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[alsa-devel] [PATCH - alsalib 1/1] autotools: fix ucm partial build
by Matthieu Crapet 14 Jan '15

14 Jan '15

When --disable-ucm configure option is specified, don't install related include file. Signed-off-by: Matthieu Crapet <mcrapet(a)gmail.com> diff --git a/include/Makefile.am b/include/Makefile.am index 0127d5c..4baa03a 100644 --- a/include/Makefile.am +++ b/include/Makefile.am @@ -5,7 +5,7 @@ alsaincludedir = ${includedir}/alsa alsainclude_HEADERS = asoundlib.h asoundef.h \ version.h global.h input.h output.h error.h \ - conf.h control.h iatomic.h use-case.h + conf.h control.h iatomic.h if BUILD_CTL_PLUGIN_EXT alsainclude_HEADERS += control_external.h @@ -17,7 +17,7 @@ if BUILD_PCM_PLUGIN alsainclude_HEADERS += pcm_plugin.h endif if BUILD_PCM_PLUGIN_RATE -alsainclude_HEADERS += pcm_rate.h +alsainclude_HEADERS += pcm_rate.h endif if BUILD_PCM_PLUGIN_EXTPLUG alsainclude_HEADERS += pcm_external.h pcm_extplug.h @@ -46,6 +46,10 @@ if BUILD_SEQ alsainclude_HEADERS += seq_event.h seq.h seqmid.h seq_midi_event.h endif +if BUILD_UCM +alsainclude_HEADERS += use-case.h +endif + if BUILD_ALISP alsainclude_HEADERS += alisp.h endif -- 1.8.5.4

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Correct rx format unit configuration
by Peter Ujfalusi 14 Jan '15

14 Jan '15

In case of capture we should not use rotation. The reverse and mask is enough to get the data align correctly from the bus to MCU: Format data from bus after reverse (XRBUF) S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| With this patch all supported formats will work for playback and capture. Reported-by: Jyri Sarha <jsarha(a)ti.com> (broken S24_3LE capture) Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6a6b2ff7d7d7..68347b55f6e1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -467,8 +467,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, { u32 fmt; u32 tx_rotate = (word_length / 4) & 0x7; - u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; + /* + * For captured data we should not rotate, inversion and masking is + * enoguh to get the data to the right position: + * Format data from bus after reverse (XRBUF) + * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| + * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| + */ + u32 rx_rotate = 0; /* * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() -- 2.1.0

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[alsa-devel] [PATCH v9 0/4] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 14 Jan '15

14 Jan '15

Based on 3.19.0-rc3. v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (4): drm/i2c: tda998x: Add DT support for audio drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter drm/i2c: tda998x: set cts_n according to the sample width .../devicetree/bindings/drm/i2c/tda998x.txt | 18 ++ drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 213 +++++++++++++++++++-- include/sound/tda998x.h | 23 +++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 175 +++++++++++++++++ 7 files changed, 424 insertions(+), 12 deletions(-) create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/tda998x.c -- 2.1.4

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[alsa-devel] [PATCH v2 00/16] ASoC: samsung: Add clk provider for I2S internal clocks
by Sylwester Nawrocki 14 Jan '15

14 Jan '15

This series is an attempt to resolve the CDCLK clock gating issue on Odroid X2/U3 as reported by Daniel Drake [1], by exposing the CDCLK gate clock (and the two other clocks) through clk API. Changes since previous version: - removed check for the i2s_opclk1 mux input clock while creating the mux and div clocks, - the DT binding documentation changes reworked (addressing review comments), - added include/dt-bindings/sound/samsung-i2s.h header file defining the clk indices, it's been put into a separate patch together with the I2S DT binding documentation updates to make merging of the ASoC and the dts patches separately easier, - a patch fixing compatible strings of I2S1, I2S2 in exynos4.dtsi is included in this series. This whole series may need more testing on other SoCs, so far I only tested it on Odroid Exynos4412 X2, with the I2S working in slave mode. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081753.… Sylwester Nawrocki (16): ASoC: samsung: i2s: Remove unused gpios field from struct i2s ASoC: samsung: i2s: samsung_i2s_get_driver_data() cleanup ASoC: samsung: i2s: Add return value checks in probe() ASoC: samsung: i2s: Request memory region in driver probe() ASoC: samsung: i2s: Move clk_get() to platform driver probe() ASoC: samsung: i2s: Move clk enable to the platform driver probe() ASoC: samsung: i2s: Add get_other_dai helper function ASoC: samsung: i2s: Remove an unneeded goto usage ASoC: samsung: i2s: Add spinlock in place of local_irq_* calls ASoC: samsung: i2s: Protect more registers with a spinlock ASoC: samsung: odroidx2: Handle I2S CDCLK clock conditionally ASoC: samsung: i2s: Add clk provider DT binding documentation ASoC: samsung: i2s: Add clock provider for the I2S internal clocks ARM: dts: Exynos4 and Odroid X2/U3 sound device nodes update ARM: dts: Switch Odroid X2/U2 to simple-audio-card ARM: dts: Fix I2S1, I2S2 compatible for exynos4 SoCs .../devicetree/bindings/sound/samsung-i2s.txt | 22 ++ arch/arm/boot/dts/exynos4.dtsi | 13 +- arch/arm/boot/dts/exynos4412-odroid-common.dtsi | 27 +- arch/arm/boot/dts/exynos4412-odroidu3.dts | 8 +- arch/arm/boot/dts/exynos4412-odroidx2.dts | 8 +- include/dt-bindings/sound/samsung-i2s.h | 8 + sound/soc/samsung/i2s.c | 362 ++++++++++++-------- sound/soc/samsung/odroidx2_max98090.c | 6 +- 8 files changed, 295 insertions(+), 159 deletions(-) create mode 100644 include/dt-bindings/sound/samsung-i2s.h -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: snd-usb-usx2y: fix incorrect indentation on if statement
by Colin King 14 Jan '15

14 Jan '15

From: Colin Ian King <colin.king(a)canonical.com> Minor style issue, statement needs an extra tab indentation on the if statement. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/usx2y/usbusx2yaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 61d5dc2..dd40ca9 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -166,7 +166,7 @@ static int usX2Y_urb_play_prepare(struct snd_usX2Y_substream *subs, /* set the buffer pointer */ urb->transfer_buffer = runtime->dma_area + subs->hwptr * usX2Y->stride; if ((subs->hwptr += count) >= runtime->buffer_size) - subs->hwptr -= runtime->buffer_size; + subs->hwptr -= runtime->buffer_size; } else urb->transfer_buffer = subs->tmpbuf; -- 2.1.4

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[alsa-devel] [PATCH - UCM 1/2] control: enable octal and hexadecimal parse
by han.lu＠intel.com 14 Jan '15

14 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index 978977d..acaf734 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -59,10 +59,10 @@ static long get_integer(const char **ptr, long min, long max) goto out; s = p; - val = strtol(s, &p, 10); + val = strtol(s, &p, 0); if (*p == '.') { p++; - strtol(p, &p, 10); + strtol(p, &p, 0); } if (*p == '%') { val = (long)convert_prange1(strtod(s, NULL), min, max); @@ -87,10 +87,10 @@ static long long get_integer64(const char **ptr, long long min, long long max) goto out; s = p; - val = strtol(s, &p, 10); + val = strtol(s, &p, 0); if (*p == '.') { p++; - strtol(p, &p, 10); + strtol(p, &p, 0); } if (*p == '%') { val = (long long)convert_prange1(strtod(s, NULL), min, max); -- 2.1.0

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[alsa-devel] [PATCH - UCM 1/2] control: enable octal and hexadecimal parse
by han.lu＠intel.com 14 Jan '15

14 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> Use zero-base for strtol(), so get_integer() and get_integer64() can parse decimal, octal and hexadecimal data from input string. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index 978977d..8d6c385 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -59,7 +59,7 @@ static long get_integer(const char **ptr, long min, long max) goto out; s = p; - val = strtol(s, &p, 10); + val = strtol(s, &p, 0); if (*p == '.') { p++; strtol(p, &p, 10); @@ -87,7 +87,7 @@ static long long get_integer64(const char **ptr, long long min, long long max) goto out; s = p; - val = strtol(s, &p, 10); + val = strtol(s, &p, 0); if (*p == '.') { p++; strtol(p, &p, 10); -- 1.9.1

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[alsa-devel] Regression with .set_bias_level in pcm512x?
by Peter Rosin 14 Jan '15

14 Jan '15

Hi! I'm working on adding support for DAIFMT_CBM_CFM and DAIFMT_CBM_CFS to pcm512x, and it works nicely in the 3.10 kernel + Atmel-patches were I have developed the code. Now I'm trying to forward-port it and test it on 3.18 + other-Atmel-patches in preparation for submission. However, there seems to be a regression in that the (pre-existing) pcm512x_set_bias_level callback is never called in the 3.18 version. I see that there has been some core work on set_bias_level, and suspect it has something to do with dapm and components, but I think someone else will find the problem much quicker than I am able to. I have feebly tried to revert 0bd2ac3dae74ee25c5ea171cb572731c7a89c248 and 68f831c2724ab72c0088471b2ed1dc99e81948ef, but that didn't help. That also seems a bit backwards. So, what needs to be done in the pcm512x driver to have set_bias_level called? Please help. Cheers, Peter

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[alsa-devel] [PATCH v3] ALSA: ASoC: soc-compress.c: fix NULL dereference
by Qais Yousef 14 Jan '15

14 Jan '15

In soc_new_compress() when rtd->dai_link->daynamic is set, we create the pcm substreams with this call: ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, 1, 0, &be_pcm); which passes 0 as capture_count leading to be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream being NULL, hence when trying to set rtd a few lines below we get an oops. Fix by using rtd->dai_link->dpcm_playback and rtd->dai_link->dpcm_capture as playback_count and capture_count to snd_pcm_new_internal(). Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- v3->v2: - fix a typo s/dynmic/dynamic/ in commit message - use if else if because either dpcm_playback or dpcm_capture should be on for compressed devices v2->v1: - use better way to fix it than just removing the line that caused the oops sound/soc/soc-compress.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 590a82f01d0b..025c38fbe3c0 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, - 1, 0, &be_pcm); + rtd->dai_link->dpcm_playback, + rtd->dai_link->dpcm_capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n", rtd->dai_link->name); @@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = be_pcm; rtd->fe_compr = 1; - be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + if (rtd->dai_link->dpcm_playback) + be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + else if (rtd->dai_link->dpcm_capture) + be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); -- 2.1.0

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[alsa-devel] [PATCH v2] ALSA: ASoC: soc-compress.c: fix NULL dereference
by Qais Yousef 14 Jan '15

14 Jan '15

In soc_new_compress() when rtd->dai_link->daynmic is set, we create the pcm substreams with this call: ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, 1, 0, &be_pcm); which passes 0 as capture_count leading to be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream being NULL, hence when trying to set rtd a few lines below we get an oops. Fix by using rtd->dai_link->dpcm_playback and rtd->dai_link->dpcm_capture as playback_count and capture_count to snd_pcm_new_internal(). Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- v2->v1: - use better way to fix it than just removing the line that caused the oops sound/soc/soc-compress.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 590a82f01d0b..27a668463ad7 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, - 1, 0, &be_pcm); + rtd->dai_link->dpcm_playback, + rtd->dai_link->dpcm_capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n", rtd->dai_link->name); @@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = be_pcm; rtd->fe_compr = 1; - be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + if (rtd->dai_link->dpcm_playback) + be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + if (rtd->dai_link->dpcm_capture) + be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); -- 2.1.0

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[alsa-devel] "ak" vs "asahi-kasei" (was: Re: [PATCH 3/3 v3] ARM: shmobile: lager: add sound support on DT)
by Geert Uytterhoeven 13 Jan '15

13 Jan '15

Hi Morimoto-san, CC devicetree, alsa-devel On Wed, Jun 11, 2014 at 9:40 AM, Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> wrote: >> On Wed, Jun 11, 2014 at 8:54 AM, Kuninori Morimoto >> <kuninori.morimoto.gx(a)renesas.com> wrote: >> > + compatible = "asahi-kasei,ak4643"; >> >> As Sergei already commented, >> Documentation/devicetree/bindings/vendor-prefixes.txt >> says >> >> ak Asahi Kasei Corp. > > I know, but, > > grep asahi-kasei sound/soc/codecs/* > sound/soc/codecs/ak4104.c: { .compatible = "asahi-kasei,ak4104", }, > sound/soc/codecs/ak4554.c: { .compatible = "asahi-kasei,ak4554" }, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata}, > sound/soc/codecs/ak5386.c: { .compatible = "asahi-kasei,ak5386", }, > > ak4642 driver can't probe by "ak"; That's bad. And all but one of these match entries have been added _after_ the addition of "ak" to vendor-prefixes.txt... Outside sound, there's also "asahi-kasei,ak8975" vs. "ak,ak8975". > My code is based on "code", not "document" Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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Re: [alsa-devel] randconfig build error with next-20150113, in sound/pci/hda/patch_realtek.c
by Takashi Iwai 13 Jan '15

13 Jan '15

At Tue, 13 Jan 2015 11:39:12 -0700, Jim Davis wrote: > > Building with the attached random configuration file, > > sound/built-in.o: In function `alc280_fixup_hp_gpio2_mic_hotkey': > /home/jim/linux/sound/pci/hda/patch_realtek.c:3512: undefined > reference to `input_allocate_device' > /home/jim/linux/sound/pci/hda/patch_realtek.c:3520: undefined > reference to `input_register_device' > /home/jim/linux/sound/pci/hda/patch_realtek.c:3522: undefined > reference to `input_free_device' > /home/jim/linux/sound/pci/hda/patch_realtek.c:3550: undefined > reference to `input_unregister_device' > sound/built-in.o: In function `gpio2_mic_hotkey_event': > /home/jim/linux/include/linux/input.h:392: undefined reference to `input_event' > /home/jim/linux/include/linux/input.h:417: undefined reference to `input_event' > /home/jim/linux/include/linux/input.h:392: undefined reference to `input_event' > /home/jim/linux/include/linux/input.h:417: undefined reference to `input_event' > make: *** [vmlinux] Error 1 Thanks. So this is the case where the audio driver is built-in while the input driver is module. I guess the cleanest workaround is just to select INPUT from SND_HDA_CODEC_REALTEK in sound/pci/hda/Kconfig. Practically seen, CONFIG_INPUT is always set, so there must be no real drawback with it. (And we can reduce a few lines as a bonus.) David, care to prepare a patch? thanks, Takashi

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[alsa-devel] [PATCH] pyalsa: alsaseq: fix memory leaks
by Hector Martin 13 Jan '15

