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

[alsa-devel] Simultaneous play/record on implicit feedback device causes 'endpoint in use' error
by Robert Giaraffa 15 Jan '20

15 Jan '20

My company (Silicon Labs) makes a CP2615 USB audio class 1.0 device which uses implicit feedback. I saw these two issues when testing with aplay/arecord/alsabat/alsaloop: * During playback-only operation, the implicit feedback (IN) endpoint was not active. * During play/record operation, the implicit feedback values were not acted on, i.e. the host always sent 192 bytes per OUT packet even if previous IN packets had been 188 or 196 bytes. This behavior was observed on these systems: RaspberryPi2: Raspbian 9 (stretch), kernel v4.14.72 x86-64: Ubuntu 18.04, kernel v4.15.0-36 I was able to get implicit feedback working properly in playback mode by adding a device-specific quirk to 'pcm.c' that is similar to existing quirks for implicit-feedback devices, e.g. M-Audio Fast Track Ultra. (Note: The following patch applies to kernel v4.14.72. In v4.18, the set_sync_ep_implicit_fb_quirk function was changed so that there is a single call to usb_ifnum_to_if(), rather than a call for each device-dependent quirk.) $ git diff pcm.c diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b9c9a19f..6fb5567b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -358,6 +358,21 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, alts = &iface->altsetting[1]; goto add_sync_ep; + case USB_ID(0x10C4, 0xEAC1): // Silicon Labs CP2615 + // Ensure CP2615 is configured in Async Mode + if (attr != USB_ENDPOINT_SYNC_ASYNC) + return 0; + + // In P16R16 async mode, AudioIN EP (IF:4 EP:0x83) used for implicit feedback + ep = 0x83; + iface = usb_ifnum_to_if(dev, 4); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + } if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && After adding the patch, I verified that implicit feedback worked properly for playback, i.e. the OUT samples per packet tracked previous IN values. However, when attempting to perform simultaneous play and record via alsabat, alsaloop, or separate aplay/arecord, the system reports the 'endpoint already in use' error when attempting to set the format for the IN endpoint that is already in use (note that this dmesg log has customized printk messages): [Oct23 10:13] snd_usb_hw_params->set_format() [ +0.000014] set_format(): dev:b3856c00 alts:b67e8824 ifnum: 3 [ +0.001654] set_format()->usb_set_interface 3:1 [ +0.000006] Creating new playback data endpoint 03 [ +0.000004] Creating new capture data endpoint 83 <Begin Playback. SYNC endpoint also starts because feedback is implicit> [ +0.000027] snd_usb_pcm_prepare->set_format() [ +0.000005] set_format(): dev:b3856c00 alts:b67e8824 ifnum: 3 [ +0.000755] Setting params for EP 03 (type 0, 12 urbs), ret=0 [ +0.000017] Setting params for EP 83 (type 0, 12 urbs), ret=0 [ +0.000004] Starting DATA_EP 03 @b9224000 [ +0.000004] snd_usb_endpoint_implicit_feedback_sink (03) is TRUE [ +0.001495] Starting SYNC_EP 83 @b9222000 <After playing for 0.5s, start recording. Failure occurs because EP83 is already in use.> [ +0.493035] snd_usb_hw_params->set_format() [ +0.000016] set_format(): dev:b3856c00 alts:b67e88a4 ifnum: 4 [ +0.001549] set_format()->usb_set_interface 4:1 [ +0.000005] Re-using EP 83 in iface 4:1 [ +0.000028] snd_usb_pcm_prepare->set_format() [ +0.000004] set_format(): dev:b3856c00 alts:b67e88a4 ifnum: 4 *** [ +0.003196] *** snd_usb_endpoint_set_params(): EP #83: already in use [ +0.000003] configure_endpoint->endpoint_set_params: err=FFFFFFF0 I bought a used Fast Track Ultra device for comparison testing, and found that it behaved the same way as the CP2615 does with the implicit feedback quirk: in playback-only mode implicit feedback works properly, but play/record causes the 'EP already in use' error. Are there any suggestions on how I can resolve this issue? Any help would be greatly appreciated! Thanks, Bob Giaraffa

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[alsa-devel] [PATCH v2 0/2] ALSA: usb-audio: fix PCM device order
by Takashi Iwai 30 Dec '19

