July 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: fireworks: add support for AudioFire2 quirk
by Takashi Sakamoto 27 Jul '15

27 Jul '15

Fireworks uses TSB43CB43(IceLynx-Micro) as its IEC 61883-1/6 interface. This chip includes ARM7 core, and loads and runs program. The firmware is stored in on-board memory and loaded every powering-on. Echo Audio ships several versions of firmwares for each model. These firmwares have each quirk and the quirk changes a sequence of packets. AudioFire2 has a quirk to transfer a first packet with non-zero in its dbc field. This causes ALSA Fireworks driver to detect discontinuity. As long as I investigated, firmware 5.7, 5.7.6 and 5.8 have this quirk. This commit adds a support for the quirk to handle AudioFire2 packets. For safe, CIP_SKIP_INIT_DBC_CHECK is applied to all versions of AudioFire2's firmwares. 02 00050002 90ffffff <- 42 0005000a 90013000 42 00050012 90014400 42 0005001a 90015800 02 0005001a 90ffffff 42 00050022 90019000 42 0005002a 9001a400 42 00050032 9001b800 02 00050032 90ffffff 42 0005003a 9001d000 42 00050042 9001e400 42 0005004a 9001f800 02 0005004a 90ffffff Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireworks/fireworks.c | 2 ++ sound/firewire/fireworks/fireworks.h | 1 + sound/firewire/fireworks/fireworks_stream.c | 3 +++ 3 files changed, 6 insertions(+) diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 2682e7e..c670db4 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -248,6 +248,8 @@ efw_probe(struct fw_unit *unit, err = get_hardware_info(efw); if (err < 0) goto error; + if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2) + efw->is_af2 = true; if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9) efw->is_af9 = true; diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 4f0201a..c33252b 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -70,6 +70,7 @@ struct snd_efw { bool resp_addr_changable; /* for quirks */ + bool is_af2; bool is_af9; u32 firmware_version; diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index c55db1b..a0762dd 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -172,6 +172,9 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw) efw->tx_stream.flags |= CIP_DBC_IS_END_EVENT; /* Fireworks reset dbc at bus reset. */ efw->tx_stream.flags |= CIP_SKIP_DBC_ZERO_CHECK; + /* AudioFire2 starts packets with non-zero dbc. */ + if (efw->is_af2) + efw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK; /* AudioFire9 always reports wrong dbs. */ if (efw->is_af9) efw->tx_stream.flags |= CIP_WRONG_DBS; -- 2.1.4

2 1

[alsa-devel] [PATCH v3 0/4] i915 to call hda driver on HDMI plug/unplug
by David Henningsson 27 Jul '15

27 Jul '15

Changes since v2 is that the patch set has diminished to not transfer connector/eld information, since it might be better that such information to be transferred by a separate call in the ordinary direction. That could be added in a later patch set. The audio_ptr is now a void* (instead of a hdac_bus*) so that the audio side can easily refactor the ptr to point to the codec instead of the bus if we would find that beneficial. David Henningsson (4): drm/i915: Add audio hotplug info callback drm/i915: Call audio hotplug notify function ALSA: hda - Dispatch incoming HDMI hotplug i915 callback ALSA: hda - Wake the codec up on hotplug notify events drivers/gpu/drm/i915/i915_drv.h | 1 + drivers/gpu/drm/i915/intel_audio.c | 23 ++++++++++++++++++++--- include/drm/i915_component.h | 5 +++++ include/sound/hdaudio.h | 4 ++++ sound/hda/hdac_i915.c | 23 +++++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 13 ++++++++++++- 6 files changed, 65 insertions(+), 4 deletions(-) -- 1.7.9.5

4 7

Re: [alsa-devel] [ALC668]: Asus N751JK - surround 21 and surround40 pulseaudio bug
by Raymond Yau 26 Jul '15