13 Jan '15

Fix a bunch of dangling reference issues. Also switch to Py_BuildValue where appropriate, since that makes it easier to build complex dictionary/tuple structures without leaking integer/string references. Signed-off-by: Hector Martin <marcan(a)marcan.st> --- I haven't looked at the other parts of pyalsa; those could probably use some looking over too. I went through alsaseq since I'm writing a long-running MIDI client and I noticed it was leaking all event objects. pyalsa/alsaseq.c | 156 +++++++++++++++++++++---------------------------------- 1 file changed, 58 insertions(+), 98 deletions(-) diff --git a/pyalsa/alsaseq.c b/pyalsa/alsaseq.c index e34d3b4..ce332ed 100644 --- a/pyalsa/alsaseq.c +++ b/pyalsa/alsaseq.c @@ -124,7 +124,9 @@ if (PyModule_AddObject(module, name, tmp) < 0) { \ return; \ } \ - PyDict_SetItem(TDICT(subtype), PyInt_FromLong(value), tmp); \ + PyObject *key = PyInt_FromLong(value); \ + PyDict_SetItem(TDICT(subtype), key, tmp); \ + Py_DECREF(key); \ } #define TCONSTRETURN(subtype, value) { \ @@ -199,8 +201,11 @@ PyDict_SetItemString(dict, name, object) /* sets a integer into the dict */ -#define SETDICTINT(name, value) \ - PyDict_SetItemString(dict, name, PyInt_FromLong(value)) +#define SETDICTINT(name, value) { \ + PyObject *val = PyInt_FromLong(value); \ + PyDict_SetItemString(dict, name, val); \ + Py_DECREF(val); \ + } /* sets note info dict (used by SeqEvent_get_data) */ #define SETDICT_NOTE3 { \ @@ -288,6 +293,7 @@ PyList_SetItem(list, i, PyInt_FromLong(t[i])); \ } \ SETDICTOBJ("ext", list); \ + Py_DECREF(list); \ } /* gets integer from python param */ @@ -1767,7 +1773,7 @@ SeqEvent_repr(SeqEventObject *self) { return PyString_FromFormat("<alsaseq.SeqEvent type=%s(%d) flags=%d tag=%d " "queue=%d time=%s(%u.%u) from=%d:%d to=%d:%d " - "at 0x%p>", + "at %p>", typestr, self->event->type, self->event->flags, self->event->tag, self->event->queue, @@ -2367,34 +2373,23 @@ _query_connections_list(snd_seq_t *handle, PyObject *list = PyList_New(0); int index = 0; - long tmplong; snd_seq_query_subscribe_set_type(query, type); snd_seq_query_subscribe_set_index(query, index); while (snd_seq_query_port_subscribers(handle, query) >= 0) { const snd_seq_addr_t *addr = snd_seq_query_subscribe_get_addr(query); - PyObject *dict = PyDict_New(); - - tmplong = snd_seq_query_subscribe_get_queue(query); - PyDict_SetItemString(dict, "queue", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_query_subscribe_get_exclusive(query); - PyDict_SetItemString(dict, "exclusive", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_query_subscribe_get_time_update(query); - PyDict_SetItemString(dict, "time_update", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_query_subscribe_get_time_real(query); - PyDict_SetItemString(dict, "time_real", PyInt_FromLong(tmplong)); - - - PyObject *tuple = PyTuple_New(3); - PyTuple_SetItem(tuple, 0, PyInt_FromLong(addr->client)); - PyTuple_SetItem(tuple, 1, PyInt_FromLong(addr->port)); - PyTuple_SetItem(tuple, 2, dict); + PyObject *tuple = Py_BuildValue( + "(ii{sisisisi})", + (int)addr->client, + (int)addr->port, + "queue", (int)snd_seq_query_subscribe_get_queue(query), + "exclusive", (int)snd_seq_query_subscribe_get_exclusive(query), + "time_update", (int)snd_seq_query_subscribe_get_time_update(query), + "time_real", (int)snd_seq_query_subscribe_get_time_real(query)); PyList_Append(list, tuple); + Py_DECREF(tuple); snd_seq_query_subscribe_set_index(query, ++index); } return list; @@ -2466,11 +2461,13 @@ Sequencer_connection_list(SequencerObject *self, PyTuple_SetItem(porttuple, 2, conntuple); PyList_Append(portlist, porttuple); + Py_DECREF(porttuple); } PyTuple_SetItem(tuple, 2, portlist); /* append list of port tuples */ PyList_Append(list, tuple); + Py_DECREF(tuple); } return list; @@ -2506,9 +2503,6 @@ Sequencer_get_client_info(SequencerObject *self, snd_seq_client_info_t *cinfo; int client_id = -1; int ret; - PyObject *tmpobj; - long tmplong; - const char * tmpchar; char *kwlist[] = { "client_id", NULL}; if (!PyArg_ParseTupleAndKeywords(args, kwds, "|i", kwlist, @@ -2532,37 +2526,21 @@ Sequencer_get_client_info(SequencerObject *self, } } - PyObject *dict = PyDict_New(); - if (dict == NULL) { - return NULL; - } - - TCONSTASSIGN(ADDR_CLIENT, client_id, tmpobj); - PyDict_SetItemString(dict, "id", tmpobj); - - tmplong = snd_seq_client_info_get_type(cinfo); - TCONSTASSIGN(CLIENT_TYPE, tmplong, tmpobj); - PyDict_SetItemString(dict, "type", tmpobj); - - tmpchar = snd_seq_client_info_get_name(cinfo); - tmpchar = (tmpchar == NULL ? "" : tmpchar); - PyDict_SetItemString(dict, "name", PyString_FromString(tmpchar)); - - tmplong = snd_seq_client_info_get_broadcast_filter(cinfo); - PyDict_SetItemString(dict, "broadcast_filter", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_client_info_get_error_bounce(cinfo); - PyDict_SetItemString(dict, "error_bounce", PyInt_FromLong(tmplong)); - - tmpchar = (const char *)snd_seq_client_info_get_event_filter(cinfo); - tmpchar = (tmpchar == NULL ? "" : tmpchar); - PyDict_SetItemString(dict, "event_filter", PyString_FromString(tmpchar)); + PyObject *d_id, *d_type; + TCONSTASSIGN(ADDR_CLIENT, client_id, d_id); + TCONSTASSIGN(CLIENT_TYPE, snd_seq_client_info_get_type(cinfo), d_type); + const char *d_name = snd_seq_client_info_get_name(cinfo); - tmplong = snd_seq_client_info_get_num_ports(cinfo); - PyDict_SetItemString(dict, "num_ports", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_client_info_get_event_lost(cinfo); - PyDict_SetItemString(dict, "event_lost", PyInt_FromLong(tmplong)); + PyObject *dict = Py_BuildValue( + "{sNsNsssisiss#sisi}", + "id", d_id, + "type", d_type, + "name", d_name == NULL ? "" : d_name, + "broadcast_filter", (int)snd_seq_client_info_get_broadcast_filter(cinfo), + "error_bounce", (int)snd_seq_client_info_get_error_bounce(cinfo), + "event_filter", snd_seq_client_info_get_event_filter(cinfo), 32, + "num_ports", (int)snd_seq_client_info_get_num_ports(cinfo), + "event_lost", (int)snd_seq_client_info_get_event_lost(cinfo)); return dict; } @@ -2594,8 +2572,6 @@ Sequencer_get_port_info(SequencerObject *self, snd_seq_client_info_t *cinfo; int port_id; int client_id; - const char *tmpchar; - long tmplong; int ret; char *kwlist[] = { "port_id", "client_id", NULL }; @@ -2612,11 +2588,6 @@ Sequencer_get_port_info(SequencerObject *self, return NULL; } - PyObject *dict = PyDict_New(); - if (dict == NULL) { - return NULL; - } - snd_seq_port_info_alloca(&pinfo); ret = snd_seq_get_any_port_info(self->handle, client_id, port_id, pinfo); @@ -2625,17 +2596,11 @@ Sequencer_get_port_info(SequencerObject *self, return NULL; } - tmpchar = snd_seq_port_info_get_name(pinfo); - tmpchar = (tmpchar == NULL ? "" : tmpchar); - PyDict_SetItemString(dict, "name", PyString_FromString(tmpchar)); - - tmplong = snd_seq_port_info_get_capability(pinfo); - PyDict_SetItemString(dict, "capability", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_port_info_get_type(pinfo); - PyDict_SetItemString(dict, "type", PyInt_FromLong(tmplong)); - - return dict; + return Py_BuildValue( + "{sssIsI}", + "name", snd_seq_port_info_get_name(pinfo), + "capability", (unsigned int)snd_seq_port_info_get_capability(pinfo), + "type", (unsigned int)snd_seq_port_info_get_type(pinfo)); } /** alsaseq.Sequencer connect_ports() method: __doc__ */ @@ -2768,7 +2733,6 @@ Sequencer_get_connect_info(SequencerObject *self, snd_seq_addr_t sender, dest; snd_seq_port_subscribe_t *sinfo; int ret; - long tmplong; if (!PyArg_ParseTuple(args, "(BB)(BB)", &(sender.client), &(sender.port), &(dest.client), &(dest.port))) { @@ -2786,21 +2750,12 @@ Sequencer_get_connect_info(SequencerObject *self, return NULL; } - PyObject *dict = PyDict_New(); - - tmplong = snd_seq_port_subscribe_get_queue(sinfo); - PyDict_SetItemString(dict, "queue", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_port_subscribe_get_exclusive(sinfo); - PyDict_SetItemString(dict, "exclusive", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_port_subscribe_get_time_update(sinfo); - PyDict_SetItemString(dict, "time_update", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_port_subscribe_get_time_real(sinfo); - PyDict_SetItemString(dict, "time_real", PyInt_FromLong(tmplong)); - - return dict; + return Py_BuildValue( + "{sisisisi}", + "queue", (int)snd_seq_port_subscribe_get_queue(sinfo), + "exclusive", (int)snd_seq_port_subscribe_get_exclusive(sinfo), + "time_update", (int)snd_seq_port_subscribe_get_time_update(sinfo), + "time_real", (int)snd_seq_port_subscribe_get_time_real(sinfo)); } /** alsaseq.Sequencer receive_events() method: __doc__ */ @@ -2878,6 +2833,7 @@ Sequencer_receive_events(SequencerObject *self, } PyList_Append(list, SeqEventObject); + Py_DECREF(SeqEventObject); maxevents --; @@ -3263,19 +3219,23 @@ Sequencer_registerpoll(SequencerObject *self, PyObject *args, PyObject *kwds) if (count <= 0) Py_RETURN_NONE; - reg = PyObject_GetAttr(pollObj, PyString_InternFromString("register")); + reg = PyObject_GetAttrString(pollObj, "register"); + if (!reg) + return NULL; for (i = 0; i < count; i++) { t = PyTuple_New(2); - if (t) { - PyTuple_SET_ITEM(t, 0, PyInt_FromLong(pfd[i].fd)); - PyTuple_SET_ITEM(t, 1, PyInt_FromLong(pfd[i].events)); - Py_XDECREF(PyObject_CallObject(reg, t)); - Py_DECREF(t); + if (!t) { + Py_DECREF(reg); + return NULL; } + PyTuple_SET_ITEM(t, 0, PyInt_FromLong(pfd[i].fd)); + PyTuple_SET_ITEM(t, 1, PyInt_FromLong(pfd[i].events)); + Py_XDECREF(PyObject_CallObject(reg, t)); + Py_DECREF(t); } - Py_XDECREF(reg); + Py_DECREF(reg); Py_RETURN_NONE; } -- 2.2.1

2 1

[alsa-devel] [PATCH] hdajackretask: Add dock hp/mic/line to simple options
by David Henningsson 13 Jan '15

13 Jan '15

This allows for simpler setting of dock headphone, dock mic, dock line out and dock line in. Also improve detection a bit so that an existing dock headphone shows up as such and not as regular headphone. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hdajackretask/sysfs-pin-configs.c | 32 ++++++++++++++++++++------------ 1 file changed, 20 insertions(+), 12 deletions(-) diff --git a/hdajackretask/sysfs-pin-configs.c b/hdajackretask/sysfs-pin-configs.c index 592ffe2..693842f 100644 --- a/hdajackretask/sysfs-pin-configs.c +++ b/hdajackretask/sysfs-pin-configs.c @@ -289,6 +289,8 @@ unsigned long actual_pin_config(pin_configs_t* pins) #define AC_DEFCFG_DEVICE_SHIFT 20 #define AC_DEFCFG_LOCATION (0x3f<<24) #define AC_DEFCFG_LOCATION_SHIFT 24 +#define AC_DEFCFG_GROSSLOC (0x3<<28) +#define AC_DEFCFG_GROSSLOC_SHIFT 28 #define AC_DEFCFG_PORT_CONN (0x3<<30) #define AC_DEFCFG_PORT_CONN_SHIFT 30 @@ -594,21 +596,27 @@ static gboolean disabled_caps(unsigned long pin_caps, unsigned long wid_caps) return TRUE; } +#define DEFAULT_MASK (AC_DEFCFG_PORT_CONN | AC_DEFCFG_GROSSLOC | AC_DEFCFG_DEVICE) + static typical_pins_t simple_typical_pins[] = { - {"Headphone", headphone_caps, 0x0321403f, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE, }, - {"Microphone", extmic_caps, 0x03a19020, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, - {"Line out (Front)", lineout_caps, 0x01014410, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE,}, - {"Line out (Center/LFE)", lineout_caps, 0x01014411, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE,}, - {"Line out (Back)", lineout_caps, 0x01014412, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE,}, - {"Line out (Side)", lineout_caps, 0x01014413, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE,}, - {"Line in", linein_caps, 0x0181344f, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, - {"Internal speaker", lineout_caps, 0x90170150, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE, }, - {"Internal speaker (LFE)", lineout_caps, 0x90170151, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE, }, - {"Internal speaker (Back)", lineout_caps, 0x90170152, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE, }, - {"Internal mic", intmic_caps, 0x90a60160, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, - {"HDMI", hdmi_caps, 0x18560070, AC_DEFCFG_PORT_CONN | AC_DEFCFG_LOCATION,}, + {"Headphone", headphone_caps, 0x0321403f, DEFAULT_MASK, }, + {"Microphone", extmic_caps, 0x03a19020, DEFAULT_MASK,}, + {"Line out (Front)", lineout_caps, 0x01014010, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Line out (Center/LFE)", lineout_caps, 0x01014011, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Line out (Back)", lineout_caps, 0x01014012, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Line out (Side)", lineout_caps, 0x01014013, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Line in", linein_caps, 0x0181304f, DEFAULT_MASK,}, + {"Internal speaker", lineout_caps, 0x90170150, DEFAULT_MASK | AC_DEFCFG_SEQUENCE, }, + {"Internal speaker (LFE)", lineout_caps, 0x90170151, DEFAULT_MASK | AC_DEFCFG_SEQUENCE, }, + {"Internal speaker (Back)", lineout_caps, 0x90170152, DEFAULT_MASK | AC_DEFCFG_SEQUENCE, }, + {"Internal mic", intmic_caps, 0x90a60160, DEFAULT_MASK,}, + {"HDMI / DisplayPort", hdmi_caps, 0x18560070, AC_DEFCFG_PORT_CONN | AC_DEFCFG_LOCATION,}, {"SPDIF out", spdifout_caps, 0x014b1180, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, {"SPDIF in", spdifin_caps, 0x01cb6190, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, + {"Dock Headphone", headphone_caps, 0x222140af, DEFAULT_MASK, }, + {"Dock Microphone", extmic_caps, 0x22a190a0, DEFAULT_MASK,}, + {"Dock Line out", lineout_caps, 0x220140b0, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Dock Line in", linein_caps, 0x228130bf, DEFAULT_MASK,}, {"Not connected", disabled_caps, 0x40f000f0, AC_DEFCFG_PORT_CONN,}, {} }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: compress_driver.h: include sound/core.h explicitly
by Qais Yousef 13 Jan '15

13 Jan '15

Fixes the following compilation error: include/sound/compress_driver.h: In function ‘snd_compr_drain_notify’: include/sound/compress_driver.h:177:2: error: implicit declaration of function ‘snd_BUG_ON’ [-Werror=implicit-function-declaration] if (snd_BUG_ON(!stream)) snd_BUG_ON() is defined in sound/core.h but the file is not included explicitly, so include it. Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- include/sound/compress_driver.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 396e8f73670a..1e2531058b7e 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -27,6 +27,7 @@ #include <linux/types.h> #include <linux/sched.h> +#include <sound/core.h> #include <sound/compress_offload.h> #include <sound/asound.h> #include <sound/pcm.h> -- 2.1.0

3 3

[alsa-devel] Configuring slave output for ALC889
by Dheeraj CVR 13 Jan '15

13 Jan '15

Hi, I am using an Intel Desktop Board DX58SO. I am using S/PDIF for HDMI Audio and there is no sound output via HDMI. I have checked with alsamixer and the volumes for the S/PDIF are stuck at 0 and I can't increase them. By looking at the commit history, I have found this commit which enables 0x10 as a slave output for 0x6 on ALC889 for the exact same board that I am using. http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/015901.html However, it appears that with the commit "ALSA: hda/realtek - Drop ALC882 desktop model quirks", the older workaround has been removed, which caused a regression. I have checked the output of alsainfo and it seems that 0x06 is still being used instead of 0x010. Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Control: name="SPDIF Phantom Jack", index=1, device=0 Pincap 0x00000010: OUT Pin Default 0x99430130: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 I was planning to edit the "patch_alc882" function to configure the "spec->gen.slave_dig_outs" and set the values to "{0x10, 0, 0}". However, I was not sure if this was the right way of doing. Please advice. I am running Arch-Linux with 3.17 kernel. -- Regards, Dheeraj CVR.

