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

[alsa-devel] [PATCH v2 0/2] ASoC: omap: sdma-pcm: modpost warning fix and include cleanup
by Peter Ujfalusi 15 May '18

15 May '18

Hi, Changes since v1: - Changed the MODULE_LICENSE from "GPL" to "GPL v2" to match the SPDX line. Since the sdma-pcm can be built as module it must have MODULE_LICENSE defined, whcih I have missed: After merging the sound-asoc tree, today's linux-next build (x86_64 allmodconfig) produced this warning: WARNING: modpost: missing MODULE_LICENSE() in sound/soc/omap/snd-soc-sdma.o see include/linux/module.h for more information WARNING: modpost: missing MODULE_LICENSE() in sound/soc/omap/snd-soc-sdma.o see include/linux/module.h for more information The first patch also going to fix runtime issue with module loading: snd_soc_sdma: Unknown symbol devm_kmalloc (err 0) snd_soc_sdma: Unknown symbol omap_dma_filter_fn (err 0) snd_soc_sdma: Unknown symbol snd_dmaengine_pcm_prepare_slave_config (err 0) snd_soc_sdma: Unknown symbol devm_snd_dmaengine_pcm_register (err 0) As these symbols are EXPORT_SYMBOL_GPL()... Introduced by commit dde637f2daf1 ("ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm") The second patch is kust a small adjustement for the include files. Regards, Peter --- Peter Ujfalusi (2): firmware: ti_sci: Correct the timeout type in ti_sci_do_xfer() ASoC: omap: sdma-pcm: Fix modpost warning drivers/firmware/ti_sci.c | 2 +- sound/soc/omap/sdma-pcm.c | 5 +++++ 2 files changed, 6 insertions(+), 1 deletion(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH 0/4] ALSA: usb: UAC3 new features.
by Jorge Sanjuan 15 May '18

15 May '18

Now that the UAC3 patch [1] has made it to linux-next I have some extra features to make a UAC3 device fully work in Linux. Including Jack insertion control that I have put on top of this other patch [2] for UAC2. Also adding support for the UAC3 Mixer Unit which is most likely to appear in most headset type devices. UAC3 devices also require to have a Basic Audio Device (BADD) in a separate config for which both Ruslan Bilovol and myself have submited different approaches[3][4] but I don't know what the final merge will be. Once there is official support for BADD, we'll need to test it with an actual UAC3 device to confirm it all wokrs. All this features are tested with an actual UAC3 device that is still in development. For this patch series, only the legacy config (#1. UAC1/UAC2) and the UAC3 config have been tested. The BADD config is only tested using and updated verison of [4]. [1]: https://patchwork.kernel.org/patch/10298179/ [2]: https://patchwork.kernel.org/patch/10305847/ [3]: https://patchwork.kernel.org/patch/10340851/ [4]: https://www.spinics.net/lists/alsa-devel/msg71617.html Based on linux-next tag: next-20180420 Jorge Sanjuan (3): ALSA: usb-audio: UAC3. Add support for mixer unit. ALSA: usb-audio: Use Class Specific EP for UAC3 devices. ALSA: usb-audio: UAC3 Add support for connector insertion. Michael Drake (1): ALSA: usb-audio: ADC3: Fix channel mapping conversion for ADC3. include/linux/usb/audio-v2.h | 7 ++ include/linux/usb/audio-v3.h | 14 ++++ include/uapi/linux/usb/audio.h | 13 ++- sound/usb/mixer.c | 175 ++++++++++++++++++++++++++++++++++++----- sound/usb/stream.c | 19 ++++- 5 files changed, 204 insertions(+), 24 deletions(-) -- 2.11.0

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[alsa-devel] [sound:for-next 62/68] sound/pci/hda/patch_ca0132.c:5062:50-51: WARNING: Use ARRAY_SIZE
by kbuild test robot 14 May '18

14 May '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 4844f51282b60f8a96b05d422f452a785df308b9 commit: 47cdf76e44e87d25b91c0f0d0539944c932941ba [62/68] ALSA: hda/ca0132: Add new control changes for SBZ + R3Di coccinelle warnings: (new ones prefixed by >>) >> sound/pci/hda/patch_ca0132.c:5062:50-51: WARNING: Use ARRAY_SIZE sound/pci/hda/patch_ca0132.c:5092:50-51: WARNING: Use ARRAY_SIZE Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 0/4] ALSA: control: optimization for position offset macro of TLV data
by Takashi Sakamoto 14 May '18