26 Jul '15

> > Alsa-info patched kernel: http://www.alsa-project.org/db/?f=cd23a3f0bb72e38d73224620e3e8ab57a73aa441 > > Chmap results: > > Type = FIXED, Channels = 2 > FL FR > Type = FIXED, Channels = 4 > FL FR NA LFE > > When unplugging the subwoofer, then chmap gives this and stays like this. > > Cannot open PCM stream hw:1 for PLAYBACK > > [Pavucontrol] -> [Configuration] -> [Built-in Audio] gives > > Analog Stereo Duplex > Analog Stereo Output > Analog Surround 2.1 Output + Analog Stereo Input > Analog Surround 2.1 Output > Analog Surround 4.0 Output + Analog Stereo Input > Analog Surround 4.0 Output > Analog Stereo Input > Off Seem pulseaudio bug since your laptop does not have surround40 chmap It is not clear how pulseaudio's 21 channel-map match with alsa's 2.1 chmap http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… [Mapping analog-surround-21] device-strings = surround21:%f channel-map = front-left,front-right,lfe paths-output = analog-output analog-output-lineout analog-output-speaker priority = 8 direction = output [Mapping analog-surround-40] device-strings = surround40:%f channel-map = front-left,front-right,rear-left,rear-right paths-output = analog-output analog-output-lineout analog-output-speaker priority = 7 direction = output > > Pulseaudio verbose log while plugging in the subwoofer: http://pastebin.com/Ya5u1qz6 https://bugs.freedesktop.org/enter_bug.cgi?product=PulseAudio

1 0

[alsa-devel] [PATCH v3] ALSA: hda - Fix the headset mic that will not work on Dell desktop machine
by woodrow.shen＠canonical.com 25 Jul '15

25 Jul '15

From: Woodrow Shen <woodrow.shen(a)canonical.com> When the headset was plugged in the Dell desktop, the mic of headset can't be detected and workable. According to the alsa-info, we found the differece between alsa and init_pin_configs on the machine, so we need to add the pin configs to make headset work. Codec: Realtek ALC3234 Vendor Id: 0x10ec0255 Subsystem Id: 0x102806bb BugLink: https://bugs.launchpad.net/bugs/1477900 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 742fc62..d94c0e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5473,6 +5473,28 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1e, 0x411111f0}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170150}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x02011020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221105f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x40000000}, + {0x14, 0x90170110}, + {0x17, 0x411111f0}, + {0x18, 0x411111f0}, + {0x19, 0x411111f0}, + {0x1a, 0x411111f0}, + {0x1b, 0x01014020}, + {0x1d, 0x4054c029}, + {0x1e, 0x411111f0}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x17, 0x90170140}, -- 2.1.4

2 1

Re: [alsa-devel] ALSA: hda - Use generic parser for Cirrus codec driver
by Dan Carpenter 25 Jul '15

25 Jul '15

Hello Takashi Iwai, The patch 1077a024812d: "ALSA: hda - Use generic parser for Cirrus codec driver" from Dec 19, 2012, leads to the following static checker warning: sound/pci/hda/patch_cirrus.c:1004 cs4210_spdif_automute() warn: we tested 'spdif_present' before and it was 'true' sound/pci/hda/patch_cirrus.c 984 static void cs4210_spdif_automute(struct hda_codec *codec, 985 struct hda_jack_callback *tbl) 986 { 987 struct cs_spec *spec = codec->spec; 988 bool spdif_present = false; 989 hda_nid_t spdif_pin = spec->gen.autocfg.dig_out_pins[0]; 990 991 /* detect on spdif is specific to CS4210 */ 992 if (!spec->spdif_detect || 993 spec->vendor_nid != CS4210_VENDOR_NID) 994 return; 995 996 spdif_present = snd_hda_jack_detect(codec, spdif_pin); 997 if (spdif_present == spec->spdif_present) 998 return; 999 1000 spec->spdif_present = spdif_present; 1001 /* SPDIF TX on/off */ 1002 if (spdif_present) ^^^^^^^^^^^^^ 1003 snd_hda_set_pin_ctl(codec, spdif_pin, 1004 spdif_present ? PIN_OUT : 0); ^^^^^^^^^^^^^ I'm not positive but I think this is buggy. We should remove the if statement. 1005 1006 cs_automute(codec); 1007 } regards, dan carpenter

2 3

[alsa-devel] [PATCH] ASoC: rt5677: Return error if devm_gpiod_get_optional return ERR_PTR
by Axel Lin 25 Jul '15

