August 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ALSA: pcm: fix multiple memory leak bugs
by Wenwen Wang 07 Aug '19

07 Aug '19

In hiface_pcm_init(), 'rt' is firstly allocated through kzalloc(). Later on, hiface_pcm_init_urb() is invoked to initialize 'rt->out_urbs[i]'. In hiface_pcm_init_urb(), 'rt->out_urbs[i].buffer' is allocated through kzalloc(). However, if hiface_pcm_init_urb() fails, both 'rt' and 'rt->out_urbs[i].buffer' are not deallocated, leading to memory leak bugs. Also, 'rt->out_urbs[i].buffer' is not deallocated if snd_pcm_new() fails. To fix the above issues, free 'rt' and 'rt->out_urbs[i].buffer'. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/usb/hiface/pcm.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 14fc1e1..9b132aa 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -599,12 +599,18 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) for (i = 0; i < PCM_N_URBS; i++) { ret = hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, hiface_pcm_out_urb_handler); - if (ret < 0) + if (ret < 0) { + for (; i >= 0; i--) + kfree(rt->out_urbs[i].buffer); + kfree(rt); return ret; + } } ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); if (ret < 0) { + for (i = 0; i < PCM_N_URBS; i++) + kfree(rt->out_urbs[i].buffer); kfree(rt); dev_err(&chip->dev->dev, "Cannot create pcm instance\n"); return ret; -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda - Workaround for crackled sound on AMD controller (1022:1457)
by Takashi Iwai 07 Aug '19