1 0

[alsa-devel] [PATCH v2] ASoC: wm8904: fix runtime warning
by Bo Shen 13 Jan '15

13 Jan '15

Correct the path name for mux to get rid of the following warning: --->8--- wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTL wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTL wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTR wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTR wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACL wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACL wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACR wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACR ---8<--- Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v2: - Correct the mux path name. - Route the mux to the pins. sound/soc/codecs/wm8904.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..110d0e6 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1076,10 +1076,13 @@ static const struct snd_soc_dapm_route adc_intercon[] = { { "Right Capture PGA", NULL, "Right Capture Mux" }, { "Right Capture PGA", NULL, "Right Capture Inverting Mux" }, - { "AIFOUTL", "Left", "ADCL" }, - { "AIFOUTL", "Right", "ADCR" }, - { "AIFOUTR", "Left", "ADCL" }, - { "AIFOUTR", "Right", "ADCR" }, + { "AIFOUTL Mux", "Left", "ADCL" }, + { "AIFOUTL Mux", "Right", "ADCR" }, + { "AIFOUTR Mux", "Left", "ADCL" }, + { "AIFOUTR Mux", "Right", "ADCR" }, + + { "AIFOUTL", NULL, "AIFOUTL Mux" }, + { "AIFOUTR", NULL, "AIFOUTR Mux" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCL", NULL, "Left Capture PGA" }, @@ -1089,14 +1092,20 @@ static const struct snd_soc_dapm_route adc_intercon[] = { }; static const struct snd_soc_dapm_route dac_intercon[] = { - { "DACL", "Right", "AIFINR" }, - { "DACL", "Left", "AIFINL" }, + { "DACL", NULL, "AIFINL" }, + { "DACL", NULL, "AIFINR" }, { "DACL", NULL, "CLK_DSP" }, - { "DACR", "Right", "AIFINR" }, - { "DACR", "Left", "AIFINL" }, + { "DACR", NULL, "AIFINL" }, + { "DACR", NULL, "AIFINR" }, { "DACR", NULL, "CLK_DSP" }, + { "DACL Mux", "Left", "DACL" }, + { "DACL Mux", "Right", "DACR" }, + + { "DACR Mux", "Left", "DACL" }, + { "DACR Mux", "Right", "DACR" }, + { "Charge pump", NULL, "SYSCLK" }, { "Headphone Output", NULL, "HPL PGA" }, -- 2.1.0.24.g4109c28

2 1

[alsa-devel] [PATCH] ASoC: wm8904: fix runtime warning
by Bo Shen 13 Jan '15

13 Jan '15

As the commit 5fe5b76 (ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets) remove the fake control. Then, the wm8904 throw out the following warning at runtime. --->8--- wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTL wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTL wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTR wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTR wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACL wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACL wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACR wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACR ---8<--- This patch is used to remove these warning. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..38582d7 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1076,10 +1076,10 @@ static const struct snd_soc_dapm_route adc_intercon[] = { { "Right Capture PGA", NULL, "Right Capture Mux" }, { "Right Capture PGA", NULL, "Right Capture Inverting Mux" }, - { "AIFOUTL", "Left", "ADCL" }, - { "AIFOUTL", "Right", "ADCR" }, - { "AIFOUTR", "Left", "ADCL" }, - { "AIFOUTR", "Right", "ADCR" }, + { "AIFOUTL", NULL, "ADCL" }, + { "AIFOUTL", NULL, "ADCR" }, + { "AIFOUTR", NULL, "ADCL" }, + { "AIFOUTR", NULL, "ADCR" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCL", NULL, "Left Capture PGA" }, @@ -1089,12 +1089,12 @@ static const struct snd_soc_dapm_route adc_intercon[] = { }; static const struct snd_soc_dapm_route dac_intercon[] = { - { "DACL", "Right", "AIFINR" }, - { "DACL", "Left", "AIFINL" }, + { "DACL", NULL, "AIFINR" }, + { "DACL", NULL, "AIFINL" }, { "DACL", NULL, "CLK_DSP" }, - { "DACR", "Right", "AIFINR" }, - { "DACR", "Left", "AIFINL" }, + { "DACR", NULL, "AIFINR" }, + { "DACR", NULL, "AIFINL" }, { "DACR", NULL, "CLK_DSP" }, { "Charge pump", NULL, "SYSCLK" }, -- 2.1.0.24.g4109c28

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[alsa-devel] [PULL] topic/i915-hda-componentized
by Daniel Vetter 13 Jan '15

13 Jan '15

Hi Takashi, So here's the stable tag with Imre's i915/hda componentized refactoring, based upon 3.19-rc4. I'll pull in the same tag into drm-intel-next. Cheers, Daniel The following changes since commit eaa27f34e91a14cdceed26ed6c6793ec1d186115: linux 3.19-rc4 (2015-01-11 12:44:53 -0800) are available in the git repository at: git://anongit.freedesktop.org/drm-intel tags/topic/i915-hda-componentized-2015-01-12 for you to fetch changes up to fcf3aac5fc307f0cae429f5844ddc25761662858: drm/i915: remove unused power_well/get_cdclk_freq api (2015-01-12 02:48:24 +0100) ---------------------------------------------------------------- Imre Deak (6): drm/i915: add dev_to_i915 helper drm/i915: add component support ALSA: hda: export struct hda_intel ALSA: hda: pass intel_hda to all i915 interface functions ALSA: hda: add component support drm/i915: remove unused power_well/get_cdclk_freq api drivers/gpu/drm/i915/i915_dma.c | 4 + drivers/gpu/drm/i915/i915_drv.c | 9 +- drivers/gpu/drm/i915/i915_drv.h | 8 ++ drivers/gpu/drm/i915/intel_audio.c | 110 +++++++++++++++++++++++ drivers/gpu/drm/i915/intel_drv.h | 2 + drivers/gpu/drm/i915/intel_runtime_pm.c | 56 ------------ include/drm/i915_component.h | 38 ++++++++ include/drm/i915_powerwell.h | 37 -------- sound/pci/hda/hda_i915.c | 154 ++++++++++++++++++++++---------- sound/pci/hda/hda_i915.h | 37 -------- sound/pci/hda/hda_intel.c | 60 ++++--------- sound/pci/hda/hda_intel.h | 71 +++++++++++++++ 12 files changed, 361 insertions(+), 225 deletions(-) create mode 100644 include/drm/i915_component.h delete mode 100644 include/drm/i915_powerwell.h delete mode 100644 sound/pci/hda/hda_i915.h create mode 100644 sound/pci/hda/hda_intel.h -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

2 1

[alsa-devel] [PATCH - UCM 1/2] control: add hexadecimal parse
by han.lu＠intel.com 13 Jan '15

13 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index 978977d..92e2f95 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -59,7 +59,10 @@ static long get_integer(const char **ptr, long min, long max) goto out; s = p; - val = strtol(s, &p, 10); + if (*p == '0' && (*(p+1) == 'x' || *(p+1) == 'X')) + val = strtol(s, &p, 16); + else + val = strtol(s, &p, 10); if (*p == '.') { p++; strtol(p, &p, 10); @@ -87,7 +90,10 @@ static long long get_integer64(const char **ptr, long long min, long long max) goto out; s = p; - val = strtol(s, &p, 10); + if (*p == '0' && (*(p+1) == 'x' || *(p+1) == 'X')) + val = strtol(s, &p, 16); + else + val = strtol(s, &p, 10); if (*p == '.') { p++; strtol(p, &p, 10); -- 1.9.1

3 5

[alsa-devel] Question on HDA Audio - resume issue
by N, Harshapriya 13 Jan '15

13 Jan '15

Hi, On running a suspend/resume stress test, after a few hundred runs, I see that resume takes a longer time (~40 sec) because of HDA audio (00:03 is the audio device in the system) . HDA audio takes around 35 sec to resume in the following case. dmesg: >>>2015-01-11T05:43:39.993107+08:00 kernel: [46300.269994] call 0000:00:03.0+ returned 0 after 35524554 usecs >>>2015-01-11T05:43:39.993109+08:00 kernel: [46300.270033] PM: Wakeup pending, aborting suspend Can anyone help understand why would HDA audio take so long to resume sporadically? -Harsha

1 0

[alsa-devel] [PATCH 0/3] Sound support for Exynos4412 Trats2 board
by Inha Song 13 Jan '15

13 Jan '15

This patch-set adds basic sound support for the Trats2 boards. It just support primary I2s and external speaker playback. Inha Song (3): ASoC: samsung: Add machine driver for Trats2 ASoC: samsung: Document Trats2 audio subsystem bindings ARM: dts: Add sound nodes for exynos4412-trats2 .../bindings/sound/samsung,trats2-wm1811.txt | 31 +++ arch/arm/boot/dts/exynos4412-trats2.dts | 38 ++++ sound/soc/samsung/Kconfig | 8 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/trats2_wm1811.c | 216 +++++++++++++++++++++ 5 files changed, 295 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,trats2-wm1811.txt create mode 100644 sound/soc/samsung/trats2_wm1811.c -- 2.0.0.390.gcb682f8

3 7

[alsa-devel] [PATCH] sound: firewire: fireworks: fireworks_command: Remove unused function
by Rickard Strandqvist 13 Jan '15

13 Jan '15

Remove the function snd_efw_command_set_resp_addr() that is not used anywhere. This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/firewire/fireworks/fireworks.h | 2 -- sound/firewire/fireworks/fireworks_command.c | 21 --------------------- 2 files changed, 23 deletions(-) diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 4f0201a..693b76b 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -189,8 +189,6 @@ enum snd_efw_transport_mode { SND_EFW_TRANSPORT_MODE_WINDOWS = 0, SND_EFW_TRANSPORT_MODE_IEC61883 = 1, }; -int snd_efw_command_set_resp_addr(struct snd_efw *efw, - u16 addr_high, u32 addr_low); int snd_efw_command_set_tx_mode(struct snd_efw *efw, enum snd_efw_transport_mode mode); int snd_efw_command_get_hwinfo(struct snd_efw *efw, diff --git a/sound/firewire/fireworks/fireworks_command.c b/sound/firewire/fireworks/fireworks_command.c index 166f805..e0cc467 100644 --- a/sound/firewire/fireworks/fireworks_command.c +++ b/sound/firewire/fireworks/fireworks_command.c @@ -175,27 +175,6 @@ end: } /* - * The address in host system for transaction response is changable when the - * device supports. struct hwinfo.flags includes its flag. The default is - * MEMORY_SPACE_EFW_RESPONSE. - */ -int snd_efw_command_set_resp_addr(struct snd_efw *efw, - u16 addr_high, u32 addr_low) -{ - __be32 addr[2]; - - addr[0] = cpu_to_be32(addr_high); - addr[1] = cpu_to_be32(addr_low); - - if (!efw->resp_addr_changable) - return -ENOSYS; - - return efw_transaction(efw, EFC_CAT_HWCTL, - EFC_CMD_HWINFO_SET_RESP_ADDR, - addr, sizeof(addr), NULL, 0); -} - -/* * This is for timestamp processing. In Windows mode, all 32bit fields of second * CIP header in AMDTP transmit packet is used for 'presentation timestamp'. In * 'no data' packet the value of this field is 0x90ffffff. -- 1.7.10.4

3 2

Re: [alsa-devel] Card name size?
by Takashi Iwai 12 Jan '15

12 Jan '15

At Sun, 11 Jan 2015 20:52:00 +0000, Alan Horstmann wrote: > > Hi, > > Just a quick question... > is there a (guaranteed) max size to the name string returned by > snd_ctl_card_info_get_name() > or is it just whatever the driver sets? It's max 32 letters (including NUL). longname and mixername are max 80, id and driver are max 16. Takashi

2 1

Re: [alsa-devel] [PATCH 00/25] line6usb cleanup
by Takashi Iwai 12 Jan '15

12 Jan '15

At Mon, 12 Jan 2015 17:35:01 +0100, Takashi Iwai wrote: > > At Sun, 11 Jan 2015 15:04:55 -0600, > Chris Rorvick wrote: > > > > > At Fri, 9 Jan 2015 23:35:46 -0600, > > > Chris Rorvick wrote: > > >> > > >> I have a TonePort UX2 that I've used for testing, meaning that some of > > >> this is really only compile-tested. > > > > > > If anyone is responsible for testing with real hardware, I'll happily > > > review. > > > > To be clear, the TonePort UX2 is real hardware. But this driver > > basically supports four classes of Line 6 devices and I'm only covering > > one of them. > > > > So this series is a first step in trying to address this. Having this > > as a single driver probably made sense when it was a separate project, > > but now that it is in-tree it seems like the POD, PODHD, TonePort, and > > Variax pieces should each be separate drivers that each depend on a core > > Line 6 driver. I think the cleanup in this series will make that > > easier. None of this is my area of expertise, though, so advice and > > feedback is very welcome. > > > > > are there any active developers for this driver? > > > > I intended to do further work. I know there is quite a bit of mundane > > checkpatch cleanup that would need to get done before this could get > > promoted, and I believe I read that it's using sysfs for stuff that > > would normally be done via an ALSA interface, and the sysfs interface > > has not been documented nor has it been justified. All stuff I thought > > I might look into. > > > > But I'm just doing this for fun so I can't promise anything. :-) > > OK, so the situation looks fairly good, we have a few active > developers and/or testers. And the current code doesn't look so > terrible despite of it being in staging directory. That said, I think > we can promote this stuff into sound/usb/line6 directory, then apply > Chris' cleanup patches, and work on it further. > > Does it sound OK for you guys? Greg? > > Once when I get approval, I'll start a new clean branch on sound.git > tree so that you guys can work on it further for 3.20 kernel. JFYI, I merged all patches Greg forwarded to me into topic/line6 branch of sound git tree. It's included to linux-next, so you can work on linux-next or my branch as well. Now the code has been moved from driver/staging/line6 to sound/usb/line6. I'm going to apply some driver rename and cleanup patches to follow to other ALSA driver standards, and think of splitting to each driver with a core line6 helper module after that. Feel free to send the further development patches to me and/or alsa-devel ML like other sound drivers. thanks, Takashi

1 0

Re: [alsa-devel] Possible regression in patch_realtek.c
by David Henningsson 12 Jan '15

