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

[RFC PATCH] soundwire: bus: Add flag to mark DPN_BlockCtrl1 as readonly
by Srinivas Kandagatla 11 Mar '20

11 Mar '20

According to SoundWire Specification Version 1.2. "A Data Port number X (in the range 0-14) which supports only one value of WordLength may implement the WordLength field in the DPX_BlockCtrl1 Register as Read-Only, returning the fixed value of WordLength in response to reads." As WSA881x interfaces in PDM mode making the only field "WordLength" in DPX_BlockCtrl1" fixed and read-only. Behaviour of writing to this register on WSA881x soundwire slave with Qualcomm Soundwire Controller is throwing up an error. Not sure how other controllers deal with writing to readonly registers, but this patch provides a way to avoid writes to DPN_BlockCtrl1 register by providing a ro_blockctrl1_reg flag in struct sdw_port_runtime. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- I will send patch for WSA881x to include this change once this patch is accepted. drivers/soundwire/bus.h | 2 ++ drivers/soundwire/stream.c | 17 ++++++++++------- 2 files changed, 12 insertions(+), 7 deletions(-) diff --git a/drivers/soundwire/bus.h b/drivers/soundwire/bus.h index 204204a26db8..791e8d14093e 100644 --- a/drivers/soundwire/bus.h +++ b/drivers/soundwire/bus.h @@ -79,6 +79,7 @@ int sdw_find_col_index(int col); * @num: Port number. For audio streams, valid port number ranges from * [1,14] * @ch_mask: Channel mask + * @ro_blockctrl1_reg: Read Only flag for DPN_BlockCtrl1 register * @transport_params: Transport parameters * @port_params: Port parameters * @port_node: List node for Master or Slave port_list @@ -89,6 +90,7 @@ int sdw_find_col_index(int col); struct sdw_port_runtime { int num; int ch_mask; + bool ro_blockctrl1_reg; struct sdw_transport_params transport_params; struct sdw_port_params port_params; struct list_head port_node; diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 00348d1fc606..4491643aeb4a 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -167,13 +167,15 @@ static int sdw_program_slave_port_params(struct sdw_bus *bus, return ret; } - /* Program DPN_BlockCtrl1 register */ - ret = sdw_write(s_rt->slave, addr2, (p_params->bps - 1)); - if (ret < 0) { - dev_err(&s_rt->slave->dev, - "DPN_BlockCtrl1 register write failed for port %d\n", - t_params->port_num); - return ret; + if (!p_rt->ro_blockctrl1_reg) { + /* Program DPN_BlockCtrl1 register */ + ret = sdw_write(s_rt->slave, addr2, (p_params->bps - 1)); + if (ret < 0) { + dev_err(&s_rt->slave->dev, + "DPN_BlockCtrl1 register write failed for port %d\n", + t_params->port_num); + return ret; + } } /* Program DPN_SampleCtrl1 register */ @@ -1195,6 +1197,7 @@ static struct sdw_port_runtime p_rt->ch_mask = port_config[port_index].ch_mask; p_rt->num = port_config[port_index].num; + p_rt->ro_blockctrl1_reg = port_config[port_index].ro_blockctrl1_reg; return p_rt; } -- 2.21.0

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[PATCH] ALSA: hda/realtek: Fix pop noise on ALC225
by Kai-Heng Feng 11 Mar '20

11 Mar '20

Commit 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") makes the ALC225 have pop noise on S3 resume and cold boot. So partially revert this commit for ALC225 to fix the regression. Fixes: 317d9313925c ("ALSA: hda/realtek - Set default power save node to 0") BugLink: https://bugs.launchpad.net/bugs/1866357 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0ac06ff1a17c..7b83b020ac3c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8051,6 +8051,8 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0225: + codec->power_save_node = 1; + /* fall through */ case 0x10ec0295: case 0x10ec0299: spec->codec_variant = ALC269_TYPE_ALC225; -- 2.17.1

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[PATCH] alsaloop: reduce cumulative error caused by non-atomic samples calculation
by Ruslan Bilovol 11 Mar '20

11 Mar '20

When doing loopback between two audio card with same sampling frequency, I noticed slow increase of pitch_diff. When I changed order of get_queued_playback_samples() vs get_queued_capture_samples(), I noticed same drift of pitch_diff but if was decreasing this time. This seems to be caused by non-atomic consecutive snd_pcm_delay() invocation for playback then for capture. snd_pcm_delay() measures delay between read/write call and actual ADC/DAC operation. So while we get this value for playback path in get_queued_playback_samples(), next call to get_queued_capture_samples() will happen a little bit later so snd_pcm_delay() may return incorrect value. Be interleaving get_queued_{playback,capture}_samples() order, we divide this small error between playback and capture paths. I do not see any issues anymore with one-way drift of pitch_diff. Signed-off-by: Ruslan Bilovol <ruslan.bilovol(a)gmail.com> --- alsaloop/pcmjob.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index b252486..1b7925a 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1951,8 +1951,16 @@ int pcmjob_pollfds_handle(struct loopback *loop, struct pollfd *fds) } if (loop->sync != SYNC_TYPE_NONE) { snd_pcm_sframes_t pqueued, cqueued; - pqueued = get_queued_playback_samples(loop); - cqueued = get_queued_capture_samples(loop); + + /* Reduce cumulative error by interleaving playback vs capture reading order */ + if (loop->total_queued_count & 1) { + pqueued = get_queued_playback_samples(loop); + cqueued = get_queued_capture_samples(loop); + } else { + cqueued = get_queued_capture_samples(loop); + pqueued = get_queued_playback_samples(loop); + } + if (verbose > 4) snd_output_printf(loop->output, "%s: queued %li/%li samples\n", loop->id, pqueued, cqueued); if (pqueued > 0) -- 1.9.1

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[PATCH 7/8] ALSA: firewire-tascam: Add missing annotation for tscm_hwdep_read_queue()
by Jules Irenge 11 Mar '20

11 Mar '20

Sparse reports a warning at tscm_hwdep_read_queue() warning: context imbalance in tscm_hwdep_read_queue() - unexpected unlock The root cause is the missing annotation at tscm_hwdep_read_queue() Add the missing __releases(&tscm->lock) annotation Signed-off-by: Jules Irenge <jbi.octave(a)gmail.com> --- sound/firewire/tascam/tascam-hwdep.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c index c29a97f6f638..9801e33e7f2a 100644 --- a/sound/firewire/tascam/tascam-hwdep.c +++ b/sound/firewire/tascam/tascam-hwdep.c @@ -36,6 +36,7 @@ static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf, static long tscm_hwdep_read_queue(struct snd_tscm *tscm, char __user *buf, long remained, loff_t *offset) + __releases(&tscm->lock) { char __user *pos = buf; unsigned int type = SNDRV_FIREWIRE_EVENT_TASCAM_CONTROL; -- 2.24.1

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[PATCH 8/8] ALSA: firewire-tascam: Add missing annotation for tscm_hwdep_read_locked()
by Jules Irenge 11 Mar '20

11 Mar '20

Sparse reports a warning at tscm_hwdep_read_locked() warning: context imbalance in tscm_hwdep_read_locked() - unexpected unlock The root cause is the missing annotation at tscm_hwdep_read_locked() Add the missing __releases(&tscm->lock) annotation Signed-off-by: Jules Irenge <jbi.octave(a)gmail.com> --- sound/firewire/tascam/tascam-hwdep.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c index 9801e33e7f2a..6f38335fe10b 100644 --- a/sound/firewire/tascam/tascam-hwdep.c +++ b/sound/firewire/tascam/tascam-hwdep.c @@ -17,6 +17,7 @@ static long tscm_hwdep_read_locked(struct snd_tscm *tscm, char __user *buf, long count, loff_t *offset) + __releases(&tscm->lock) { struct snd_firewire_event_lock_status event = { .type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS, -- 2.24.1

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Sample project request...
by GitHub issues - opened 11 Mar '20

11 Mar '20

alsa-project/alsa-lib issue #35 was opened from mnini: Can you make sample of alsa processor that is have input and output in the code? I know only syntax of C and I want to program DSP for linux output sound. Issue URL : https://github.com/alsa-project/alsa-lib/issues/35 Repository URL: https://github.com/alsa-project/alsa-lib

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How to model 3xADAU1761 codec in TDM8 with 2xSAI in ALSA ?
by Rolf Peder Klemetsen 10 Mar '20