25 Jul '15

If devm_gpiod_get_optional() return ERR_PTR, it means something wrong so request gpio fails. We had better return error in such case. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5677.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8d70911..3a074b6 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5133,14 +5133,14 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, if (IS_ERR(rt5677->pow_ldo2)) { ret = PTR_ERR(rt5677->pow_ldo2); dev_err(&i2c->dev, "Failed to request POW_LDO2: %d\n", ret); - rt5677->pow_ldo2 = 0; + return PTR_ERR(rt5677->pow_ldo2); } rt5677->reset_pin = devm_gpiod_get_optional(&i2c->dev, "realtek,reset", GPIOD_OUT_HIGH); if (IS_ERR(rt5677->reset_pin)) { ret = PTR_ERR(rt5677->reset_pin); dev_err(&i2c->dev, "Failed to request RESET: %d\n", ret); - rt5677->reset_pin = 0; + return PTR_ERR(rt5677->reset_pin); } if (rt5677->pow_ldo2 || rt5677->reset_pin) { -- 2.1.0

1 0

Re: [alsa-devel] writing an alsa driver
by Marco Bodega 24 Jul '15

24 Jul '15

Hi all, I own an Asus x205ta too and I really want to help to make this audio card works. I'm using linux next with Michele Curti's patch (as of 23 june) Here kernel .config: CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH=m CONFIG_SND_SOC_INTEL_BYTCR_RT5645_MACH=m # CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH is no CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH=m # CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH is not set # CONFIG_SND_SOC_INTEL_BROADWELL_MACH is not set CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH=m CONFIG_SND_SOC_INTEL_BYTCR_RT5645_MACH=m bo@X205TA:~$ uname -a Linux X205TA 4.2.0-rc2-next-20150717+ #3 SMP Wed Jul 22 15:31:59 CEST 2015 x86_64 x86_64 x86_64 GNU/Linux Output of lsmod: snd_soc_sst_bytcr_rt5645 16384 0 snd_intel_sst_acpi 16384 1 snd_intel_sst_core 73728 1 snd_intel_sst_acpi snd_soc_rt5645 139264 1 snd_soc_sst_mfld_platform 90112 3 snd_intel_sst_core snd_pcm 102400 4 snd_soc_rt5645,snd_soc_core,snd_soc_sst_mfld_platform,snd_soc_sst_bytcr_rt5645 bo@X205TA:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: baytrailcraudio [baytrailcraudio], device 0: Baytrail Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 bo@@X205TA:~$ aplay -D hw:0 -c 2 -r 48000 -f S16_LE /dev/urandomPlaying raw data '/dev/urandom' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: 125000 PERIOD_SIZE: 6000 PERIOD_BYTES: 24000 PERIODS: 4 BUFFER_TIME: 500000 BUFFER_SIZE: 24000 BUFFER_BYTES: 96000 TICK_TIME: 0 bo@X205TA:~$ cat /proc/interrupts | grep sst 6: 0 0 0 0 IO-APIC 28-fasteoi intel_sst_driver bo@X205TA:~$ ll /lib/firmware/intel/fw* -rw-r--r-- 1 root root 647078 lug 22 17:56 /lib/firmware/intel/fw_sst_0f28.bin -rw-r--r-- 1 root root 265684 lug 22 17:56 /lib/firmware/intel/fw_sst_0f28.bin-48kHz_i2s_master -rw-r--r-- 1 root root 701614 lug 22 17:56 /lib/firmware/intel/fw_sst_22a8.bin I read that the firmware used by bytx205_rt5645 is intel/fw_sst_0f28.bin: in fact in sst_acpi i have: {"10EC5648", "X205", "bytx205_rt5645", NULL, "intel/fw_sst_0f28.bin", &byt_rvp_platform_data }, but it's bounded to sst1 as i see from disassembling acpi data: cat dsdt.dsl: [...] Device (SSP1) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "SSPX0000" /* Intel SSP Device */) // _HID: Hardware ID Name (_CID, "SSPX0000" /* Intel SSP Device */) // _CID: Compatible ID [...] Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001D, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x00000018, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x00000019, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001A, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001B, } Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001C, } [...] however the fw is tied to work with SSP0. What can i do? I tried to connect parts in routing: #DSP routing amixer -c0 sset 'codec_out0 mix 0 pcm0_in' on amixer -c0 sset 'media0_out mix 0 media1_in' on amixer -c0 sset 'media1_in Gain 0' 80% amixer -c0 sset 'media1_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'media1_in Gain 0' off amixer -c0 sset 'pcm0_in Gain 0' 80% amixer -c0 sset 'pcm0_in Gain 0 Ramp Delay' 50 amixer -c0 sset 'pcm0_in Gain 0' off amixer -c0 sset 'codec_out0 Gain 0' 80% amixer -c0 sset 'codec_out0 Gain 0 Ramp Delay' 50 amixer -c0 sset 'codec_out0 Gain 0' off #Codec routin, if you have different one please modfiy amixer -c0 sset "DAC MIXL INF1" on amixer -c0 sset "DAC MIXR INF1" on amixer -c0 sset "Stereo DAC MIXL DAC L1" on amixer -c0 sset "Stereo DAC MIXR DAC R1" on amixer -c0 sset "SPK MIXL DAC L1" on amixer -c0 sset "SPK MIXR DAC R1" on amixer -c0 sset "SPOL MIX SPKVOL L" on amixer -c0 sset "SPOR MIX SPKVOL R" on amixer -c0 sset "SPOL MIX DAC L1" on amixer -c0 sset "SPOR MIX DAC R1" on amixer -c0 sset "Speaker L" on amixer -c0 sset "Speaker R" on amixer -c0 sset "Headphone" off amixer -c0 sset "Ext Spk" on and the attached dmesg log appeared I get no sound but at least no heat :) Is everything that I've done correct? I see in these output logs many of the errors and bugs that the other users found. Thanks for the time spent Marco Output of dmesg:

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.2
by Mark Brown 24 Jul '15

24 Jul '15

The following changes since commit 52721d9d3334c1cb1f76219a161084094ec634dc: Linux 4.2-rc3 (2015-07-19 14:45:02 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-fix-v4.2-rc3 for you to fetch changes up to 996034b117b467709dec7811ef134063934fa626: Merge remote-tracking branches 'asoc/fix/sgtl5000', 'asoc/fix/topology' and 'asoc/fix/zx' into asoc-linus (2015-07-24 16:18:22 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.2 A lot of small fixes here, a few to the core: - Fix for binding DAPM stream widgets on devices with prefixes assigned to them - Minor fixes for the newly added topology interfaces - Locking and memory leak fixes for DAPM - Driver specific fixes ---------------------------------------------------------------- Axel Lin (2): ASoC: sgtl5000: Fix up define for SGTL5000_SMALL_POP ASoC: pcm1681: Fix setting de-emphasis sampling rate selection Ben Zhang (1): ASoC: ssm4567: Keep TDM_BCLKS in ssm4567_set_dai_fmt Charles Keepax (1): ASoC: dapm: Fix deadlock on auto-disable mux controls Fang, Yang A (1): ASoC: Intel: fix incorrect widget name Jeeja KP (1): ASoC: topology: Fix to add dapm mixer info Juergen Borleis (1): ASoC: fsl-ssi: Fix bitclock calculation for master mode Koro Chen (2): ASoC: mediatek: Fix unbalanced calls to runtime suspend/resume ASoC: mediatek: Use platform_of_node for machine drivers Krzysztof Kozlowski (2): ASoC: zx: i2s: Fix devm_ioremap_resource return value check ASoC: zx: spdif: Fix devm_ioremap_resource return value check Lars-Peter Clausen (4): ASoC: dapm: Lock during userspace access ASoC: dapm: Fix kcontrol widget name memory management ASoC: Free card DAPM context on snd_soc_instantiate_card() error path ASoC: dapm: Don't add prefix to widget stream name Liam Girdwood (1): ASoC: topology: Fix TLV size calculation. Mark Brown (6): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm1681' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branch 'asoc/fix/ssm4567' into asoc-linus Merge remote-tracking branches 'asoc/fix/fsl-ssi', 'asoc/fix/intel', 'asoc/fix/intel-kconfig' and 'asoc/fix/mediatek' into asoc-linus Merge remote-tracking branches 'asoc/fix/sgtl5000', 'asoc/fix/topology' and 'asoc/fix/zx' into asoc-linus Nicolas Boichat (1): ASoC: rt5645: Check if codec is initialized in workqueue handler Shilpa Sreeramalu (1): ASoC: Intel: Get correct usage_count value to load firmware Vinod Koul (2): ASoC: topology: fix typos in topology header ASoC: Intel: use CONFIG_SND_SOC for intel boards .../devicetree/bindings/sound/mt8173-max98090.txt | 2 ++ .../bindings/sound/mt8173-rt5650-rt5676.txt | 2 ++ include/uapi/sound/asoc.h | 4 +-- sound/soc/codecs/pcm1681.c | 2 +- sound/soc/codecs/rt5645.c | 3 ++ sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/ssm4567.c | 8 ++++- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/sst/sst_drv_interface.c | 14 ++++----- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 +-- sound/soc/mediatek/mt8173-max98090.c | 17 ++++++++--- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 19 +++++++++--- sound/soc/mediatek/mtk-afe-pcm.c | 2 ++ sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 35 +++++++++++----------- sound/soc/soc-topology.c | 23 +++++++------- sound/soc/zte/zx296702-i2s.c | 4 +-- sound/soc/zte/zx296702-spdif.c | 4 +-- 19 files changed, 93 insertions(+), 57 deletions(-)