12 Jan '15

On 2015-01-11 22:11, Thanos Kyritsis wrote: > Hello, > > I hope it's ok to e-mail you directly instead of going through an > official bug tracker, sorry in advance otherwise. Ok. I have added alsa-devel to CC. > I'd like your opinion on a patch of yours that dates a year back: > > https://git.kernel.org/cgit/linux/kernel/git/stable/linux-stable.git/commit… > https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1248116 > > This patch has a negative impact for my hardware. I've completely lost > the kernel's front panel jack auto-mute functionality. > > In detail: my motherboard is Gigabyte Z68AP-D3, utilizing ALC889 > chipset (http://www.gigabyte.com/products/product-page.aspx?pid=3897) > with PCI id: 0x1458, 0xa002. > > Without this patch (that is kernel up to 3.13.x), all is OK, when I > plug headphones via the front panel jack, the kernel auto-mutes the > speaker-phone jack. > With the patch (that is kernel 3.14.x and all later versions), > auto-mute does not work. Plugging headphones results on sound getting > through both jacks simultaneously. > I can, of course, control the volume/mute of these two jacks > independently via CLI alsamixer, but I think this is a regression. Do > you agree ? I'm noticing that the patch's purpose was to fix a > slightly different problem, though. > Is it possible to satisfy both requirements for both motherboards ? > > Can I do something to further help ? If you'd like me to comment on > bug #1248116, that will be no problem of course. > > Thank you in advance for your help and for your contribution towards a > better kernel and distros :-) Okay. So the real question is if there's a way we can tell the working and the faulty ones apart. Until we have found such a way, I'll be hesitant to revert the patch. Here's alsa-info for a faulty one: http://www.alsa-project.org/db/?f=435c404946687b445f6c60da5edc5c3f083bc1da Both Codec SSID and PCI SSID are 1458:a002 on that machine, codec vendor ID is 10ec:0899 - maybe we should only apply this quirk for certain codec vendor IDs, then? Anyhow, you can bring back jack detection for yourself (as a workaround) by reconfiguring the codec. This is easiest done through hda-jack-retask (available in the alsa-tools repository). In hda-jack-retask click "advanced override", then make sure pin 0x1b has jack detect available. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 2

[alsa-devel] [PATCH] ASoC: rt5677: Add the slot_width "25" support in the TDM mode
by Oder Chiou 12 Jan '15

12 Jan '15

Add the slot_width "25" support in the TDM mode for the Intel platform. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 32c368c..af34ffc 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4123,6 +4123,7 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, val |= (1 << 8); break; case 24: + case 25: val |= (2 << 8); break; case 32: @@ -4136,9 +4137,14 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, switch (dai->id) { case RT5677_AIF1: snd_soc_update_bits(codec, RT5677_TDM1_CTRL1, 0x1f00, val); + if (slot_width == 25) + snd_soc_update_bits(codec, RT5677_DIG_MISC, 0x8000, + 0x8000); break; case RT5677_AIF2: snd_soc_update_bits(codec, RT5677_TDM2_CTRL1, 0x1f00, val); + if (slot_width == 25) + snd_soc_update_bits(codec, RT5677_DIG_MISC, 0x80, 0x80); break; default: break; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 4/5] sound: dmasound_atari: Remove obsolete IRQ_TYPE_SLOW
by Geert Uytterhoeven 12 Jan '15

12 Jan '15

IRQ_TYPE_SLOW is no longer used by the Atari platform interrupt code since commit 734085651c9b80aa ("[PATCH] m68k: convert atari irq code") in v2.6.18-rc1, so drop it. Note that its value has been reused for a different purpose (IRQ_TYPE_NONE) since commit 6a6de9ef5850d063 ("[PATCH] genirq: core") in v2.6.18-rc1. Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: alsa-devel(a)alsa-project.org --- sound/oss/dmasound/dmasound_atari.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 13c214466d3b9f0e..1c56bf58eff906e0 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -851,7 +851,7 @@ static int __init AtaIrqInit(void) st_mfp.tim_dt_a = 1; /* Cause interrupt after first event. */ st_mfp.tim_ct_a = 8; /* Turn on event counting. */ /* Register interrupt handler. */ - if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound", + if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, 0, "DMA sound", AtaInterrupt)) return 0; st_mfp.int_en_a |= 0x20; /* Turn interrupt on. */ -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] alsactl/init: Change the "basic rules" description
by David Henningsson 12 Jan '15

12 Jan '15

We now enable more outputs than just the main speaker, so update the basic rules to reflect the actual situation. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- alsactl/init/default | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/alsactl/init/default b/alsactl/init/default index 5fb0113..64f933f 100644 --- a/alsactl/init/default +++ b/alsactl/init/default @@ -3,7 +3,7 @@ # # Basic rules are: # - keep volumes at minimal level, but sound should be hearable -# - enable just main speakers for playback and main microphone for recording +# - enable standard outputs for playback and main microphone for recording # # ************************************************************************** -- 1.9.1

2 2

[alsa-devel] [PATCH] ALSA: rme*: Use snd_pcm_format_t
by Takashi Iwai 12 Jan '15

12 Jan '15

Fix sparse warnings below by using snd_pcm_format_t properly: sound/pci/rme32.c:682:60: sparse: incorrect type in argument 2 (different base types) sound/pci/rme96.c:1006:69: sparse: incorrect type in argument 2 (different base types) ..... Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/rme32.c | 2 +- sound/pci/rme96.c | 6 ++---- 2 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 6c60dcd2e5a1..1a7affad7164 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -632,7 +632,7 @@ snd_rme32_setframelog(struct rme32 * rme32, int n_channels, int is_playback) } } -static int snd_rme32_setformat(struct rme32 * rme32, int format) +static int snd_rme32_setformat(struct rme32 *rme32, snd_pcm_format_t format) { switch (format) { case SNDRV_PCM_FORMAT_S16_LE: diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index e33e79eadf83..236ac1d48184 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -922,8 +922,7 @@ snd_rme96_setframelog(struct rme96 *rme96, } static int -snd_rme96_playback_setformat(struct rme96 *rme96, - int format) +snd_rme96_playback_setformat(struct rme96 *rme96, snd_pcm_format_t format) { switch (format) { case SNDRV_PCM_FORMAT_S16_LE: @@ -940,8 +939,7 @@ snd_rme96_playback_setformat(struct rme96 *rme96, } static int -snd_rme96_capture_setformat(struct rme96 *rme96, - int format) +snd_rme96_capture_setformat(struct rme96 *rme96, snd_pcm_format_t format) { switch (format) { case SNDRV_PCM_FORMAT_S16_LE: -- 2.2.1

1 0

Re: [alsa-devel] [FFADO-devel] [RFC] libhinawa: a light-weight I/O library for status/transactions to ALSA FireWire devices
by Jonathan Woithe 12 Jan '15

12 Jan '15

Hi Takashi S On Tue, Sep 23, 2014 at 08:39:34PM +0900, Takashi Sakamoto wrote: > For some reasons (I describe later), I start to develop a light-weight > I/O library for status/transactions to ALSA FireWire devices. > https://github.com/takaswie/libhinawa > > This library gives ways to use ALSA FireWire driver functionality such as > lock/unlock kernel streaming, EFW transaction. This is one of my > solutions to some issues between ALSA/FFADO. As I understand it, this library will be a thin wrapper around whatever kernel interface is used to communicate these details out to userspace (I think byte-type mixer controls are the preferred avenue at present). In principle I have no problem with this approach so long as it remains lightweight. The situation to avoid is the one we had with libraw1394; when the new stack came about the preferred way to interact with it switched back to the native kernel interface since (as I understand the situation) it was felt that libraw1394 didn't really provide a worthwhile API abstraction. Out of interest, is there any signficance in the library's name "hinawa"? > I hope to merge this library into alsa-tools package in future, and this > is a first RFC for it. It's under development and I require your > comments about its APIs and structure ... I will try to find some time to look at the current proposal in the next week or so. > Dice notification is an actual example. Dice based devices transfer > notification to a specific address on host controller. The address space > is exclusive resources on an system. For Dice notification, ALSA Dice > driver gives a way to utilize it for applications As well as Fireworks > situation, some userspace stuffs are needed to use this functionality. For sure. One thing to keep in mind is that while this is an issue for DICE devices it may not apply to others. The library API should be structured to keep in mind that not all features will be needed across the board. We don't want to burden all subsequent firewire-audio streaming drivers with baggage which is only required for a small subset. Regards jonathan

2 2

[alsa-devel] [PATCHv3 1/2] ALSA: fix emu8000 DRAM sizing for AWE64 Value
by David Flater 12 Jan '15

12 Jan '15

Applicable to any kernel since 2013: The special case added in commit 1338fc97d07a did not handle the possibility that the address space on an AWE64 Value would wrap around at 512 KiB. That is what it does, so the memory is still not detected on those cards. Fix that with a logic clean-up that eliminates the need for a special case. Signed-off-by: David Flater <dave(a)flaterco.com> --- History: 2015-01-11 v3: In response to feedback from Takashi Iwai, Patch 1: Repeat code to avoid goto middle of loop. Patch 2: Use pr_info not printk. Retested on CT4390 (4 MiB) and CT4380 (512 KiB). 2015-01-09 v2: In response to feedback from Takashi Iwai, 1. Optimize for diff size. 1a. Use goto to avoid indenting and repeating code. 1b. Jettison new debugging printouts. 2. Split printk into second patch. Retested on CT4390 (4 MiB) and CT4380 (512 KiB). 2015-01-08 v1 patch sent to LKML, Alsa Devel and maintainers. Tested on unexpanded CT4390 (4 MiB), CT4520 (512 KiB), and CT4380 (512 KiB). sound/isa/sb/emu8000.c | 31 +++++++++++++++---------------- 1 file changed, 15 insertions(+), 16 deletions(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 45fcdff..d25e384 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -378,13 +378,12 @@ init_arrays(struct snd_emu8000 *emu) static void size_dram(struct snd_emu8000 *emu) { - int i, size, detected_size; + int i, size; if (emu->dram_checked) return; size = 0; - detected_size = 0; /* write out a magic number */ snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE); @@ -392,10 +391,19 @@ size_dram(struct snd_emu8000 *emu) EMU8000_SMALW_WRITE(emu, EMU8000_DRAM_OFFSET); EMU8000_SMLD_WRITE(emu, UNIQUE_ID1); snd_emu8000_init_fm(emu); /* This must really be here and not 2 lines back even */ + snd_emu8000_write_wait(emu); - while (size < EMU8000_MAX_DRAM) { + /* + * Detect first 512 KiB. If a write succeeds at the beginning of a + * 512 KiB page we assume that the whole page is there. + */ + EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); + EMU8000_SMLD_READ(emu); /* discard stale data */ + if (EMU8000_SMLD_READ(emu) != UNIQUE_ID1) + goto skip_detect; /* No RAM */ + snd_emu8000_read_wait(emu); - size += 512 * 1024; /* increment 512kbytes */ + for (size = 512 * 1024; size < EMU8000_MAX_DRAM; size += 512 * 1024) { /* Write a unique data on the test address. * if the address is out of range, the data is written on @@ -431,18 +439,9 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_read_wait(emu); /* Otherwise, it's valid memory. */ - detected_size = size + 512 * 1024; - } - - /* Distinguish 512 KiB from 0. */ - if (detected_size == 0) { - snd_emu8000_read_wait(emu); - EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); - EMU8000_SMLD_READ(emu); /* discard stale data */ - if (EMU8000_SMLD_READ(emu) == UNIQUE_ID1) - detected_size = 512 * 1024; } +skip_detect: /* wait until FULL bit in SMAxW register is false */ for (i = 0; i < 10000; i++) { if ((EMU8000_SMALW_READ(emu) & 0x80000000) == 0) @@ -455,9 +454,9 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", - emu->port1, detected_size/1024); + emu->port1, size/1024); - emu->mem_size = detected_size; + emu->mem_size = size; emu->dram_checked = 1; } -- 1.8.4

2 5

[alsa-devel] [PATCH 1/2] lib: devres: add a helper function for ioremap_wc
by Abhilash Kesavan 12 Jan '15

12 Jan '15

Implement a resource managed writecombine ioremap function. Signed-off-by: Abhilash Kesavan <a.kesavan(a)samsung.com> --- Documentation/driver-model/devres.txt | 1 + include/linux/io.h | 2 ++ lib/devres.c | 28 ++++++++++++++++++++++++++++ 3 files changed, 31 insertions(+) diff --git a/Documentation/driver-model/devres.txt b/Documentation/driver-model/devres.txt index b5ab416..0f80cee 100644 --- a/Documentation/driver-model/devres.txt +++ b/Documentation/driver-model/devres.txt @@ -274,6 +274,7 @@ IOMAP devm_ioport_unmap() devm_ioremap() devm_ioremap_nocache() + devm_ioremap_wc() devm_ioremap_resource() : checks resource, requests memory region, ioremaps devm_iounmap() pcim_iomap() diff --git a/include/linux/io.h b/include/linux/io.h index fa02e55..42b33f0 100644 --- a/include/linux/io.h +++ b/include/linux/io.h @@ -64,6 +64,8 @@ void __iomem *devm_ioremap(struct device *dev, resource_size_t offset, resource_size_t size); void __iomem *devm_ioremap_nocache(struct device *dev, resource_size_t offset, resource_size_t size); +void __iomem *devm_ioremap_wc(struct device *dev, resource_size_t offset, + resource_size_t size); void devm_iounmap(struct device *dev, void __iomem *addr); int check_signature(const volatile void __iomem *io_addr, const unsigned char *signature, int length); diff --git a/lib/devres.c b/lib/devres.c index 0f1dd2e..e8e1738 100644 --- a/lib/devres.c +++ b/lib/devres.c @@ -72,6 +72,34 @@ void __iomem *devm_ioremap_nocache(struct device *dev, resource_size_t offset, EXPORT_SYMBOL(devm_ioremap_nocache); /** + * devm_ioremap_wc - Managed ioremap_wc() + * @dev: Generic device to remap IO address for + * @offset: BUS offset to map + * @size: Size of map + * + * Managed ioremap_wc(). Map is automatically unmapped on driver detach. + */ +void __iomem *devm_ioremap_wc(struct device *dev, resource_size_t offset, + resource_size_t size) +{ + void __iomem **ptr, *addr; + + ptr = devres_alloc(devm_ioremap_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return NULL; + + addr = ioremap_wc(offset, size); + if (addr) { + *ptr = addr; + devres_add(dev, ptr); + } else + devres_free(ptr); + + return addr; +} +EXPORT_SYMBOL(devm_ioremap_wc); + +/** * devm_iounmap - Managed iounmap() * @dev: Generic device to unmap for * @addr: Address to unmap -- 1.7.9.5

9 14

[alsa-devel] [PATCH] ASoC: ad193x: Don't power down DAC in CODEC probe
by Lars-Peter Clausen 12 Jan '15

12 Jan '15

The DAC powerdown bit is managed by DAPM. Manually powering down the DAC in the CODEC probe function may cause unnecessary power state transitions which can lead to click and pop noises. So leave the DAC powerdown bit in its default poweron-reset state and let DAPM do all the management. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad193x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 387530b..17c9535 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -333,8 +333,8 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_DAC_CHNL_MUTE, 0x0); /* de-emphasis: 48kHz, powedown dac */ regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); - /* powerdown dac, dac in tdm mode */ - regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x41); + /* dac in tdm mode */ + regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); /* high-pass filter enable */ regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); /* sata delay=1, adc aux mode */ -- 1.8.0

2 1

[alsa-devel] [PATCH v4 0/6] sanitize hda/i915 interface using the component fw
by Imre Deak 12 Jan '15