07 Aug '19

A long-time problem on the recent AMD chip (1022:1457) with Realtek codecs (X370, X470, B450, etc) was the crackled or distorted sound for capture streams, as well as occasional playback hiccups. After lengthy debugging sessions, the workaround we've found is like below: - Set up the proper driver caps for this controller, similar as the other AMD controller. - Correct the DMA position reporting with the fixed FIFO size, which is similar like as workaround used for VIA chip set. - Even after the position correction, PA still shows mysterious stalls of playback streams when a capture is triggered in timer-scheduled mode. Since we have no clear way to eliminate the stall, pass the BATCH PCM flag for PA to suppress the tsched mode as a temporary workaround. This patch implements the workarounds. For the driver caps, it defines a new preset, AXZ_DCAPS_PRESET_AMD_SB. It enables the FIFO- corrected position reporting (corresponding to the new position_fix=6) and enforces the SNDRV_PCM_INFO_BATCH flag. Note that the current implementation is merely a workaround. Hopefully we'll find a better alternative in future, especially about removing the BATCH flag hack again. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195303 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 7 +++++ sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 63 +++++++++++++++++++++++++++++++++++++++++- 3 files changed, 70 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 2fbdde239936..48d863736b3c 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -613,6 +613,13 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) 20, 178000000); + /* by some reason, the playback stream stalls on PulseAudio with + * tsched=1 when a capture stream triggers. Until we figure out the + * real cause, disable tsched mode by telling the PCM info flag. + */ + if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) + runtime->hw.info |= SNDRV_PCM_INFO_BATCH; + if (chip->align_buffer_size) /* constrain buffer sizes to be multiple of 128 bytes. This is more efficient in terms of memory diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index baa15374fbcb..f2a6df5e6bcb 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -31,7 +31,7 @@ /* 14 unused */ #define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ #define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ -/* 17 unused */ +#define AZX_DCAPS_AMD_WORKAROUND (1 << 17) /* AMD-specific workaround */ #define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ #define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1e14d7270adf..a6d8c0d77b84 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -64,6 +64,7 @@ enum { POS_FIX_VIACOMBO, POS_FIX_COMBO, POS_FIX_SKL, + POS_FIX_FIFO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -135,7 +136,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "DMA pointer read method." - "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+)."); + "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+, 6 = FIFO)."); module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); @@ -332,6 +333,11 @@ enum { #define AZX_DCAPS_PRESET_ATI_HDMI_NS \ (AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF) +/* quirks for AMD SB */ +#define AZX_DCAPS_PRESET_AMD_SB \ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\ + AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME) + /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ (AZX_DCAPS_NO_MSI | AZX_DCAPS_CORBRP_SELF_CLEAR |\ @@ -841,6 +847,49 @@ static unsigned int azx_via_get_position(struct azx *chip, return bound_pos + mod_dma_pos; } +#define AMD_FIFO_SIZE 32 + +/* get the current DMA position with FIFO size correction */ +static unsigned int azx_get_pos_fifo(struct azx *chip, struct azx_dev *azx_dev) +{ + struct snd_pcm_substream *substream = azx_dev->core.substream; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int pos, delay; + + pos = snd_hdac_stream_get_pos_lpib(azx_stream(azx_dev)); + if (!runtime) + return pos; + + runtime->delay = AMD_FIFO_SIZE; + delay = frames_to_bytes(runtime, AMD_FIFO_SIZE); + if (azx_dev->insufficient) { + if (pos < delay) { + delay = pos; + runtime->delay = bytes_to_frames(runtime, pos); + } else { + azx_dev->insufficient = 0; + } + } + + /* correct the DMA position for capture stream */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (pos < delay) + pos += azx_dev->core.bufsize; + pos -= delay; + } + + return pos; +} + +static int azx_get_delay_from_fifo(struct azx *chip, struct azx_dev *azx_dev, + unsigned int pos) +{ + struct snd_pcm_substream *substream = azx_dev->core.substream; + + /* just read back the calculated value in the above */ + return substream->runtime->delay; +} + static unsigned int azx_skl_get_dpib_pos(struct azx *chip, struct azx_dev *azx_dev) { @@ -1417,6 +1466,7 @@ static int check_position_fix(struct azx *chip, int fix) case POS_FIX_VIACOMBO: case POS_FIX_COMBO: case POS_FIX_SKL: + case POS_FIX_FIFO: return fix; } @@ -1433,6 +1483,10 @@ static int check_position_fix(struct azx *chip, int fix) dev_dbg(chip->card->dev, "Using VIACOMBO position fix\n"); return POS_FIX_VIACOMBO; } + if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) { + dev_dbg(chip->card->dev, "Using FIFO position fix\n"); + return POS_FIX_FIFO; + } if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) { dev_dbg(chip->card->dev, "Using LPIB position fix\n"); return POS_FIX_LPIB; @@ -1453,6 +1507,7 @@ static void assign_position_fix(struct azx *chip, int fix) [POS_FIX_VIACOMBO] = azx_via_get_position, [POS_FIX_COMBO] = azx_get_pos_lpib, [POS_FIX_SKL] = azx_get_pos_skl, + [POS_FIX_FIFO] = azx_get_pos_fifo, }; chip->get_position[0] = chip->get_position[1] = callbacks[fix]; @@ -1467,6 +1522,9 @@ static void assign_position_fix(struct azx *chip, int fix) azx_get_delay_from_lpib; } + if (fix == POS_FIX_FIFO) + chip->get_delay[0] = chip->get_delay[1] = + azx_get_delay_from_fifo; } /* @@ -2447,6 +2505,9 @@ static const struct pci_device_id azx_ids[] = { /* AMD Hudson */ { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + /* AMD, X370 & co */ + { PCI_DEVICE(0x1022, 0x1457), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* AMD Stoney */ { PCI_DEVICE(0x1022, 0x157a), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | -- 2.16.4

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[alsa-devel] [PATCH] ALSA: pcm: fix a memory leak bug
by Wenwen Wang 07 Aug '19

07 Aug '19

In hiface_pcm_init(), 'rt' is firstly allocated through kzalloc(). Later on, hiface_pcm_init_urb() is invoked to initialize 'rt->out_urbs[i]'. However, if the initialization fails, 'rt' is not deallocated, leading to a memory leak bug. To fix the above issue, free 'rt' before returning the error. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/usb/hiface/pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 14fc1e1..5dbcd0d 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -599,8 +599,10 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) for (i = 0; i < PCM_N_URBS; i++) { ret = hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, hiface_pcm_out_urb_handler); - if (ret < 0) + if (ret < 0) { + kfree(rt); return ret; + } } ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); -- 2.7.4

2 2

[alsa-devel] [PATCH] ALSA: usb-midi: fix a memory leak bug
by Wenwen Wang 07 Aug '19

