December 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [snd_hda_intel] snd_hda_intel causes high CPU lockup and system instability if mic disabled in BIOS on Lenovo P50
by Alexander Kappner 07 Jan '19

07 Jan '19

My Lenovo P50 laptop has a BIOS option to disable the microphone. When this option gets chosen, the snd_hda_intel driver causes high CPU load on a single kworker thread, spinning on "process_unsol_events", and system instability. This behavior occurs from the time that the snd_hda_intel module is loaded, irrespective of whether anything is attempting to access the mic. The sound output still works. When in this state, the module cannot be removed cleanly; attempting to remove it (even without rmmod -f) triggers an oops. I have attached two exemplary dmesg outputs. Strangely enough, the exact location of the oops varies, but further up the call chain, I always see process_unsol_events. When the mic is not disabled in the BIOS, the module works stable, regardless whether or not the mic is muted in ALSA. I wasn't able to pinpoint the root cause. Any pointers on where to start? Much appreciated. Best regards Alexander First dmesg: [31592.691573] BUG: unable to handle kernel NULL pointer dereference at 000000000000020c [31592.691600] IP: mux_select+0x1d/0x190 [snd_hda_codec_generic] [31592.691601] PGD 0 P4D 0 [31592.691604] Oops: 0000 [#1] PREEMPT SMP [31592.691606] Modules linked in: snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic snd_hda_intel(-) snd_hda_codec snd_hwdep snd_hda_core snd_pcm snd_timer zfs(PO) zunicode(PO) zlua(PO) nvidia_drm(PO) nvidia_modeset(PO) nvidia(PO) iwlmvm mac80211 qmi_wwan iwlwifi cfg80211 thinkpad_acpi rfkill zcommon(PO) znvpair(PO) zavl(PO) icp(PO) spl(O) sdhci_pci cqhci sdhci mmc_block rtsx_pci_sdmmc rtsx_pci mfd_core [31592.691639] CPU: 3 PID: 11044 Comm: kworker/3:1 Tainted: P O 4.16.0-rc7-00206-gf461da3-dirty #33 [31592.691640] Hardware name: LENOVO 20ENCTO1WW/20ENCTO1WW, BIOS N1EET62W (1.35 ) 11/10/2016 [31592.691643] Workqueue: events process_unsol_events [snd_hda_core] [31592.691647] RIP: 0010:mux_select+0x1d/0x190 [snd_hda_codec_generic] [31592.691648] RSP: 0018:ffffc9000f0dfe08 EFLAGS: 00010292 [31592.691650] RAX: 0000000000000001 RBX: ffff880e44737300 RCX: 00000000ffffffff [31592.691651] RDX: 0000000000000000 RSI: 0000000000000000 RDI: ffff880ee6368800 [31592.691652] RBP: 0000000000000000 R08: 0000000000000f09 R09: 0000000000000000 [31592.691653] R10: 0000000000000018 R11: 0000000000000003 R12: ffff880ee6368800 [31592.691655] R13: 0000000000000000 R14: ffff881053ce6100 R15: 0ffff881053ce610 [31592.691656] FS: 0000000000000000(0000) GS:ffff881053cc0000(0000) knlGS:0000000000000000 [31592.691658] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [31592.691659] CR2: 000000000000020c CR3: 000000000220a004 CR4: 00000000003606e0 [31592.691661] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [31592.691662] DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 [31592.691662] Call Trace: [31592.691668] ? snd_hda_jack_detect_state+0x4b/0x60 [snd_hda_codec] [31592.691672] call_jack_callback.isra.1+0x20/0xa0 [snd_hda_codec] [31592.691676] snd_hda_jack_unsol_event+0x62/0x70 [snd_hda_codec] [31592.691679] process_unsol_events+0x64/0x80 [snd_hda_core] [31592.691682] process_one_work+0x13e/0x380 [31592.691684] worker_thread+0x62/0x3e0 [31592.691686] ? rescuer_thread+0x2d0/0x2d0 [31592.691689] kthread+0x10d/0x140 [31592.691690] ? kthread_bind+0x10/0x10 [31592.691693] ret_from_fork+0x35/0x40 [31592.691695] Code: e2 01 e9 77 fb ff ff c3 66 0f 1f 44 00 00 0f 1f 44 00 00 41 57 41 56 41 55 41 54 49 89 fc 55 53 48 83 ec 08 48 8b af 08 04 00 00 <8b> 85 0c 02 00 00 85 c0 74 73 89 d3 41 89 f5 8d 50 ff 39 d8 89 [31592.691721] RIP: mux_select+0x1d/0x190 [snd_hda_codec_generic] RSP: ffffc9000f0dfe08 [31592.691723] CR2: 000000000000020c [31592.691725] ---[ end trace dee155e55b20eab8 ]--- Second dmesg: [115121.084841] BUG: unable to handle kernel NULL pointer dereference at 00000000000005f8 [115121.084844] PGD 0 P4D 0 [115121.084847] Oops: 0000 [#1] PREEMPT SMP [115121.084850] CPU: 2 PID: 12228 Comm: kworker/2:1 Tainted: P O 4.18.0-rc2-00223-g1904148-dirty #39 [115121.084851] Hardware name: LENOVO 20ENCTO1WW/20ENCTO1WW, BIOS N1EET62W (1.35 ) 11/10/2016 [115121.084856] Workqueue: events process_unsol_events [snd_hda_core] [115121.084859] RIP: 0010:set_path_power+0x24/0x120 [snd_hda_codec_generic] [115121.084860] Code: 66 0f 1f 44 00 00 0f 1f 44 00 00 41 57 41 89 f7 41 56 41 55 41 54 55 48 89 fd 53 48 83 ec 08 4c 8b b7 08 04 00 00 89 54 24 04 <41> 8b be f8 05 00 00 49 8b b6 08 06 00 00 85 ff 0f 84 ce 00 00 00 [115121.084901] RSP: 0018:ffffc900243e3de8 EFLAGS: 00010296 [115121.084903] RAX: 0000000000000000 RBX: 0000000000000018 RCX: 00000000ffffffff [115121.084904] RDX: 0000000000000000 RSI: 0000000000000018 RDI: ffff880f993a2800 [115121.084906] RBP: ffff880f993a2800 R08: 0000000000000f09 R09: 0000000000000000 [115121.084907] R10: 0000000000000019 R11: 0000000000000000 R12: 0000000000000000 [115121.084908] R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000018 [115121.084910] FS: 0000000000000000(0000) GS:ffff881053c80000(0000) knlGS:0000000000000000 [115121.084911] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [115121.084912] CR2: 00000000000005f8 CR3: 000000000260a004 CR4: 00000000003606e0 [115121.084914] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [115121.084915] DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 [115121.084916] Call Trace: [115121.084922] pin_power_callback+0x4f/0x80 [snd_hda_codec_generic] [115121.084926] call_jack_callback.isra.1+0x20/0xa0 [snd_hda_codec] [115121.084930] snd_hda_jack_unsol_event+0x62/0x70 [snd_hda_codec] [115121.084933] process_unsol_events+0x64/0x80 [snd_hda_core] [115121.084937] process_one_work+0x157/0x390 [115121.084939] worker_thread+0x64/0x3e0 [115121.084942] ? rescuer_thread+0x2e0/0x2e0 [115121.084944] kthread+0x10d/0x140 [115121.084946] ? kthread_bind+0x10/0x10 [115121.084948] ret_from_fork+0x35/0x40 [115121.084951] Modules linked in: snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic snd_hda_intel(-) snd_hda_codec snd_hwdep snd_hda_core snd_pcm snd_timer tun nvidia_modeset(PO) nvidia(PO) iwlmvm mac80211 iwlwifi btusb btrtl btbcm btintel bluetooth cfg80211 qmi_wwan ecdh_generic thinkpad_acpi rfkill sdhci_pci cqhci sdhci mmc_block rtsx_pci_sdmmc rtsx_pci mfd_core [115121.084969] CR2: 00000000000005f8 [115121.084971] ---[ end trace 010d684599325728 ]--- [115121.084975] RIP: 0010:set_path_power+0x24/0x120 [snd_hda_codec_generic] [115121.084976] Code: 66 0f 1f 44 00 00 0f 1f 44 00 00 41 57 41 89 f7 41 56 41 55 41 54 55 48 89 fd 53 48 83 ec 08 4c 8b b7 08 04 00 00 89 54 24 04 <41> 8b be f8 05 00 00 49 8b b6 08 06 00 00 85 ff 0f 84 ce 00 00 00 [115121.085000] RSP: 0018:ffffc900243e3de8 EFLAGS: 00010296 [115121.085001] RAX: 0000000000000000 RBX: 0000000000000018 RCX: 00000000ffffffff [115121.085002] RDX: 0000000000000000 RSI: 0000000000000018 RDI: ffff880f993a2800 [115121.085003] RBP: ffff880f993a2800 R08: 0000000000000f09 R09: 0000000000000000 [115121.085004] R10: 0000000000000019 R11: 0000000000000000 R12: 0000000000000000 [115121.085005] R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000018 [115121.085006] FS: 0000000000000000(0000) GS:ffff881053c80000(0000) knlGS:0000000000000000 [115121.085007] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [115121.085008] CR2: 00000000000005f8 CR3: 000000000260a004 CR4: 00000000003606e0 [115121.085009] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [115121.085010] DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400