12 Jan '15

This is v4 of [1] addressing the review comments from Takashi and Jani. [1] http://lists.freedesktop.org/archives/intel-gfx/2014-December/056992.html Imre Deak (6): drm/i915: add dev_to_i915 helper drm/i915: add component support ALSA: hda: export struct hda_intel ALSA: hda: pass intel_hda to all i915 interface functions ALSA: hda: add component support drm/i915: remove unused power_well/get_cdclk_freq api drivers/gpu/drm/i915/i915_dma.c | 4 + drivers/gpu/drm/i915/i915_drv.c | 9 +- drivers/gpu/drm/i915/i915_drv.h | 8 ++ drivers/gpu/drm/i915/intel_audio.c | 110 +++++++++++++++++++++++ drivers/gpu/drm/i915/intel_drv.h | 2 + drivers/gpu/drm/i915/intel_runtime_pm.c | 56 ------------ include/drm/i915_component.h | 38 ++++++++ include/drm/i915_powerwell.h | 37 -------- sound/pci/hda/hda_i915.c | 152 ++++++++++++++++++++++---------- sound/pci/hda/hda_i915.h | 37 -------- sound/pci/hda/hda_intel.c | 60 ++++--------- sound/pci/hda/hda_intel.h | 71 +++++++++++++++ 12 files changed, 360 insertions(+), 224 deletions(-) create mode 100644 include/drm/i915_component.h delete mode 100644 include/drm/i915_powerwell.h delete mode 100644 sound/pci/hda/hda_i915.h create mode 100644 sound/pci/hda/hda_intel.h -- 2.1.0

3 9

Re: [alsa-devel] Regression for Lenovo X240 sound in 3.18
by Raymond Yau 12 Jan '15

12 Jan '15

> > > >control.14 { > >iface CARD > >name 'Mic Jack' > >value false > >comment { access read type BOOLEAN count 1 } } > > > >What is the usage of this jack detect control if conventional mic is not > >supported ? > > "jack detect" reads like "detect if something is present", and that > is overly general, which could also mean it is used to detect > when to turn off the built-in speaker. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Seem those dock stations also have combo jack but the patch did not create dock headset mic jack / dock headset mic phantom jack ThinkPad® Ultra Dock(40A20090XX), ThinkPad® Pro Dock (40A10065XX), ThinkPad® Dock (40A00065) http://support.lenovo.com/en/documents/pd028010 http://support.lenovo.com/en/documents/pd028011 1x Stereo/Mic Combo Audio Port Refer to the maunal of those dock station, the icon near the combo jack is also a headset

1 0

[alsa-devel] [PATCHv3 2/2] ALSA: log emu8000 DRAM size at level INFO
by David Flater 11 Jan '15

11 Jan '15

Applicable after PATCHv3 1/2: Detected sound font memory goes unreported unless the kernel was built with ALSA debugging enabled. Elevate that to a pr_info. Signed-off-by: David Flater <dave(a)flaterco.com> --- History: 2015-01-11 v3: In response to feedback from Takashi Iwai, Patch 1: Repeat code to avoid goto middle of loop. Patch 2: Use pr_info not printk. Retested on CT4390 (4 MiB) and CT4380 (512 KiB). 2015-01-09 v2: In response to feedback from Takashi Iwai, 1. Optimize for diff size. 1a. Use goto to avoid indenting and repeating code. 1b. Jettison new debugging printouts. 2. Split printk into second patch. Retested on CT4390 (4 MiB) and CT4380 (512 KiB). 2015-01-08 v1 patch sent to LKML, Alsa Devel and maintainers. Tested on unexpanded CT4390 (4 MiB), CT4520 (512 KiB), and CT4380 (512 KiB). sound/isa/sb/emu8000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index d25e384..96e9d94 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -453,7 +453,7 @@ skip_detect: snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_CLOSE); snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); - snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", + pr_info("EMU8000 [0x%lx]: %d KiB on-board DRAM detected\n", emu->port1, size/1024); emu->mem_size = size; -- 1.8.4

1 0

[alsa-devel] [PATCH] ALSA: lx6464es: lx_core: Remove some unused functions
by Rickard Strandqvist 11 Jan '15

11 Jan '15

Removes some functions that are not used anywhere: lx_buffer_cancel() lx_buffer_free() lx_stream_sample_position() lx_stream_state() lx_pipe_sample_count() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/lx6464es/lx_core.c | 107 ------------------------------------------ sound/pci/lx6464es/lx_core.h | 10 ---- 2 files changed, 117 deletions(-) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index f3d6202..5621cd7 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -582,33 +582,6 @@ int lx_pipe_pause(struct lx6464es *chip, u32 pipe, int is_capture) } -int lx_pipe_sample_count(struct lx6464es *chip, u32 pipe, int is_capture, - u64 *rsample_count) -{ - int err; - u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); - - mutex_lock(&chip->msg_lock); - lx_message_init(&chip->rmh, CMD_0A_GET_PIPE_SPL_COUNT); - - chip->rmh.cmd[0] |= pipe_cmd; - chip->rmh.stat_len = 2; /* need all words here! */ - - err = lx_message_send_atomic(chip, &chip->rmh); /* don't sleep! */ - - if (err != 0) - dev_err(chip->card->dev, - "could not query pipe's sample count\n"); - else { - *rsample_count = ((u64)(chip->rmh.stat[0] & MASK_SPL_COUNT_HI) - << 24) /* hi part */ - + chip->rmh.stat[1]; /* lo part */ - } - - mutex_unlock(&chip->msg_lock); - return err; -} - int lx_pipe_state(struct lx6464es *chip, u32 pipe, int is_capture, u16 *rstate) { int err; @@ -714,46 +687,6 @@ int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, return err; } -int lx_stream_state(struct lx6464es *chip, u32 pipe, int is_capture, - int *rstate) -{ - int err; - u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); - - mutex_lock(&chip->msg_lock); - lx_message_init(&chip->rmh, CMD_0E_GET_STREAM_SPL_COUNT); - - chip->rmh.cmd[0] |= pipe_cmd; - - err = lx_message_send_atomic(chip, &chip->rmh); - - *rstate = (chip->rmh.stat[0] & SF_START) ? START_STATE : PAUSE_STATE; - - mutex_unlock(&chip->msg_lock); - return err; -} - -int lx_stream_sample_position(struct lx6464es *chip, u32 pipe, int is_capture, - u64 *r_bytepos) -{ - int err; - u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); - - mutex_lock(&chip->msg_lock); - lx_message_init(&chip->rmh, CMD_0E_GET_STREAM_SPL_COUNT); - - chip->rmh.cmd[0] |= pipe_cmd; - - err = lx_message_send_atomic(chip, &chip->rmh); - - *r_bytepos = ((u64) (chip->rmh.stat[0] & MASK_SPL_COUNT_HI) - << 32) /* hi part */ - + chip->rmh.stat[1]; /* lo part */ - - mutex_unlock(&chip->msg_lock); - return err; -} - /* low-level buffer handling */ int lx_buffer_give(struct lx6464es *chip, u32 pipe, int is_capture, u32 buffer_size, u32 buf_address_lo, u32 buf_address_hi, @@ -803,46 +736,6 @@ int lx_buffer_give(struct lx6464es *chip, u32 pipe, int is_capture, return err; } -int lx_buffer_free(struct lx6464es *chip, u32 pipe, int is_capture, - u32 *r_buffer_size) -{ - int err; - u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); - - mutex_lock(&chip->msg_lock); - lx_message_init(&chip->rmh, CMD_11_CANCEL_BUFFER); - - chip->rmh.cmd[0] |= pipe_cmd; - chip->rmh.cmd[0] |= MASK_BUFFER_ID; /* ask for the current buffer: the - * microblaze will seek for it */ - - err = lx_message_send_atomic(chip, &chip->rmh); - - if (err == 0) - *r_buffer_size = chip->rmh.stat[0] & MASK_DATA_SIZE; - - mutex_unlock(&chip->msg_lock); - return err; -} - -int lx_buffer_cancel(struct lx6464es *chip, u32 pipe, int is_capture, - u32 buffer_index) -{ - int err; - u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); - - mutex_lock(&chip->msg_lock); - lx_message_init(&chip->rmh, CMD_11_CANCEL_BUFFER); - - chip->rmh.cmd[0] |= pipe_cmd; - chip->rmh.cmd[0] |= buffer_index; - - err = lx_message_send_atomic(chip, &chip->rmh); - - mutex_unlock(&chip->msg_lock); - return err; -} - /* low-level gain/peak handling * diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h index 0cc140c..f95ed1b 100644 --- a/sound/pci/lx6464es/lx_core.h +++ b/sound/pci/lx6464es/lx_core.h @@ -120,8 +120,6 @@ int lx_dsp_get_mac(struct lx6464es *chip); int lx_pipe_allocate(struct lx6464es *chip, u32 pipe, int is_capture, int channels); int lx_pipe_release(struct lx6464es *chip, u32 pipe, int is_capture); -int lx_pipe_sample_count(struct lx6464es *chip, u32 pipe, int is_capture, - u64 *rsample_count); int lx_pipe_state(struct lx6464es *chip, u32 pipe, int is_capture, u16 *rstate); int lx_pipe_stop(struct lx6464es *chip, u32 pipe, int is_capture); int lx_pipe_start(struct lx6464es *chip, u32 pipe, int is_capture); @@ -133,10 +131,6 @@ int lx_pipe_wait_for_idle(struct lx6464es *chip, u32 pipe, int is_capture); /* low-level stream handling */ int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, u32 pipe, int is_capture); -int lx_stream_state(struct lx6464es *chip, u32 pipe, int is_capture, - int *rstate); -int lx_stream_sample_position(struct lx6464es *chip, u32 pipe, int is_capture, - u64 *r_bytepos); int lx_stream_set_state(struct lx6464es *chip, u32 pipe, int is_capture, enum stream_state_t state); @@ -168,10 +162,6 @@ int lx_buffer_ask(struct lx6464es *chip, u32 pipe, int is_capture, int lx_buffer_give(struct lx6464es *chip, u32 pipe, int is_capture, u32 buffer_size, u32 buf_address_lo, u32 buf_address_hi, u32 *r_buffer_index); -int lx_buffer_free(struct lx6464es *chip, u32 pipe, int is_capture, - u32 *r_buffer_size); -int lx_buffer_cancel(struct lx6464es *chip, u32 pipe, int is_capture, - u32 buffer_index); /* low-level gain/peak handling */ int lx_level_unmute(struct lx6464es *chip, int is_capture, int unmute); -- 1.7.10.4

3 3

[alsa-devel] No internal speakers on laptop, suspend-to-ram and resume makes them work
by Wolfy 11 Jan '15

11 Jan '15

I have a Panasonic Toughbook CF-52 with intel hda and AD1833. No sound from internal speakers, but sound from headphones jack works correctly. Based on a forum post I found regarding the same issue in Win7, I tried suspending to ram and resuming, and sound works afterwards, until the next reboot. debianuser (very very helpful!) on irc said I should post this here, as different settings after resume shouldn't be, suggesting a bug. Plus, sound not working for a pretty large laptop brand suggests a fix/quirk of some kind is needed, even if the real bug is in the laptop. Some notes / things tried: Headphones output seems to work correctly on boot. No alsamixer sliders/muting/switches result in sound from speakers. Googling shows this affects a pretty wide variety of panasonic laptops, not just mine or my model. Googling shows this happens in win7 (but not earlier windows versions) unless a "Hotkey Appendix Driver" from panasonic is some random combination (depending on post) of installed, uninstalled and reinstalled. The keyboard does have a few audio hotkeys, but they seem to be software only, and are simply grabbed by xfce's volumed, where they seem to do exactly what they should. But apparently the panasonic windows one does something else with audio from within the hotkey driver? Perhaps it mutes it by toggling some non-obvious bit? Or despite being named "hotkey", it also handles the headphones jack automute? Googling found some reports saying that booting into windows (with the proper extra driver installed) then rebooting into linux will make sound work "for a while", even over reboots. I do not have windows to test these posts. This suggests it's something being saved somewhere that survives poweroff. Googling didn't find anyone with this problem who managed to make sound work correctly without windows. (not good!) It always works (internal speakers come on) after a suspend-to-ram and resume. It always stops working after a restart, suspend-to-disk or any other kind of reboot. Reloading all snd* modules makes no change. If it's not working, it stays not working, if it's working, it stays working. Suspending and resuming fixes it even if all the snd* modules are unloaded before suspending and loaded after resuming. When working (after suspend and resume), the labels shown in alsamixer are wrong - for example, Headphones now controls the internal speakers, while the Speakers slider does exactly nothing. Loading the module with generic, nofixup, ref models did nothing. I installed hda-jack-retask to see if I could make it work that way, but I was unable to make hda-jack-retask do anything at all - even setting every pin to not connected results in absolutely no change in operation. One forum thread on this problem, with some hdajackretask screenshots, is: http://www.toughbooktalk.com/viewtopic.php?f=39&t=1909 I've attached three alsa-info.sh files. The "broken" one is with sound not working. I then suspended and resumed, then generated the "working" one. The third is a diff between them, to save people time trying to spot changes. Thanks, --Randy

1 0

[alsa-devel] [PATCHv2 1/2] ALSA: fix emu8000 DRAM sizing for AWE64 Value
by David Flater 11 Jan '15

11 Jan '15

Applicable to any kernel since 2013: The special case added in commit 1338fc97d07a did not handle the possibility that the address space on an AWE64 Value would wrap around at 512 KiB. That is what it does, so the memory is still not detected on those cards. Fix that with a logic clean-up that eliminates the need for a special case. Signed-off-by: David Flater <dave(a)flaterco.com> --- History: 2015-01-09 v2: In response to feedback from Takashi Iwai, 1. Optimize for diff size. 1a. Use goto to avoid indenting and repeating code. 1b. Jettison new debugging printouts. 2. Split printk into second patch. Retested on CT4390 (4 MiB) and CT4380 (512 KiB). 2015-01-08 v1 patch sent to LKML, Alsa Devel and maintainers. Tested on unexpanded CT4390 (4 MiB), CT4520 (512 KiB), and CT4380 (512 KiB). CT4380 is commonly said to come with 1 MiB of DRAM, but Creative's AWE Control app agreed that mine has only 512 KiB. It has the same memory chip as the CT4520. The affected function first appeared in alsa-driver-0.3.0 and was merged in linux-2.5.5. Its somewhat different ancestor was in sound/oss/awe_wave.c. sound/isa/sb/emu8000.c | 33 +++++++++++++++------------------ 1 file changed, 15 insertions(+), 18 deletions(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 45fcdff..3dcf80e 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -378,13 +378,12 @@ init_arrays(struct snd_emu8000 *emu) static void size_dram(struct snd_emu8000 *emu) { - int i, size, detected_size; + int i, size; if (emu->dram_checked) return; size = 0; - detected_size = 0; /* write out a magic number */ snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE); @@ -392,11 +391,12 @@ size_dram(struct snd_emu8000 *emu) EMU8000_SMALW_WRITE(emu, EMU8000_DRAM_OFFSET); EMU8000_SMLD_WRITE(emu, UNIQUE_ID1); snd_emu8000_init_fm(emu); /* This must really be here and not 2 lines back even */ + snd_emu8000_write_wait(emu); + + goto size_dram_ID1_check; /* If that fails, we have no RAM. */ while (size < EMU8000_MAX_DRAM) { - size += 512 * 1024; /* increment 512kbytes */ - /* Write a unique data on the test address. * if the address is out of range, the data is written on * 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is @@ -424,23 +424,20 @@ size_dram(struct snd_emu8000 *emu) * wraps back to the beginning; so check to see if the * initial value has been overwritten. */ + + size_dram_ID1_check: EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); EMU8000_SMLD_READ(emu); /* discard stale data */ if (EMU8000_SMLD_READ(emu) != UNIQUE_ID1) - break; /* we must have wrapped around */ + break; snd_emu8000_read_wait(emu); - /* Otherwise, it's valid memory. */ - detected_size = size + 512 * 1024; - } - - /* Distinguish 512 KiB from 0. */ - if (detected_size == 0) { - snd_emu8000_read_wait(emu); - EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); - EMU8000_SMLD_READ(emu); /* discard stale data */ - if (EMU8000_SMLD_READ(emu) == UNIQUE_ID1) - detected_size = 512 * 1024; + /* + * Otherwise, it's valid memory. If a write succeeds at + * the beginning of a 512 KiB page we assume that the whole + * page is there. + */ + size += 512 * 1024; /* increment 512 KiB */ } /* wait until FULL bit in SMAxW register is false */ @@ -455,9 +452,9 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", - emu->port1, detected_size/1024); + emu->port1, size/1024); - emu->mem_size = detected_size; + emu->mem_size = size; emu->dram_checked = 1; } -- 1.8.4