30 Dec '19

Hi, this is a rewrite of the old USB-audio fix patch I obviously overlooked to adapt to 5.3 kernel with a preliminary cleanup patch. Totally untested, so any test / feedback would be appreciated. thanks, Takashi === Alexander Tsoy (1): ALSA: usb-audio: fix PCM device order Takashi Iwai (1): ALSA: usb-audio: Unify audioformat release code sound/usb/stream.c | 64 +++++++++++++++++++++++++++++++++++++++--------------- 1 file changed, 47 insertions(+), 17 deletions(-) -- 2.16.4

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[alsa-devel] Surround speaker connection on Acer 8951G
by Sergey 'Jin' Bostandzhyan 28 Nov '19

28 Nov '19

Hi, I am not sure if this is the right place to ask or if I should just submit a bugreport, but you guys helped me out with a similar problem about 10 years ago, so I'll try here first. If I'm wrong here, then I apologize, just let me know. I have an Acer Aspire Ethos 8951G notebook which has 6 built in speakers for 5.1 surround sound, it's a Realtek ALC669X. The problem is that only two speakers are recognized by default. I tried fiddling around with hdajackretast and it does make a difference, but I somehow could not find a step-by-step approach on how to figure out what is what and how to map everything correctly, so some guidance would be very much appreciated. I even ended up in situations where I got channels supporting only mute which would influence muting of other channels, I really got lost there.. Here's my alsa-info output: http://alsa-project.org/db/?f=3adde87164f5fc349c3c78b211ee63e416ebf10b It would be great if we could figure it out and make it work by default. Basically that's what we did 10 years ago, back then I had an Acer Aspire 8920G with a ALC889 which had the same problem that got fixed by you and these days it just works out of the box :) Kind regards, Jin

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[alsa-devel] Focusrite Scarlett 6i6 gen1 - input handling fix
by Jens Verwiebe 19 Nov '19

19 Nov '19

The Scarlett 6i6 has no padding on rear inputs 3/4 but a gainstage. This patch introduces this functionality as to be seen in the mac or windows scarlett control. The correct address could already be found in the dump info, but was never used. Cheers ... Jens Signed-off-by: Jens Verwiebe <info(a)jensverwiebe.de> diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 7438e7c..f3bdac3 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -152,6 +152,7 @@ enum { SCARLETT_OUTPUTS, SCARLETT_SWITCH_IMPEDANCE, SCARLETT_SWITCH_PAD, + SCARLETT_SWITCH_GAIN, }; enum { @@ -202,6 +203,15 @@ struct scarlett_device_info { } }; +static const struct scarlett_mixer_elem_enum_info opt_gain = { + .start = 0, + .len = 2, + .offsets = {}, + .names = (char const * const []){ + "Lo", "Hi" + } +}; + static const struct scarlett_mixer_elem_enum_info opt_impedance = { .start = 0, .len = 2, @@ -664,8 +674,8 @@ static int add_output_ctls(struct usb_mixer_interface *mixer, { .num = 1, .type = SCARLETT_SWITCH_PAD, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 3, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 4, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 3, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, + { .num = 4, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, }, .matrix_mux_init = { @@ -895,6 +905,15 @@ static int scarlett_controls_create_generic(struct usb_mixer_interface *mixer, if (err < 0) return err; break; + case SCARLETT_SWITCH_GAIN: + sprintf(mx, "Input %d Gain Switch", ctl->num); + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, + scarlett_ctl_enum_resume, 0x01, + 0x08, ctl->num, USB_MIXER_S16, 1, mx, + &opt_gain, &elem); + if (err < 0) + return err; + break; } } -- Jens Verwiebe Allerskehre 44 - 22309 Hamburg Tel.: +49 40 68 78 50 mobile: +49 172 400 49 07 mailto: info(a)jensverwiebe.de web: http://www.jensverwiebe.de

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[alsa-devel] [PATCH] ASoC: rsnd: dma: fix SSI9 4/5/6/7 busif dma address
by twischer＠de.adit-jv.com 22 Oct '19