07 Aug '19

In __snd_usbmidi_create(), a MIDI streaming interface structure is allocated through kzalloc() and the pointer is saved to 'umidi'. Later on, the endpoint structures are created by invoking snd_usbmidi_create_endpoints_midiman() or snd_usbmidi_create_endpoints(), depending on the type of the audio quirk type. However, if the creation fails, the allocated 'umidi' is not deallocated, leading to a memory leak bug. To fix the above issue, free 'umidi' before returning the error. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/usb/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b737f0e..22db37f 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2476,7 +2476,7 @@ int __snd_usbmidi_create(struct snd_card *card, else err = snd_usbmidi_create_endpoints(umidi, endpoints); if (err < 0) - goto exit; + goto free_midi; usb_autopm_get_interface_no_resume(umidi->iface); -- 2.7.4

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[alsa-devel] [PATCH v2 0/7] Add support for new SAI IP version
by Daniel Baluta 07 Aug '19

07 Aug '19

So far SAI IPs integrated with imx6 only supported one data line. Starting with imx7 and imx8 SAI integration support up to 8 data lines. New SAI IP version introduces two new registers (Version and Parmeter registers) which are placed at the beginning of register address space. For this reason we need to fix the register's address. Changes since v1: - removed patches from Lucas as they were already accepted - addressed comments from Lucas and Nicolin regarding device tree property naming - removed comment saying that "datalines" must be always consecutively enabled (this is not true, checked with IP owner) - added new patch to document newly introduced compatbile strings - removed patch introducing combined mode as I will still need some time to figure out how to properly allow users to set it. Nicolin, Unfortunately I couldn't find any clean solution on handling registers address shifts. As mentioned in patch 5/7 Tx/Rx data registers and Tx/Rx FIFO registers keep their addresses while others are shifted by 8 bytes. Even if I could create two regmaps as suggested I will still need to update each call of regmap_functions. Daniel Baluta (7): ASoC: fsl_sai: Add registers definition for multiple datalines ASoC: fsl_sai: Update Tx/Rx channel enable mask ASoC: fsl_sai: Add support to enable multiple data lines ASoC: dt-bindings: Document dl-mask property ASoC: fsl_sai: Add support for SAI new version ASoC: fsl_sai: Add support for imx7ulp/imx8mq ASoC: dt-bindings: Introduce compatible strings for 7ULP and 8MQ .../devicetree/bindings/sound/fsl-sai.txt | 10 +- sound/soc/fsl/fsl_sai.c | 331 ++++++++++++------ sound/soc/fsl/fsl_sai.h | 82 +++-- 3 files changed, 293 insertions(+), 130 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH 00/28] ASoC: cleanup patches for soc-core
by Kuninori Morimoto 07 Aug '19

07 Aug '19

Hi Mark I want to add multi CPU DAI support to ALSA SoC. But I noticed that we want to cleanup ALSA SoC before that. These patches will do nothing from "technical" point of view. Just for cleanup Kuninori Morimoto (28): ASoC: soc-core: use device_register() ASoC: soc-core: set component->debugfs_root NULL ASoC: soc-core: add comment for for_each_xxx ASoC: soc-core: check return value of snd_soc_add_dai_link() ASoC: soc-core: don't use for_each_card_links_safe() at snd_soc_find_dai_link() ASoC: soc-core: reuse rtdcom at snd_soc_rtdcom_add() ASoC: soc-core: don't check widget for snd_soc_dapm_new_controls() ASoC: soc-core: don't check controls for snd_soc_add_component_controls() ASoC: soc-core: don't check routes for snd_soc_dapm_add_routes() ASoC: soc-core: don't check controls for snd_soc_add_card_controls() ASoC: soc-core: call snd_soc_dapm_debugfs_init() at soc_init_card_debugfs() ASoC: soc-core: remove unneeded list_empty() check for snd_soc_try_rebind_card() ASoC: soc-core: remove verbose debug message from soc_bind_dai_link() ASoC: soc-core: remove duplicated soc_is_dai_link_bound() ASoC: soc-core: tidyup for card->deferred_resume_work ASoC: soc-core: initialize rtd->list ASoC: soc-core: define soc_dpcm_debugfs_add() for non CONFIG_DEBUG_FS ASoC: soc-core: dai_link check under soc_dpcm_debugfs_add() ASoC: soc-core: merge snd_soc_initialize_card_lists() ASoC: soc-core: remove unneeded dai_link check from snd_soc_remove_dai_link() ASoC: soc-core: add NOTE to snd_soc_rtdcom_lookup() ASoC: soc-core: don't call snd_soc_component_set_jack() ASoC: soc-core: don't alloc memory carelessly when creating debugfs ASoC: soc-core: soc_cleanup_card_resources() become void ASoC: soc-core: initialize component list ASoC: soc-core: initialize list at one place ASoC: soc-dai: use bit field for bus_control ASoC: soc-topology: use for_each_component_dais() at remove_dai() include/sound/soc-dai.h | 3 +- include/sound/soc-dpcm.h | 9 ++- include/sound/soc.h | 15 +--- sound/soc/soc-core.c | 196 ++++++++++++++++++++++------------------------- sound/soc/soc-pcm.c | 3 + sound/soc/soc-topology.c | 2 +- 6 files changed, 108 insertions(+), 120 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: Intel: boards: Add Cometlake Dialog Maxim machine driver
by mac.chiang＠intel.com 06 Aug '19