2 6

[alsa-devel] Need help fixing pop/click artifacts in an ASOC driver
by Dimitris Papavasiliou 05 Jan '19

05 Jan '19

Hi, A (commercial) DAC board, I use with a Raspberry Pi is plagued by clicks and pops and I'm trying to fix it at the driver level. I have mostly fixed it, but I could use some guidance on the problem described below. I apologize in advance for the long read, but the situation is a bit convoluted (or my understanding of it is). The board (a HifiBerry DAC+ Pro) uses a PCM5122 chip which is driven by the pcm512x codec driver, but the Pro version also includes a set of external clocks on the board (one each for 48Khz and 44.1Khz sample rates and multiples thereof). There are two kinds of clicks and pops: on the one hand the board clicks when the device is opened/closed, mostly because the pcm512x codec driver doesn't support the digital_mute operation. These are relatively mild clicks, which are an annoyance at most and I have a separate patch for the pcm512x, that addresses them, which I'm currently testing and will (hopefully) submit soon. A separate case is a pop that's generated when the clock source is changed. This pop is independent of the digital volume setting and very loud. As far as I could determine, when the hw_params callback specified in the snd_soc_dai_link struct is called, it figures out what clock is needed for the giver sample rate and powers the appropriate clock through a GPIO on the PCM5122. This essentially means that the external clock source fed into the 5122 is stopped and restarted at a different rate, potentially while the DAC chip is operating. Later the hw_params callback defined in the pcm512x driver is called and it reconfigure the clock dividers for the new rate. It is at this point that the pop seems to be generated, provided that the card is not powered down or muted. If I comment out either the clock power-up/down in the dai_link hw_params, or the clock divider reconfiguration in the codec's hw_params, no pop is generated. Now, the digital_mute patch described before, also happens to fix this loud pop. I'm not sure whether that's a happy coincidence that might cease to be the case in the future though and was considering fixing the pop problem separately, at its source: the clock change. One way to fix the problem is to switch the 5122 into standby before changing the clock source in the dai_link hw_params callback and switch it back into normal operation afterwards. This seems to work regardless of whether digital_mute is implemented or not, but has the following problem: The register that's used to put the chip into/out of standby, is also used by the pcm512x driver, in the set_bias_level operation. Since the hw_params operation is, as far as I can tell, not atomic, I'm concerned that race conditions may arise, causing the card to be left powered up, when it should be off and vice-versa. Is there some way to synchronize the use of common chip registers in the DAI and codec drivers? Is the digital_mute somehow guaranteed to fix the pop originating at the DAI/machine driver level (whatever the hw_params operation in the dai_link struct is supposed to be)? Am I perhaps missing something entirely? Any help or guidance is welcome. Thanks, Dimitris

5 41

[alsa-devel] [PATCH 1/1] conf/ucm: chtrt5650: Add UCM config for chtrt5650
by Russell Parker 04 Jan '19

04 Jan '19

Add a UCM configuration for the rt5650 codec. Tested on a Samsung Chromebook 3. Adapted with minor modifications from GitHub user evan-a-a's gist: https://gist.github.com/evan-a-a/86b2a698708074530e2d0ee7c6498767 Signed-off-by: Russell Parker <russell.parker7(a)gmail.com> diff --git a/configure.ac b/configure.ac index d0cab2d6..d524ff4b 100644 --- a/configure.ac +++ b/configure.ac @@ -737,6 +737,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/chtnau8824/Makefile \ src/conf/ucm/chtrt5645/Makefile \ src/conf/ucm/chtrt5645-mono-speaker-analog-mic/Makefile \ + src/conf/ucm/chtrt5650/Makefile \ src/conf/ucm/cube-i1_TF-Defaultstring-CherryTrailCR/Makefile \ src/conf/ucm/DAISY-I2S/Makefile \ src/conf/ucm/DB410c/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 85c15090..985c460c 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -21,6 +21,7 @@ bytcr-rt5651-stereo-spk-in12-mic \ chtnau8824 \ chtrt5645 \ chtrt5645-mono-speaker-analog-mic \ +chtrt5650 \ cube-i1_TF-Defaultstring-CherryTrailCR \ DAISY-I2S \ DB410c \ diff --git a/src/conf/ucm/chtrt5650/HiFi.conf b/src/conf/ucm/chtrt5650/HiFi.conf new file mode 100644 index 00000000..cc3d64bf --- /dev/null +++ b/src/conf/ucm/chtrt5650/HiFi.conf @@ -0,0 +1,203 @@ + +SectionVerb { + # ALSA PCM + Value { + TQ "HiFi" + + # ALSA PCM device for HiFi + PlaybackPCM "hw:chtrt5650" + CapturePCM "hw:chtrt5650" + } + + EnableSequence [ + cdev "hw:chtrt5650" + + <platforms/bytcr/PlatformEnableSeq.conf> + + # Output Configuration + cset "name='DAC1 L Mux' IF1 DAC" + cset "name='DAC1 R Mux' IF1 DAC" + cset "name='DAC1 MIXL DAC1 Switch' 1" + cset "name='DAC1 MIXR DAC1 Switch' 1" + cset "name='Stereo DAC MIXL DAC L1 Switch' 1" + cset "name='Stereo DAC MIXR DAC R1 Switch' 1" + cset "name='Speaker HWEQ' 1,164,237,135,1,165,0,0,1,166,237,135,1,167,0,0,1,192,30,196,1,193,0,0,1,194,30,196,1,195,0,0,1,196,31,180,1,197,0,75,1,198,31,180,1,199,31,180,1,200,0,75,1,201,31,180,0,177,51,224" + cset "name='DAC1 Playback Volume' 77,77" + + cset "name='Speaker ClassD Playback Volume' 4" + + cset "name='DAC L2 Mux' IF2 DAC" + cset "name='DAC R2 Mux' IF2 DAC" + cset "name='Mono DAC MIXL DAC L2 Switch' on" + cset "name='Mono DAC MIXR DAC R2 Switch' on" + cset "name='DAC2 Playback Switch' on" + + cset "name='HPOVOL MIXL DAC1 Switch' on" + cset "name='HPOVOL MIXR DAC1 Switch' on" + cset "name='HPOVOL MIXL DAC2 Switch' on" + cset "name='HPOVOL MIXR DAC2 Switch' on" + cset "name='HPO MIX HPVOL Switch' on" + cset "name='HPOVOL L Switch' on" + cset "name='HPOVOL R Switch' on" + + cset "name='SPK MIXL DAC L1 Switch' on" + cset "name='SPK MIXR DAC R1 Switch' on" + cset "name='SPK MIXL DAC L2 Switch' on" + cset "name='SPK MIXR DAC R2 Switch' on" + cset "name='SPOL MIX SPKVOL L Switch' on" + cset "name='SPOR MIX SPKVOL R Switch' on" + cset "name='SPKVOL L Switch' on" + cset "name='SPKVOL R Switch' on" + + # Input Configuration + cset "name='Stereo1 DMIC Mux' 0" + cset "name='Stereo1 ADC2 Mux' 1" + cset "name='ADC Capture Switch' on" + cset "name='ADC Capture Volume' 74" + cset "name='ADC Boost Capture Volume' 3" + cset "name='Mono ADC Capture Volume' 74" + cset "name='Mono ADC Boost Capture Volume' 2" + cset "name='IN Capture Volume' 63" + cset "name='RT5650 IF1 ADC Mux' 0" + cset "name='I2S2 Func Switch' off" + ] + + DisableSequence [ + cdev "hw:chtrt5650" + + # Disable audio output path + cset "name='codec_out1 mix 0 pcm0_in Switch' off" + cset "name='media0_out mix 0 media1_in Switch' off" + + cset "name='media1_in Gain 0 Switch' off" + cset "name='pcm0_in Gain 0 Switch' off" + cset "name='codec_out1 Gain 0 Switch' off" + + # Disable audio input path + cset "name='pcm1_out mix 0 media_loop2_in Switch' off" + cset "name='media_loop2_out mix 0 codec_in0 Switch' off" + + cset "name='media_loop2_out Gain 0 Switch' off" + cset "name='pcm1_out Gain 0 Switch' off" + cset "name='codec_in0 Gain 0 Switch' off" + ] +} + +SectionDevice."Speaker" { + Comment "Speaker" + + Value { + PlaybackChannels "2" + } + + ConflictingDevice [ + "Headphones" + ] + + EnableSequence [ + cdev "hw:chtrt5650" + + cset "name='Headphone Switch' off" + cset "name='Headphone Channel Switch' off" + + cset "name='Ext Spk Switch' on" + cset "name='Speaker Channel Switch' on" + cset "name='Speaker Playback Volume' 31" + ] + + DisableSequence [ + cdev "hw:chtrt5650" + + cset "name='Ext Spk Switch' off" + cset "name='Speaker Channel Switch' off" + ] +} + +SectionDevice."Headphones" { + Comment "Headphones" + + Value { + PlaybackChannels "2" + JackControl "Headphone Jack" + JackHWMute "Speaker" + } + + ConflictingDevice [ + "Speaker" + ] + + EnableSequence [ + cdev "hw:chtrt5650" + + cset "name='Ext Spk Switch' off" + cset "name='Speaker Channel Switch' off" + + cset "name='Headphone Switch' on" + cset "name='Headphone Channel Switch' on" + cset "name='Headphone Playback Volume' 31" + ] + + DisableSequence [ + cdev "hw:chtrt5650" + + cset "name='Headphone Switch' off" + cset "name='Headphone Channel Switch' off" + ] +} + +SectionDevice."Mic".0 { + Comment "Internal Microphone" + + Value { + CaptureChannels "2" + CapturePriority "150" + } + + EnableSequence [ + cdev "hw:chtrt5650" + + cset "name='Int Mic Switch' on" + cset "name='Sto1 ADC MIXL ADC2 Switch' on" + cset "name='Sto1 ADC MIXR ADC2 Switch' on" + ] + + DisableSequence [ + cdev "hw:chtrt5650" + + cset "name='Sto1 ADC MIXL ADC2 Switch' off" + cset "name='Sto1 ADC MIXR ADC2 Switch' off" + cset "name='Int Mic Switch' off" + ] +} + +SectionDevice."HSMic".0 { + Comment "Headset Microphone" + + Value { + CaptureChannels "2" + JackControl "Headset Mic Jack" + JackHWMute "Mic" + } + + EnableSequence [ + cdev "hw:chtrt5650" + + cset "name='Headset Mic Switch' on" + cset "name='RECMIXL BST1 Switch' on" + cset "name='RECMIXR BST1 Switch' on" + cset "name='Sto1 ADC MIXL ADC1 Switch' on" + cset "name='Sto1 ADC MIXR ADC1 Switch' on" + cset "name='IN1 Boost' 1" + ] + + DisableSequence [ + cdev "hw:chtrt5650" + + cset "name='Headset Mic Switch' off" + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' off" + cset "name='Sto1 ADC MIXL ADC1 Switch' off" + cset "name='Sto1 ADC MIXR ADC1 Switch' off" + cset "name='IN1 Boost' 0" + ] +} diff --git a/src/conf/ucm/chtrt5650/Makefile.am b/src/conf/ucm/chtrt5650/Makefile.am new file mode 100644 index 00000000..f506cd77 --- /dev/null +++ b/src/conf/ucm/chtrt5650/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/chtrt5650 +ucm_DATA = chtrt5650.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/chtrt5650/chtrt5650.conf b/src/conf/ucm/chtrt5650/chtrt5650.conf new file mode 100644 index 00000000..dfc36c89 --- /dev/null +++ b/src/conf/ucm/chtrt5650/chtrt5650.conf @@ -0,0 +1,5 @@ +Comment "Intel SoC Audio Device" +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" +} -- 2.17.2 (Apple Git-113)