14 May '18

Hi, This patchset is a supplement for a commit 08f9f4485f21 ('ALSA: core api: define offsets for TLV items') for items below: - add position offset macro for db-minmax and db-linear types of TLV data - optimization with these macros Takashi Sakamoto (4): ALSA: control: complement TLV macro for db-minmax and db-linear types ALSA: vmaster: use position offset macro of TLV data ALSA: hda: use position offset macro of TLV data ALSA: isight: use position offset macro of TLV data include/uapi/sound/tlv.h | 8 ++++++++ sound/core/vmaster.c | 16 +++++++++------- sound/firewire/isight.c | 10 ++++++---- sound/pci/hda/hda_codec.c | 22 +++++++++++----------- sound/pci/hda/hda_generic.c | 2 +- 5 files changed, 35 insertions(+), 23 deletions(-) -- 2.14.1

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[alsa-devel] [PATCH] Clevo P950ER ALC1220 Fixup
by Jeremy Soller 14 May '18

14 May '18

This adds support for the P950ER, which has the same required fixup as the P950HR, but has a different PCI ID. Signed-off-by: Jeremy Soller <jeremy(a)system76.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2dd34dd77447..01a6643fc7d4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2363,6 +2363,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1558, 0x9501, "Clevo P950HR", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1558, 0x95e2, "Clevo P950ER", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530),

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[alsa-devel] [PATCH v14 0/8] ASoC: sun4i-codec: Add Line-In, FM-In, Mic 2, Capture Source, Differential Line-In
by Danny Milosavljevic 14 May '18

14 May '18

This patchset adds some mixer controls to sun4i-codec for the Allwinner A10 and the Allwinner A20. It also adds a mux for the capture source and the PGA for the MIC2 preamp. Where possible, it uses SOC_DAPM_DOUBLE in order to cut down on the number of distinct controls in alsamixer. v14 changes compared to v13 are: - Merged some of the patches together if it made sense to test them together. - Use snd_soc_component_driver. - Moved SUN4I_CODEC_DAC_ACTL_LFMS, SUN4I_CODEC_DAC_ACTL_RFMS to the correct patch in the series. - Kept "Left Mixer Left DAC Playback Switch", "Right Mixer Left DAC Playback Switch" and "Right Mixer Right DAC Playback Switch" unchanged compared to the released version - for backward compatibility. v13 changes compared to v12 are: - Added my "Signed-off-by". - Clarified some commit message text. v12 changes compared to v11 are: - Split up patchset in another way. - Renamed "Mic1 Capture Volume" to "Mic1 Boost Volume". - Renamed "Mic2 Capture Volume" to "Mic2 Boost Volume". - Renamed "Line Capture Volume" to "Line Boost Volume". - Renamed "Differential Line Capture Switch" to "Differential Line Source". v11 changes compared to v10 are: - Split up patchset. - Fixed typo in Differential Line Capture Switch. - Renamed "Non-Differential" value to "Stereo". - Removed duplicate PA Volume mixer control. v10 changes compared to v9 are: - Use SOC_DAPM_DOUBLE where possible and it makes sense in order to cut down on the number of controls. v9 changes compared to v8 are: - added Line Differential Capture Switch. - split Capture Source into Left Capture Select, Right Capture Select. - added Line Capture Volume. - rename "sun4i_codec_widgets" to "sun4i_codec_controls" for consistency with the struct field it's used in. - rename "Line-In" to "Line". - rename "Power Amplifier Playback Volume" to "Headphone Playback Volume". v8 changes compared to v7 are: - fixed the routes for line and mic capturing. v7 changes compared to v6 are: - preparation for different A20, A10 controls is now in an extra patch. - all register definitions are now at the top. - sun7i-specific things (A20-specific things) are now less grouped-together. - rename "Power Amplifier Volume" to "Power Amplifier Playback Volume". v6 changes compared to v5 are: - Mic preamplifier special cases for A20 and A10 now are now not icky: There are two different _widget arrays and the probe() function now selects the right one to pass to snd_soc_register_codec() unmodified. - sun7i-specific things (A20-specific things) are now grouped together. v5 changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. v4 changes compared to v3 are: - names of the input are not uppercase anymore. - bit index constants are now named as in the A20 user manual v1.4. - added Mic1-In, Mac2-In. - added Mic1 and Mic2 Pre-Amplifiers. v3 changes compared to v2 are: - added DAPM routes. v2 changes compared to v1 are: - moved Line-In and FM-In playback switches to their respective sun4i_codec_*_mixer_controls. v1 changes: - added linein, fmin output volumes and switches. Danny Milosavljevic (8): ASoC: sun4i-codec: Add MIC2 Pre-Amplifier, Mic2, Mic Playback Volume ASoC: sun4i-codec: Add Mic1 Boost Volume, Mic2 Boost Volume ASoC: sun4i-codec: Merge sun4i_codec_left_mixer_controls and sun4i_codec_right_mixer_controls into sun4i_codec_mixer_controls. ASoC: sun4i-codec: Add Mic1 Playback Switch, Mic2 Playback Switch. ASoC: sun4i-codec: Add FM Playback Volume, FM Left, FM Right, FM Playback Switch ASoC: sun4i-codec: Add Line Playback Volume, Line Boost Volume, Line Right, Line Left, Line Playback Switch ASoC: sun4i-codec: Add Differential Line Source. ASoC: sun4i-codec: Add Left Capture Select, Right Capture Select. sound/soc/sunxi/sun4i-codec.c | 233 +++++++++++++++++++++++++++++++++++++----- 1 file changed, 209 insertions(+), 24 deletions(-)