06 Aug '19

From: Mac Chiang <mac.chiang(a)intel.com> enable Cometlake support with: SSP0 for DA7219 headphone codec SSP1 for MAX98357a speaker amp codec Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/Kconfig | 31 +++++++------ sound/soc/intel/boards/kbl_da7219_max98357a.c | 55 ++++++++++++++++++++--- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 12 +++++ 3 files changed, 80 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 50bf149..545aef8 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -322,19 +322,6 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". -config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH - tristate "KBL with DA7219 and MAX98357A in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_DA7219 - select SND_SOC_MAX98357A - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for DA7219 + MAX98357A I2S audio codec. - Say Y if you have such a device. - config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH tristate "KBL with DA7219 and MAX98927 in I2S Mode" depends on I2C && ACPI @@ -363,6 +350,24 @@ config SND_SOC_INTEL_KBL_RT5660_MACH endif ## SND_SOC_INTEL_KBL +if SND_SOC_INTEL_KBL || SND_SOC_SOF_COMETLAKE_LP + +config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH + tristate "KBL with DA7219 and MAX98357A in I2S Mode" + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_DA7219 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for DA7219 + MAX98357A I2S audio codec for + Kabylake/Cometlake platforms. + Say Y if you have such a device. + +endif ## SND_SOC_INTEL_KBL || SND_SOC_SOF_COMETLAKE_LP + if SND_SOC_INTEL_GLK || (SND_SOC_SOF_GEMINILAKE && SND_SOC_SOF_HDA_LINK) config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a889..fe3ac70 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -9,6 +9,7 @@ * RT5663 codecs */ +#include <asm/cpu_device_id.h> #include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> @@ -17,6 +18,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include "../../codecs/da7219.h" #include "../../codecs/hdac_hdmi.h" #include "../../codecs/da7219-aad.h" @@ -210,7 +212,11 @@ static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) if (!pcm) return -ENOMEM; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + pcm->device = dai->id; +#else pcm->device = device; +#endif pcm->codec_dai = dai; list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); @@ -587,9 +593,17 @@ static struct snd_soc_card kabylake_audio_card_da7219_m98357a = { .late_probe = kabylake_card_late_probe, }; +static const struct x86_cpu_id cml_ids[] = { + { X86_VENDOR_INTEL, 6, 0x8E }, /* Cometlake CPU_ID */ + {} +}; + static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; + struct snd_soc_acpi_mach *mach; + const char *platform_name; + int ret; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -600,17 +614,46 @@ static int kabylake_audio_probe(struct platform_device *pdev) kabylake_audio_card = (struct snd_soc_card *)pdev->id_entry->driver_data; + kabylake_audio_card = &kabylake_audio_card_da7219_m98357a; + kabylake_audio_card->dev = &pdev->dev; snd_soc_card_set_drvdata(kabylake_audio_card, ctx); + + if (x86_match_cpu(cml_ids)) { + unsigned int i; + + kabylake_audio_card->name = "cmlda7219max"; + + for (i = 0; i < ARRAY_SIZE(kabylake_dais); i++) { + /* MAXIM_CODEC is connected to SSP1. */ + if (!strcmp(kabylake_dais[i].cpus->dai_name, + KBL_MAXIM_CODEC_DAI)) { + kabylake_dais[i].name = "SSP1-Codec"; + kabylake_dais[i].cpus->dai_name = "SSP1 Pin"; + } + /* DIALOG_CODEC is connected to SSP0 */ + else if (!strcmp(kabylake_dais[i].cpus->dai_name, + KBL_DIALOG_CODEC_DAI)) { + kabylake_dais[i].name = "SSP0-Codec"; + kabylake_dais[i].cpus->dai_name = "SSP0 Pin"; + } + } + } + + mach = (&pdev->dev)->platform_data; + platform_name = mach->mach_params.platform; + + ret = snd_soc_fixup_dai_links_platform_name(kabylake_audio_card, + platform_name); + if (ret) + return ret; + return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); } static const struct platform_device_id kbl_board_ids[] = { - { - .name = "kbl_da7219_max98357a", - .driver_data = - (kernel_ulong_t)&kabylake_audio_card_da7219_m98357a, - }, + {.name = "kbl_da7219_max98357a",}, + {.name = "cml_da7219_max98357a",}, { } }; @@ -628,5 +671,7 @@ module_platform_driver(kabylake_audio) /* Module information */ MODULE_DESCRIPTION("Audio Machine driver-DA7219 & MAX98357A in I2S mode"); MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:kbl_da7219_max98357a"); +MODULE_ALIAS("platform:cml_da7219_max98357a"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index c36c0aa..4ea32b2 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs cml_codecs = { .codecs = {"10EC5682"} }; +static struct snd_soc_acpi_codecs cml_spk_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { { .id = "INT34C2", @@ -29,6 +34,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { .sof_tplg_filename = "sof-cnl-rt274.tplg", }, { + .id = "DLGS7219", + .drv_name = "cml_da7219_max98357a", + .quirk_data = &cml_spk_codecs, + .sof_fw_filename = "sof-cnl.ri", + .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + }, + { .id = "MX98357A", .drv_name = "sof_rt5682", .quirk_data = &cml_codecs, -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda - Don't override global PCM hw info flag
by Takashi Iwai 06 Aug '19