3 4

[alsa-devel] [RFC PATCH 00/10] Add support for DMA and audio codec of F1C100s
by Mesih Kilinc 04 Jan '19

04 Jan '19

This is RFC patchset for Allwinner suniv F1C100s to support DMA and audio codec. Allwinner F1C100s has a audio codec that has necessary digital and analog parts. It has r-l headphone output and microphone, line, r-l FM inputs. ADC can capture any inputs and also output channels via mux. Any input channels or DAC samples can feed output channels. Add support for this audio codec. F1C100s utilizes DMA channels to send and receive ADC-DAC samples. So DMA support needed. Patch 1~5 adds support for DMA. Suniv F1C100s has very similar DMA to sun4i. But there is some dissimilarities also. Suniv features a DMA reset bit in clock control unit. It has smaller number of DMA channels. Several registers has different addresses. It's max burst size is 4 instead of 8. Also DMA endpoint numbers are different. Patch 6 adds DMA max burst option to sun4i-codec. Patch 7~8 Add support for suniv F1C100s audio codec. Patch 9 adds audio codec to suniv-f1c100s.dtsi Patch 10 adds audio codec support to Lichee Pi Nano board. Thanks! Mesih Kilinc (10): dma-engine: sun4i: Add a quirk to support different chips dma-engine: sun4i: Add has_reset option to quirk dt-bindings: dmaengine: Add Allwinner suniv F1C100s DMA dma-engine: sun4i: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for DMA ASoC: sun4i-codec: Add DMA Max Burst field dt-bindigs: sound: Add Allwinner suniv F1C100s Audio Codec ASoC: sun4i-codec: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for Audio Codec ARM: dts: suniv: f1c100s: Activate Audio Codec for Lichee Pi Nano .../devicetree/bindings/dma/sun4i-dma.txt | 4 +- .../devicetree/bindings/sound/sun4i-codec.txt | 5 + arch/arm/boot/dts/suniv-f1c100s-licheepi-nano.dts | 8 + arch/arm/boot/dts/suniv-f1c100s.dtsi | 25 ++ drivers/dma/Kconfig | 4 +- drivers/dma/sun4i-dma.c | 221 ++++++++++-- sound/soc/sunxi/sun4i-codec.c | 371 ++++++++++++++++++++- 7 files changed, 601 insertions(+), 37 deletions(-) -- 2.7.4

4 16

[alsa-devel] [RFC PATCH] ASoC: fsl: Add Audio Mixer CPU DAI driver
by Viorel Suman 04 Jan '19