10 Mar '20

Hello, I am working on bringup of a custom board that has three ADAU1761 codecs, and a NXP LS1028A SoC. The codecs are supposed to be configured in TDM8 mode and share a common playback I2S bus connected to SAI4, and a common capture I2S bus connected to SAI3. What is the recommended way of modelling this setup in the Linux device tree ? So far I have tried the following: &i2c0 { status = "okay"; i2c_codec1: adau1761@38 { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x38>; tdm-offset = <0>; clocks = <&adau1761_mclk>; clock-names = "mclk"; }; i2c_codec2: adau1761@39 { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x39>; tdm-offset = <16>; clocks = <&adau1761_mclk>; clock-names = "mclk"; }; i2c_codec3: adau1761@3a { compatible = "adi,adau1761"; #sound-dai-cells=<0>; reg = <0x3a>; tdm-offset = <32>; clocks = <&adau1761_mclk>; clock-names = "mclk"; }; }; At first I attempted creating three simple-audio-card's. That attempted failed since sharing the same cpu (SAI3 and SAI4) failed with no ALSA device as a result. Then I attempted creating a single simple-audio-card with two dai-links (one for playback and one for capture), each containing three codec sub-nodes. That doesn't really work either. Can someone please point me in the right direction? Many of the parameters that I have put in the dts have no effect and are assumed not supported. Regards Rolf sound { status = "okay"; model = "Super Sound System"; compatible = "simple-audio-card"; simple-audio-card,name = "sound"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&p_snd_codec_o1>; simple-audio-card,frame-master = <&p_snd_codec_o1>; simple-audio-card,widgets = "Microphone", "Mic In", "Headphone", "Headphone Out", "Line", "Line In", "Line", "Line Out"; simple-audio-card,routing = "Line Out", "LOUT", "Line Out", "ROUT", "Headphone Out", "LHP", "Headphone Out", "RHP", "Mic In", "MICBIAS", "LINN", "Mic In", "RINN", "Mic In", "LINP", "Mic In", "RINP", "Mic In", "LAUX", "Line In", "RAUX", "Line In"; playback_link: simple-audio-card,dai-link@0 { format = "i2s"; convert-channels = <8>; bitclock-master = <&p_snd_codec_o1>; frame-master = <&p_snd_codec_o1>; name-prefix = "p"; snd_dai_playback: cpu@0 { cpu-dai-name = "sai4_0"; sound-dai-name = "sai4_0"; sound-dai = <&sai4 0>; dai-tdm-slot-num = <8>; dai-tdm-slot-width = <16>; dai-tdm-slot-tx-mask = <1 0 1 0 1 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o1: codec@0 { sound-dai = <&i2c_codec1>; frame-master; bitclock-master; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c1"; dai-tdm-slot-tx-mask = <1 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o2: codec@1 { sound-dai = <&i2c_codec2>; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c2"; dai-tdm-slot-tx-mask = <0 0 1 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; p_snd_codec_o3: codec@2 { sound-dai = <&i2c_codec3>; clocks=<&adau1761_mclk>; clock-frequency = <12288000>; clock-names = "mclk"; name-prefix = "c3"; dai-tdm-slot-tx-mask = <0 0 0 0 1 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 0 0 0>; }; }; capture_link: simple-audio-card,dai-link@1 { format = "i2s"; convert-channels = <8>; bitclock-master = <&c_snd_codec_o1>; frame-master = <&c_snd_codec_o1>; name-prefix = "c"; snd_dai_capture: cpu@0 { sound-dai = <&sai3>; dai-tdm-slot-num = <8>; dai-tdm-slot-width = <16>; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 1 0 1 0 1 0 0>; }; c_snd_codec_o1: codec@0 { sound-dai = <&i2c_codec1>; frame-master; bitclock-master; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c1"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 1 0 0 0 0 0 0>; }; c_snd_codec_o2: codec@1 { sound-dai = <&i2c_codec2>; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c2"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 1 0 0 0 0>; }; c_snd_codec_o3: codec@2 { sound-dai = <&i2c_codec3>; clocks=<&adau1761_mclk>; clock-names = "mclk"; clock-frequency = <12288000>; name-prefix = "c3"; dai-tdm-slot-tx-mask = <0 0 0 0 0 0 0 0>; dai-tdm-slot-rx-mask = <0 0 0 0 0 1 0 0>; }; };

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[PATCH 0/3] ASoC: rt5682: fix Kconfig/compilation issues when I2C is not selected
by Pierre-Louis Bossart 10 Mar '20

10 Mar '20

RT5682 can work in I2C or SoundWire mode, but the code will not compile if I2C is not selected. Reported-by: kbuild test robot <lkp(a)intel.com> Pierre-Louis Bossart (3): ASoC: rt5682: fix unmet dependencies ASoC: rt5682: fix compilation issues without I2C ASoC: rt5682-sdw: fix 'defined but not used' pm functions sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/rt5682-sdw.c | 4 ++-- sound/soc/codecs/rt5682.c | 10 ++++++++++ 3 files changed, 13 insertions(+), 3 deletions(-) -- 2.20.1

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Re: [PATCH] dt-bindings: sound: Convert cirrus, cs42l51 to json-schema
by Rob Herring 10 Mar '20

10 Mar '20

On Fri, 28 Feb 2020 16:27:06 +0100, Benjamin Gaignard wrote: > Convert cirrus,cs42l51 to yaml format. > > Signed-off-by: Benjamin Gaignard <benjamin.gaignard(a)st.com> > --- > .../devicetree/bindings/sound/cirrus,cs42l51.yaml | 69 ++++++++++++++++++++++ > .../devicetree/bindings/sound/cs42l51.txt | 33 ----------- > 2 files changed, 69 insertions(+), 33 deletions(-) > create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l51.yaml > delete mode 100644 Documentation/devicetree/bindings/sound/cs42l51.txt > Reviewed-by: Rob Herring <robh(a)kernel.org>

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[PATCH for-5.7] ASoC: pcm: Fix (again) possible buffer overflow in dpcm state sysfs output
by Takashi Iwai 10 Mar '20

10 Mar '20

This is re-applying the fix that went into 5.6 (commit 6c89ffea60aa) as the changes were wiped out after merging the other code refactoring. Basically the same changes, just replacing the suspicious calls of snprintf() with scnprintf(). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-pcm.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e7915adaaf0a..34ee54ea7b30 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -66,16 +66,16 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, unsigned long flags; /* FE state */ - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, "[%s - %s]\n", fe->dai_link->name, stream ? "Capture" : "Playback"); - offset += snprintf(buf + offset, size - offset, "State: %s\n", + offset += scnprintf(buf + offset, size - offset, "State: %s\n", dpcm_state_string(fe->dpcm[stream].state)); if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, "Hardware Params: " "Format = %s, Channels = %d, Rate = %d\n", snd_pcm_format_name(params_format(params)), @@ -83,10 +83,10 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_rate(params)); /* BEs state */ - offset += snprintf(buf + offset, size - offset, "Backends:\n"); + offset += scnprintf(buf + offset, size - offset, "Backends:\n"); if (list_empty(&fe->dpcm[stream].be_clients)) { - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, " No active DSP links\n"); goto out; } @@ -96,16 +96,16 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, "- %s\n", be->dai_link->name); - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, " State: %s\n", dpcm_state_string(be->dpcm[stream].state)); if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP)) - offset += snprintf(buf + offset, size - offset, + offset += scnprintf(buf + offset, size - offset, " Hardware Params: " "Format = %s, Channels = %d, Rate = %d\n", snd_pcm_format_name(params_format(params)), -- 2.16.4

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[PATCH] MAINTAINERS: Update Cirrus Logic codec driver maintainers
by Schulman, James 10 Mar '20

10 Mar '20

Brian & Paul are no longer active audio codec driver maintainers. Update list to reflect myself and David Rhodes as the active maintainers. Signed-off-by: James Schulman <james.schulman(a)cirrus.com> --- MAINTAINERS | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 6158a143a13e..05379f8bc8e1 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4021,8 +4021,8 @@ F: Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt F: sound/soc/codecs/cros_ec_codec.* CIRRUS LOGIC AUDIO CODEC DRIVERS -M: Brian Austin <brian.austin(a)cirrus.com> -M: Paul Handrigan <Paul.Handrigan(a)cirrus.com> +M: James Schulman <james.schulman(a)cirrus.com> +M: David Rhodes <david.rhodes(a)cirrus.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained F: sound/soc/codecs/cs* -- 2.17.1

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[PATCH] ASoC: brcm: Add DSL/PON SoC audio driver
by Kevin Li 10 Mar '20