2 2

[alsa-devel] [PATCH - alsa-lib 1/1] include/global.h: missing struct on FreeBSD
by rumko＠rumko.net 11 Jan '15

11 Jan '15

From: Rumko <rumko(a)rumko.net> On FreeBSD, struct timespec is not accessible through <time.h>, but through <sys/tiespec.h>. Add specific include for that header on that platform only. Signed-off-by: Rumko <rumko(a)rumko.net> diff --git a/include/global.h b/include/global.h index 16a26dc..70be1f9 100644 --- a/include/global.h +++ b/include/global.h @@ -31,6 +31,10 @@ /* for timeval and timespec */ #include <time.h> +#if defined(__FreeBSD__) +#include <sys/timespec.h> +#endif + #ifdef __cplusplus extern "C" { #endif -- 2.2.1

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[alsa-devel] [PATCH] ALSA: ice1712: wm8766: Remove some unused functions
by Rickard Strandqvist 11 Jan '15

11 Jan '15

Removes some functions that are not used anywhere: snd_wm8766_set_power() snd_wm8766_set_master_mode() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/ice1712/wm8766.c | 16 ---------------- sound/pci/ice1712/wm8766.h | 2 -- 2 files changed, 18 deletions(-) diff --git a/sound/pci/ice1712/wm8766.c b/sound/pci/ice1712/wm8766.c index 21b373b..f7ac8d5 100644 --- a/sound/pci/ice1712/wm8766.c +++ b/sound/pci/ice1712/wm8766.c @@ -183,22 +183,6 @@ void snd_wm8766_set_if(struct snd_wm8766 *wm, u16 dac) snd_wm8766_write(wm, WM8766_REG_IFCTRL, val | dac); } -void snd_wm8766_set_master_mode(struct snd_wm8766 *wm, u16 mode) -{ - u16 val = wm->regs[WM8766_REG_DACCTRL3] & ~WM8766_DAC3_MSTR_MASK; - - mode &= WM8766_DAC3_MSTR_MASK; - snd_wm8766_write(wm, WM8766_REG_DACCTRL3, val | mode); -} - -void snd_wm8766_set_power(struct snd_wm8766 *wm, u16 power) -{ - u16 val = wm->regs[WM8766_REG_DACCTRL3] & ~WM8766_DAC3_POWER_MASK; - - power &= WM8766_DAC3_POWER_MASK; - snd_wm8766_write(wm, WM8766_REG_DACCTRL3, val | power); -} - void snd_wm8766_volume_restore(struct snd_wm8766 *wm) { u16 val = wm->regs[WM8766_REG_DACR1]; diff --git a/sound/pci/ice1712/wm8766.h b/sound/pci/ice1712/wm8766.h index c119f84..18c8d9d 100644 --- a/sound/pci/ice1712/wm8766.h +++ b/sound/pci/ice1712/wm8766.h @@ -155,8 +155,6 @@ struct snd_wm8766 { void snd_wm8766_init(struct snd_wm8766 *wm); void snd_wm8766_resume(struct snd_wm8766 *wm); void snd_wm8766_set_if(struct snd_wm8766 *wm, u16 dac); -void snd_wm8766_set_master_mode(struct snd_wm8766 *wm, u16 mode); -void snd_wm8766_set_power(struct snd_wm8766 *wm, u16 power); void snd_wm8766_volume_restore(struct snd_wm8766 *wm); int snd_wm8766_build_controls(struct snd_wm8766 *wm); -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: lola: lola_mixer: Remove some unused functions
by Rickard Strandqvist 11 Jan '15

11 Jan '15

Removes some functions that are not used anywhere: lola_restore_mixer() lola_save_mixer() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/lola/lola.h | 2 -- sound/pci/lola/lola_mixer.c | 43 ------------------------------------------- 2 files changed, 45 deletions(-) diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h index f0b1000..9125916 100644 --- a/sound/pci/lola/lola.h +++ b/sound/pci/lola/lola.h @@ -513,8 +513,6 @@ int lola_init_mixer_widget(struct lola *chip, int nid); void lola_free_mixer(struct lola *chip); int lola_create_mixer(struct lola *chip); int lola_setup_all_analog_gains(struct lola *chip, int dir, bool mute); -void lola_save_mixer(struct lola *chip); -void lola_restore_mixer(struct lola *chip); int lola_set_src_config(struct lola *chip, unsigned int src_mask, bool update); /* proc */ diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c index 782f4d8..47ac93a 100644 --- a/sound/pci/lola/lola_mixer.c +++ b/sound/pci/lola/lola_mixer.c @@ -348,49 +348,6 @@ int lola_setup_all_analog_gains(struct lola *chip, int dir, bool mute) return lola_codec_flush(chip); } -void lola_save_mixer(struct lola *chip) -{ - /* mute analog output */ - if (chip->mixer.array_saved) { - /* store contents of mixer array */ - memcpy_fromio(chip->mixer.array_saved, chip->mixer.array, - sizeof(*chip->mixer.array)); - } - lola_setup_all_analog_gains(chip, PLAY, true); /* output mute */ -} - -void lola_restore_mixer(struct lola *chip) -{ - int i; - - /*lola_reset_setups(chip);*/ - if (chip->mixer.array_saved) { - /* restore contents of mixer array */ - memcpy_toio(chip->mixer.array, chip->mixer.array_saved, - sizeof(*chip->mixer.array)); - /* inform micro-controller about all restored values - * and ignore return values - */ - for (i = 0; i < chip->mixer.src_phys_ins; i++) - lola_codec_write(chip, chip->mixer.nid, - LOLA_VERB_SET_SOURCE_GAIN, - i, 0); - for (i = 0; i < chip->mixer.src_stream_outs; i++) - lola_codec_write(chip, chip->mixer.nid, - LOLA_VERB_SET_SOURCE_GAIN, - chip->mixer.src_stream_out_ofs + i, 0); - for (i = 0; i < chip->mixer.dest_stream_ins; i++) - lola_codec_write(chip, chip->mixer.nid, - LOLA_VERB_SET_DESTINATION_GAIN, - i, 0); - for (i = 0; i < chip->mixer.dest_phys_outs; i++) - lola_codec_write(chip, chip->mixer.nid, - LOLA_VERB_SET_DESTINATION_GAIN, - chip->mixer.dest_phys_out_ofs + i, 0); - lola_codec_flush(chip); - } -} - /* */ -- 1.7.10.4

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[alsa-devel] [PATCH 0/4] ASoC: Make set_tdm_slot() semantics consistent
by Lars-Peter Clausen 11 Jan '15

11 Jan '15

This is something we shortly discussed during the ALSA mini conference in Duesseldorf last year. We want to do more useful stuff with the snd_soc_set_tdm_slot() API and build generic infrastructure on top of it. But there are currently two different incompatible semantics of how the tx_mask and rx_mask parameters of snd_soc_dai_set_tdm_slot() are used. If we want to be able to use the API in a generic way we need to fix this inconsistency. Both interpretations agree that the slot masks should be represented as bitmasks which bit 0 representing the first slot, bit 1 the second slot and so on. But where they differ is in the meaning of value of the bit. Some implementations assume that if a bit is set the channel is enabled and if the bit is cleared the channel is disabled, others have inverse semantics, they assume that if a bit is cleared the channel is enabled and if the bit is set the channel is disabled. The former is used by most drivers and later is only used by some the Freescale DAI drivers and the MC13783 CODEC driver. This appears to be a result of Freescale DAI drivers directly passing the masks through to the registers which expect the inverted semantics. And the MC13783 appears to have picked up the same meaning since it is used on a board which has a Freescale DAI on the CPU side of the DAI link. This series updates the MC13783 and Freescale DAI drivers to use the same semantics as other drivers and also updates the documentation of the snd_soc_dai_set_tdm_slot() function to be more specific on the intended semantics. - Lars Lars-Peter Clausen (4): ASoC: mc13783: Update set_tdm_slot() semantics ASoC: fsl: Update set_tdm_slot() semantics ASoC: fsl: Remove fsl_asoc_xlate_tdm_slot_mask() ASoC: Update snd_soc_dai_set_tdm_slot() documentation sound/soc/codecs/mc13783.c | 10 +++++----- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_esai.c | 3 +++ sound/soc/fsl/fsl_ssi.c | 4 ++-- sound/soc/fsl/fsl_utils.c | 27 --------------------------- sound/soc/fsl/fsl_utils.h | 3 --- sound/soc/fsl/imx-mc13783.c | 5 ++--- sound/soc/fsl/imx-ssi.c | 5 ++--- sound/soc/fsl/wm1133-ev1.c | 4 ++-- sound/soc/soc-core.c | 20 ++++++++++++++++---- 10 files changed, 33 insertions(+), 50 deletions(-) -- 1.8.0

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[alsa-devel] [PATCHv2 2/2] ALSA: log emu8000 DRAM size at level INFO
by David Flater 11 Jan '15

11 Jan '15

Applicable after PATCHv2 1/2: Detected sound font memory goes unreported unless the kernel was built with ALSA debugging enabled. Elevate that to a printk at level INFO. Signed-off-by: David Flater <dave(a)flaterco.com> --- History: 2015-01-09 v2: In response to feedback from Takashi Iwai, 1. Optimize for diff size. 1a. Use goto to avoid indenting. 1b. Jettison new debugging printouts. 2. Split printk into second patch. Retested on CT4390 (4 MiB) and CT4380 (512 KiB). 2015-01-08 v1 patch sent to LKML, Alsa Devel and maintainers. Tested on unexpanded CT4390 (4 MiB), CT4520 (512 KiB), and CT4380 (512 KiB). sound/isa/sb/emu8000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 3dcf80e..7429a51 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -451,7 +451,7 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_CLOSE); snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); - snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", + snd_printk(KERN_INFO "EMU8000 [0x%lx]: %d KiB on-board DRAM detected\n", emu->port1, size/1024); emu->mem_size = size; -- 1.8.4

2 1

[alsa-devel] [PATCH] fm801: remove FSF address
by Andy Shevchenko 11 Jan '15

11 Jan '15

The FSF address is subject to change. So, we remove it from the header. While here, remove a line with my old and non-functional email from the header. Moreover, nowadays we have a Git VCS that contains a history of changes. Signed-off-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> --- sound/pci/fm801.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 52555bda..1fdd92b6 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -2,8 +2,6 @@ * The driver for the ForteMedia FM801 based soundcards * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> * - * Support FM only card by Andy Shevchenko <andy(a)smile.org.ua> - * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -14,10 +12,6 @@ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * */ #include <linux/delay.h> -- 1.8.3.101.g727a46b

2 1

[alsa-devel] [PATCH] ALSA: hda - fixup input_free_device called after input_unregister_device
by David Henningsson 11 Jan '15

11 Jan '15

Input_unregister_device will internally free the device, so input_free_device should not be called. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 09d2131..a50e15e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3548,7 +3548,6 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, break; case HDA_FIXUP_ACT_FREE: input_unregister_device(spec->kb_dev); - input_free_device(spec->kb_dev); spec->kb_dev = NULL; } #endif -- 1.9.1

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Re: [alsa-devel] Regression for Lenovo X240 sound in 3.18
by Raymond Yau 11 Jan '15

11 Jan '15

> > > >If Combo audio connector does not support conventional microphone, this mean > >the driver should not create mic jack control for node 0x1a > > > >http://support.lenovo.com/us/en/documents/pd029567 > > > >Connect headphones or a headset that has a 3.5-mm (0.14-inch), 4-pole plug > >to the combo audio connector to listen to the sound from the computer. > > I do not have any 4-pole (3-rings) equipment to test. > A conventional 3-pole (2-ring) microphone certainly does not do its job. > > Then again, I ditched unpowered microphones (most external ones are). > The internal mic next to the internal webcam is powered and provides > better gain. > > The Mic control in alsamixer should stay. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… The debug version of driver message indicate whether the codec can differentiate headphone , headset (ctia or omtp) or not http://voices.canonical.com/david.henningsson/2014/03/07/headset-jacks-on-n… http://bazaar.launchpad.net/~unity-settings-daemon-team/unity-settings-daem… Headphone Mic Jack - indicates headphone and mic-in mode share the same jack, i e, not two separate jacks. Hardware cannot distinguish between a headphone and a mic. Headset Mic Phantom Jack - indicates headset jack where hardware can not distinguish between headphones and headsets Headset Mic Jack - indicates headset jack where hardware can distinguish between headphones and headsets. There is no use popping up a dialog in this case, unless we already need to do this for the mic-in mode. control.14 { iface CARD name 'Mic Jack' value false comment { access read type BOOLEAN count 1 } } What is the usage of this jack detect control if conventional mic is not supported ?

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Re: [alsa-devel] ALSA: hda - Enable mic mute hotkey and LEDs for an HP machine
by Dan Carpenter 10 Jan '15

10 Jan '15

Hello David Henningsson, The patch 33f4acd3b214: "ALSA: hda - Enable mic mute hotkey and LEDs for an HP machine" from Jan 7, 2015, leads to the following static checker warning: sound/pci/hda/patch_realtek.c:3551 alc280_fixup_hp_gpio2_mic_hotkey() error: don't call input_free_device() after input_unregister_device() sound/pci/hda/patch_realtek.c 3545 switch (action) { 3546 case HDA_FIXUP_ACT_PROBE: 3547 spec->init_amp = ALC_INIT_DEFAULT; 3548 break; 3549 case HDA_FIXUP_ACT_FREE: 3550 input_unregister_device(spec->kb_dev); 3551 input_free_device(spec->kb_dev); I think this is a double free. 3552 spec->kb_dev = NULL; 3553 } 3554 #endif 3555 } regards, dan carpenter

1 0

[alsa-devel] [PATCH 1/8] ASoC: corgi: Automatically disconnect non-connected pins
by Lars-Peter Clausen 09 Jan '15

09 Jan '15

All DAPM input and output pins of the wm8994 are either used in the card's DAPM routing table or are marked as not connected. Set the fully_routed flag of the card instead of manually marking the unused inputs and outputs as not connected. This makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/pxa/corgi.c | 16 +--------------- 1 file changed, 1 insertion(+), 15 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index b7cd0a7..3580d10 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -259,20 +259,6 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = { corgi_set_spk), }; -/* - * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device - */ -static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_nc_pin(dapm, "LLINEIN"); - snd_soc_dapm_nc_pin(dapm, "RLINEIN"); - - return 0; -} - /* corgi digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link corgi_dai = { .name = "WM8731", @@ -281,7 +267,6 @@ static struct snd_soc_dai_link corgi_dai = { .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", .codec_name = "wm8731.0-001b", - .init = corgi_wm8731_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ops = &corgi_ops, @@ -300,6 +285,7 @@ static struct snd_soc_card corgi = { .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets), .dapm_routes = corgi_audio_map, .num_dapm_routes = ARRAY_SIZE(corgi_audio_map), + .fully_routed = true, }; static int corgi_probe(struct platform_device *pdev) -- 1.8.0