04 Jan '19

This patch implements Audio Mixer CPU DAI driver for NXP iMX8 SOCs. The Audio Mixer is a on-chip functional module that allows mixing of two audio streams into a single audio stream. Audio Mixer datasheet is available here: https://www.nxp.com/docs/en/reference-manual/IMX8DQXPRM.pdf Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- .../devicetree/bindings/sound/fsl,amix.txt | 45 ++ sound/soc/fsl/Kconfig | 7 + sound/soc/fsl/Makefile | 3 + sound/soc/fsl/fsl_amix.c | 554 +++++++++++++++++++++ sound/soc/fsl/fsl_amix.h | 101 ++++ 5 files changed, 710 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,amix.txt create mode 100644 sound/soc/fsl/fsl_amix.c create mode 100644 sound/soc/fsl/fsl_amix.h diff --git a/Documentation/devicetree/bindings/sound/fsl,amix.txt b/Documentation/devicetree/bindings/sound/fsl,amix.txt new file mode 100644 index 0000000..049144f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,amix.txt @@ -0,0 +1,45 @@ +NXP Audio Mixer (AMIX). + +The Audio Mixer is a on-chip functional module that allows mixing of two +audio streams into a single audio stream. Audio Mixer has two input serial +audio interfaces. These are driven by two Synchronous Audio interface +modules (SAI). Each input serial interface carries 8 audio channels in its +frame in TDM manner. Mixer mixes audio samples of corresponding channels +from two interfaces into a single sample. Before mixing, audio samples of +two inputs can be attenuated based on configuration. The output of the +Audio Mixer is also a serial audio interface. Like input interfaces it has +the same TDM frame format. This output is used to drive the serial DAC TDM +interface of audio codec and also sent to the external pins along with the +receive path of normal audio SAI module for readback by the CPU. + +The output of Audio mixer can be selected from any of the three streams + - serial audio input 1 + - serial audio input 2 + - Mixed audio + +Mixing operation is independent of audio sample rate but the two audio +input streams must have same audio sample rate with same number of channels +in TDM frame to be eligible for mixing. + +Device driver required properties: +================================= + - compatible : Compatible list, contains "fsl,imx8qm-amix" + + - reg : Offset and length of the register set for the device. + + - clocks : Must contain an entry for each entry in clock-names. + + - clock-names : Must include the "ipg" for register access. + + - power-domains : Must contain the phandle to the AMIX power domain node + +Device driver configuration example: +====================================== + amix: amix@59840000 { + compatible = "fsl,imx8qm-amix"; + reg = <0x0 0x59840000 0x0 0x10000>; + clocks = <&clk IMX8QXP_AUD_AMIX_IPG>; + clock-names = "ipg"; + power-domains = <&pd_amix>; + }; + diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 2e75b5bc..0b08eb2 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -24,6 +24,13 @@ config SND_SOC_FSL_SAI This option is only useful for out-of-tree drivers since in-tree drivers select it automatically. +config SND_SOC_FSL_AMIX + tristate "Audio Mixer (AMIX) module support" + select REGMAP_MMIO + help + Say Y if you want to add Audio Mixer (AMIX) + support for the NXP iMX CPUs. + config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module (SSI) support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index de94fa0..3263634 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -12,6 +12,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support +snd-soc-fsl-amix-objs := fsl_amix.o snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o snd-soc-fsl-sai-objs := fsl_sai.o @@ -21,6 +22,8 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-esai-objs := fsl_esai.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o + +obj-$(CONFIG_SND_SOC_FSL_AMIX) += snd-soc-fsl-amix.o obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o diff --git a/sound/soc/fsl/fsl_amix.c b/sound/soc/fsl/fsl_amix.c new file mode 100644 index 0000000..d752029 --- /dev/null +++ b/sound/soc/fsl/fsl_amix.c @@ -0,0 +1,554 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * NXP AMIX ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright 2017 NXP + */ + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/pm_runtime.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "fsl_amix.h" + +#define SOC_ENUM_SINGLE_S(xreg, xshift, xtexts) \ + SOC_ENUM_SINGLE(xreg, xshift, ARRAY_SIZE(xtexts), xtexts) + +static const char + *tdm_sel[] = { "TDM1", "TDM2", }, + *mode_sel[] = { "Disabled", "TDM1", "TDM2", "Mixed", }, + *width_sel[] = { "16b", "18b", "20b", "24b", "32b", }, + *pol_sel[] = { "Positive edge", "Negative edge", }, + *endis_sel[] = { "Disabled", "Enabled", }, + *updn_sel[] = { "Downward", "Upward", }, + *mask_sel[] = { "Unmask", "Mask", }; + +static const struct soc_enum fsl_amix_enum[] = { +/* FSL_AMIX_CTR enums */ +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_MIXCLK_SHIFT, tdm_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_OUTSRC_SHIFT, mode_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_OUTWIDTH_SHIFT, width_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_OUTCKPOL_SHIFT, pol_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_MASKRTDF_SHIFT, mask_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_MASKCKDF_SHIFT, mask_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_SYNCMODE_SHIFT, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_CTR, FSL_AMIX_CTR_SYNCSRC_SHIFT, tdm_sel), +/* FSL_AMIX_ATCR0 enums */ +SOC_ENUM_SINGLE_S(FSL_AMIX_ATCR0, 0, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_ATCR0, 1, updn_sel), +/* FSL_AMIX_ATCR1 enums */ +SOC_ENUM_SINGLE_S(FSL_AMIX_ATCR1, 0, endis_sel), +SOC_ENUM_SINGLE_S(FSL_AMIX_ATCR1, 1, updn_sel), +}; + +struct fsl_amix_state { + u8 tdms; + u8 clk; + char msg[64]; +}; + +static const struct fsl_amix_state prms[4][4] = {{ + /* DIS->DIS, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* DIS->TDM1*/ + { .tdms = 1, .clk = 1, .msg = "DIS->TDM1: TDM1 not started!\n" }, + /* DIS->TDM2*/ + { .tdms = 2, .clk = 2, .msg = "DIS->TDM2: TDM2 not started!\n" }, + /* DIS->MIX */ + { .tdms = 3, .clk = 0, .msg = "DIS->MIX: Please start both TDMs!\n" } +}, { /* TDM1->DIS */ + { .tdms = 1, .clk = 0, .msg = "TDM1->DIS: TDM1 not started!\n" }, + /* TDM1->TDM1, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* TDM1->TDM2 */ + { .tdms = 3, .clk = 2, .msg = "TDM1->TDM2: Please start both TDMs!\n" }, + /* TDM1->MIX */ + { .tdms = 3, .clk = 0, .msg = "TDM1->MIX: Please start both TDMs!\n" } +}, { /* TDM2->DIS */ + { .tdms = 2, .clk = 0, .msg = "TDM2->DIS: TDM2 not started!\n" }, + /* TDM2->TDM1 */ + { .tdms = 3, .clk = 1, .msg = "TDM2->TDM1: Please start both TDMs!\n" }, + /* TDM2->TDM2, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" }, + /* TDM2->MIX */ + { .tdms = 3, .clk = 0, .msg = "TDM2->MIX: Please start both TDMs!\n" } +}, { /* MIX->DIS */ + { .tdms = 3, .clk = 0, .msg = "MIX->DIS: Please start both TDMs!\n" }, + /* MIX->TDM1 */ + { .tdms = 3, .clk = 1, .msg = "MIX->TDM1: Please start both TDMs!\n" }, + /* MIX->TDM2 */ + { .tdms = 3, .clk = 2, .msg = "MIX->TDM2: Please start both TDMs!