self_clearing_bits(cdns, __func__, false, 0); + /* Check suspend status */ if (sdw_cdns_is_clock_stop(cdns)) { dev_dbg(cdns->dev, "Clock is already stopped\n"); diff --git a/drivers/soundwire/cadence_master.h b/drivers/soundwire/cadence_master.h index 6c039d456ba8..e587aede63bf 100644 --- a/drivers/soundwire/cadence_master.h +++ b/drivers/soundwire/cadence_master.h @@ -187,4 +187,8 @@ int cdns_bus_conf(struct sdw_bus *bus, struct sdw_bus_params *params); int cdns_set_sdw_stream(struct snd_soc_dai *dai, void *stream, bool pcm, int direction); + +void sdw_cdns_check_self_clearing_bits(struct sdw_cdns *cdns, const char *string, + bool initial_delay, int reset_iterations); + #endif /* __SDW_CADENCE_H */ diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 15668d6fecd6..fb9c23e13206 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -23,6 +23,7 @@ #include "intel.h" #define INTEL_MASTER_SUSPEND_DELAY_MS 3000 +#define INTEL_MASTER_RESET_ITERATIONS 10 /* * debug/config flags for the Intel SoundWire Master. @@ -1393,6 +1394,8 @@ int intel_link_startup(struct auxiliary_device *auxdev) goto err_interrupt; } } + sdw_cdns_check_self_clearing_bits(cdns, __func__, + true, INTEL_MASTER_RESET_ITERATIONS); /* Register DAIs */ ret = intel_register_dai(sdw); @@ -1709,6 +1712,8 @@ static int __maybe_unused intel_resume(struct device *dev) return ret; } } + sdw_cdns_check_self_clearing_bits(cdns, __func__, + true, INTEL_MASTER_RESET_ITERATIONS); /* * after system resume, the pm_runtime suspend() may kick in @@ -1793,6 +1798,9 @@ static int __maybe_unused intel_resume_runtime(struct device *dev) return ret; } } + sdw_cdns_check_self_clearing_bits(cdns, "intel_resume_runtime TEARDOWN", + true, INTEL_MASTER_RESET_ITERATIONS); + } else if (clock_stop_quirks & SDW_INTEL_CLK_STOP_BUS_RESET) { ret = intel_init(sdw); if (ret) { @@ -1866,6 +1874,9 @@ static int __maybe_unused intel_resume_runtime(struct device *dev) } } } + sdw_cdns_check_self_clearing_bits(cdns, "intel_resume_runtime BUS_RESET", + true, INTEL_MASTER_RESET_ITERATIONS); + } else if (!clock_stop_quirks) { clock_stop0 = sdw_cdns_is_clock_stop(&sdw->cdns); @@ -1889,6 +1900,9 @@ static int __maybe_unused intel_resume_runtime(struct device *dev) dev_err(dev, "unable to resume master during resume\n"); return ret; } + + sdw_cdns_check_self_clearing_bits(cdns, "intel_resume_runtime no_quirks", + true, INTEL_MASTER_RESET_ITERATIONS); } else { dev_err(dev, "%s clock_stop_quirks %x unsupported\n", __func__, clock_stop_quirks); -- 2.17.1

2 1

[5.14 regression] "ALSA: core: Add continuous and vmalloc mmap ops" breaks Intel SST audio
by Hans de Goede 01 Aug '21

01 Aug '21

Hi All;, As discussed in the "5.14 regression, Intel SST (BYT) audio no longer works" thread, audio on X86 devices using the Intel SST driver no longer works with 5.14. After poking at this for a while I've found the culprit: 30b7ba6972d5 ("ALSA: core: Add continuous and vmalloc mmap ops") If I revert that single commit then everything works fine again with 5.14. This is with a Fedora 34 userspace using pipewire as audiodaemon Regards, Hans

2 3

Incorrect device setting for M-Audio Audiophile USB sound card
by GitHub issues - opened 31 Jul '21