2 1

[alsa-devel] [PATCH RFT] ASoC: cs4265: Fix setting dai format for Left/Right Justified
by Axel Lin 24 Jul '15

24 Jul '15

The settings in current code does not match the datasheet, fix it. DAC Control - Address 03h DAC Digital Interface Format (Bits 5:4) DAC_DIF1 DAC_DIF0 Description 0 0 Left Justified, up to 24-bit data (default) 0 1 I²S, up to 24-bit data 1 0 Right-Justified, 16-bit Data 1 1 Right-Justified, 24-bit Data Transmitter Control 2 - Address 12h Transmitter Digital Interface Format (Bits 7:6) Tx_DIF1 Tx_DIF0 Description Format Figure 0 0 Left Justified, up to 24-bit data (default) 0 1 I²S, up to 24-bit data 1 0 Right-Justified, 16-bit Data 1 1 Right-Justified, 24-bit Data Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi, I don't have the h/w to test, so please test if this patch works. Thanks, Axel sound/soc/codecs/cs4265.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index d1a77c7..e42a5038 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -457,14 +457,14 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_RIGHT_J: if (params_width(params) == 16) { snd_soc_update_bits(codec, CS4265_DAC_CTL, - CS4265_DAC_CTL_DIF, (1 << 5)); + CS4265_DAC_CTL_DIF, (2 << 4)); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 7)); + CS4265_SPDIF_CTL2_DIF, (2 << 6)); } else { snd_soc_update_bits(codec, CS4265_DAC_CTL, - CS4265_DAC_CTL_DIF, (3 << 5)); + CS4265_DAC_CTL_DIF, (3 << 4)); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 7)); + CS4265_SPDIF_CTL2_DIF, (3 << 6)); } break; case SND_SOC_DAIFMT_LEFT_J: @@ -473,7 +473,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, CS4265_ADC_CTL, CS4265_ADC_DIF, 0); snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, - CS4265_SPDIF_CTL2_DIF, (1 << 6)); + CS4265_SPDIF_CTL2_DIF, 0); break; default: -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: cs4265: CS4265_INT_STATUS is readable register
by Axel Lin 24 Jul '15

24 Jul '15

CS4265_INT_STATUS was missed in cs4265_readable_register, fix it. The register 0x01 ~ 0x12 are readable, use CS4265_CHIP_ID ... CS4265_SPDIF_CTL2 syntax for better readability. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs4265.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index e42a5038..55db19d 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -60,23 +60,7 @@ static const struct reg_default cs4265_reg_defaults[] = { static bool cs4265_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS4265_PWRCTL: - case CS4265_DAC_CTL: - case CS4265_ADC_CTL: - case CS4265_MCLK_FREQ: - case CS4265_SIG_SEL: - case CS4265_CHB_PGA_CTL: - case CS4265_CHA_PGA_CTL: - case CS4265_ADC_CTL2: - case CS4265_DAC_CHA_VOL: - case CS4265_DAC_CHB_VOL: - case CS4265_DAC_CTL2: - case CS4265_SPDIF_CTL1: - case CS4265_SPDIF_CTL2: - case CS4265_INT_MASK: - case CS4265_STATUS_MODE_MSB: - case CS4265_STATUS_MODE_LSB: - case CS4265_CHIP_ID: + case CS4265_CHIP_ID ... CS4265_SPDIF_CTL2: return true; default: return false; -- 2.1.0

2 1