\n" }, + /* MIX->MIX, do nothing */ + { .tdms = 0, .clk = 0, .msg = "" } +}, }; + +static int fsl_amix_state_trans(struct snd_soc_component *comp, + unsigned int *mask, unsigned int *ctr, + const struct fsl_amix_state prm) +{ + struct fsl_amix *priv = snd_soc_component_get_drvdata(comp); + /* Enforce all required TDMs are started */ + if ((priv->tdms & prm.tdms) != prm.tdms) { + dev_dbg(comp->dev, prm.msg); + return -EINVAL; + } + + switch (prm.clk) { + case 1: + case 2: + /* Set mix clock */ + (*mask) |= FSL_AMIX_CTR_MIXCLK_MASK; + (*ctr) |= FSL_AMIX_CTR_MIXCLK(prm.clk - 1); + break; + default: + break; + } + + return 0; +} + +static int fsl_amix_put_mix_clk_src(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct fsl_amix *priv = snd_soc_component_get_drvdata(comp); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int reg_val, val, mix_clk; + int ret = 0; + + /* Get current state */ + ret = snd_soc_component_read(comp, FSL_AMIX_CTR, &reg_val); + if (ret) + return ret; + + mix_clk = ((reg_val & FSL_AMIX_CTR_MIXCLK_MASK) + >> FSL_AMIX_CTR_MIXCLK_SHIFT); + val = snd_soc_enum_item_to_val(e, item[0]); + + dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val); + + /** + * Ensure the current selected mixer clock is available + * for configuration propagation + */ + if (!(priv->tdms & BIT(mix_clk))) { + dev_err(comp->dev, + "Started TDM%d needed for config propagation!\n", + mix_clk + 1); + return -EINVAL; + } + + if (!(priv->tdms & BIT(val))) { + dev_err(comp->dev, + "The selected clock source has no TDM%d enabled!\n", + val + 1); + return -EINVAL; + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + +static int fsl_amix_put_out_src(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + struct fsl_amix *priv = snd_soc_component_get_drvdata(comp); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + u32 out_src, mix_clk; + unsigned int reg_val, val, mask = 0, ctr = 0; + int ret = 0; + + /* Get current state */ + ret = snd_soc_component_read(comp, FSL_AMIX_CTR, &reg_val); + if (ret) + return ret; + + /* "From" state */ + out_src = ((reg_val & FSL_AMIX_CTR_OUTSRC_MASK) + >> FSL_AMIX_CTR_OUTSRC_SHIFT); + mix_clk = ((reg_val & FSL_AMIX_CTR_MIXCLK_MASK) + >> FSL_AMIX_CTR_MIXCLK_SHIFT); + + /* "To" state */ + val = snd_soc_enum_item_to_val(e, item[0]); + + dev_dbg(comp->dev, "TDMs=x%08x, val=x%08x\n", priv->tdms, val); + + /* Check if state is changing ... */ + if (out_src == val) + return 0; + /** + * Ensure the current selected mixer clock is available + * for configuration propagation + */ + if (!(priv->tdms & BIT(mix_clk))) { + dev_err(comp->dev, + "Started TDM%d needed for config propagation!\n", + mix_clk + 1); + return -EINVAL; + } + + /* Check state transition constraints */ + ret = fsl_amix_state_trans(comp, &mask, &ctr, prms[out_src][val]); + if (ret) + return ret; + + /* Complete transition to new state */ + mask |= FSL_AMIX_CTR_OUTSRC_MASK; + ctr |= FSL_AMIX_CTR_OUTSRC(val); + + return snd_soc_component_update_bits(comp, FSL_AMIX_CTR, mask, ctr); +} + +static const struct snd_kcontrol_new fsl_amix_snd_controls[] = { + /* FSL_AMIX_CTR controls */ + SOC_ENUM_EXT("Mixing Clock Source", fsl_amix_enum[0], + snd_soc_get_enum_double, fsl_amix_put_mix_clk_src), + SOC_ENUM_EXT("Output Source", fsl_amix_enum[1], + snd_soc_get_enum_double, fsl_amix_put_out_src), + SOC_ENUM("Output Width", fsl_amix_enum[2]), + SOC_ENUM("Output Clock Polarity", fsl_amix_enum[3]), + SOC_ENUM("Frame Rate Diff Error", fsl_amix_enum[4]), + SOC_ENUM("Clock Freq Diff Error", fsl_amix_enum[5]), + SOC_ENUM("Sync Mode Config", fsl_amix_enum[6]), + SOC_ENUM("Sync Mode Clk Source", fsl_amix_enum[7]), + /* TDM1 Attenuation controls */ + SOC_ENUM("TDM1 Attenuation", fsl_amix_enum[8]), + SOC_ENUM("TDM1 Attenuation Direction", fsl_amix_enum[9]), + SOC_SINGLE("TDM1 Attenuation Step Divider", FSL_AMIX_ATCR0, + 2, 0x00fff, 0), + SOC_SINGLE("TDM1 Attenuation Initial Value", FSL_AMIX_ATIVAL0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Up Factor", FSL_AMIX_ATSTPUP0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Down Factor", FSL_AMIX_ATSTPDN0, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM1 Attenuation Step Target", FSL_AMIX_ATSTPTGT0, + 0, 0x3ffff, 0), + /* TDM2 Attenuation controls */ + SOC_ENUM("TDM2 Attenuation", fsl_amix_enum[10]), + SOC_ENUM("TDM2 Attenuation Direction", fsl_amix_enum[11]), + SOC_SINGLE("TDM2 Attenuation Step Divider", FSL_AMIX_ATCR1, + 2, 0x00fff, 0), + SOC_SINGLE("TDM2 Attenuation Initial Value", FSL_AMIX_ATIVAL1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Up Factor", FSL_AMIX_ATSTPUP1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Down Factor", FSL_AMIX_ATSTPDN1, + 0, 0x3ffff, 0), + SOC_SINGLE("TDM2 Attenuation Step Target", FSL_AMIX_ATSTPTGT1, + 0, 0x3ffff, 0), +}; + +static int fsl_amix_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *comp = dai->component; + u32 mask = 0, ctr = 0; + + /* AMIX is working in DSP_A format only */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + break; + default: + return -EINVAL; + } + + /* For playback the AMIX is slave, and for record is master */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + /* Output data will be written on positive edge of the clock */ + ctr |= FSL_AMIX_CTR_OUTCKPOL(0); + break; + case SND_SOC_DAIFMT_NB_NF: + /* Output data will be written on negative edge of the clock */ + ctr |= FSL_AMIX_CTR_OUTCKPOL(1); + break; + default: + return -EINVAL; + } + + mask |= FSL_AMIX_CTR_OUTCKPOL_MASK; + + return snd_soc_component_update_bits(comp, FSL_AMIX_CTR, mask, ctr); +} + +static int fsl_amix_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsl_amix *priv = snd_soc_dai_get_drvdata(dai); + + /* Capture stream shall not be handled */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + priv->tdms |= BIT(dai->driver->id); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + priv->tdms &= ~BIT(dai->driver->id); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops fsl_amix_dai_ops = { + .set_fmt = fsl_amix_dai_set_fmt, + .trigger = fsl_amix_dai_trigger, +}; + +static struct snd_soc_dai_driver fsl_amix_dai[] = { + { + .id = 0, + .name = "amix-0", + .playback = { + .stream_name = "AMIX-Playback-0", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AMIX_FORMATS, + }, + .capture = { + .stream_name = "AMIX-Capture-0", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AMIX_FORMATS, + }, + .ops = &fsl_amix_dai_ops, + }, + { + .id = 1, + .name = "amix-1", + .playback = { + .stream_name = "AMIX-Playback-1", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AMIX_FORMATS, + }, + .capture = { + .stream_name = "AMIX-Capture-1", + .channels_min = 8, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = FSL_AMIX_FORMATS, + }, + .ops = &fsl_amix_dai_ops, + }, +}; + +static const struct snd_soc_component_driver fsl_amix_component = { + .name = "fsl-amix-dai", + .controls = fsl_amix_snd_controls, + .num_controls = ARRAY_SIZE(fsl_amix_snd_controls), +}; + +static bool fsl_amix_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_AMIX_CTR: + case FSL_AMIX_STR: + case FSL_AMIX_ATCR0: + case FSL_AMIX_ATIVAL0: + case FSL_AMIX_ATSTPUP0: + case FSL_AMIX_ATSTPDN0: + case FSL_AMIX_ATSTPTGT0: + case FSL_AMIX_ATTNVAL0: + case FSL_AMIX_ATSTP0: + case FSL_AMIX_ATCR1: + case FSL_AMIX_ATIVAL1: + case FSL_AMIX_ATSTPUP1: + case FSL_AMIX_ATSTPDN1: + case FSL_AMIX_ATSTPTGT1: + case FSL_AMIX_ATTNVAL1: + case FSL_AMIX_ATSTP1: + return true; + default: + return false; + } +} + +static bool fsl_amix_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_AMIX_CTR: + case FSL_AMIX_ATCR0: + case FSL_AMIX_ATIVAL0: + case FSL_AMIX_ATSTPUP0: + case FSL_AMIX_ATSTPDN0: + case FSL_AMIX_ATSTPTGT0: + case FSL_AMIX_ATCR1: + case FSL_AMIX_ATIVAL1: + case FSL_AMIX_ATSTPUP1: + case FSL_AMIX_ATSTPDN1: + case FSL_AMIX_ATSTPTGT1: + return true; + default: + return false; + } +} + +static const struct reg_default fsl_amix_reg[] = { + { FSL_AMIX_CTR, 0x00060 }, + { FSL_AMIX_STR, 0x00003 }, + { FSL_AMIX_ATCR0, 0x00000 }, + { FSL_AMIX_ATIVAL0, 0x3FFFF }, + { FSL_AMIX_ATSTPUP0, 0x2AAAA }, + { FSL_AMIX_ATSTPDN0, 0x30000 }, + { FSL_AMIX_ATSTPTGT0, 0x00010 }, + { FSL_AMIX_ATTNVAL0, 0x00000 }, + { FSL_AMIX_ATSTP0, 0x00000 }, + { FSL_AMIX_ATCR1, 0x00000 }, + { FSL_AMIX_ATIVAL1, 0x3FFFF }, + { FSL_AMIX_ATSTPUP1, 0x2AAAA }, + { FSL_AMIX_ATSTPDN1, 0x30000 }, + { FSL_AMIX_ATSTPTGT1, 0x00010 }, + { FSL_AMIX_ATTNVAL1, 0x00000 }, + { FSL_AMIX_ATSTP1, 0x00000 }, +}; + +static const struct regmap_config fsl_amix_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = FSL_AMIX_ATSTP1, + .reg_defaults = fsl_amix_reg, + .num_reg_defaults = ARRAY_SIZE(fsl_amix_reg), + .readable_reg = fsl_amix_readable_reg, + .writeable_reg = fsl_amix_writeable_reg, + .cache_type = REGCACHE_FLAT, +}; + +static int fsl_amix_probe(struct platform_device *pdev) +{ + struct fsl_amix *priv; + struct resource *res; + void __iomem *regs; + int ret; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->pdev = pdev; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) + return PTR_ERR(regs); + + priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "ipg", regs, + &fsl_amix_regmap_config); + if (IS_ERR(priv->regmap)) { + dev_err(&pdev->dev, "failed to init regmap\n"); + return PTR_ERR(priv->regmap); + } + + priv->ipg_clk = devm_clk_get(&pdev->dev, "ipg"); + if (IS_ERR(priv->ipg_clk)) { + dev_err(&pdev->dev, "failed to get ipg clock\n"); + return PTR_ERR(priv->ipg_clk); + } + + platform_set_drvdata(pdev, priv); + pm_runtime_enable(&pdev->dev); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_amix_component, + fsl_amix_dai, + ARRAY_SIZE(fsl_amix_dai)); + if (ret) { + dev_err(&pdev->dev, "failed to register ASoC DAI\n"); + return ret; + } + + return 0; +} + +#ifdef CONFIG_PM +static int fsl_amix_runtime_resume(struct device *dev) +{ + struct fsl_amix *priv = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(priv->ipg_clk); + if (ret) { + dev_err(dev, "Failed to enable IPG clock: %d\n", ret); + return ret; + } + + regcache_cache_only(priv->regmap, false); + regcache_mark_dirty(priv->regmap); + + return regcache_sync(priv->regmap); +} + +static int fsl_amix_runtime_suspend(struct device *dev) +{ + struct fsl_amix *priv = dev_get_drvdata(dev); + + regcache_cache_only(priv->regmap, true); + + clk_disable_unprepare(priv->ipg_clk); + + return 0; +} +#endif /* CONFIG_PM */ + +static const struct dev_pm_ops fsl_amix_pm = { + SET_RUNTIME_PM_OPS(fsl_amix_runtime_suspend, + fsl_amix_runtime_resume, + NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) +}; + +static const struct of_device_id fsl_amix_ids[] = { + { .compatible = "fsl,imx8qm-amix", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, fsl_amix_ids); + +static struct platform_driver fsl_amix_driver = { + .probe = fsl_amix_probe, + .driver = { + .name = "fsl-amix", + .of_match_table = fsl_amix_ids, + .pm = &fsl_amix_pm, + }, +}; +module_platform_driver(fsl_amix_driver); + +MODULE_DESCRIPTION("NXP AMIX ASoC DAI driver"); +MODULE_AUTHOR("Viorel Suman <viorel.suman(a)nxp.com>"); +MODULE_ALIAS("platform:fsl-amix"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_amix.h b/sound/soc/fsl/fsl_amix.h new file mode 100644 index 0000000..5312683 --- /dev/null +++ b/sound/soc/fsl/fsl_amix.h @@ -0,0 +1,101 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * NXP AMIX ALSA SoC Digital Audio Interface (DAI) driver + * + * Copyright 2017 NXP + */ + +#ifndef __FSL_AMIX_H +#define __FSL_AMIX_H + +#define FSL_AMIX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) +/* AMIX Registers */ +#define FSL_AMIX_CTR 0x200 /* Control */ +#define FSL_AMIX_STR 0x204 /* Status */ + +#define FSL_AMIX_ATCR0 0x208 /* Attenuation Control */ +#define FSL_AMIX_ATIVAL0 0x20c /* Attenuation Initial Value */ +#define FSL_AMIX_ATSTPUP0 0x210 /* Attenuation step up factor */ +#define FSL_AMIX_ATSTPDN0 0x214 /* Attenuation step down factor */ +#define FSL_AMIX_ATSTPTGT0 0x218 /* Attenuation step target */ +#define FSL_AMIX_ATTNVAL0 0x21c /* Attenuation Value */ +#define FSL_AMIX_ATSTP0 0x220 /* Attenuation step number */ + +#define FSL_AMIX_ATCR1 0x228 /* Attenuation Control */ +#define FSL_AMIX_ATIVAL1 0x22c /* Attenuation Initial Value */ +#define FSL_AMIX_ATSTPUP1 0x230 /* Attenuation step up factor */ +#define FSL_AMIX_ATSTPDN1 0x234 /* Attenuation step down factor */ +#define FSL_AMIX_ATSTPTGT1 0x238 /* Attenuation step target */ +#define FSL_AMIX_ATTNVAL1 0x23c /* Attenuation Value */ +#define FSL_AMIX_ATSTP1 0x240 /* Attenuation step number */ + +/* AMIX Control Register */ +#define FSL_AMIX_CTR_MIXCLK_SHIFT 0 +#define FSL_AMIX_CTR_MIXCLK_MASK BIT(FSL_AMIX_CTR_MIXCLK_SHIFT) +#define FSL_AMIX_CTR_MIXCLK(i) ((i) << FSL_AMIX_CTR_MIXCLK_SHIFT) +#define FSL_AMIX_CTR_OUTSRC_SHIFT 1 +#define FSL_AMIX_CTR_OUTSRC_MASK (0x3 << FSL_AMIX_CTR_OUTSRC_SHIFT) +#define FSL_AMIX_CTR_OUTSRC(i) (((i) << FSL_AMIX_CTR_OUTSRC_SHIFT) \ + & FSL_AMIX_CTR_OUTSRC_MASK) +#define FSL_AMIX_CTR_OUTWIDTH_SHIFT 3 +#define FSL_AMIX_CTR_OUTWIDTH_MASK (0x7 << FSL_AMIX_CTR_OUTWIDTH_SHIFT) +#define FSL_AMIX_CTR_OUTWIDTH(i) (((i) << FSL_AMIX_CTR_OUTWIDTH_SHIFT) \ + & FSL_AMIX_CTR_OUTWIDTH_MASK) +#define FSL_AMIX_CTR_OUTCKPOL_SHIFT 6 +#define FSL_AMIX_CTR_OUTCKPOL_MASK BIT(FSL_AMIX_CTR_OUTCKPOL_SHIFT) +#define FSL_AMIX_CTR_OUTCKPOL(i) ((i) << FSL_AMIX_CTR_OUTCKPOL_SHIFT) +#define FSL_AMIX_CTR_MASKRTDF_SHIFT 7 +#define FSL_AMIX_CTR_MASKRTDF_MASK BIT(FSL_AMIX_CTR_MASKRTDF_SHIFT) +#define FSL_AMIX_CTR_MASKRTDF(i) ((i) << FSL_AMIX_CTR_MASKRTDF_SHIFT) +#define FSL_AMIX_CTR_MASKCKDF_SHIFT 8 +#define FSL_AMIX_CTR_MASKCKDF_MASK BIT(FSL_AMIX_CTR_MASKCKDF_SHIFT) +#define FSL_AMIX_CTR_MASKCKDF(i) ((i) << FSL_AMIX_CTR_MASKCKDF_SHIFT) +#define FSL_AMIX_CTR_SYNCMODE_SHIFT 9 +#define FSL_AMIX_CTR_SYNCMODE_MASK BIT(FSL_AMIX_CTR_SYNCMODE_SHIFT) +#define FSL_AMIX_CTR_SYNCMODE(i) ((i) << FSL_AMIX_CTR_SYNCMODE_SHIFT) +#define FSL_AMIX_CTR_SYNCSRC_SHIFT 10 +#define FSL_AMIX_CTR_SYNCSRC_MASK BIT(FSL_AMIX_CTR_SYNCSRC_SHIFT) +#define FSL_AMIX_CTR_SYNCSRC(i) ((i) << FSL_AMIX_CTR_SYNCSRC_SHIFT) + +/* AMIX Status Register */ +#define FSL_AMIX_STR_RATEDIFF BIT(0) +#define FSL_AMIX_STR_CLKDIFF BIT(1) +#define FSL_AMIX_STR_MIXSTAT_SHIFT 2 +#define FSL_AMIX_STR_MIXSTAT_MASK (0x3 << FSL_AMIX_STR_MIXSTAT_SHIFT) +#define FSL_AMIX_STR_MIXSTAT(i) ((i & FSL_AMIX_STR_MIXSTAT_MASK) \ + >> FSL_AMIX_STR_MIXSTAT_SHIFT) +/* AMIX Attenuation Control Register */ +#define FSL_AMIX_ATCR_AT_EN BIT(0) +#define FSL_AMIX_ATCR_AT_UPDN BIT(1) +#define FSL_AMIX_ATCR_ATSTPDIF_SHIFT 2 +#define FSL_AMIX_ATCR_ATSTPDFI_MASK (0xfff << FSL_AMIX_ATCR_ATSTPDIF_SHIFT) + +/* AMIX Attenuation Initial Value Register */ +#define FSL_AMIX_ATIVAL_ATINVAL_MASK 0x3FFFF + +/* AMIX Attenuation Step Up Factor Register */ +#define FSL_AMIX_ATSTPUP_ATSTEPUP_MASK 0x3FFFF + +/* AMIX Attenuation Step Down Factor Register */ +#define FSL_AMIX_ATSTPDN_ATSTEPDN_MASK 0x3FFFF + +/* AMIX Attenuation Step Target Register */ +#define FSL_AMIX_ATSTPTGT_ATSTPTG_MASK 0x3FFFF + +/* AMIX Attenuation Value Register */ +#define FSL_AMIX_ATTNVAL_ATCURVAL_MASK 0x3FFFF + +/* AMIX Attenuation Step Number Register */ +#define FSL_AMIX_ATSTP_STPCTR_MASK 0x3FFFF + +#define FSL_AMIX_MAX_DAIS 2 +struct fsl_amix { + struct platform_device *pdev; + struct regmap *regmap; + struct clk *ipg_clk; + u8 tdms; +}; + +#endif /* __FSL_AMIX_H */ -- 2.7.4