31 Jul '21

alsa-project/alsa-lib issue #166 was opened from michele-perrone: ### M-Audio Audiophile USB - fix needed There is a problem with the ALSA configuration of the "M-Audio Audiophile USB" sound card. From the linux kernel documentation (https://www.kernel.org/doc/html/latest/sound/cards/audiophile-usb.html) you can see that this card has four audio interfaces: * Analog 2-channel Input (Ai) * Analog 2-channel Output (Ao) * Digital 2-channel Input (Di) * Digital 2-channel Output (Do) These interfaces are distributed over three devices of the same card: * hw:USBtm,0 is Ao in playback and Di in capture * hw:USBtm,1 is Do in playback and Ai in capture * hw:USBtm,2 is Do in AC3/DTS passthrough mode Now, this is where ALSA makes a bit of a mess. If you look at the output of `aplay - L` (see attachment aplay_out.txt), you can see that the analog audio output is associated to CARD=USBtm,DEV=0 (which is correct), **but** you can also see that two digital outputs are assigned, one to CARD=USBtm,DEV=0 (**wrong!**) and another one to CARD=USBtm,DEV=1 (correct). Speaking now of `arecord - L` (see attachment arecord_out.txt), there is only one input assigned to CARD=USBtm,DEV=0, and it is marked as "Front output / input" instead of SPDIF digital input. And most importantly, the analog input configuration, which should correspond to CARD=USBtm,DEV=1, is completely absent. The result of this is that audio applications relying on ALSA don't know where to pick the correct inputs and outputs. If want both Ao and Ai to work in Jack, I have to select hw:USBtm,1 for the input and hw:USBtm,0 for the output. Alternatively, I have to create an `.asoundrc` file with the `asym` module (see attachment) and then run Jack with the name of my virtual device. However, this is just a workaround. I'm not familiar with the alsa source but I know my way around C code. If you tell me where to look, I may be able to help fixing this. Cheers! Michele Attachments: [aplay_out.txt](https://github.com/alsa-project/alsa-lib/files/6911115/aplay… [arecord_out.txt](https://github.com/alsa-project/alsa-lib/files/6911116/are… [asoundrc.txt](https://github.com/alsa-project/alsa-lib/files/6911119/asound… Issue URL : https://github.com/alsa-project/alsa-lib/issues/166 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: pcxhr: use __func__ to get funcion's name in an output message
by Jason Wang 31 Jul '21