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Re: [alsa-devel] [PATCH v2 09/11] docs: Fix some broken references
by Masami Hiramatsu 14 May '18

14 May '18

On Wed, 9 May 2018 10:18:52 -0300 Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> wrote: > As we move stuff around, some doc references are broken. Fix some of > them via this script: > ./scripts/documentation-file-ref-check --fix-rst > > Manually checked if the produced result is valid, removing a few > false-positives. The part of kprobes and trace_kprobe changes look good to me :) Acked-by: Masami Hiramatsu <mhiramat(a)kernel.org> Thank you, > > Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> > --- > .../admin-guide/kernel-parameters.txt | 4 ++-- > .../bindings/input/rotary-encoder.txt | 2 +- > Documentation/driver-api/gpio/consumer.rst | 2 +- > Documentation/kprobes.txt | 4 ++-- > Documentation/trace/coresight.txt | 2 +- > Documentation/trace/events.rst | 2 +- > Documentation/trace/ftrace-uses.rst | 2 +- > Documentation/trace/histogram.txt | 2 +- > Documentation/trace/intel_th.rst | 2 +- > Documentation/trace/tracepoint-analysis.rst | 6 +++--- > Documentation/translations/ja_JP/howto.rst | 4 ++-- > .../translations/zh_CN/magic-number.txt | 4 ++-- > .../zh_CN/video4linux/omap3isp.txt | 4 ++-- > MAINTAINERS | 20 +++++++++---------- > arch/Kconfig | 2 +- > arch/arm/include/asm/cacheflush.h | 2 +- > arch/arm64/include/asm/cacheflush.h | 2 +- > arch/microblaze/include/asm/cacheflush.h | 2 +- > arch/um/Kconfig.um | 2 +- > arch/unicore32/include/asm/cacheflush.h | 2 +- > arch/x86/entry/vsyscall/vsyscall_64.c | 2 +- > arch/xtensa/include/asm/cacheflush.h | 4 ++-- > block/Kconfig | 2 +- > certs/Kconfig | 2 +- > crypto/asymmetric_keys/asymmetric_type.c | 2 +- > crypto/asymmetric_keys/signature.c | 2 +- > drivers/char/Kconfig | 2 +- > drivers/clk/clk.c | 4 ++-- > drivers/clk/ingenic/cgu.h | 2 +- > drivers/gpu/vga/Kconfig | 2 +- > drivers/gpu/vga/vgaarb.c | 2 +- > drivers/input/joystick/Kconfig | 10 +++++----- > drivers/input/joystick/iforce/Kconfig | 2 +- > drivers/input/joystick/walkera0701.c | 2 +- > drivers/input/misc/Kconfig | 4 ++-- > drivers/input/misc/rotary_encoder.c | 2 +- > drivers/input/mouse/Kconfig | 6 +++--- > drivers/input/mouse/alps.c | 2 +- > drivers/input/touchscreen/wm97xx-core.c | 2 +- > drivers/lightnvm/pblk-rb.c | 2 +- > drivers/md/bcache/Kconfig | 2 +- > drivers/md/bcache/btree.c | 2 +- > drivers/md/bcache/extents.c | 2 +- > drivers/media/dvb-core/dvb_ringbuffer.c | 2 +- > drivers/media/pci/meye/Kconfig | 2 +- > drivers/media/platform/pxa_camera.c | 4 ++-- > .../soc_camera/sh_mobile_ceu_camera.c | 2 +- > drivers/media/radio/Kconfig | 2 +- > drivers/media/radio/si470x/Kconfig | 2 +- > drivers/media/usb/dvb-usb-v2/lmedm04.c | 2 +- > drivers/media/usb/zr364xx/Kconfig | 2 +- > drivers/parport/Kconfig | 6 +++--- > drivers/staging/media/bcm2048/TODO | 2 +- > include/keys/asymmetric-subtype.h | 2 +- > include/keys/asymmetric-type.h | 2 +- > include/linux/assoc_array.h | 2 +- > include/linux/assoc_array_priv.h | 2 +- > include/linux/circ_buf.h | 2 +- > include/linux/ftrace.h | 2 +- > include/linux/rculist_nulls.h | 2 +- > include/uapi/linux/prctl.h | 2 +- > include/xen/interface/io/kbdif.h | 2 +- > kernel/cgroup/cpuset.c | 2 +- > kernel/trace/Kconfig | 16 +++++++-------- > lib/Kconfig | 2 +- > security/selinux/hooks.c | 2 +- > sound/core/Kconfig | 4 ++-- > sound/drivers/Kconfig | 4 ++-- > sound/pci/Kconfig | 10 +++++----- > tools/include/uapi/linux/prctl.h | 2 +- > tools/lib/api/fs/fs.c | 2 +- > tools/memory-model/README | 10 +++++----- > tools/perf/util/bpf-prologue.c | 2 +- > .../config/custom-timeline-functions.cfg | 4 ++-- > 74 files changed, 120 insertions(+), 120 deletions(-) > -- Masami Hiramatsu <mhiramat(a)kernel.org>