4 8

[alsa-devel] [PATCH 1/2] ASoC: Intel: sst: Fallback to BYT-CR if IRQ 5 is missing
by Stephan Gerhold 02 Jan '19

02 Jan '19

Some devices detected as BYT-T by the PMIC-type based detection have only a single IRQ listed in the 80860F28 ACPI device. This causes -ENXIO later when attempting to get the IRQ at index 5. It turns out these devices behave more like BYT-CR devices, and using the IRQ at index 0 makes sound work correctly. This patch adds a fallback for these devices to is_byt_cr(): If there is no IRQ resource at index 5, treating the device as BYT-T is guaranteed to fail later, so we can safely treat these devices as BYT-CR without breaking any working device. Link: http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143176.h… Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Moved the "Detected Baytrail-CR platform" message to is_byt_cr() so we can log a different message if the fallback is used. Tested this on my device as-is, and simulated a "normal" BYT-T and BYT-CR device (copied their IRQs to a custom DSDT). sound/soc/intel/atom/sst/sst_acpi.c | 24 ++++++++++++++++++------ 1 file changed, 18 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 3a95ebbfc45d..755a396121ff 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -255,10 +255,22 @@ static int is_byt(void) return status; } -static int is_byt_cr(struct device *dev, bool *bytcr) +static int is_byt_cr(struct platform_device *pdev, bool *bytcr) { + struct device *dev = &pdev->dev; int status = 0; + if (platform_get_resource(pdev, IORESOURCE_IRQ, 5) == NULL) { + /* + * Some devices detected as BYT-T have only a single IRQ listed, + * causing platform_get_irq with index 5 to return -ENXIO. + * The correct IRQ in this case is at index 0, as used on BYT-CR. + */ + dev_info(dev, "Falling back to Baytrail-CR platform\n"); + *bytcr = true; + return status; + } + if (IS_ENABLED(CONFIG_IOSF_MBI)) { u32 bios_status; @@ -278,10 +290,12 @@ static int is_byt_cr(struct device *dev, bool *bytcr) /* bits 26:27 mirror PMIC options */ bios_status = (bios_status >> 26) & 3; - if ((bios_status == 1) || (bios_status == 3)) + if ((bios_status == 1) || (bios_status == 3)) { + dev_info(dev, "Detected Baytrail-CR platform\n"); *bytcr = true; - else + } else { dev_info(dev, "BYT-CR not detected\n"); + } } } else { dev_info(dev, "IOSF_MBI not enabled, no BYT-CR detection\n"); @@ -333,10 +347,8 @@ static int sst_acpi_probe(struct platform_device *pdev) if (ret < 0) return ret; - ret = is_byt_cr(dev, &bytcr); + ret = is_byt_cr(pdev, &bytcr); if (!(ret < 0 || !bytcr)) { - dev_info(dev, "Detected Baytrail-CR platform\n"); - /* override resource info */ byt_rvp_platform_data.res_info = &bytcr_res_info; } -- 2.20.1

3 10

[alsa-devel] [PATCH v4 08/11] ASoC: Intel: atom: Make PCI dependency explicit
by Sinan Kaya 02 Jan '19

02 Jan '19

Code does unconditional select for IOSF_MBI. IOSF_MBI driver depends on CONFIG_PCI set but this is not specified anywhere. Fixes: 5d32a66541c46 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set") Signed-off-by: Sinan Kaya <okaya(a)kernel.org> --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2fd1b61e8331..b0764b2fe001 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI - depends on X86 && ACPI + depends on X86 && ACPI && PCI select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH -- 2.19.0

4 11

[alsa-devel] [PATCH 1/4] conf/ucm: bytcr-rt5651: Enable Stereo? ADC MIXL ADC? switches when enabling inputs
by Hans de Goede 01 Jan '19

01 Jan '19

Explicitly enable the right 'Stereo? ADC MIXL ADC?' when selecting an input, rather then just enabling them all in ucm/codecs/rt5651/EnableSeq.conf . This is a preparation patch for adding digital mic support. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/conf/ucm/codecs/rt5651/EnableSeq.conf | 16 ++++++++-------- src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf | 4 ++++ src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf | 4 ++++ src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf | 4 ++++ src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf | 4 ++++ 5 files changed, 24 insertions(+), 8 deletions(-) diff --git a/src/conf/ucm/codecs/rt5651/EnableSeq.conf b/src/conf/ucm/codecs/rt5651/EnableSeq.conf index 6fb3c1f2..db2fd328 100644 --- a/src/conf/ucm/codecs/rt5651/EnableSeq.conf +++ b/src/conf/ucm/codecs/rt5651/EnableSeq.conf @@ -14,12 +14,14 @@ cset "name='Stereo DAC MIXL DAC L1 Switch' on" cset "name='Stereo DAC MIXL DAC R1 Switch' off" cset "name='Stereo DAC MIXR DAC L1 Switch' off" cset "name='Stereo DAC MIXR DAC R1 Switch' on" -cset "name='Stereo1 ADC MIXR ADC1 Switch' on" -cset "name='Stereo1 ADC MIXR ADC2 Switch' on" -cset "name='Stereo2 ADC MIXL ADC1 Switch' on" -cset "name='Stereo2 ADC MIXL ADC2 Switch' on" -cset "name='Stereo2 ADC MIXR ADC1 Switch' on" -cset "name='Stereo2 ADC MIXR ADC2 Switch' on" +cset "name='Stereo1 ADC MIXL ADC1 Switch' off" +cset "name='Stereo1 ADC MIXR ADC1 Switch' off" +cset "name='Stereo1 ADC MIXL ADC2 Switch' off" +cset "name='Stereo1 ADC MIXR ADC2 Switch' off" +cset "name='Stereo2 ADC MIXL ADC1 Switch' off" +cset "name='Stereo2 ADC MIXR ADC1 Switch' off" +cset "name='Stereo2 ADC MIXL ADC2 Switch' off" +cset "name='Stereo2 ADC MIXR ADC2 Switch' off" cset "name='Headphone Switch' off" cset "name='Speaker Switch' off" @@ -41,8 +43,6 @@ cset "name='IN3 Boost' 1" # input cset "name='Stereo1 ADC L1 Mux' ADC" cset "name='Stereo1 ADC R1 Mux' ADC" -cset "name='Stereo1 ADC MIXL ADC1 Switch' on" -cset "name='Stereo1 ADC MIXR ADC1 Switch' on" cset "name='RECMIXL BST1 Switch' off" cset "name='RECMIXR BST1 Switch' off" cset "name='RECMIXL BST3 Switch' off" diff --git a/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf index 85e25a5c..b2ce17b6 100644 --- a/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf +++ b/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf @@ -12,10 +12,14 @@ SectionDevice."InternalMic-IN1" { cset "name='Internal Mic Switch' on" cset "name='RECMIXL BST1 Switch' on" cset "name='RECMIXR BST1 Switch' on" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" ] DisableSequence [ cdev "hw:bytcrrt5651" + cset "name='Stereo1 ADC MIXL ADC1 Switch' off" + cset "name='Stereo1 ADC MIXR ADC1 Switch' off" cset "name='Internal Mic Switch' off" cset "name='RECMIXL BST1 Switch' off" cset "name='RECMIXR BST1 Switch' off" diff --git a/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf index f1cc17fe..e8041912 100644 --- a/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf +++ b/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf @@ -14,10 +14,14 @@ SectionDevice."InternalMic-IN12" { cset "name='RECMIXR BST1 Switch' on" cset "name='RECMIXL BST2 Switch' on" cset "name='RECMIXR BST2 Switch' on" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" ] DisableSequence [ cdev "hw:bytcrrt5651" + cset "name='Stereo1 ADC MIXL ADC1 Switch' off" + cset "name='Stereo1 ADC MIXR ADC1 Switch' off" cset "name='Internal Mic Switch' off" cset "name='RECMIXL BST1 Switch' off" cset "name='RECMIXR BST1 Switch' off" diff --git a/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf index 18f74e1a..048fcc95 100644 --- a/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf +++ b/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf @@ -12,10 +12,14 @@ SectionDevice."InternalMic-IN2" { cset "name='Internal Mic Switch' on" cset "name='RECMIXL BST2 Switch' on" cset "name='RECMIXR BST2 Switch' on" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" ] DisableSequence [ cdev "hw:bytcrrt5651" + cset "name='Stereo1 ADC MIXL ADC1 Switch' off" + cset "name='Stereo1 ADC MIXR ADC1 Switch' off" cset "name='Internal Mic Switch' off" cset "name='RECMIXL BST2 Switch' off" cset "name='RECMIXR BST2 Switch' off" diff --git a/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf b/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf index bfd0b009..712d14bb 100644 --- a/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf +++ b/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf @@ -12,10 +12,14 @@ SectionDevice."HeadsetMic-IN3" { cset "name='Headset Mic Switch' on" cset "name='RECMIXL BST3 Switch' on" cset "name='RECMIXR BST3 Switch' on" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" ] DisableSequence [ cdev "hw:bytcrrt5651" + cset "name='Stereo1 ADC MIXL ADC1 Switch' off" + cset "name='Stereo1 ADC MIXR ADC1 Switch' off" cset "name='Headset Mic Switch' off" cset "name='RECMIXL BST3 Switch' off" cset "name='RECMIXR BST3 Switch' off" -- 2.20.1