10 Mar '20

This patch adds Broadcom DSL/PON SoC audio driver with Whistler I2S block. The SoC supported by this patch are BCM63158B0,BCM63178 and BCM47622/6755. Signed-off-by: Kevin Li <kevin-ke.li(a)broadcom.com> --- sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 4 + sound/soc/bcm/bcm63xx-i2s-whistler.c | 328 ++++++++++++++++++ sound/soc/bcm/bcm63xx-i2s.h | 93 +++++ sound/soc/bcm/bcm63xx-pcm-whistler.c | 488 +++++++++++++++++++++++++++ 5 files changed, 922 insertions(+) create mode 100644 sound/soc/bcm/bcm63xx-i2s-whistler.c create mode 100644 sound/soc/bcm/bcm63xx-i2s.h create mode 100644 sound/soc/bcm/bcm63xx-pcm-whistler.c diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig index 0037e96aa228..4218057b0874 100644 --- a/sound/soc/bcm/Kconfig +++ b/sound/soc/bcm/Kconfig @@ -17,3 +17,12 @@ config SND_SOC_CYGNUS Cygnus chips (bcm958300, bcm958305, bcm911360) If you don't know what to do here, say N. + +config SND_BCM63XX_I2S_WHISTLER + tristate "SoC Audio support for the Broadcom BCM63XX I2S module" + select REGMAP_MMIO + help + Say Y if you want to add support for ASoC audio on Broadcom + DSL/PON chips (bcm63158, bcm63178) + + If you don't know what to do here, say N diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile index b81fa421ec27..7c2d7899603b 100644 --- a/sound/soc/bcm/Makefile +++ b/sound/soc/bcm/Makefile @@ -9,3 +9,7 @@ snd-soc-cygnus-objs := cygnus-pcm.o cygnus-ssp.o obj-$(CONFIG_SND_SOC_CYGNUS) += snd-soc-cygnus.o +# BCM63XX Platform Support +snd-soc-63xx-objs := bcm63xx-i2s-whistler.o bcm63xx-pcm-whistler.o + +obj-$(CONFIG_SND_BCM63XX_I2S_WHISTLER) += snd-soc-63xx.o \ No newline at end of file diff --git a/sound/soc/bcm/bcm63xx-i2s-whistler.c b/sound/soc/bcm/bcm63xx-i2s-whistler.c new file mode 100644 index 000000000000..c37135f0b85c --- /dev/null +++ b/sound/soc/bcm/bcm63xx-i2s-whistler.c @@ -0,0 +1,328 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * linux/sound/bcm/bcm63xx-i2s-whistler.c + * BCM63xx whistler i2s driver + + * Copyright (c) 2020 Broadcom Corporation + * Author: Kevin-Ke Li <kevin-ke.li(a)broadcom.com> + */ + +#include <linux/clk.h> +#include <linux/dma-mapping.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "bcm63xx-i2s.h" + +#define DRV_NAME "brcm-i2s" + +static bool brcm_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_TX_CFG ... I2S_TX_DESC_IFF_LEN: + case I2S_TX_CFG_2 ... I2S_RX_DESC_IFF_LEN: + case I2S_RX_CFG_2 ... I2S_REG_MAX: + return true; + default: + return false; + } +} + +static bool brcm_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_TX_CFG ... I2S_REG_MAX: + return true; + default: + return false; + } +} + +static bool brcm_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case I2S_TX_CFG: + case I2S_TX_IRQ_CTL: + case I2S_TX_DESC_IFF_ADDR: + case I2S_TX_DESC_IFF_LEN: + case I2S_TX_DESC_OFF_ADDR: + case I2S_TX_DESC_OFF_LEN: + case I2S_TX_CFG_2: + case I2S_RX_CFG: + case I2S_RX_IRQ_CTL: + case I2S_RX_DESC_OFF_ADDR: + case I2S_RX_DESC_OFF_LEN: + case I2S_RX_DESC_IFF_LEN: + case I2S_RX_DESC_IFF_ADDR: + case I2S_RX_CFG_2: + return true; + default: + return false; + } +} + +static const struct regmap_config brcm_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = I2S_REG_MAX, + .writeable_reg = brcm_i2s_wr_reg, + .readable_reg = brcm_i2s_rd_reg, + .volatile_reg = brcm_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int bcm63xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + return 0; +} + +static int bcm63xx_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai); + + ret = clk_set_rate(i2s_priv->i2s_clk, params_rate(params)); + if (ret < 0) + dev_err(i2s_priv->dev, + "Can't set sample rate, err: %d\n", ret); + + return ret; +} + +static int bcm63xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + return 0; +} + +static int bcm63xx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + unsigned int slaveMode; + struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai); + struct regmap *regmap_i2s = i2s_priv->regmap_i2s; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(regmap_i2s, I2S_TX_CFG, + I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT | + I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE, + I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT | + I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE); + regmap_write(regmap_i2s, I2S_TX_IRQ_CTL, 0); + regmap_write(regmap_i2s, I2S_TX_IRQ_IFF_THLD, 0); + regmap_write(regmap_i2s, I2S_TX_IRQ_OFF_THLD, 1); + + regmap_read(regmap_i2s, I2S_RX_CFG_2, &slaveMode); + if (slaveMode & I2S_RX_SLAVE_MODE_MASK) + regmap_update_bits(regmap_i2s, I2S_TX_CFG_2, + I2S_TX_SLAVE_MODE_MASK, + I2S_TX_MASTER_MODE); + else + regmap_update_bits(regmap_i2s, I2S_TX_CFG_2, + I2S_TX_SLAVE_MODE_MASK, + I2S_TX_SLAVE_MODE); + } else { + regmap_update_bits(regmap_i2s, I2S_RX_CFG, + I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT | + I2S_RX_CLOCK_ENABLE, + I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT | + I2S_RX_CLOCK_ENABLE); + regmap_write(regmap_i2s, I2S_RX_IRQ_CTL, 0); + regmap_write(regmap_i2s, I2S_RX_IRQ_IFF_THLD, 0); + regmap_write(regmap_i2s, I2S_RX_IRQ_OFF_THLD, 1); + + regmap_read(regmap_i2s, I2S_TX_CFG_2, &slaveMode); + if (slaveMode & I2S_TX_SLAVE_MODE_MASK) + regmap_update_bits(regmap_i2s, I2S_RX_CFG_2, + I2S_RX_SLAVE_MODE_MASK, 0); + else + regmap_update_bits(regmap_i2s, I2S_RX_CFG_2, + I2S_RX_SLAVE_MODE_MASK, + I2S_RX_SLAVE_MODE); + } + return 0; +} + +static void bcm63xx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + unsigned int enabled, slavemode; + struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai); + struct regmap *regmap_i2s = i2s_priv->regmap_i2s; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(regmap_i2s, I2S_TX_CFG, + I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT | + I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE, 0); + regmap_write(regmap_i2s, I2S_TX_IRQ_CTL, 1); + regmap_write(regmap_i2s, I2S_TX_IRQ_IFF_THLD, 4); + regmap_write(regmap_i2s, I2S_TX_IRQ_OFF_THLD, 4); + + regmap_read(regmap_i2s, I2S_TX_CFG_2, &slavemode); + slavemode = slavemode & I2S_TX_SLAVE_MODE_MASK; + if (!slavemode) { + regmap_read(regmap_i2s, I2S_RX_CFG, &enabled); + enabled = enabled & I2S_RX_ENABLE_MASK; + if (enabled) + regmap_update_bits(regmap_i2s, I2S_RX_CFG_2, + I2S_RX_SLAVE_MODE_MASK, + I2S_RX_MASTER_MODE); + } + regmap_update_bits(regmap_i2s, I2S_TX_CFG_2, + I2S_TX_SLAVE_MODE_MASK, + I2S_TX_SLAVE_MODE); + } else { + regmap_update_bits(regmap_i2s, I2S_RX_CFG, + I2S_RX_IN_R | I2S_RX_DATA_ALIGNMENT | + I2S_RX_CLOCK_ENABLE, 0); + regmap_write(regmap_i2s, I2S_RX_IRQ_CTL, 1); + regmap_write(regmap_i2s, I2S_RX_IRQ_IFF_THLD, 4); + regmap_write(regmap_i2s, I2S_RX_IRQ_OFF_THLD, 4); + + regmap_read(regmap_i2s, I2S_RX_CFG_2, &slavemode); + slavemode = slavemode & I2S_RX_SLAVE_MODE_MASK; + if (!slavemode) { + regmap_read(regmap_i2s, I2S_TX_CFG, &enabled); + enabled = enabled & I2S_TX_ENABLE_MASK; + if (enabled) + regmap_update_bits(regmap_i2s, I2S_TX_CFG_2, + I2S_TX_SLAVE_MODE_MASK, + I2S_TX_MASTER_MODE); + } + + regmap_update_bits(regmap_i2s, I2S_RX_CFG_2, + I2S_RX_SLAVE_MODE_MASK, I2S_RX_SLAVE_MODE); + } +} + +static const struct snd_soc_dai_ops bcm63xx_i2s_dai_ops = { + .startup = bcm63xx_i2s_startup, + .shutdown = bcm63xx_i2s_shutdown, + .hw_params = bcm63xx_i2s_hw_params, + .set_fmt = bcm63xx_i2s_set_fmt, + .set_sysclk = bcm63xx_i2s_set_sysclk, +}; + +static struct snd_soc_dai_driver bcm63xx_i2s_dai = { + .name = DRV_NAME, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &bcm63xx_i2s_dai_ops, + .symmetric_rates = 1, + .symmetric_channels = 1, +}; + +static const struct snd_soc_component_driver bcm63xx_i2s_component = { + .name = "bcm63xx", +}; + +static int bcm63xx_i2s_dev_probe(struct platform_device *pdev) +{ + int ret = 0; + void __iomem *regs; + struct resource *r_mem, *region; + struct bcm_i2s_priv *i2s_priv; + struct regmap *regmap_i2s; + struct clk *i2s_clk; + + i2s_priv = devm_kzalloc(&pdev->dev, sizeof(*i2s_priv), GFP_KERNEL); + if (!i2s_priv) + return -ENOMEM; + + i2s_clk = devm_clk_get(&pdev->dev, "i2sclk"); + if (IS_ERR(i2s_clk)) { + dev_err(&pdev->dev, "%s: cannot get a brcm clock: %ld\n", + __func__, PTR_ERR(i2s_clk)); + return PTR_ERR(i2s_clk); + } + + r_mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!r_mem) { + dev_err(&pdev->dev, "Unable to get register resource.\n"); + return -ENODEV; + } + + region = devm_request_mem_region(&pdev->dev, r_mem->start, + resource_size(r_mem), DRV_NAME); + if (!region) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + return -EBUSY; + } + + regs = devm_ioremap_resource(&pdev->dev, r_mem); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); + return ret; + } + + regmap_i2s = devm_regmap_init_mmio(&pdev->dev, + regs, &brcm_i2s_regmap_config); + if (IS_ERR(regmap_i2s)) + return PTR_ERR(regmap_i2s); + + regmap_update_bits(regmap_i2s, I2S_MISC_CFG, + I2S_PAD_LVL_LOOP_DIS_MASK, + I2S_PAD_LVL_LOOP_DIS_ENABLE); + + ret = devm_snd_soc_register_component(&pdev->dev, + &bcm63xx_i2s_component, + &bcm63xx_i2s_dai, 1); + if (ret) { + dev_err(&pdev->dev, "failed to register the dai\n"); + return ret; + } + + i2s_priv->dev = &pdev->dev; + i2s_priv->i2s_clk = i2s_clk; + i2s_priv->regmap_i2s = regmap_i2s; + dev_set_drvdata(&pdev->dev, i2s_priv); + + ret = bcm63xx_soc_platform_probe(pdev, i2s_priv); + if (ret) + dev_err(&pdev->dev, "failed to register the pcm\n"); + + return ret; +} + +static int bcm63xx_i2s_dev_remove(struct platform_device *pdev) +{ + bcm63xx_soc_platform_remove(pdev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id snd_soc_bcm_audio_match[] = { + {.compatible = "brcm,bcm63xx-i2s"}, + { } +}; +#endif + +static struct platform_driver bcm63xx_i2s_driver = { + .driver = { + .name = DRV_NAME, + .of_match_table = of_match_ptr(snd_soc_bcm_audio_match), + }, + .probe = bcm63xx_i2s_dev_probe, + .remove = bcm63xx_i2s_dev_remove, +}; + +module_platform_driver(bcm63xx_i2s_driver); + +MODULE_AUTHOR("Kevin,Li <kevin-ke.li(a)broadcom.com>"); +MODULE_DESCRIPTION("Broadcom DSL XPON ASOC I2S Interface"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/bcm/bcm63xx-i2s.h b/sound/soc/bcm/bcm63xx-i2s.h new file mode 100644 index 000000000000..695e9ba88544 --- /dev/null +++ b/sound/soc/bcm/bcm63xx-i2s.h @@ -0,0 +1,93 @@ +/* SPDX-License-Identifier: GPL-2.0-only*/ +/* + * linux/sound/soc/bcm/bcm63xx-i2s.h + * + * Copyright (c) 2020 Broadcom Corporation + * Author: Kevin-Ke Li <kevin-ke.li(a)broadcom.com> + */ + +#ifndef __BCM63XX_I2S_H +#define __BCM63XX_I2S_H + +#define I2S_DESC_FIFO_DEPTH 8 +#define I2S_MISC_CFG (0x003C) +#define I2S_PAD_LVL_LOOP_DIS_MASK (1 << 2) +#define I2S_PAD_LVL_LOOP_DIS_ENABLE I2S_PAD_LVL_LOOP_DIS_MASK + +#define I2S_TX_ENABLE_MASK (1 << 31) +#define I2S_TX_ENABLE I2S_TX_ENABLE_MASK +#define I2S_TX_OUT_R (1 << 19) +#define I2S_TX_DATA_ALIGNMENT (1 << 2) +#define I2S_TX_DATA_ENABLE (1 << 1) +#define I2S_TX_CLOCK_ENABLE (1 << 0) + +#define I2S_TX_DESC_OFF_LEVEL_SHIFT 12 +#define I2S_TX_DESC_OFF_LEVEL_MASK (0x0F << I2S_TX_DESC_OFF_LEVEL_SHIFT) +#define I2S_TX_DESC_IFF_LEVEL_SHIFT 8 +#define I2S_TX_DESC_IFF_LEVEL_MASK (0x0F << I2S_TX_DESC_IFF_LEVEL_SHIFT) +#define I2S_TX_DESC_OFF_INTR_EN_MSK (1 << 1) +#define I2S_TX_DESC_OFF_INTR_EN I2S_TX_DESC_OFF_INTR_EN_MSK + +#define I2S_TX_CFG (0x0000) +#define I2S_TX_IRQ_CTL (0x0004) +#define I2S_TX_IRQ_EN (0x0008) +#define I2S_TX_IRQ_IFF_THLD (0x000c) +#define I2S_TX_IRQ_OFF_THLD (0x0010) +#define I2S_TX_DESC_IFF_ADDR (0x0014) +#define I2S_TX_DESC_IFF_LEN (0x0018) +#define I2S_TX_DESC_OFF_ADDR (0x001C) +#define I2S_TX_DESC_OFF_LEN (0x0020) +#define I2S_TX_CFG_2 (0x0024) +#define I2S_TX_SLAVE_MODE_SHIFT 13 +#define I2S_TX_SLAVE_MODE_MASK (1 << I2S_TX_SLAVE_MODE_SHIFT) +#define I2S_TX_SLAVE_MODE I2S_TX_SLAVE_MODE_MASK +#define I2S_TX_MASTER_MODE 0 +#define I2S_TX_INTR_MASK 0x0F + +#define I2S_RX_ENABLE_MASK (1 << 31) +#define I2S_RX_ENABLE I2S_RX_ENABLE_MASK +#define I2S_RX_IN_R (1 << 19) +#define I2S_RX_DATA_ALIGNMENT (1 << 2) +#define I2S_RX_CLOCK_ENABLE (1 << 0) + +#define I2S_RX_DESC_OFF_LEVEL_SHIFT 12 +#define I2S_RX_DESC_OFF_LEVEL_MASK (0x0F << I2S_RX_DESC_OFF_LEVEL_SHIFT) +#define I2S_RX_DESC_IFF_LEVEL_SHIFT 8 +#define I2S_RX_DESC_IFF_LEVEL_MASK (0x0F << I2S_RX_DESC_IFF_LEVEL_SHIFT) +#define I2S_RX_DESC_OFF_INTR_EN_MSK (1 << 1) +#define I2S_RX_DESC_OFF_INTR_EN I2S_RX_DESC_OFF_INTR_EN_MSK + +#define I2S_RX_CFG (0x0040) /* 20c0 */ +#define I2S_RX_IRQ_CTL (0x0044) +#define I2S_RX_IRQ_EN (0x0048) +#define I2S_RX_IRQ_IFF_THLD (0x004C) +#define I2S_RX_IRQ_OFF_THLD (0x0050) +#define I2S_RX_DESC_IFF_ADDR (0x0054) +#define I2S_RX_DESC_IFF_LEN (0x0058) +#define I2S_RX_DESC_OFF_ADDR (0x005C) +#define I2S_RX_DESC_OFF_LEN (0x0060) +#define I2S_RX_CFG_2 (0x0064) +#define I2S_RX_SLAVE_MODE_SHIFT 13 +#define I2S_RX_SLAVE_MODE_MASK (1 << I2S_RX_SLAVE_MODE_SHIFT) +#define I2S_RX_SLAVE_MODE I2S_RX_SLAVE_MODE_MASK +#define I2S_RX_MASTER_MODE 0 +#define I2S_RX_INTR_MASK 0x0F + +#define I2S_REG_MAX 0x007C + +struct bcm_i2s_priv { + struct device *dev; + struct resource *r_irq; + struct regmap *regmap_i2s; + struct clk *i2s_clk; + struct snd_pcm_substream *play_substream; + struct snd_pcm_substream *capture_substream; + struct i2s_dma_desc *play_dma_desc; + struct i2s_dma_desc *capture_dma_desc; +}; + +extern int bcm63xx_soc_platform_probe(struct platform_device *pdev, + struct bcm_i2s_priv *i2s_priv); +extern int bcm63xx_soc_platform_remove(struct platform_device *pdev); + +#endif diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c new file mode 100644 index 000000000000..fdf383a9a99a --- /dev/null +++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c @@ -0,0 +1,488 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * linux/sound/bcm/bcm63xx-pcm-whistler.c + * BCM63xx whistler pcm interface + + * Copyright (c) 2020 Broadcom Corporation + * Author: Kevin-Ke Li <kevin-ke.li(a)broadcom.com> + */ + +#include <linux/dma-mapping.h> +#include <linux/io.h> +#include <linux/module.h> +#include <sound/pcm_params.h> +#include <linux/regmap.h> +#include <linux/of_device.h> +#include <sound/soc.h> +#include "bcm63xx-i2s.h" + + +struct i2s_dma_desc { + unsigned char *dma_area; + dma_addr_t dma_addr; + unsigned int dma_len; +}; + +struct bcm63xx_runtime_data { + int dma_len; + dma_addr_t dma_addr; + dma_addr_t dma_addr_next; +}; + +static const struct snd_pcm_hardware bcm63xx_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S32_LE, /* support S32 only */ + .period_bytes_max = 8192 - 32, + .periods_min = 1, + .periods_max = PAGE_SIZE/sizeof(struct i2s_dma_desc), + .buffer_bytes_max = 128 * 1024, + .fifo_size = 32, +}; + +static int bcm63xx_pcm_hw_params(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct i2s_dma_desc *dma_desc; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + dma_desc = kzalloc(sizeof(*dma_desc), GFP_NOWAIT); + if (!dma_desc) + return -ENOMEM; + + snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_desc); + + return 0; +} + +static int bcm63xx_pcm_hw_free(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct i2s_dma_desc *dma_desc; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dma_desc = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + kfree(dma_desc); + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int bcm63xx_pcm_trigger(struct snd_soc_component *component, + struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd; + struct bcm_i2s_priv *i2s_priv; + struct regmap *regmap_i2s; + + rtd = substream->private_data; + i2s_priv = dev_get_drvdata(rtd->cpu_dai->dev); + regmap_i2s = i2s_priv->regmap_i2s; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + regmap_update_bits(regmap_i2s, + I2S_TX_IRQ_EN, + I2S_TX_DESC_OFF_INTR_EN, + I2S_TX_DESC_OFF_INTR_EN); + regmap_update_bits(regmap_i2s, + I2S_TX_CFG, + I2S_TX_ENABLE_MASK, + I2S_TX_ENABLE); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_write(regmap_i2s, + I2S_TX_IRQ_EN, + 0); + regmap_update_bits(regmap_i2s, + I2S_TX_CFG, + I2S_TX_ENABLE_MASK, + 0); + break; + default: + ret = -EINVAL; + } + } else { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + regmap_update_bits(regmap_i2s, + I2S_RX_IRQ_EN, + I2S_RX_DESC_OFF_INTR_EN_MSK, + I2S_RX_DESC_OFF_INTR_EN); + regmap_update_bits(regmap_i2s, + I2S_RX_CFG, + I2S_RX_ENABLE_MASK, + I2S_RX_ENABLE); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_update_bits(regmap_i2s, + I2S_RX_IRQ_EN, + I2S_RX_DESC_OFF_INTR_EN_MSK, + 0); + regmap_update_bits(regmap_i2s, + I2S_RX_CFG, + I2S_RX_ENABLE_MASK, + 0); + break; + default: + ret = -EINVAL; + } + } + return ret; +} + +static int bcm63xx_pcm_prepare(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct i2s_dma_desc *dma_desc; + struct regmap *regmap_i2s; + struct bcm_i2s_priv *i2s_priv; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + uint32_t regaddr_desclen, regaddr_descaddr; + + dma_desc = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + dma_desc->dma_len = snd_pcm_lib_period_bytes(substream); + dma_desc->dma_addr = runtime->dma_addr; + dma_desc->dma_area = runtime->dma_area; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regaddr_desclen = I2S_TX_DESC_IFF_LEN; + regaddr_descaddr = I2S_TX_DESC_IFF_ADDR; + } else { + regaddr_desclen = I2S_RX_DESC_IFF_LEN; + regaddr_descaddr = I2S_RX_DESC_IFF_ADDR; + } + + i2s_priv = dev_get_drvdata(rtd->cpu_dai->dev); + regmap_i2s = i2s_priv->regmap_i2s; + + regmap_write(regmap_i2s, regaddr_desclen, dma_desc->dma_len); + regmap_write(regmap_i2s, regaddr_descaddr, dma_desc->dma_addr); + + return 0; +} + +static snd_pcm_uframes_t +bcm63xx_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + snd_pcm_uframes_t x; + struct bcm63xx_runtime_data *prtd = substream->runtime->private_data; + + if ((void *)prtd->dma_addr_next == NULL) + prtd->dma_addr_next = substream->runtime->dma_addr; + + x = bytes_to_frames(substream->runtime, + prtd->dma_addr_next - substream->runtime->dma_addr); + + return x == substream->runtime->buffer_size ? 0 : x; +} + +static int bcm63xx_pcm_mmap(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_wc(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); + +} + +static int bcm63xx_pcm_open(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct bcm63xx_runtime_data *prtd; + + runtime->hw = bcm63xx_pcm_hardware; + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); + if (ret) + goto out; + + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); + if (ret) + goto out; + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + ret = -ENOMEM; + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (!prtd) + goto out; + + runtime->private_data = prtd; + return 0; +out: + return ret; +} + +static int bcm63xx_pcm_close(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct bcm63xx_runtime_data *prtd = runtime->private_data; + + kfree(prtd); + return 0; +} + +static irqreturn_t i2s_dma_isr(int irq, void *bcm_i2s_priv) +{ + unsigned int availdepth, ifflevel, offlevel, int_status, val_1, val_2; + struct bcm63xx_runtime_data *prtd; + struct snd_pcm_substream *substream; + struct snd_pcm_runtime *runtime; + struct regmap *regmap_i2s; + struct i2s_dma_desc *dma_desc; + struct snd_soc_pcm_runtime *rtd; + struct bcm_i2s_priv *i2s_priv; + + i2s_priv = (struct bcm_i2s_priv *)bcm_i2s_priv; + regmap_i2s = i2s_priv->regmap_i2s; + + /* rx */ + regmap_read(regmap_i2s, I2S_RX_IRQ_CTL, &int_status); + + if (int_status & I2S_RX_DESC_OFF_INTR_EN_MSK) { + substream = i2s_priv->capture_substream; + runtime = substream->runtime; + rtd = substream->private_data; + prtd = runtime->private_data; + dma_desc = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + offlevel = (int_status & I2S_RX_DESC_OFF_LEVEL_MASK) >> + I2S_RX_DESC_OFF_LEVEL_SHIFT; + while (offlevel) { + regmap_read(regmap_i2s, I2S_RX_DESC_OFF_ADDR, &val_1); + regmap_read(regmap_i2s, I2S_RX_DESC_OFF_LEN, &val_2); + offlevel--; + } + prtd->dma_addr_next = val_1 + val_2; + ifflevel = (int_status & I2S_RX_DESC_IFF_LEVEL_MASK) >> + I2S_RX_DESC_IFF_LEVEL_SHIFT; + + availdepth = I2S_DESC_FIFO_DEPTH - ifflevel; + while (availdepth) { + dma_desc->dma_addr += + snd_pcm_lib_period_bytes(substream); + dma_desc->dma_area += + snd_pcm_lib_period_bytes(substream); + if (dma_desc->dma_addr - runtime->dma_addr >= + runtime->dma_bytes) { + dma_desc->dma_addr = runtime->dma_addr; + dma_desc->dma_area = runtime->dma_area; + } + + prtd->dma_addr = dma_desc->dma_addr; + regmap_write(regmap_i2s, I2S_RX_DESC_IFF_LEN, + snd_pcm_lib_period_bytes(substream)); + regmap_write(regmap_i2s, I2S_RX_DESC_IFF_ADDR, + dma_desc->dma_addr); + availdepth--; + } + + snd_pcm_period_elapsed(substream); + + /* Clear interrupt by writing 0 */ + regmap_update_bits(regmap_i2s, I2S_RX_IRQ_CTL, + I2S_RX_INTR_MASK, 0); + } + + /* tx */ + regmap_read(regmap_i2s, I2S_TX_IRQ_CTL, &int_status); + + if (int_status & I2S_TX_DESC_OFF_INTR_EN_MSK) { + substream = i2s_priv->play_substream; + runtime = substream->runtime; + rtd = substream->private_data; + prtd = runtime->private_data; + dma_desc = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + offlevel = (int_status & I2S_TX_DESC_OFF_LEVEL_MASK) >> + I2S_TX_DESC_OFF_LEVEL_SHIFT; + while (offlevel) { + regmap_read(regmap_i2s, I2S_TX_DESC_OFF_ADDR, &val_1); + regmap_read(regmap_i2s, I2S_TX_DESC_OFF_LEN, &val_2); + prtd->dma_addr_next = val_1 + val_2; + offlevel--; + } + + ifflevel = (int_status & I2S_TX_DESC_IFF_LEVEL_MASK) >> + I2S_TX_DESC_IFF_LEVEL_SHIFT; + availdepth = I2S_DESC_FIFO_DEPTH - ifflevel; + + while (availdepth) { + dma_desc->dma_addr += + snd_pcm_lib_period_bytes(substream); + dma_desc->dma_area += + snd_pcm_lib_period_bytes(substream); + + if (dma_desc->dma_addr - runtime->dma_addr >= + runtime->dma_bytes) { + dma_desc->dma_addr = runtime->dma_addr; + dma_desc->dma_area = runtime->dma_area; + } + + prtd->dma_addr = dma_desc->dma_addr; + regmap_write(regmap_i2s, I2S_TX_DESC_IFF_LEN, + snd_pcm_lib_period_bytes(substream)); + regmap_write(regmap_i2s, I2S_TX_DESC_IFF_ADDR, + dma_desc->dma_addr); + availdepth--; + } + + snd_pcm_period_elapsed(substream); + + /* Clear interrupt by writing 0 */ + regmap_update_bits(regmap_i2s, I2S_TX_IRQ_CTL, + I2S_TX_INTR_MASK, 0); + } + + return IRQ_HANDLED; +} + +static int bcm63xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = bcm63xx_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + buf->area = dma_alloc_wc(pcm->card->dev, + size, &buf->addr, + GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + return 0; +} + +static int bcm63xx_soc_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + struct bcm_i2s_priv *i2s_priv; + int ret; + + i2s_priv = dev_get_drvdata(rtd->cpu_dai->dev); + + of_dma_configure(pcm->card->dev, pcm->card->dev->of_node, 1); + + ret = dma_coerce_mask_and_coherent(pcm->card->dev, DMA_BIT_MASK(32)); + if (ret) + goto out; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = bcm63xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + + i2s_priv->play_substream = + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = bcm63xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + i2s_priv->capture_substream = + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + } + +out: + return ret; +} + +static void bcm63xx_pcm_free_dma_buffers(struct snd_soc_component *component, + struct snd_pcm *pcm) +{ + int stream; + struct snd_dma_buffer *buf; + struct snd_pcm_substream *substream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_wc(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static const struct snd_soc_component_driver bcm63xx_soc_platform = { + .open = bcm63xx_pcm_open, + .close = bcm63xx_pcm_close, + .hw_params = bcm63xx_pcm_hw_params, + .hw_free = bcm63xx_pcm_hw_free, + .prepare = bcm63xx_pcm_prepare, + .trigger = bcm63xx_pcm_trigger, + .pointer = bcm63xx_pcm_pointer, + .mmap = bcm63xx_pcm_mmap, + .pcm_construct = bcm63xx_soc_pcm_new, + .pcm_destruct = bcm63xx_pcm_free_dma_buffers, +}; + +int bcm63xx_soc_platform_probe(struct platform_device *pdev, + struct bcm_i2s_priv *i2s_priv) +{ + int ret; + + i2s_priv->r_irq = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!i2s_priv->r_irq) { + dev_err(&pdev->dev, "Unable to get register irq resource.\n"); + return -ENODEV; + } + + ret = devm_request_irq(&pdev->dev, i2s_priv->r_irq->start, i2s_dma_isr, + i2s_priv->r_irq->flags, "i2s_dma", (void *)i2s_priv); + if (ret) { + dev_err(&pdev->dev, + "i2s_init: failed to request interrupt.ret=%d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(&pdev->dev, + &bcm63xx_soc_platform, NULL, 0); +} + +int bcm63xx_soc_platform_remove(struct platform_device *pdev) +{ + return 0; +} + +MODULE_AUTHOR("Kevin,Li <kevin-ke.li(a)broadcom.com>"); +MODULE_DESCRIPTION("Broadcom DSL XPON ASOC PCM Interface"); +MODULE_LICENSE("GPL v2"); -- 2.25.1