22 Oct '19

From: Jiada Wang <jiada_wang(a)mentor.com> Currently each SSI unit 's busif dma address is calculated by following calculation formulation: 0xec540000 + 0x1000 * id + busif / 4 * 0xA000 + busif % 4 * 0x400 But according to user manual 41.1.4 Register Configuration ssi9 4/5/6/7 busif data register address (SSI9_4_BUSIF/SSI9_5_BUSIF/SSI9_6_BUSIF/SSI9_7_BUSIF) are out of this rule. This patch updates the calculation formulation to correct ssi9 4/5/6/7 busif data register address Fixes: commit 5e45a6fab3b9 ("ASoc: rsnd: dma: Calculate dma address with consider of BUSIF") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- sound/soc/sh/rcar/dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 0324a5c..28f65eb 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -508,10 +508,10 @@ static struct rsnd_mod_ops rsnd_dmapp_ops = { #define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) #define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) -#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_N(addr, i, j) RDMA_SSIU_I_N(addr, i, j) -#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_P(addr, i, j) RDMA_SSIU_I_P(addr, i, j) #define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) -- 2.7.4

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[alsa-devel] [PATCH 1/2] ASoC: adau1761: Add PGA Slew time control
by Alexandru Ardelean 01 Oct '19

01 Oct '19

From: Lars-Peter Clausen <lars(a)metafoo.de> The PGA Slew Time control allows to configure the rate with which the PGA gain control ramps up/down to the target setting. The PGA slew control is done via the ALC Control 0 register. There are 2 bits on that reg, that control PGA slew time, while the other bits control parts of the ALC (automatic level control) block. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- sound/soc/codecs/adau1761.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..428bf95ea648 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -29,6 +29,7 @@ #define ADAU1761_REC_MIXER_RIGHT1 0x400d #define ADAU1761_LEFT_DIFF_INPUT_VOL 0x400e #define ADAU1761_RIGHT_DIFF_INPUT_VOL 0x400f +#define ADAU1761_ALC_CTRL0 0x4011 #define ADAU1761_PLAY_LR_MIXER_LEFT 0x4020 #define ADAU1761_PLAY_MIXER_LEFT0 0x401c #define ADAU1761_PLAY_MIXER_LEFT1 0x401d @@ -72,6 +73,7 @@ static const struct reg_default adau1761_reg_defaults[] = { { ADAU1761_REC_MIXER_RIGHT0, 0x00 }, { ADAU1761_REC_MIXER_RIGHT1, 0x00 }, { ADAU1761_LEFT_DIFF_INPUT_VOL, 0x00 }, + { ADAU1761_ALC_CTRL0, 0x00 }, { ADAU1761_RIGHT_DIFF_INPUT_VOL, 0x00 }, { ADAU1761_PLAY_LR_MIXER_LEFT, 0x00 }, { ADAU1761_PLAY_MIXER_LEFT0, 0x00 }, @@ -148,6 +150,21 @@ static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_capture_bias_enum, ADAU17X1_REC_POWER_MGMT, 1, 0x3, adau1761_bias_select_text, adau1761_bias_select_values); +static const unsigned int adau1761_pga_slew_time_values[] = { + 3, 0, 1, 2, +}; + +static const char * const adau1761_pga_slew_time_text[] = { + "Off", + "24 ms", + "48 ms", + "96 ms", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_pga_slew_time_enum, + ADAU1761_ALC_CTRL0, 6, 0x3, adau1761_pga_slew_time_text, + adau1761_pga_slew_time_values); + static const struct snd_kcontrol_new adau1761_jack_detect_controls[] = { SOC_SINGLE("Speaker Auto-mute Switch", ADAU1761_DIGMIC_JACKDETECT, 4, 1, 0), @@ -162,6 +179,8 @@ static const struct snd_kcontrol_new adau1761_differential_mode_controls[] = { SOC_DOUBLE_R_TLV("PGA Boost Capture Volume", ADAU1761_REC_MIXER_LEFT1, ADAU1761_REC_MIXER_RIGHT1, 3, 2, 0, adau1761_pga_boost_tlv), + + SOC_ENUM("PGA Capture Slew Time", adau1761_pga_slew_time_enum), }; static const struct snd_kcontrol_new adau1761_single_mode_controls[] = { @@ -633,6 +652,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg) case ADAU1761_DEJITTER: case ADAU1761_CLK_ENABLE0: case ADAU1761_CLK_ENABLE1: + case ADAU1761_ALC_CTRL0: return true; default: break; -- 2.20.1

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[alsa-devel] [PATCH v5 0/5] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 18 Sep '19