2 6

[alsa-devel] [PATCH v2] staging: bcm2835-audio: interpolate audio delay
by Mike Brady 01 Jan '19

01 Jan '19

When the BCM2835 audio output is used, userspace sees a jitter up to 10ms in the audio position, aka "delay" -- the number of frames that must be output before a new frame would be played. Make this a bit nicer for userspace by interpolating the position using the CPU clock. The overhead is small -- an extra ktime_get() every time a GPU message is sent -- and another call and a few calculations whenever the delay is sought from userland. At 48,000 frames per second, i.e. approximately 20 microseconds per frame, it would take a clock inaccuracy of 20 microseconds in 10 milliseconds -- 2,000 parts per million -- to result in an inaccurate estimate, whereas crystal- or resonator-based clocks typically have an inaccuracy of 10s to 100s of parts per million. Signed-off-by: Mike Brady <mikebrady(a)eircom.net> --- Changes in v2 -- remove inappropriate addition of SNDRV_PCM_INFO_BATCH flag .../vc04_services/bcm2835-audio/bcm2835-pcm.c | 20 +++++++++++++++++++ .../vc04_services/bcm2835-audio/bcm2835.h | 1 + 2 files changed, 21 insertions(+) diff --git a/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c b/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c index e66da11af5cf..9053b996cada 100644 --- a/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c +++ b/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c @@ -74,6 +74,7 @@ void bcm2835_playback_fifo(struct bcm2835_alsa_stream *alsa_stream, atomic_set(&alsa_stream->pos, pos); alsa_stream->period_offset += bytes; + alsa_stream->interpolate_start = ktime_get(); if (alsa_stream->period_offset >= alsa_stream->period_size) { alsa_stream->period_offset %= alsa_stream->period_size; snd_pcm_period_elapsed(substream); @@ -243,6 +244,7 @@ static int snd_bcm2835_pcm_prepare(struct snd_pcm_substream *substream) atomic_set(&alsa_stream->pos, 0); alsa_stream->period_offset = 0; alsa_stream->draining = false; + alsa_stream->interpolate_start = ktime_get(); return 0; } @@ -292,6 +294,24 @@ snd_bcm2835_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct bcm2835_alsa_stream *alsa_stream = runtime->private_data; + ktime_t now = ktime_get(); + + /* Give userspace better delay reporting by interpolating between GPU + * notifications, assuming audio speed is close enough to the clock + * used for ktime + */ + + if ((ktime_to_ns(alsa_stream->interpolate_start)) && + (ktime_compare(alsa_stream->interpolate_start, now) < 0)) { + u64 interval = + (ktime_to_ns(ktime_sub(now, + alsa_stream->interpolate_start))); + u64 frames_output_in_interval = + div_u64((interval * runtime->rate), 1000000000); + snd_pcm_sframes_t frames_output_in_interval_sized = + -frames_output_in_interval; + runtime->delay = frames_output_in_interval_sized; + } return snd_pcm_indirect_playback_pointer(substream, &alsa_stream->pcm_indirect, diff --git a/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h b/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h index e13435d1c205..595ad584243f 100644 --- a/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h +++ b/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h @@ -78,6 +78,7 @@ struct bcm2835_alsa_stream { unsigned int period_offset; unsigned int buffer_size; unsigned int period_size; + ktime_t interpolate_start; struct bcm2835_audio_instance *instance; int idx; -- 2.17.1

4 17

[alsa-devel] [PATCH] aplay: support no period wakeup option in argument
by S.j. Wang 01 Jan '19

01 Jan '19

In the case that alsa driver can't support period wakeup, we need to set the no period wakeup flag Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- aplay/aplay.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/aplay/aplay.c b/aplay/aplay.c index efc1eb4cae3a..4f562bfe2884 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -137,6 +137,7 @@ static int use_strftime = 0; volatile static int recycle_capture_file = 0; static long term_c_lflag = -1; static int dump_hw_params = 0; +static int no_period_wakeup = 0; static int fd = -1; static off64_t pbrec_count = LLONG_MAX, fdcount; @@ -243,6 +244,7 @@ _("Usage: %s [OPTION]... [FILE]...\n" " --use-strftime apply the strftime facility to the output file name\n" " --dump-hw-params dump hw_params of the device\n" " --fatal-errors treat all errors as fatal\n" +" --no-period-wakeup set no period wakeup flag if necessary\n" ) , command); printf(_("Recognized sample formats are:")); @@ -429,6 +431,7 @@ enum { OPT_USE_STRFTIME, OPT_DUMP_HWPARAMS, OPT_FATAL_ERRORS, + OPT_NO_PERIOD_WAKEUP, }; /* @@ -516,6 +519,7 @@ int main(int argc, char *argv[]) {"interactive", 0, 0, 'i'}, {"dump-hw-params", 0, 0, OPT_DUMP_HWPARAMS}, {"fatal-errors", 0, 0, OPT_FATAL_ERRORS}, + {"no-period-wakeup", 0, 0, OPT_NO_PERIOD_WAKEUP}, #ifdef CONFIG_SUPPORT_CHMAP {"chmap", 1, 0, 'm'}, #endif @@ -799,6 +803,9 @@ int main(int argc, char *argv[]) case OPT_FATAL_ERRORS: fatal_errors = 1; break; + case OPT_NO_PERIOD_WAKEUP: + no_period_wakeup = 1; + break; #ifdef CONFIG_SUPPORT_CHMAP case 'm': channel_map = snd_pcm_chmap_parse_string(optarg); @@ -1396,6 +1403,12 @@ static void set_params(void) &buffer_frames); } assert(err >= 0); + + if (no_period_wakeup) { + err = snd_pcm_hw_params_set_period_wakeup(handle, params, 0); + assert(err >= 0); + } + monotonic = snd_pcm_hw_params_is_monotonic(params); can_pause = snd_pcm_hw_params_can_pause(params); err = snd_pcm_hw_params(handle, params); -- 1.9.1

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