06 Aug '19

The commit bfcba288b97f ("ALSA - hda: Add support for link audio time reporting") introduced the conditional PCM hw info setup, but it overwrites the global azx_pcm_hw object. This will cause a problem if any other HD-audio controller, as it'll inherit the same bit flag although another controller doesn't support that feature. Fix the bug by setting the PCM hw info flag locally. Fixes: bfcba288b97f ("ALSA - hda: Add support for link audio time reporting") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index c8d1b4316245..2fbdde239936 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -598,11 +598,9 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) } runtime->private_data = azx_dev; - if (chip->gts_present) - azx_pcm_hw.info = azx_pcm_hw.info | - SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME; - runtime->hw = azx_pcm_hw; + if (chip->gts_present) + runtime->hw.info |= SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME; runtime->hw.channels_min = hinfo->channels_min; runtime->hw.channels_max = hinfo->channels_max; runtime->hw.formats = hinfo->formats; -- 2.16.4

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[alsa-devel] [GIT PULL] ASoC fixes for v5.3
by Mark Brown 06 Aug '19

06 Aug '19

The following changes since commit 87a6fe80d56be85bda093c4dd8a3e6535bda7e4a: ASoC: SOF: Intel: implement runtime idle for CNL/APL (2019-07-06 12:23:47 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-fix-v5.3-rc3-2 for you to fetch changes up to 7c0767643f3b6b0dd2cda923ae37a18590d431cf: ASoC: max98373: Remove executable bits (2019-08-06 11:15:06 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.3 Incremental fix removing executable bits added in a prior patch accidentally. ---------------------------------------------------------------- Cezary Rojewski (1): MAINTAINERS: Update Intel ASoC drivers maintainers Charles Keepax (1): ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks Cheng-Yi Chiang (1): ASoC: rockchip: Fix mono capture Christophe JAILLET (1): ASoC: Intel: Fix some acpi vs apci typo in somme comments Enric Balletbo i Serra (1): SoC: rockchip: rockchip_max98090: Enable MICBIAS for headset keypress detection Hans de Goede (1): ASoC: Intel: bytcht_es8316: Add quirk for Irbis NB41 netbook Joe Perches (1): ASoC: rt1308: Remove executable attribute from source files Kai Vehmanen (1): ASoC: SOF: Intel: hda: remove misleading error trace from IRQ thread Kirill Marinushkin (1): ASoC: Relocate my e-mail to .com domain zone Kuninori Morimoto (3): ASoC: simple_card_utils.h: care NULL dai at asoc_simple_debug_dai() ASoC: simple-card-utils: care no Platform for DPCM ASoC: audio-graph-card: add missing const at graph_get_dai_id() Marcus Cooper (1): ASoC: sun4i-i2s: Incorrect SR and WSS computation Mark Brown (1): ASoC: max98373: Remove executable bits Masahiro Yamada (1): ASoC: SOF: use __u32 instead of uint32_t in uapi headers Peter Ujfalusi (2): ASoC: ti: davinci-mcasp: Fix clk PDIR handling for i2s master mode ASoC: ti: davinci-mcasp: Correct slot_width posed constraint Ricard Wanderlof (1): ASoC: Fail card instantiation if DAI format setup fails Shengjiu Wang (1): ASoC: cs42xx8: Fix MFREQ selection issue for async mode Shuming Fan (1): ASoC: rt1011: fix