18 Sep '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. Changes from v4 to v5: - synopsys/Kconfig: Remove the incorrect dependency change in v4. - rockchip/Kconfig: Add dependency of hdmi-codec when it is really need for jack support. Cheng-Yi Chiang (5): ASoC: hdmi-codec: Add an op to set callback function for plug event drm: bridge: dw-hdmi: Report connector status using callback drm: dw-hdmi-i2s: Use fixed id for codec device ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip_max98090: Add HDMI jack support .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 13 +- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 ++++++- include/drm/bridge/dw_hdmi.h | 4 + include/sound/hdmi-codec.h | 17 +++ sound/soc/codecs/hdmi-codec.c | 46 +++++++ sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 3 +- sound/soc/rockchip/rockchip_max98090.c | 116 ++++++++++++++---- 8 files changed, 216 insertions(+), 27 deletions(-) -- 2.22.0.510.g264f2c817a-goog

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[alsa-devel] [PATCH] ASoC: rsnd: fixup 6ch settings to 8ch
by Kuninori Morimoto 30 Aug '19

30 Aug '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd need to use 8ch clock settings for 6ch for TDM. Otherwise, it can't work correctly. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 18 +++++++++++++----- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/ssi.c | 2 ++ 3 files changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4fe83e61..37cb615 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -300,6 +300,18 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, return chan; } +int rsnd_channel_normalization(int chan) +{ + if ((chan > 8) || (chan < 0)) + return 0; + + /* TDM Extend Mode needs 8ch */ + if (chan == 6) + chan = 8; + + return chan; +} + int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, struct snd_pcm_hw_params *params) { @@ -312,11 +324,7 @@ int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, if (rsnd_runtime_is_multi_ssi(io)) chan /= rsnd_rdai_ssi_lane_get(rdai); - /* TDM Extend Mode needs 8ch */ - if (chan == 6) - chan = 8; - - return chan; + return rsnd_channel_normalization(chan); } int rsnd_runtime_is_multi_ssi(struct rsnd_dai_stream *io) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 0e6ef4e1..7727add 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -446,6 +446,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, struct device_node *playback, struct device_node *capture); +int rsnd_channel_normalization(int chan); #define rsnd_runtime_channel_original(io) \ rsnd_runtime_channel_original_with_params(io, NULL) int rsnd_runtime_channel_original_with_params(struct rsnd_dai_stream *io, diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index f5afab6..44bda21 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -303,6 +303,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_runtime_is_tdm_split(io)) chan = rsnd_io_converted_chan(io); + chan = rsnd_channel_normalization(chan); + main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx); if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); -- 2.7.4

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[alsa-devel] [BUG] bdw-rt5650 DSP boot timeout
by Jon Flatley 28 Aug '19

28 Aug '19

I've been working on upstreaming the bdw-rt5650 machine driver for the Acer Chromebase 24 (buddy). There seems to be an issue when first setting the hardware controls that appears to be crashing the DSP: [ 51.424554] haswell-pcm-audio haswell-pcm-audio: FW loaded, mailbox readback FW info: type 01, - version: 00.00, build 77, source commit id: 876ac6906f31a43b6772b23c7c983ce9dcb18a19 ... [ 84.924666] haswell-pcm-audio haswell-pcm-audio: error: audio DSP boot timeout IPCD 0x0 IPCX 0x0 [ 85.260655] haswell-pcm-audio haswell-pcm-audio: ipc: --message timeout-- ipcx 0x83000000 isr 0x00000000 ipcd 0x00000000 imrx 0x7fff0000 [ 85.273609] haswell-pcm-audio haswell-pcm-audio: error: stream commit failed [ 85.279746] System PCM: error: failed to commit stream -110 [ 85.285388] haswell-pcm-audio haswell-pcm-audio: ASoC: haswell-pcm-audio hw params failed: -110 [ 85.293963] System PCM: ASoC: hw_params FE failed -110 This happens roughly 50% of the time when first setting hardware controls after a reboot. The other 50% of the time the DSP comes up just fine and audio works fine thereafter. Adding "#define DEBUG 1" to sound/soc/intel/haswell/sst-haswell-ipc.c makes the issue occur much less frequently in my testing. Seems like a subtle timing issue. There were timing issues encountered during the bringup of the 2015 chromebook pixel (samus) which uses the bdw-rt5677 machine driver. Those were slightly different, and manifested during repeated arecords. Both devices use the same revision of the sst2 firmware. Any ideas for how to debug this? Thanks, Jon