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[alsa-devel] [PATCH] ASoC: ttc-dkb: Remove unnecessary snd_soc_dapm_disable_pin() calls
by Lars-Peter Clausen 09 Jan '15

09 Jan '15

The "Headset Mic 2" and the "Headset Stereophone" widget are managed by the jack detection logic. Their will be set depending on whether something is connected to the jack or not. There is no need to manually change the state beforehand as it will be overwritten anyway. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/pxa/ttc-dkb.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index e3d7257a..5001dbb 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -76,10 +76,6 @@ static const struct snd_soc_dapm_route ttc_audio_map[] = { static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); - snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); /* Headset jack detection */ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: pcm512x: Fix DSP program selection
by Peter Rosin 09 Jan '15

09 Jan '15

From: Peter Rosin <peda(a)axentia.se> The DSP programs are listed out of order. Signed-off-by: Peter Rosin <peda(a)axentia.se> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/pcm512x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 0c8aefab404c..640c99198cda 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -188,8 +188,8 @@ static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0); static const char * const pcm512x_dsp_program_texts[] = { "FIR interpolation with de-emphasis", "Low latency IIR with de-emphasis", - "Fixed process flow", "High attenuation with de-emphasis", + "Fixed process flow", "Ringing-less low latency FIR", }; -- 1.7.10.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: rt5677: Modify the behavior that updates the PLL parameter.
by Oder Chiou 09 Jan '15

09 Jan '15

The patch modified the behavior that updates the PLL parameter. It set the update bit before the PLL power up. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 18 ++++++++++++++---- 1 file changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8900221..d27630a 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -2174,10 +2174,14 @@ static int rt5677_set_pll1_event(struct snd_soc_dapm_widget *w, struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); switch (event) { - case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_PRE_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x2); + break; + + case SND_SOC_DAPM_POST_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x0); break; + default: return 0; } @@ -2192,10 +2196,14 @@ static int rt5677_set_pll2_event(struct snd_soc_dapm_widget *w, struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); switch (event) { - case SND_SOC_DAPM_POST_PMU: + case SND_SOC_DAPM_PRE_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x2); + break; + + case SND_SOC_DAPM_POST_PMU: regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x0); break; + default: return 0; } @@ -2303,9 +2311,11 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w, static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT, - 0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU), + 0, rt5677_set_pll1_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("PLL2", RT5677_PWR_ANLG2, RT5677_PWR_PLL2_BIT, - 0, rt5677_set_pll2_event, SND_SOC_DAPM_POST_PMU), + 0, rt5677_set_pll2_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5677_ASRC_1, 0, 0, NULL, 0), -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: Intel: Split hsw_pcm_data for playback and capture
by Jie Yang 09 Jan '15

09 Jan '15

There may be 2 pcm streams for a same DAI at most, and these 2 streams should have different hsw_pcm_data, e.g. they have different persistent data, so here we need split hsw_pcm_data for playback and capture, to make sure they won't be mixed and keep cleaned. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 91 ++++++++++++++++++++++++++------------- 1 file changed, 61 insertions(+), 30 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 5448f79..ad7f4a5 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -135,7 +135,17 @@ struct hsw_priv_data { struct snd_dma_buffer dmab[HSW_PCM_COUNT][2]; /* DAI data */ - struct hsw_pcm_data pcm[HSW_PCM_COUNT]; + struct hsw_pcm_data pcm[HSW_PCM_COUNT][2]; +}; + + +/* static mappings between PCMs and modules - may be dynamic in future */ +static struct hsw_pcm_module_map mod_map[] = { + {HSW_PCM_DAI_ID_SYSTEM, 0, SST_HSW_MODULE_PCM_SYSTEM}, + {HSW_PCM_DAI_ID_OFFLOAD0, 0, SST_HSW_MODULE_PCM}, + {HSW_PCM_DAI_ID_OFFLOAD1, 0, SST_HSW_MODULE_PCM}, + {HSW_PCM_DAI_ID_LOOPBACK, 1, SST_HSW_MODULE_PCM_REFERENCE}, + {HSW_PCM_DAI_ID_SYSTEM, 1, SST_HSW_MODULE_PCM_CAPTURE}, }; static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data); @@ -168,9 +178,14 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); - struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; u32 volume; + int dai, stream; + + dai = mod_map[mc->reg].dai_id; + stream = mod_map[mc->reg].stream; + pcm_data = &pdata->pcm[dai][stream]; mutex_lock(&pcm_data->mutex); pm_runtime_get_sync(pdata->dev); @@ -212,9 +227,14 @@ static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); - struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; u32 volume; + int dai, stream; + + dai = mod_map[mc->reg].dai_id; + stream = mod_map[mc->reg].stream; + pcm_data = &pdata->pcm[dai][stream]; mutex_lock(&pcm_data->mutex); pm_runtime_get_sync(pdata->dev); @@ -309,7 +329,7 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ - SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 0, 0, 8, + SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), }; @@ -353,7 +373,7 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; struct sst_module *module_data; struct sst_dsp *dsp; @@ -362,7 +382,10 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, enum sst_hsw_stream_path_id path_id; u32 rate, bits, map, pages, module_id; u8 channels; - int ret; + int ret, dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; /* check if we are being called a subsequent time */ if (pcm_data->allocated) { @@ -552,8 +575,12 @@ static int hsw_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; + int dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -597,11 +624,16 @@ static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; snd_pcm_uframes_t offset; uint64_t ppos; - u32 position = sst_hsw_get_dsp_position(hsw, pcm_data->stream); + u32 position; + int dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; + position = sst_hsw_get_dsp_position(hsw, pcm_data->stream); offset = bytes_to_frames(runtime, position); ppos = sst_hsw_get_dsp_presentation_position(hsw, pcm_data->stream); @@ -618,8 +650,10 @@ static int hsw_pcm_open(struct snd_pcm_substream *substream) snd_soc_platform_get_drvdata(rtd->platform); struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; + int dai; - pcm_data = &pdata->pcm[rtd->cpu_dai->id]; + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; mutex_lock(&pcm_data->mutex); pm_runtime_get_sync(pdata->dev); @@ -648,9 +682,12 @@ static int hsw_pcm_close(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); - struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct hsw_pcm_data *pcm_data; struct sst_hsw *hsw = pdata->hsw; - int ret; + int ret, dai; + + dai = mod_map[rtd->cpu_dai->id].dai_id; + pcm_data = &pdata->pcm[dai][substream->stream]; mutex_lock(&pcm_data->mutex); ret = sst_hsw_stream_reset(hsw, pcm_data->stream); @@ -685,15 +722,6 @@ static struct snd_pcm_ops hsw_pcm_ops = { .page = snd_pcm_sgbuf_ops_page, }; -/* static mappings between PCMs and modules - may be dynamic in future */ -static struct hsw_pcm_module_map mod_map[] = { - {HSW_PCM_DAI_ID_SYSTEM, 0, SST_HSW_MODULE_PCM_SYSTEM}, - {HSW_PCM_DAI_ID_OFFLOAD0, 0, SST_HSW_MODULE_PCM}, - {HSW_PCM_DAI_ID_OFFLOAD1, 0, SST_HSW_MODULE_PCM}, - {HSW_PCM_DAI_ID_LOOPBACK, 1, SST_HSW_MODULE_PCM_REFERENCE}, - {HSW_PCM_DAI_ID_SYSTEM, 1, SST_HSW_MODULE_PCM_CAPTURE}, -}; - static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) { struct sst_hsw *hsw = pdata->hsw; @@ -701,7 +729,7 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) int i; for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; /* create new runtime module, use same offset if recreated */ pcm_data->runtime = sst_hsw_runtime_module_create(hsw, @@ -716,7 +744,7 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) err: for (--i; i >= 0; i--) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; sst_hsw_runtime_module_free(pcm_data->runtime); } @@ -729,7 +757,7 @@ static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) int i; for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; sst_hsw_runtime_module_free(pcm_data->runtime); } @@ -757,7 +785,10 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) return ret; } } - priv_data->pcm[rtd->cpu_dai->id].hsw_pcm = pcm; + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) + priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_PLAYBACK].hsw_pcm = pcm; + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) + priv_data->pcm[rtd->cpu_dai->id][SNDRV_PCM_STREAM_CAPTURE].hsw_pcm = pcm; return ret; } @@ -866,10 +897,9 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) /* allocate DSP buffer page tables */ for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { - mutex_init(&priv_data->pcm[i].mutex); - /* playback */ if (hsw_dais[i].playback.channels_min) { + mutex_init(&priv_data->pcm[i][SNDRV_PCM_STREAM_PLAYBACK].mutex); ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dma_dev, PAGE_SIZE, &priv_data->dmab[i][0]); if (ret < 0) @@ -878,6 +908,7 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) /* capture */ if (hsw_dais[i].capture.channels_min) { + mutex_init(&priv_data->pcm[i][SNDRV_PCM_STREAM_CAPTURE].mutex); ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dma_dev, PAGE_SIZE, &priv_data->dmab[i][1]); if (ret < 0) @@ -1076,7 +1107,7 @@ static void hsw_pcm_complete(struct device *dev) } for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; if (!pcm_data->substream) continue; @@ -1110,7 +1141,7 @@ static int hsw_pcm_prepare(struct device *dev) return 0; /* suspend all active streams */ for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; if (!pcm_data->substream) continue; @@ -1129,7 +1160,7 @@ static int hsw_pcm_prepare(struct device *dev) /* preserve persistent memory */ for (i = 0; i < ARRAY_SIZE(mod_map); i++) { - pcm_data = &pdata->pcm[i]; + pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; if (!pcm_data->substream) continue; -- 1.9.1

2 1

[alsa-devel] [PATCH 00/33] Simplify PM callbacks of PCI drivers
by Takashi Iwai 09 Jan '15

09 Jan '15

This is a series of patches to get rid of superfluous calls found in suspend/resume callbacks of PCI drivers. Just systematically to each driver code. Takashi

2 34

[alsa-devel] [PATCH v2] ALSA: usb-audio: Add support for Akai MPC Element USB MIDI controller
by Paul Bonser 09 Jan '15

09 Jan '15

From: Paul Bonser <misterpib(a)gmail.com> The Akai MPC Element incorrectly reports its bInterfaceClass as 255, but otherwise implements the USB MIDI spec correctly. This adds a quirks-table.h entry which allows the device to be recognized as a standard USB MIDI device. Signed-off-by: Paul Bonser <misterpib(a)gmail.com> --- diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8590a32..e869b6c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2516,6 +2516,28 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + /* Akai MPC Element */ + USB_DEVICE(0x09e8, 0x0021), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = & (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, + /* TerraTec devices */ { USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012),

3 5

[alsa-devel] unload Audio drivers while playback stream is active case kernel crash
by Wang, Jiada (ESD) 09 Jan '15