31 Jul '21

Prefer using '"%s...", __func__' to get current function's name in an output message. Signed-off-by: Jason Wang <wangborong(a)cdjrlc.com> --- sound/pci/pcxhr/pcxhr.c | 47 ++++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 24 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 96b9371c201a..242bd7e04b3e 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -366,7 +366,7 @@ static int pcxhr_sub_set_clock(struct pcxhr_mgr *mgr, mgr->codec_speed = speed; /* save new codec speed */ } - dev_dbg(&mgr->pci->dev, "pcxhr_sub_set_clock to %dHz (realfreq=%d)\n", + dev_dbg(&mgr->pci->dev, "%s to %dHz (realfreq=%d)\n", __func__, rate, realfreq); return 0; } @@ -499,7 +499,7 @@ static int pcxhr_set_stream_state(struct snd_pcxhr *chip, else { if (stream->status != PCXHR_STREAM_STATUS_SCHEDULE_STOP) { dev_err(chip->card->dev, - "pcxhr_set_stream_state CANNOT be stopped\n"); + "%s CANNOT be stopped\n", __func__); return -EINVAL; } start = 0; @@ -524,7 +524,7 @@ static int pcxhr_set_stream_state(struct snd_pcxhr *chip, err = pcxhr_send_msg(chip->mgr, &rmh); if (err) dev_err(chip->card->dev, - "ERROR pcxhr_set_stream_state err=%x;\n", err); + "ERROR %s err=%x;\n", __func__, err); stream->status = start ? PCXHR_STREAM_STATUS_STARTED : PCXHR_STREAM_STATUS_STOPPED; return err; @@ -570,7 +570,7 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) break; default: dev_err(chip->card->dev, - "error pcxhr_set_format() : unknown format\n"); + "error %s() : unknown format\n", __func__); return -EINVAL; } @@ -615,7 +615,7 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) err = pcxhr_send_msg(chip->mgr, &rmh); if (err) dev_err(chip->card->dev, - "ERROR pcxhr_set_format err=%x;\n", err); + "ERROR %s err=%x;\n", __func__, err); return err; } @@ -630,7 +630,7 @@ static int pcxhr_update_r_buffer(struct pcxhr_stream *stream) stream_num = is_capture ? 0 : subs->number; dev_dbg(chip->card->dev, - "pcxhr_update_r_buffer(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n", + "%s(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n", __func__, is_capture ? 'c' : 'p', chip->chip_idx, (void *)(long)subs->runtime->dma_addr, subs->runtime->dma_bytes, subs->number); @@ -721,21 +721,20 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) } if (capture_mask == 0 && playback_mask == 0) { mutex_unlock(&mgr->setup_mutex); - dev_err(&mgr->pci->dev, "pcxhr_start_linked_stream : no pipes\n"); + dev_err(&mgr->pci->dev, "%s : no pipes\n", __func__); return; } - dev_dbg(&mgr->pci->dev, "pcxhr_start_linked_stream : " - "playback_mask=%x capture_mask=%x\n", - playback_mask, capture_mask); + dev_dbg(&mgr->pci->dev, "%s : playback_mask=%x capture_mask=%x\n", + __func__, playback_mask, capture_mask); /* synchronous stop of all the pipes concerned */ err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 0); if (err) { mutex_unlock(&mgr->setup_mutex); - dev_err(&mgr->pci->dev, "pcxhr_start_linked_stream : " + dev_err(&mgr->pci->dev, "%s : " "error stop pipes (P%x C%x)\n", - playback_mask, capture_mask); + __func__, playback_mask, capture_mask); return; } @@ -778,9 +777,9 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 1); if (err) { mutex_unlock(&mgr->setup_mutex); - dev_err(&mgr->pci->dev, "pcxhr_start_linked_stream : " + dev_err(&mgr->pci->dev, "%s : " "error start pipes (P%x C%x)\n", - playback_mask, capture_mask); + __func__, playback_mask, capture_mask); return; } @@ -889,7 +888,7 @@ static int pcxhr_hardware_timer(struct pcxhr_mgr *mgr, int start) } err = pcxhr_send_msg(mgr, &rmh); if (err < 0) - dev_err(&mgr->pci->dev, "error pcxhr_hardware_timer err(%x)\n", + dev_err(&mgr->pci->dev, "error %s err(%x)\n", __func__, err); return err; } @@ -904,7 +903,7 @@ static int pcxhr_prepare(struct snd_pcm_substream *subs) int err = 0; dev_dbg(chip->card->dev, - "pcxhr_prepare : period_size(%lx) periods(%x) buffer_size(%lx)\n", + "%s : period_size(%lx) periods(%x) buffer_size(%lx)\n", __func__, subs->runtime->period_size, subs->runtime->periods, subs->runtime->buffer_size); @@ -997,12 +996,12 @@ static int pcxhr_open(struct snd_pcm_substream *subs) runtime->hw = pcxhr_caps; if( subs->stream == SNDRV_PCM_STREAM_PLAYBACK ) { - dev_dbg(chip->card->dev, "pcxhr_open playback chip%d subs%d\n", - chip->chip_idx, subs->number); + dev_dbg(chip->card->dev, "%s playback chip%d subs%d\n", + __func__, chip->chip_idx, subs->number); stream = &chip->playback_stream[subs->number]; } else { - dev_dbg(chip->card->dev, "pcxhr_open capture chip%d subs%d\n", - chip->chip_idx, subs->number); + dev_dbg(chip->card->dev, "%s capture chip%d subs%d\n", + __func__, chip->chip_idx, subs->number); if (mgr->mono_capture) runtime->hw.channels_max = 1; else @@ -1011,8 +1010,8 @@ static int pcxhr_open(struct snd_pcm_substream *subs) } if (stream->status != PCXHR_STREAM_STATUS_FREE){ /* streams in use */ - dev_err(chip->card->dev, "pcxhr_open chip%d subs%d in use\n", - chip->chip_idx, subs->number); + dev_err(chip->card->dev, "%s chip%d subs%d in use\n", + __func__, chip->chip_idx, subs->number); mutex_unlock(&mgr->setup_mutex); return -EBUSY; } @@ -1077,7 +1076,7 @@ static int pcxhr_close(struct snd_pcm_substream *subs) mutex_lock(&mgr->setup_mutex); - dev_dbg(chip->card->dev, "pcxhr_close chip%d subs%d\n", + dev_dbg(chip->card->dev, "%s chip%d subs%d\n", __func__, chip->chip_idx, subs->number); /* sample rate released */ @@ -1572,7 +1571,7 @@ static int pcxhr_probe(struct pci_dev *pci, /* init setup mutex*/ mutex_init(&mgr->setup_mutex); - mgr->prmh = kmalloc(sizeof(*mgr->prmh) + + mgr->prmh = kmalloc(sizeof(*mgr->prmh) + sizeof(u32) * (PCXHR_SIZE_MAX_LONG_STATUS - PCXHR_SIZE_MAX_STATUS), GFP_KERNEL); -- 2.32.0