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[alsa-devel] [RFC PATCH 00/40] soundwire: updates for 5.4
by Pierre-Louis Bossart 23 Aug '19

23 Aug '19

The existing upstream code allows for SoundWire devices to be enumerated and managed by the bus, but streaming is not currently supported. Bard Liao, Rander Wang and I did quite a bit of integration/validation work to close this gap and we now have SoundWire streaming + basic power managemement on Intel CometLake and IceLake reference boards. These changes are still preliminary and should not be merged as is, but it's time to start reviews. While the number of patches is quite large, each of the changes is quite small. SOF driver changes will be submitted shortly as well but are still being validated. ClockStop modes and synchronized playback on multiple links are not supported for now and will likely be part of the next cycle (dependencies on codec drivers and multi-cpu DAI support). Acknowledgements: This work would not have been possible without the support of Slawomir Blauciak and Tomasz Lauda on the SOF side, currently being reviewed, see https://github.com/thesofproject/sof/pull/1638 Comments and feedback welcome! Bard liao (1): soundwire: include mod_devicetable.h to avoid compiling warnings Pierre-Louis Bossart (38): soundwire: add debugfs support soundwire: cadence_master: add debugfs register dump soundwire: cadence_master: align debugfs to 8 digits soundwire: intel: add debugfs register dump soundwire: intel: move interrupt enable after interrupt handler registration soundwire: intel: prevent possible dereference in hw_params soundwire: intel: fix channel number reported by hardware soundwire: intel: remove BIOS work-arounds soundwire: cadence_master: fix usage of CONFIG_UPDATE soundwire: cadence_master: remove useless wrapper soundwire: cadence_master: simplify bus clash interrupt clear soundwire: cadence_master: revisit interrupt settings soundwire: cadence_master: fix register definition for SLAVE_STATE soundwire: cadence_master: fix definitions for INTSTAT0/1 soundwire: cadence_master: handle multiple status reports per Slave soundwire: cadence_master: improve startup sequence with link hw_reset soundwire: bus: use runtime_pm_get_sync/pm when enabled soundwire: bus: split handling of Device0 events soundwire: bus: improve dynamic debug comments for enumeration soundwire: prototypes for suspend/resume soundwire: export helpers to find row and column values soundwire: stream: fix disable sequence soundwire: cadence_master: use BIOS defaults for frame shape soundwire: intel: use BIOS information to set clock dividers soundwire: Add Intel resource management algorithm soundwire: intel: handle disabled links soundwire: intel_init: add kernel module parameter to filter out links soundwire: cadence_master: add kernel parameter to override interrupt mask soundwire: intel: move shutdown() callback and don't export symbol soundwire: intel: add helper for initialization soundwire: intel: Add basic power management support soundwire: intel: ignore disabled links for suspend/resume soundwire: intel: export helper to exit reset soundwire: intel: disable interrupts on suspend soundwire: cadence_master: add hw_reset capability in debugfs soundwire: cadence_master: make clock stop exit configurable on init soundwire: intel: add pm_runtime support soundwire: intel: add delay on restart for enumeration Rander Wang (1): soundwire: cadence_master: fix divider setting in clock register drivers/soundwire/Makefile | 4 +- drivers/soundwire/algo_dynamic_allocation.c | 403 ++++++++++++++++++++ drivers/soundwire/bus.c | 44 ++- drivers/soundwire/bus.h | 77 +++- drivers/soundwire/bus_type.c | 3 + drivers/soundwire/cadence_master.c | 365 ++++++++++++++---- drivers/soundwire/cadence_master.h | 12 +- drivers/soundwire/debugfs.c | 156 ++++++++ drivers/soundwire/intel.c | 381 +++++++++++++++++- drivers/soundwire/intel_init.c | 14 + drivers/soundwire/slave.c | 1 + drivers/soundwire/stream.c | 53 ++- include/linux/soundwire/sdw.h | 15 + 13 files changed, 1414 insertions(+), 114 deletions(-) create mode 100644 drivers/soundwire/algo_dynamic_allocation.c create mode 100644 drivers/soundwire/debugfs.c -- 2.20.1

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