DC calibration offset not applying Stephan Gerhold (1): ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI links Tzung-Bi Shih (1): ASoC: max98357a: use mdelay for sdmode-delay Vijendar Mukunda (2): ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driver ASoC: amd: acp3x: use dma address for acp3x dma driver Wen Yang (6): ASoC: simple-card: fix an use-after-free in simple_dai_link_of_dpcm() ASoC: simple-card: fix an use-after-free in simple_for_each_link() ASoC: audio-graph-card: fix use-after-free in graph_dai_link_of_dpcm() ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id() ASoC: samsung: odroid: fix an use-after-free issue for codec ASoC: samsung: odroid: fix a double-free issue for cpu_dai Wenwen Wang (1): ASoC: dapm: fix a memory leak bug fengchunguo (1): ASoC: max98373: add 88200 and 96000 sampling rate support MAINTAINERS | 3 +- include/sound/simple_card_utils.h | 4 + include/uapi/sound/sof/fw.h | 16 +-- include/uapi/sound/sof/header.h | 14 +-- sound/soc/amd/raven/acp3x-pcm-dma.c | 20 ++-- sound/soc/codecs/cs42xx8.c | 116 +++++++++++++++++---- sound/soc/codecs/max98357a.c | 25 +---- sound/soc/codecs/max98373.c | 6 ++ sound/soc/codecs/max98373.h | 2 + sound/soc/codecs/pcm3060-i2c.c | 4 +- sound/soc/codecs/pcm3060-spi.c | 4 +- sound/soc/codecs/pcm3060.c | 4 +- sound/soc/codecs/pcm3060.h | 2 +- sound/soc/codecs/rt1011.c | 4 +- sound/soc/codecs/rt1308.c | 0 sound/soc/codecs/rt1308.h | 0 sound/soc/generic/audio-graph-card.c | 30 +++--- sound/soc/generic/simple-card-utils.c | 7 ++ sound/soc/generic/simple-card.c | 26 ++--- sound/soc/intel/boards/bytcht_es8316.c | 8 ++ sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-glk-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-hda-match.c | 2 +- .../intel/common/soc-acpi-intel-hsw-bdw-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-skl-match.c | 2 +- sound/soc/qcom/apq8016_sbc.c | 16 +-- sound/soc/rockchip/rockchip_i2s.c | 5 +- sound/soc/rockchip/rockchip_max98090.c | 32 ++++++ sound/soc/samsung/odroid.c | 8 +- sound/soc/soc-core.c | 7 +- sound/soc/soc-dapm.c | 10 +- sound/soc/sof/intel/cnl.c | 4 +- sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sunxi/sun4i-i2s.c | 4 +- sound/soc/ti/davinci-mcasp.c | 46 ++++++-- 40 files changed, 304 insertions(+), 147 deletions(-) mode change 100755 => 100644 sound/soc/codecs/rt1308.c mode change 100755 => 100644 sound/soc/codecs/rt1308.h

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[alsa-devel] Possible bug report
by Harry Bloomberg 06 Aug '19

06 Aug '19

While running an amateur radio application on a Raspberry Pi 3B+ with the stretch version of Raspberrian, I see the following in my syslog: Aug 1 17:31:06 raspberrypi pulseaudio[853]: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 1 17:31:06 raspberrypi pulseaudio[853]: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Aug 1 17:31:06 raspberrypi pulseaudio[853]: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. So...I am reporting this to the developers as instructed by the error message. What additional information do you need? I'm not exactly sure what I was doing to trigger this error. I may have been exiting the application. Thanks, Harry Bloomberg

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