3 11

[PATCH 0/9] ASoC: soc-pcm cleanup step6
by Kuninori Morimoto 10 Mar '20

10 Mar '20

Hi Mark We are using plural form for for_each_xxx() macro. But, for_each_rtd_codec/cpu_dai() are out of this rule. 1) - 7) are for it. 8) - 9) add new for_each_card_xxx() macros. Kuninori Morimoto (9): 1) ASoC: soc.h: add for_each_rtd_codecs/cpus_dai() macro 2) ASoC: Intel: use for_each_rtd_codecs/cpus_dai() macro 3) ASoC: mediatek: use for_each_rtd_codecs/cpus_dai() macro 4) ASoC: meson: use for_each_rtd_codecs/cpus_dai() macro 5) ASoC: qcom: use for_each_rtd_codecs/cpus_dai() macro 6) ASoC: soc: use for_each_rtd_codecs/cpus_dai() macro 7) ASoC: soc.h: remove non plural form for_each_xxx macro 8) ASoC: soc-dapm: add for_each_card_dapms() macro 9) ASoC: soc-dapm: add for_each_card_widgets() macro drivers/soundwire/qcom.c | 2 +- include/sound/soc.h | 28 ++-- sound/soc/intel/boards/cml_rt1011_rt5682.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 4 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 4 +- sound/soc/meson/axg-card.c | 2 +- sound/soc/meson/meson-card-utils.c | 2 +- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/sdm845.c | 8 +- sound/soc/soc-core.c | 16 +-- sound/soc/soc-dapm.c | 53 ++++---- sound/soc/soc-pcm.c | 124 +++++++++--------- sound/soc/soc-topology.c | 2 +- 18 files changed, 134 insertions(+), 125 deletions(-) -- 2.17.1