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[alsa-devel] sound: aplay\arecord audio frame flow
by Muni Sekhar 14 May '18

14 May '18

[ Please keep me in CC as I'm not subscribed to the list] Hi All, Is there anyway in the kernel sound subsystem to know when the application calls ‘aplay’ and anything starts playing? Also for arecord\aplay, Is there any way from kernel's point of view to know when it receives for example the first and last audio frames? -- Thanks, Sekhar

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[alsa-devel] [PATCH V2] ASoC: amd: modified dma pointer callback implementation
by Vijendar Mukunda 14 May '18

14 May '18

Modified dma pointer callback implementation. hw ptr calculated based on System Memory to ACP SRAM dma channel transfer count in case of playback. Increased min period size for playback to 4096 to avoid underruns. In case of capture, hw ptr calculated based on dma transfer count for ACP SRAM to System Memory Dma channel. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v1->v2: removed hard coded values in acp dma pointer callback declared dma varaibles in rtd structure sound/soc/amd/acp-pcm-dma.c | 66 +++++++++++++++++++-------------------------- sound/soc/amd/acp.h | 15 +++-------- 2 files changed, 31 insertions(+), 50 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 906eb77..34a88ab 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -28,7 +28,7 @@ #define PLAYBACK_MIN_NUM_PERIODS 2 #define PLAYBACK_MAX_NUM_PERIODS 2 #define PLAYBACK_MAX_PERIOD_SIZE 16384 -#define PLAYBACK_MIN_PERIOD_SIZE 1024 +#define PLAYBACK_MIN_PERIOD_SIZE 4096 #define CAPTURE_MIN_NUM_PERIODS 2 #define CAPTURE_MAX_NUM_PERIODS 2 #define CAPTURE_MAX_PERIOD_SIZE 16384 @@ -854,10 +854,9 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = TO_BLUETOOTH; rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH8; rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH9; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_BT_TRANSMIT_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_BT_TRANSMIT_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_8; + rtd->cur_trans_cnt = mmACP_DMA_CUR_TRANS_CNT_8; + rtd->end_dma_dscr = PLAYBACK_END_DMA_DESCR_CH8; adata->play_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: @@ -875,10 +874,9 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = TO_ACP_I2S_1; rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH12; rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH13; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_TRANSMIT_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_TRANSMIT_BYTE_CNT_LOW; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_12; + rtd->cur_trans_cnt = mmACP_DMA_CUR_TRANS_CNT_12; + rtd->end_dma_dscr = PLAYBACK_END_DMA_DESCR_CH12; adata->play_i2ssp_stream = substream; } } else { @@ -891,10 +889,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_BLUETOOTH; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH11; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW; + rtd->dma_cntl = mmACP_DMA_CNTL_10; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_10; + rtd->cur_trans_cnt = mmACP_DMA_CUR_TRANS_CNT_10; + rtd->end_dma_dscr = CAPTURE_END_DMA_DESCR_CH10; adata->capture_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: @@ -914,10 +912,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_ACP_I2S_1; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_LOW; + rtd->dma_cntl = mmACP_DMA_CNTL_14; + rtd->dma_curr_dscr = mmACP_DMA_CUR_DSCR_14; + rtd->cur_trans_cnt = mmACP_DMA_CUR_TRANS_CNT_14; + rtd->end_dma_dscr = CAPTURE_END_DMA_DESCR_CH14; adata->capture_i2ssp_stream = substream; } } @@ -955,22 +953,13 @@ static int acp_dma_hw_free(struct