2 1

[PATCH] ALSA: core: Fix double calls of snd_card_free() via devres
by Takashi Iwai 31 Jul '21

31 Jul '21

At the transition to the devres-managed card release, we've put the check of double-free at trigger_card_release(). But this wasn't enough, as the code path calls snd_card_free() again, and it would lead to the doubly snd_card_free() calls. Actually the v1 patch was correct to handle this, but I forgot that corner case and moved the check to the more obvious place as I thought it's clearer. But, as usual, devils live in details. This patch corrects the check of the double-free to the right place, with a bit more comments. Fixes: e8ad415b7a55 ("ALSA: core: Add managed card creation") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/init.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index e985185ebc91..ac335f5906c6 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -605,6 +605,15 @@ int snd_card_free(struct snd_card *card) DECLARE_COMPLETION_ONSTACK(released); int ret; + /* The call of snd_card_free() is allowed from various code paths; + * a manual call from the driver and the call via devres_free, and + * we need to avoid double-free. Moreover, the release via devres + * may call snd_card_free() twice due to its nature, we need to have + * the check here at the beginning. + */ + if (card->releasing) + return 0; + card->release_completion = &released; ret = snd_card_free_when_closed(card); if (ret) @@ -813,10 +822,7 @@ EXPORT_SYMBOL_GPL(snd_card_add_dev_attr); static void trigger_card_free(void *data) { - struct snd_card *card = data; - - if (!card->releasing) - snd_card_free(data); + snd_card_free(data); } /** -- 2.26.2

1 0

[PATCH] ASoC: amd: Drop superfluous mmap callbacks
by Takashi Iwai 30 Jul '21

30 Jul '21

All ASoC AMD drivers just call the standard mmap handler, hence those are superfluous. Let's drop them. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/amd/acp-pcm-dma.c | 8 -------- sound/soc/amd/raven/acp3x-pcm-dma.c | 8 -------- sound/soc/amd/renoir/acp3x-pdm-dma.c | 8 -------- 3 files changed, 24 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 143155a840ac..8f202423e6ce 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1035,13 +1035,6 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_soc_component *component, return bytes_to_frames(runtime, pos); } -static int acp_dma_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return snd_pcm_lib_default_mmap(substream, vma); -} - static int acp_dma_prepare(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -1205,7 +1198,6 @@ static const struct snd_soc_component_driver acp_asoc_platform = { .hw_params = acp_dma_hw_params, .trigger = acp_dma_trigger, .pointer = acp_dma_pointer, - .mmap = acp_dma_mmap, .prepare = acp_dma_prepare, .pcm_construct = acp_dma_new, }; diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 8148b0d22e88..5f24546eeed0 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -320,13 +320,6 @@ static int acp3x_dma_new(struct snd_soc_component *component, return 0; } -static int acp3x_dma_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return snd_pcm_lib_default_mmap(substream, vma); -} - static int acp3x_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -370,7 +363,6 @@ static const struct snd_soc_component_driver acp3x_i2s_component = { .close = acp3x_dma_close, .hw_params = acp3x_dma_hw_params, .pointer = acp3x_dma_pointer, - .mmap = acp3x_dma_mmap, .pcm_construct = acp3x_dma_new, }; diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index bd20622b0933..9988a50a81b0 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -291,13 +291,6 @@ static int acp_pdm_dma_new(struct snd_soc_component *component, return 0; } -static int acp_pdm_dma_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return snd_pcm_lib_default_mmap(substream, vma); -} - static int acp_pdm_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -375,7 +368,6 @@ static const struct snd_soc_component_driver acp_pdm_component = { .close = acp_pdm_dma_close, .hw_params = acp_pdm_dma_hw_params, .pointer = acp_pdm_dma_pointer, - .mmap = acp_pdm_dma_mmap, .pcm_construct = acp_pdm_dma_new, }; -- 2.26.2

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