4 20

[PATCH] ASoC: SOF: Fix probe point getter
by Cezary Rojewski 10 Mar '20

10 Mar '20

Firmware API changes which introduced 'num_elems' param in several probe structs such as sof_ipc_probe_dma_add_params also impacted getter for both, DMA and probe points. All struct handlers except for sof_ipc_probe_info_params have been updated. Align said handler too to calculate payload size correctly. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Fixes: f3b433e4699f ("ASoC: SOF: Implement Probe IPC API") --- sound/soc/sof/probe.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/probe.c b/sound/soc/sof/probe.c index 2b2f3dcfc7e9..c38169fe00c5 100644 --- a/sound/soc/sof/probe.c +++ b/sound/soc/sof/probe.c @@ -95,13 +95,17 @@ static int sof_ipc_probe_info(struct snd_sof_dev *sdev, unsigned int cmd, if (!reply->num_elems) goto exit; - bytes = reply->num_elems * sizeof(reply->dma[0]); + if (cmd == SOF_IPC_PROBE_DMA_INFO) + bytes = sizeof(reply->dma[0]); + else + bytes = sizeof(reply->desc[0]); + bytes *= reply->num_elems; *params = kmemdup(&reply->dma[0], bytes, GFP_KERNEL); if (!*params) { ret = -ENOMEM; goto exit; } - *num_params = msg.num_elems; + *num_params = reply->num_elems; exit: kfree(reply); -- 2.17.1

3 2

[PATCH 0/2] ASoC: soc-core: disable checks on dapm_add_routes for legacy devices
by Pierre-Louis Bossart 10 Mar '20

10 Mar '20

Since v5.4, the card probe ends when one or more DAPM routes cannot be added. This is a good thing, but legacy devices have incomplete routes that cannot practically be fixed by distributing a new topology file, so users lose all audio functionality by updating their kernel. Introduce an optional flag to allow for a backwards-compatible behavior to keep supporting these devices. For now only set this flag for the one device where an issue was reported. If desired, we could set it for all SKL machine drivers by default. Pierre-Louis Bossart (2): ASoC: soc-core: disable route checks for legacy devices ASoC: Intel: skl_nau88l25_ssm4567: disable route checks include/sound/soc.h | 1 + sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 1 + sound/soc/soc-core.c | 28 ++++++++++++++++--- 3 files changed, 26 insertions(+), 4 deletions(-) -- 2.20.1

3 5

[asoc:for-5.7 149/182] sound/soc/codecs/rt5682.c:3698:1: error: type defaults to 'int' in declaration of 'module_i2c_driver'
by kbuild test robot 10 Mar '20

10 Mar '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.7 head: 62f9ed5f8768d2425461737d77b83f888b525c06 commit: 03f6fc6de9192f4e4209ceee0e92f5947d44fc0a [149/182] ASoC: rt5682: Add the soundwire support config: sparc-randconfig-a001-20200310 (attached as .config) compiler: sparc-linux-gcc (GCC) 7.5.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 03f6fc6de9192f4e4209ceee0e92f5947d44fc0a # save the attached .config to linux build tree GCC_VERSION=7.5.0 make.cross ARCH=sparc If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All error/warnings (new ones prefixed by >>): >> sound/soc/codecs/rt5682.c:3698:1: warning: data definition has no type or storage class module_i2c_driver(rt5682_i2c_driver); ^~~~~~~~~~~~~~~~~ >> sound/soc/codecs/rt5682.c:3698:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] >> sound/soc/codecs/rt5682.c:3698:1: warning: parameter names (without types) in function declaration sound/soc/codecs/rt5682.c:3688:26: warning: 'rt5682_i2c_driver' defined but not used [-Wunused-variable] static struct i2c_driver rt5682_i2c_driver = { ^~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +3698 sound/soc/codecs/rt5682.c 0ddce71c21f03f Bard Liao 2018-06-07 3687 0ddce71c21f03f Bard Liao 2018-06-07 3688 static struct i2c_driver rt5682_i2c_driver = { 0ddce71c21f03f Bard Liao 2018-06-07 3689 .driver = { 0ddce71c21f03f Bard Liao 2018-06-07 3690 .name = "rt5682", 0ddce71c21f03f Bard Liao 2018-06-07 3691 .of_match_table = of_match_ptr(rt5682_of_match), 0ddce71c21f03f Bard Liao 2018-06-07 3692 .acpi_match_table = ACPI_PTR(rt5682_acpi_match), 0ddce71c21f03f Bard Liao 2018-06-07 3693 }, 0ddce71c21f03f Bard Liao 2018-06-07 3694 .probe = rt5682_i2c_probe, 0ddce71c21f03f Bard Liao 2018-06-07 3695 .shutdown = rt5682_i2c_shutdown, 0ddce71c21f03f Bard Liao 2018-06-07 3696 .id_table = rt5682_i2c_id, 0ddce71c21f03f Bard Liao 2018-06-07 3697 }; 0ddce71c21f03f Bard Liao 2018-06-07 @3698 module_i2c_driver(rt5682_i2c_driver); 0ddce71c21f03f Bard Liao 2018-06-07 3699 :::::: The code at line 3698 was first introduced by commit :::::: 0ddce71c21f03fd19867c4939d3ca710f37cdf1a ASoC: rt5682: add rt5682 codec driver :::::: TO: Bard Liao <bardliao(a)realtek.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

[PATCH v2] ALSA: pcm: oss: Simplify plugin frame size calculations
by Takashi Iwai 09 Mar '20

09 Mar '20

Both snd_pcm_plug_client_size() and snd_pcm_plug_slave_size() do the almost same calculations of calling src_frames() and dst_frames() in the chain, but just to the different directions with each other. This patch simplifies those functions. Now they return -EINVAL for the invalid direction, but practically seen, there is no functional changes at all. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_plugin.c | 120 +++++++++++++++++++++----------------------- 1 file changed, 56 insertions(+), 64 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index c9401832967c..58642fecd15c 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -196,82 +196,74 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin) return 0; } -snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames) +static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug, + snd_pcm_sframes_t frames) { - struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; - int stream; + struct snd_pcm_plugin *plugin, *plugin_next; - if (snd_BUG_ON(!plug)) - return -ENXIO; - if (drv_frames == 0) - return 0; - stream = snd_pcm_plug_stream(plug); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - plugin = snd_pcm_plug_last(plug); - while (plugin && drv_frames > 0) { - if (drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; - plugin_prev = plugin->prev; - if (plugin->src_frames) - drv_frames = plugin->src_frames(plugin, drv_frames); - plugin = plugin_prev; + plugin = snd_pcm_plug_first(plug); + while (plugin && frames > 0) { + plugin_next = plugin->next; + if (plugin->dst_frames) { + frames = plugin->dst_frames(plugin, frames); + if (frames < 0) + return frames; } - } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { - plugin = snd_pcm_plug_first(plug); - while (plugin && drv_frames > 0) { - plugin_next = plugin->next; - if (plugin->dst_frames) - drv_frames = plugin->dst_frames(plugin, drv_frames); - if (drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; - plugin = plugin_next; + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; + plugin = plugin_next; + } + return frames; +} + +static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug, + snd_pcm_sframes_t frames) +{ + struct snd_pcm_plugin *plugin, *plugin_prev; + + plugin = snd_pcm_plug_last(plug); + while (plugin && frames > 0) { + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; + plugin_prev = plugin->prev; + if (plugin->src_frames) { + frames = plugin->src_frames(plugin, frames); + if (frames < 0) + return frames; } - } else + plugin = plugin_prev; + } + return frames; +} + +snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames) +{ + if (snd_BUG_ON(!plug)) + return -ENXIO; + switch (snd_pcm_plug_stream(plug)) { + case SNDRV_PCM_STREAM_PLAYBACK: + return calc_src_frames(plug, drv_frames); + case SNDRV_PCM_STREAM_CAPTURE: + return calc_dst_frames(plug, drv_frames); + default: snd_BUG(); - return drv_frames; + return -EINVAL; + } } snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames) { - struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; - snd_pcm_sframes_t frames; - int stream; - if (snd_BUG_ON(!plug)) return -ENXIO; - if (clt_frames == 0) - return 0; - frames = clt_frames; - stream = snd_pcm_plug_stream(plug); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - plugin = snd_pcm_plug_first(plug); - while (plugin && frames > 0) { - plugin_next = plugin->next; - if (plugin->dst_frames) { - frames = plugin->dst_frames(plugin, frames); - if (frames < 0) - return frames; - } - if (frames > plugin->buf_frames) - frames = plugin->buf_frames; - plugin = plugin_next; - } - } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { - plugin = snd_pcm_plug_last(plug); - while (plugin) { - if (frames > plugin->buf_frames) - frames = plugin->buf_frames; - plugin_prev = plugin->prev; - if (plugin->src_frames) { - frames = plugin->src_frames(plugin, frames); - if (frames < 0) - return frames; - } - plugin = plugin_prev; - } - } else + switch (snd_pcm_plug_stream(plug)) { + case SNDRV_PCM_STREAM_PLAYBACK: + return calc_dst_frames(plug, clt_frames); + case SNDRV_PCM_STREAM_CAPTURE: + return calc_src_frames(plug, clt_frames); + default: snd_BUG(); - return frames; + return -EINVAL; + } } static int snd_pcm_plug_formats(const struct snd_mask *mask, -- 2.16.4