snd_pcm_substream *substream) return snd_pcm_lib_free_pages(substream); } -static u64 acp_get_byte_count(struct audio_substream_data *rtd) -{ - union acp_dma_count byte_count; - - byte_count.bcount.high = acp_reg_read(rtd->acp_mmio, - rtd->byte_cnt_high_reg_offset); - byte_count.bcount.low = acp_reg_read(rtd->acp_mmio, - rtd->byte_cnt_low_reg_offset); - return byte_count.bytescount; -} - static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) { u32 buffersize; u32 pos = 0; - u64 bytescount = 0; + u32 dma_count = 0; + u16 dscr; + u32 period_bytes; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -979,11 +968,15 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) return -EINVAL; buffersize = frames_to_bytes(runtime, runtime->buffer_size); - bytescount = acp_get_byte_count(rtd); - - if (bytescount > rtd->bytescount) - bytescount -= rtd->bytescount; - pos = do_div(bytescount, buffersize); + period_bytes = frames_to_bytes(runtime, runtime->period_size); + if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) && + acp_reg_read(rtd->acp_mmio, rtd->dma_cntl) == 0) + return 0; + dscr = acp_reg_read(rtd->acp_mmio, rtd->dma_curr_dscr); + dma_count = acp_reg_read(rtd->acp_mmio, rtd->cur_trans_cnt); + if (dscr == rtd->end_dma_dscr) + dma_count += period_bytes; + pos = dma_count % buffersize; return bytes_to_frames(runtime, pos); } @@ -1016,7 +1009,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) { int ret; u32 loops = 4000; - u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *prtd = substream->private_data; @@ -1030,9 +1022,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - bytescount = acp_get_byte_count(rtd); - if (rtd->bytescount == 0) - rtd->bytescount = bytescount; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { acp_dma_start(rtd->acp_mmio, rtd->ch1, false); while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & @@ -1066,7 +1055,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_stop(rtd->acp_mmio, rtd->ch2); ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); } - rtd->bytescount = 0; break; default: ret = -EINVAL; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 9cd3e96..4740804 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -131,12 +131,13 @@ struct audio_substream_data { u16 destination; u16 dma_dscr_idx_1; u16 dma_dscr_idx_2; + u16 end_dma_dscr; u32 pte_offset; u32 sram_bank; - u32 byte_cnt_high_reg_offset; - u32 byte_cnt_low_reg_offset; + u32 dma_cntl; + u32 dma_curr_dscr; + u32 cur_trans_cnt; uint64_t size; - u64 bytescount; void __iomem *acp_mmio; }; @@ -157,14 +158,6 @@ struct acp_platform_info { u16 i2s_instance; }; -union acp_dma_count { - struct { - u32 low; - u32 high; - } bcount; - u64 bytescount; -}; - enum { ACP_TILE_P1 = 0, ACP_TILE_P2, -- 2.7.4

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[alsa-devel] [PATCH -next] ASoC: omap: add sdma-pcm.c MODULE_LICENSE
by Randy Dunlap 14 May '18

14 May '18

From: Randy Dunlap <rdunlap(a)infradead.org> ASoC: omap: add sdma-pcm.c MODULE_LICENSE Fixes this build warning: WARNING: modpost: missing MODULE_LICENSE() in sound/soc/omap/snd-soc-sdma.o Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/omap/sdma-pcm.c | 3 +++ 1 file changed, 3 insertions(+) --- linux-next-20180510.orig/sound/soc/omap/sdma-pcm.c +++ linux-next-20180510/sound/soc/omap/sdma-pcm.c @@ -9,6 +9,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> +#include <linux/module.h> #include <linux/omap-dma.h> #include "sdma-pcm.h" @@ -66,3 +67,5 @@ int sdma_pcm_platform_register(struct de return devm_snd_dmaengine_pcm_register(dev, config, flags); } EXPORT_SYMBOL_GPL(sdma_pcm_platform_register); + +MODULE_LICENSE("GPL v2");

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