1 0

[PATCH] ALSA: pcm: oss: Simplify plugin frame size calculations
by Takashi Iwai 09 Mar '20

09 Mar '20

Both snd_pcm_plug_client_size() and snd_pcm_plug_slave_size() do the almost same calculations of calling src_frames() and dst_frames() in the chain, but just to the different directions with each other. This patch simplifies those functions. Now they return -EINVAL for the invalid direction, but practically seen, there is no functional changes at all. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_plugin.c | 120 +++++++++++++++++++++----------------------- 1 file changed, 56 insertions(+), 64 deletions(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index c9401832967c..3978f6d99076 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -196,82 +196,74 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin) return 0; } -snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames) +static snd_pcm_sframes_t calc_dst_frames(struct snd_pcm_substream *plug, + snd_pcm_uframes_t frames) { - struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; - int stream; + struct snd_pcm_plugin *plugin, *plugin_next; - if (snd_BUG_ON(!plug)) - return -ENXIO; - if (drv_frames == 0) - return 0; - stream = snd_pcm_plug_stream(plug); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - plugin = snd_pcm_plug_last(plug); - while (plugin && drv_frames > 0) { - if (drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; - plugin_prev = plugin->prev; - if (plugin->src_frames) - drv_frames = plugin->src_frames(plugin, drv_frames); - plugin = plugin_prev; + plugin = snd_pcm_plug_first(plug); + while (plugin && frames > 0) { + plugin_next = plugin->next; + if (plugin->dst_frames) { + frames = plugin->dst_frames(plugin, frames); + if (frames < 0) + return frames; } - } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { - plugin = snd_pcm_plug_first(plug); - while (plugin && drv_frames > 0) { - plugin_next = plugin->next; - if (plugin->dst_frames) - drv_frames = plugin->dst_frames(plugin, drv_frames); - if (drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; - plugin = plugin_next; + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; + plugin = plugin_next; + } + return frames; +} + +static snd_pcm_sframes_t calc_src_frames(struct snd_pcm_substream *plug, + snd_pcm_uframes_t frames) +{ + struct snd_pcm_plugin *plugin, *plugin_prev; + + plugin = snd_pcm_plug_last(plug); + while (plugin) { + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; + plugin_prev = plugin->prev; + if (plugin->src_frames) { + frames = plugin->src_frames(plugin, frames); + if (frames < 0) + return frames; } - } else + plugin = plugin_prev; + } + return frames; +} + +snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames) +{ + if (snd_BUG_ON(!plug)) + return -ENXIO; + switch (snd_pcm_plug_stream(plug)) { + case SNDRV_PCM_STREAM_PLAYBACK: + return calc_src_frames(plug, drv_frames); + case SNDRV_PCM_STREAM_CAPTURE: + return calc_dst_frames(plug, drv_frames); + default: snd_BUG(); - return drv_frames; + return -EINVAL; + } } snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames) { - struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; - snd_pcm_sframes_t frames; - int stream; - if (snd_BUG_ON(!plug)) return -ENXIO; - if (clt_frames == 0) - return 0; - frames = clt_frames; - stream = snd_pcm_plug_stream(plug); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - plugin = snd_pcm_plug_first(plug); - while (plugin && frames > 0) { - plugin_next = plugin->next; - if (plugin->dst_frames) { - frames = plugin->dst_frames(plugin, frames); - if (frames < 0) - return frames; - } - if (frames > plugin->buf_frames) - frames = plugin->buf_frames; - plugin = plugin_next; - } - } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { - plugin = snd_pcm_plug_last(plug); - while (plugin) { - if (frames > plugin->buf_frames) - frames = plugin->buf_frames; - plugin_prev = plugin->prev; - if (plugin->src_frames) { - frames = plugin->src_frames(plugin, frames); - if (frames < 0) - return frames; - } - plugin = plugin_prev; - } - } else + switch (snd_pcm_plug_stream(plug)) { + case SNDRV_PCM_STREAM_PLAYBACK: + return calc_dst_frames(plug, clt_frames); + case SNDRV_PCM_STREAM_CAPTURE: + return calc_src_frames(plug, clt_frames); + default: snd_BUG(); - return frames; + return -EINVAL; + } } static int snd_pcm_plug_formats(const struct snd_mask *mask, -- 2.16.4

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about alc880 no sound at ecs g420 and eapd coef
by 尤晓杰 09 Mar '20

09 Mar '20

dear friend, I am using ecs g420 laptop, use options snd-hda-intel model=generic the speaker and mic correctly, 2 pop sounds,when the kernel load(left channel) and the x11 load(right channel), but with standard way only one pop sound when kernel load and silent always. the mic is ok. I have got datasheet of alc880 and see the eapd function, but in your code I see this need coef parameter. so what should I write using verb to open eapd?the physical connexion in summary: an externel amplifier in charge of jack sense to mute speaker, connected with the 0x14 hp out, 0x18 mic in,0x1c cd in, 0x1e spif out,0x1d pcbeep in , no jack sense. eapd shared with spdif in. all are correct with default of datasheet's describe. I use verb to dectt each pin widget's capability , take 0x14 as exaple, no eapd capability.To set 0x14's eapd with hda-verb /dev/snd/hwC0D0 0x70c 0x2, failed to read it again alwas 0. and bios also reported 0x16, I think it is bios's bug. if not changing source code , to use "Early Patching" method to deal with it without changing source code, is it feasible? could you give me some idea? thanks!!!! dmesg: [ 8.009944] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC880: line_outs=2 (0x14/0x16/0x0/0x0/0x0) type:hp [ 8.014023] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.017999] snd_hda_codec_realtek hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.021625] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 8.024255] snd_hda_codec_realtek hdaudioC0D0: dig-out=0x1e/0x0 [ 8.026741] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 8.038468] mc: Linux media interface: v0.10 [ 8.067515] snd_hda_codec_realtek hdaudioC0D0: Mic=0x18 [ 8.078984] snd_hda_codec_realtek hdaudioC0D0: CD=0x1c [ 8.290306] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card0/input13 Node 0x02 [Audio Output] wcaps 0x411: Stereo Device: name="ALC880 Analog", type="Audio", device=0 Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: setting=D0, actual=D0 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x02 0x0b Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 Control: name="CD Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=1, idx=4, ofs=0 Control: name="CD Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=1, idx=4, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=1, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=1, idx=5, ofs=0 Amp-In caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=1 Amp-In vals: [0x41 0x41] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x41 0x41] [0x41 0x41] [0x80 0x80] [0x80 0x80] Connection: 8 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x01211110: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP Unsolicited: tag=0x00, enabled=0 Connection: 1 0x0c Codec: 0x10ec0880 Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10ec0880 Subsystem Id: 0x10191205 Revision Id: 0x100800 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0

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[RESEND RFC PATCH] soundwire: bus: Add flag to mark DPN_BlockCtrl1 as readonly
by Srinivas Kandagatla 09 Mar '20

09 Mar '20

According to SoundWire Specification Version 1.2. "A Data Port number X (in the range 0-14) which supports only one value of WordLength may implement the WordLength field in the DPX_BlockCtrl1 Register as Read-Only, returning the fixed value of WordLength in response to reads." As WSA881x interfaces in PDM mode making the only field "WordLength" in DPX_BlockCtrl1" fixed and read-only. Behaviour of writing to this register on WSA881x soundwire slave with Qualcomm Soundwire Controller is throwing up an error. Not sure how other controllers deal with writing to readonly registers, but this patch provides a way to avoid writes to DPN_BlockCtrl1 register by providing a ro_blockctrl1_reg flag in struct sdw_port_runtime. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Sorry resending this one as I missed header file changes in my previous patch. I will send patch for WSA881x to include this change once this patch is accepted. drivers/soundwire/bus.h | 2 ++ drivers/soundwire/stream.c | 17 ++++++++++------- include/linux/soundwire/sdw.h | 1 + 3 files changed, 13 insertions(+), 7 deletions(-) diff --git a/drivers/soundwire/bus.h b/drivers/soundwire/bus.h index 204204a26db8..791e8d14093e 100644 --- a/drivers/soundwire/bus.h +++ b/drivers/soundwire/bus.h @@ -79,6 +79,7 @@ int sdw_find_col_index(int col); * @num: Port number. For audio streams, valid port number ranges from * [1,14] * @ch_mask: Channel mask + * @ro_blockctrl1_reg: Read Only flag for DPN_BlockCtrl1 register * @transport_params: Transport parameters * @port_params: Port parameters * @port_node: List node for Master or Slave port_list @@ -89,6 +90,7 @@ int sdw_find_col_index(int col); struct sdw_port_runtime { int num; int ch_mask; + bool ro_blockctrl1_reg; struct sdw_transport_params transport_params; struct sdw_port_params port_params; struct list_head port_node; diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 00348d1fc606..4491643aeb4a 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -167,13 +167,15 @@ static int sdw_program_slave_port_params(struct sdw_bus *bus, return ret; } - /* Program DPN_BlockCtrl1 register */ - ret = sdw_write(s_rt->slave, addr2, (p_params->bps - 1)); - if (ret < 0) { - dev_err(&s_rt->slave->dev, - "DPN_BlockCtrl1 register write failed for port %d\n", - t_params->port_num); - return ret; + if (!p_rt->ro_blockctrl1_reg) { + /* Program DPN_BlockCtrl1 register */ + ret = sdw_write(s_rt->slave, addr2, (p_params->bps - 1)); + if (ret < 0) { + dev_err(&s_rt->slave->dev, + "DPN_BlockCtrl1 register write failed for port %d\n", + t_params->port_num); + return ret; + } } /* Program DPN_SampleCtrl1 register */ @@ -1195,6 +1197,7 @@ static struct sdw_port_runtime p_rt->ch_mask = port_config[port_index].ch_mask; p_rt->num = port_config[port_index].num; + p_rt->ro_blockctrl1_reg = port_config[port_index].ro_blockctrl1_reg; return p_rt; } diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index ee349a4c5349..8b130855acd1 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -842,6 +842,7 @@ void sdw_delete_bus_master(struct sdw_bus *bus); struct sdw_port_config { unsigned int num; unsigned int ch_mask; + bool ro_blockctrl1_reg; }; /** -- 2.21.0

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[PATCH] ASoC: simple-card-utils: use for_each_pcm_streams()
by Kuninori Morimoto 09 Mar '20

09 Mar '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We already have for_each_pcm_streams() macro. Let's use it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 1ab3745547e0..189c4db020cd 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -347,7 +347,7 @@ static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd, } /* Assumes the capabilities are the same for all supported streams */ - for (stream = 0; stream < 2; stream++) { + for_each_pcm_streams(stream) { ret = snd_soc_runtime_calc_hw(rtd, &hw, stream); if (ret == 0) break; -- 2.17.1

2 1

[PATCH] ASoC: amd: Adding TDM support in hw_params.
by Ravulapati Vishnu vardhan rao 09 Mar '20

09 Mar '20

TDM related settings for ACP registers in hw_params. When TDM mode is enabled, Hw_params needs to read and write from/to respective TX/RX (ACP_(I2S/BT)TDM_(TX/RX)FRMT) registers. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-i2s.c | 44 ++++++++++++----------------------------- 1 file changed, 13 insertions(+), 31 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index 91a3881..3a3c47e 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -42,7 +42,7 @@ static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, u32 rx_mask, int slots, int slot_width) { struct i2s_dev_data *adata; - u32 val, reg_val, frmt_reg, frm_len; + u32 frm_len; u16 slot_len; adata = snd_soc_dai_get_drvdata(cpu_dai); @@ -64,36 +64,7 @@ static int acp3x_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } - - /* Enable I2S/BT channels TDM, respective TX/RX frame lengths.*/ - frm_len = FRM_LEN | (slots << 15) | (slot_len << 18); - if (adata->substream_type == SNDRV_PCM_STREAM_PLAYBACK) { - switch (adata->i2s_instance) { - case I2S_BT_INSTANCE: - reg_val = mmACP_BTTDM_ITER; - frmt_reg = mmACP_BTTDM_TXFRMT; - break; - case I2S_SP_INSTANCE: - default: - reg_val = mmACP_I2STDM_ITER; - frmt_reg = mmACP_I2STDM_TXFRMT; - } - } else { - switch (adata->i2s_instance) { - case I2S_BT_INSTANCE: - reg_val = mmACP_BTTDM_IRER; - frmt_reg = mmACP_BTTDM_RXFRMT; - break; - case I2S_SP_INSTANCE: - default: - reg_val = mmACP_I2STDM_IRER; - frmt_reg = mmACP_I2STDM_RXFRMT; - } - } - val = rv_readl(adata->acp3x_base + reg_val); - rv_writel(val | 0x2, adata->acp3x_base + reg_val); - rv_writel(frm_len, adata->acp3x_base + frmt_reg); adata->tdm_fmt = frm_len; return 0; } @@ -105,12 +76,14 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *prtd; struct snd_soc_card *card; struct acp3x_platform_info *pinfo; + struct i2s_dev_data *adata; u32 val; - u32 reg_val; + u32 reg_val, frmt_reg; prtd = substream->private_data; rtd = substream->runtime->private_data; card = prtd->card; + adata = snd_soc_dai_get_drvdata(dai); pinfo = snd_soc_card_get_drvdata(card); if (pinfo) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -141,21 +114,30 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: reg_val = mmACP_BTTDM_ITER; + frmt_reg = mmACP_BTTDM_TXFRMT; break; case I2S_SP_INSTANCE: default: reg_val = mmACP_I2STDM_ITER; + frmt_reg = mmACP_I2STDM_TXFRMT; } } else { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: reg_val = mmACP_BTTDM_IRER; + frmt_reg = mmACP_BTTDM_RXFRMT; break; case I2S_SP_INSTANCE: default: reg_val = mmACP_I2STDM_IRER; + frmt_reg = mmACP_I2STDM_RXFRMT; } } + if (adata->tdm_mode) { + val = rv_readl(rtd->acp3x_base + reg_val); + rv_writel(val | 0x2, rtd->acp3x_base + reg_val); + rv_writel(adata->tdm_fmt, rtd->acp3x_base + frmt_reg); + } val = rv_readl(rtd->acp3x_base + reg_val); val = val | (rtd->xfer_resolution << 3); rv_writel(val, rtd->acp3x_base + reg_val); -- 2.7.4

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Applied "ASoC: jz4740-i2s: Fix divider written at incorrect offset in register" to the asoc tree
by Mark Brown 09 Mar '20

09 Mar '20

The patch ASoC: jz4740-i2s: Fix divider written at incorrect offset in register has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 9401d5aa328e64617d87abd59af1c91cace4c3e4 Mon Sep 17 00:00:00 2001 From: Paul Cercueil <paul(a)crapouillou.net> Date: Fri, 6 Mar 2020 23:29:27 +0100 Subject: [PATCH] ASoC: jz4740-i2s: Fix divider written at incorrect offset in register The 4-bit divider value was written at offset 8, while the jz4740 programming manual locates it at offset 0. Fixes: 26b0aad80a86 ("ASoC: jz4740: Add dynamic sampling rate support to jz4740-i2s") Signed-off-by: Paul Cercueil <paul(a)crapouillou.net> Cc: stable(a)vger.kernel.org Link: https://lore.kernel.org/r/20200306222931.39664-2-paul@crapouillou.net Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/jz4740/jz4740-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 9d5405881209..434737b2b2b2 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -83,7 +83,7 @@ #define JZ_AIC_I2S_STATUS_BUSY BIT(2) #define JZ_AIC_CLK_DIV_MASK 0xf -#define I2SDIV_DV_SHIFT 8 +#define I2SDIV_DV_SHIFT 0 #define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT) #define I2SDIV_IDV_SHIFT 8 #define I2SDIV_IDV_MASK (0xf << I2SDIV_IDV_SHIFT) -- 2.20.1

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[PATCH Fix for Kingston HyperX Amp (0951:16d8) 0/1] snd-usb-audio: Fix Kingston HyperX Amp (0951:16d8).
by crwulff＠gmail.com 09 Mar '20

09 Mar '20

From: Chris Wulff <crwulff(a)gmail.com> The newer version of the HyperX Amp USB audio sound card appears to use two separate interfaces for controls and audio. This doesn't work well with the current linux kernel (tested with 5.4). Interface 0 has just a mute control, while interface 2 has mute and volume controls for both playback and capture. This appears to cause a couple different issues. The first is that the wrong interface is used to query most of the controls, resulting in lots of messages like this: usb 1-3: 12:0: cannot get min/max values for control 2 (id 12) The second problem is that since it is enumerated as two separate interfaces, the device gets registered and then the capture stream gets merged into the already-registered pcm device (which results in the udev events not being generated for the capture stream and no /dev/snd/pcmC#D#c file.) This patch fixes both those problems, and allows the device to be used with ALSA programs, but pulseaudio does not enumerate the capture device correctly since it is pcmC#D1c and there is no pcmC#D0c. Adding a symlink from pcmC#D0c -> pcmC#D1c fixes pulseaudio but I'm not sure exactly why pulseaudio can't find the device without it.

2 2

[PATCH] ALSA: line6: Fix endless MIDI read loop
by Takashi Iwai 09 Mar '20

09 Mar '20

The MIDI input event parser of the LINE6 driver may enter into an endless loop when the unexpected data sequence is given, as it tries to continue the secondary bytes without termination. Also, when the input data is too short, the parser returns a negative error, while the caller doesn't handle it properly. This would lead to the unexpected behavior as well. This patch addresses those issues by checking the return value correctly and handling the one-byte event in the parser properly. The bug was reported by syzkaller. Reported-by: syzbot+cce32521ee0a824c21f7(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000033087059f8f8fa3@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/line6/driver.c | 2 +- sound/usb/line6/midibuf.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index b5a3f754a4f1..4f096685ed65 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -305,7 +305,7 @@ static void line6_data_received(struct urb *urb) line6_midibuf_read(mb, line6->buffer_message, LINE6_MIDI_MESSAGE_MAXLEN); - if (done == 0) + if (done <= 0) break; line6->message_length = done; diff --git a/sound/usb/line6/midibuf.c b/sound/usb/line6/midibuf.c index 8d6eefa0d936..6a70463f82c4 100644 --- a/sound/usb/line6/midibuf.c +++ b/sound/usb/line6/midibuf.c @@ -159,7 +159,7 @@ int line6_midibuf_read(struct midi_buffer *this, unsigned char *data, int midi_length_prev = midibuf_message_length(this->command_prev); - if (midi_length_prev > 0) { + if (midi_length_prev > 1) { midi_length = midi_length_prev - 1; repeat = 1; } else -- 2.16.4

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