09 Jan '15

Hi Community Sorry previous email doesn't have formatted crash log. I am using i.MX6Q sabreSD board, which have imx_wm892 machine driver, wm8962 codec and SSI CPU DAI, I got Kernel crash when unloading audio drivers (playback stream is active) modprobe -r snd_soc_imx_wm8962 modprobe -r snd_soc_fsl_ssi modprobe -r snd_soc_wm8962 ######################### [ 31.363894] wm8962 0-001a: Failed to read DSPCLK: -1 [ 31.386320] wm8962 0-001a: Failed to read DCS status: -1 [ 31.417252] Unable to handle kernel NULL pointer dereference at virtual address 0000001c [ 31.425380] pgd = 80004000 [ 31.428154] [0000001c] *pgd=00000000 [ 31.431796] Internal error: Oops: 17 [#1] SMP ARM [ 31.436527] Modules linked in: snd_soc_imx_audmux snd_soc_wm8962 snd_soc_fsl_ssi imx_pcm_fiq evbug] [ 31.448764] CPU: 3 PID: 380 Comm: kworker/3:3 Not tainted 3.19.0-rc3-00069-g11c8f01-dirty #8 [ 31.457210] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 31.463763] Workqueue: events_power_efficient close_delayed_work [ 31.469792] task: be20ce00 ti: bbe20000 task.ti: bbe20000 [ 31.475202] PC is at soc_dapm_dai_stream_event+0x14/0xac [ 31.480522] LR is at snd_soc_dapm_stream_event+0x64/0x90 [ 31.485842] pc : [<8052b298>] lr : [<8052e5cc>] psr: 60000013 [ 31.485842] sp : bbe21e38 ip : bbe21e50 fp : bbe21e4c [ 31.497324] r10: 00000001 r9 : 00000000 r8 : bd449ccc [ 31.502555] r7 : 00000000 r6 : 00000002 r5 : 00000002 r4 : 00000000 [ 31.509087] r3 : 00000000 r2 : 00000002 r1 : 00000000 r0 : 00000000 [ 31.515622] Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel [ 31.522936] Control: 10c5387d Table: 4d42004a DAC: 00000015 [ 31.528689] Process kworker/3:3 (pid: 380, stack limit = 0xbbe20238) [ 31.535048] Stack: (0xbbe21e38 to 0xbbe22000) [ 31.539412] 1e20: 00000000 ba945810 [ 31.547599] 1e40: bbe21e74 bbe21e50 8052e5cc 8052b290 00000000 ba945dd4 ba94581c ba945810 [ 31.555785] 1e60: bbe21eb0 be7c0c00 bbe21e94 bbe21e78 80530038 8052e574 8052ffec bd585680 [ 31.563971] 1e80: ba945dd4 be7bd380 bbe21eec bbe21e98 8003ef74 8052fff8 00000001 00000000 [ 31.572158] 1ea0: 8003ef08 8003f294 00000000 00000000 811d1670 80b33330 00000000 808b61c0 [ 31.580344] 1ec0: 806bac24 bd585680 be7bd3b0 809c7882 bd585698 00000008 be7bd380 be7bd380 [ 31.588529] 1ee0: bbe21f24 bbe21ef0 8003f388 8003edec bdad17c0 bd585680 8003f228 00000000 [ 31.596715] 1f00: bdad17c0 bd585680 8003f228 00000000 00000000 00000000 bbe21fac bbe21f28 [ 31.604902] 1f20: 800446cc 8003f234 bbe21f44 00000000 80062798 bd585680 00000000 00000000 [ 31.613087] 1f40: dead4ead ffffffff ffffffff 809c9f34 00000000 00000000 80838390 bbe21f5c [ 31.621275] 1f60: bbe21f5c 00000000 00000000 dead4ead ffffffff ffffffff 809c9f34 00000000 [ 31.629461] 1f80: 00000000 80838390 bbe21f88 bbe21f88 bdad17c0 800445f0 00000000 00000000 [ 31.637648] 1fa0: 00000000 bbe21fb0 8000ed68 800445fc 00000000 00000000 00000000 00000000 [ 31.645834] 1fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 31.654019] 1fe0: 00000000 00000000 00000000 00000000 00000013 00000000 00000000 00000000 [ 31.662200] Backtrace: [ 31.664680] [<8052b284>] (soc_dapm_dai_stream_event) from [<8052e5cc>] (snd_soc_dapm_stream_event+) [ 31.674512] r5:ba945810 r4:00000000 [ 31.678134] [<8052e568>] (snd_soc_dapm_stream_event) from [<80530038>] (close_delayed_work+0x4c/0x) [ 31.687358] r8:be7c0c00 r7:bbe21eb0 r6:ba945810 r5:ba94581c r4:ba945dd4 r3:00000000 [ 31.695206] [<8052ffec>] (close_delayed_work) from [<8003ef74>] (process_one_work+0x194/0x40c) [ 31.703821] r6:be7bd380 r5:ba945dd4 r4:bd585680 r3:8052ffec [ 31.709558] [<8003ede0>] (process_one_work) from [<8003f388>] (worker_thread+0x160/0x48c) [ 31.717739] r10:be7bd380 r9:be7bd380 r8:00000008 r7:bd585698 r6:809c7882 r5:be7bd3b0 [ 31.725659] r4:bd585680 [ 31.728220] [<8003f228>] (worker_thread) from [<800446cc>] (kthread+0xdc/0xf8) [ 31.735447] r10:00000000 r9:00000000 r8:00000000 r7:8003f228 r6:bd585680 r5:bdad17c0 [ 31.743365] r4:00000000 [ 31.745931] [<800445f0>] (kthread) from [<8000ed68>] (ret_from_fork+0x14/0x2c) [ 31.753158] r7:00000000 r6:00000000 r5:800445f0 r4:bdad17c0 [ 31.758895] Code: e92dd830 e24cb004 e3510000 e1a05002 (0590401c) [ 31.765088] ---[ end trace c7913ba4763772ee ]--- Seems it's because when playback stream is active soc_cleanup_card_resources() -> snd_card_free() Will trigger soc_pcm_close() which will queue close_delayed_work(), and then when close_delayed_work() is scheduled the necessary resource has already been released. I tried to set rtd->pmdown_time to 0 in soc_cleanup_card_resources() to avoid any new delayed_work be queued. BUT when unload audio drivers, I got another kernel crash ######################### [ 28.234740] Unable to handle kernel paging request at virtual address 6faaaf90 [ 28.242075] pgd = bbde0000 [ 28.244818] [6faaaf90] *pgd=00000000 [ 28.248511] Internal error: Oops: 5 [#1] SMP ARM [ 28.253155] Modules linked in: snd_soc_imx_wm8962(-) snd_soc_imx_audmux snd_soc_wm8962 snd_soc_fsl] [ 28.267055] CPU: 1 PID: 188 Comm: pulseaudio Not tainted 3.19.0-rc3-00069-g11c8f01-dirty #7 [ 28.275435] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 28.281991] task: bc3e1d40 ti: bbc1e000 task.ti: bbc1e000 [ 28.287431] PC is at dapm_seq_insert+0x34/0xf4 [ 28.291902] LR is at dapm_power_widgets+0x234/0x798 [ 28.296809] pc : [<8052ad74>] lr : [<8052d090>] psr: 600f0113 [ 28.296809] sp : bbc1fdd0 ip : 809bdd90 fp : bbc1fdec [ 28.308308] r10: bbc3bb90 r9 : bc0405c4 r8 : bc0405ac [ 28.313554] r7 : bbc1fe24 r6 : 809bdd90 r5 : bbc3b480 r4 : bbc3b134 [ 28.320093] r3 : bbc3b0c0 r2 : 00000000 r1 : bbc1fe1c r0 : bbc3bb40 [ 28.326628] Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user [ 28.333771] Control: 10c5387d Table: 4bde004a DAC: 00000015 [ 28.339524] Process pulseaudio (pid: 188, stack limit = 0xbbc1e238) [ 28.345797] Stack: (0xbbc1fdd0 to 0xbbc20000) [ 28.350162] fdc0: 00000000 bbc3bb40 bbc3bba4 bbc1fe24 [ 28.358349] fde0: bbc1fe5c bbc1fdf0 8052d090 8052ad4c 80062798 8006259c bbc1fe5c 00000002 [ 28.366535] fe00: bc04046c bc0405bc bbc1fe1c bbc1fe14 bbc1fe44 bbc1fe14 bbc1fe14 bbc3b134 [ 28.374721] fe20: bbc3b134 bbc1fe24 bbc1fe24 bbc3bb40 00000004 00000001 bd916010 00000002 [ 28.382907] fe40: 00000000 bc0404cc 00000000 bd91601c bbc1fe84 bbc1fe60 8052e5f4 8052ce68 [ 28.391093] fe60: 00000000 bd916010 00000000 bc2c5a00 bbebb800 bdbe720c bbc1feac bbc1fe88 [ 28.399279] fe80: 805317cc 8052e57c bc2c5a00 bc2c5a00 bc3d9b80 bbdf7640 bc2c4ae8 00000008 [ 28.407465] fea0: bbc1fec4 bbc1feb0 8051b610 80531664 bc2c4a00 bc2c5a00 bbc1feec bbc1fec8 [ 28.415651] fec0: 8051b6a0 8051b5c0 8051b664 bbdf75c0 809bd508 bc3d9b80 bdba8f30 bdba8f30 [ 28.423837] fee0: bbc1ff0c bbc1fef0 8050eb2c 8051b670 bc3d9b80 bdba8f30 bd9b9850 bb8f75c0 [ 28.432023] ff00: bbc1ff4c bbc1ff10 800ef5bc 8050ea50 00000000 00000000 80042d68 bc3d9b88 [ 28.440208] ff20: bc3d9b80 bc3e20f0 00000000 809c9f14 bc3e1d40 00000000 bbc1e000 00000000 [ 28.448393] ff40: bbc1ff5c bbc1ff50 800ef760 800ef538 bbc1ff84 bbc1ff60 80042ec0 800ef75c [ 28.456579] ff60: bc3d9b80 8000ee64 bbc1ffb0 00000000 00000006 8000ee64 bbc1ffac bbc1ff88 [ 28.464765] ff80: 80011884 80042e14 8000ecac 00000004 00000000 0116aa50 0116ab58 0118e100 [ 28.472950] ffa0: 00000000 bbc1ffb0 8000ecf8 800117dc 00000000 76fcf084 76fcf4c0 725fcc94 [ 28.481136] ffc0: 0116aa50 0116ab58 0118e100 00000006 01118168 4ce839e4 00000000 00000000 [ 28.489322] ffe0: 00000000 7e957978 4e1e6d24 4e1e6d34 800f0010 00000016 4eff6821 4eff6c21 [ 28.497501] Backtrace: [ 28.499980] [<8052ad40>] (dapm_seq_insert) from [<8052d090>] (dapm_power_widgets+0x234/0x798) [ 28.508509] r7:bbc1fe24 r6:bbc3bba4 r5:bbc3bb40 r4:00000000 [ 28.514251] [<8052ce5c>] (dapm_power_widgets) from [<8052e5f4>] (snd_soc_dapm_stream_event+0x84/0x) [ 28.523473] r10:bd91601c r9:00000000 r8:bc0404cc r7:00000000 r6:00000002 r5:bd916010 [ 28.531393] r4:00000001 [ 28.533959] [<8052e570>] (snd_soc_dapm_stream_event) from [<805317cc>] (soc_pcm_close+0x174/0x28c) [ 28.542920] r8:bdbe720c r7:bbebb800 r6:bc2c5a00 r5:00000000 r4:bd916010 r3:00000000 [ 28.550764] [<80531658>] (soc_pcm_close) from [<8051b610>] (snd_pcm_release_substream+0x5c/0xb0) [ 28.559552] r10:00000008 r8:bc2c4ae8 r7:bbdf7640 r6:bc3d9b80 r5:bc2c5a00 r4:bc2c5a00 [ 28.567479] [<8051b5b4>] (snd_pcm_release_substream) from [<8051b6a0>] (snd_pcm_release+0x3c/0x88) [ 28.576440] r5:bc2c5a00 r4:bc2c4a00 [ 28.580062] [<8051b664>] (snd_pcm_release) from [<8050eb2c>] (snd_disconnect_release+0xe8/0xf8) [ 28.588764] r8:bdba8f30 r7:bdba8f30 r6:bc3d9b80 r5:809bd508 r4:bbdf75c0 r3:8051b664 [ 28.596604] [<8050ea44>] (snd_disconnect_release) from [<800ef5bc>] (__fput+0x90/0x1d4) [ 28.604611] r7:bb8f75c0 r6:bd9b9850 r5:bdba8f30 r4:bc3d9b80 [ 28.610345] [<800ef52c>] (__fput) from [<800ef760>] (____fput+0x10/0x14) [ 28.617050] r10:00000000 r9:bbc1e000 r8:00000000 r7:bc3e1d40 r6:809c9f14 r5:00000000 [ 28.624968] r4:bc3e20f0 [ 28.627535] [<800ef750>] (____fput) from [<80042ec0>] (task_work_run+0xb8/0xe8) [ 28.634865] [<80042e08>] (task_work_run) from [<80011884>] (do_work_pending+0xb4/0xd4) [ 28.642786] r8:8000ee64 r7:00000006 r6:00000000 r5:bbc1ffb0 r4:8000ee64 r3:bc3d9b80 [ 28.650624] [<800117d0>] (do_work_pending) from [<8000ecf8>] (work_pending+0xc/0x20) [ 28.658371] r6:0118e100 r5:0116ab58 r4:0116aa50 [ 28.663047] Code: e5905000 e3520000 e24c6070 01a0600c (e7965105) [ 28.669220] ---[ end trace 69b76cbdebd80b05 ]--- >From the backtrace, seems it's because when soc_pcm_close() is called, it will power widget due to stream event, but at that time, all widgets have already been freed by soc_remove_dai_links() Currently I am out of idea how to fix this issue, As I think this is a generic issue not i.MX platform specific, Someone in community may has already find this issue, Could anyone give some inputs/suggestion to fix this issue? Regards & Thanks, Jiada

1 0

[alsa-devel] [PATCH] ALSA: fix emu8000 DRAM sizing for AWE64 Value
by David Flater 09 Jan '15

09 Jan '15

Applicable to any kernel since 2013: The special case added in commit 1338fc97d07a did not handle the possibility that the address space on an AWE64 Value would wrap around at 512 KiB. That is what it does, so the memory is still not detected on those cards. Fix that with a logic clean-up that eliminates the need for a special case. Also log the amount of memory detected at level INFO and add sufficiently verbose debugging to diagnose any additional faults of this kind. Tested on unexpanded CT4390 (4 MiB), CT4520 (512 KiB), and CT4380 (512 KiB). The latter is commonly said to come with 1 MiB of DRAM, but Creative's AWE Control app agreed that mine has only 512 KiB. It has the same memory chip as the CT4520. Signed-off-by: David Flater <dave(a)flaterco.com> --- History: 2015-01-08 v1 patch sent to LKML, Alsa Devel and maintainers. The affected function first appeared in alsa-driver-0.3.0 and was merged in linux-2.5.5. Its somewhat different ancestor was in sound/oss/awe_wave.c. AFAICT, the manufacturer never disclosed the right way to do it. In the AWE32 Developer's Information Pack (1994-1996) the RAM sizing function was implemented in object files with a license that prohibited even disassembly. sound/isa/sb/emu8000.c | 127 +++++++++++++++++++++++++++++-------------------- 1 file changed, 76 insertions(+), 51 deletions(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 45fcdff..3aa2250 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -378,13 +378,13 @@ init_arrays(struct snd_emu8000 *emu) static void size_dram(struct snd_emu8000 *emu) { - int i, size, detected_size; + int i, size; + unsigned short rdback; if (emu->dram_checked) return; size = 0; - detected_size = 0; /* write out a magic number */ snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE); @@ -392,55 +392,81 @@ size_dram(struct snd_emu8000 *emu) EMU8000_SMALW_WRITE(emu, EMU8000_DRAM_OFFSET); EMU8000_SMLD_WRITE(emu, UNIQUE_ID1); snd_emu8000_init_fm(emu); /* This must really be here and not 2 lines back even */ - - while (size < EMU8000_MAX_DRAM) { - - size += 512 * 1024; /* increment 512kbytes */ - - /* Write a unique data on the test address. - * if the address is out of range, the data is written on - * 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is - * changed by this data. - */ - /*snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE);*/ - EMU8000_SMALW_WRITE(emu, EMU8000_DRAM_OFFSET + (size>>1)); - EMU8000_SMLD_WRITE(emu, UNIQUE_ID2); - snd_emu8000_write_wait(emu); - - /* - * read the data on the just written DRAM address - * if not the same then we have reached the end of ram. - */ - /*snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_READ);*/ - EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET + (size>>1)); - /*snd_emu8000_read_wait(emu);*/ - EMU8000_SMLD_READ(emu); /* discard stale data */ - if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2) - break; /* no memory at this address */ + snd_emu8000_write_wait(emu); + + /* If that didn't work, we have no RAM at all. */ + EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); + rdback = EMU8000_SMLD_READ(emu); + snd_printdd("EMU8000 [0x%lx]: initial discard data = %04hx\n", + emu->port1, rdback); + rdback = EMU8000_SMLD_READ(emu); + snd_printdd("EMU8000 [0x%lx]: initial readback = %04hx\n", + emu->port1, rdback); + if (rdback == UNIQUE_ID1) { snd_emu8000_read_wait(emu); /* - * If it is the same it could be that the address just - * wraps back to the beginning; so check to see if the - * initial value has been overwritten. + * If a write succeeds at the beginning of a 512 KiB page we + * assume that the whole page is there. */ - EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); - EMU8000_SMLD_READ(emu); /* discard stale data */ - if (EMU8000_SMLD_READ(emu) != UNIQUE_ID1) - break; /* we must have wrapped around */ - snd_emu8000_read_wait(emu); - - /* Otherwise, it's valid memory. */ - detected_size = size + 512 * 1024; - } - - /* Distinguish 512 KiB from 0. */ - if (detected_size == 0) { - snd_emu8000_read_wait(emu); - EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); - EMU8000_SMLD_READ(emu); /* discard stale data */ - if (EMU8000_SMLD_READ(emu) == UNIQUE_ID1) - detected_size = 512 * 1024; + size = 512 * 1024; + + while (size < EMU8000_MAX_DRAM) { + /* + * Write a unique data on the test address. If the + * address is out of range, the data is written on + * 0x200000(=EMU8000_DRAM_OFFSET). Then the id word + * is changed by this data. + */ + EMU8000_SMALW_WRITE(emu, EMU8000_DRAM_OFFSET + + (size>>1)); + EMU8000_SMLD_WRITE(emu, UNIQUE_ID2); + snd_emu8000_write_wait(emu); + + /* + * Read the data on the just written DRAM address. + * If not the same then we have reached the end of RAM. + */ + EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET + + (size>>1)); + rdback = EMU8000_SMLD_READ(emu); + snd_printdd("EMU8000 [0x%lx]: ID2 discard data = %04hx\n", + emu->port1, rdback); + rdback = EMU8000_SMLD_READ(emu); + snd_printdd("EMU8000 [0x%lx]: ID2 readback = %04hx\n", + emu->port1, rdback); + if (rdback != UNIQUE_ID2) { + snd_printdd("EMU8000 [0x%lx]: ID2 break\n", + emu->port1); + break; /* no memory at this address */ + } + snd_emu8000_read_wait(emu); + + /* + * If it is the same it could be that the address just + * wraps back to the beginning, so check to see if the + * initial value has been overwritten. + */ + EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); + rdback = EMU8000_SMLD_READ(emu); + snd_printdd("EMU8000 [0x%lx]: ID1 discard data = %04hx\n", + emu->port1, rdback); + rdback = EMU8000_SMLD_READ(emu); + snd_printdd("EMU8000 [0x%lx]: ID1 readback = %04hx\n", + emu->port1, rdback); + if (rdback != UNIQUE_ID1) { + snd_printdd("EMU8000 [0x%lx]: ID1 break\n", + emu->port1); + break; /* may have wrapped around */ + } + snd_emu8000_read_wait(emu); + + /* Otherwise, it's valid memory. */ + size += 512 * 1024; + } + } else { + snd_printdd("EMU8000 [0x%lx]: initial test failed\n", + emu->port1); } /* wait until FULL bit in SMAxW register is false */ @@ -454,10 +480,9 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_CLOSE); snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); - snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", - emu->port1, detected_size/1024); - - emu->mem_size = detected_size; + snd_printk(KERN_INFO "sbawe [0x%lx]: %d B on-board DRAM detected\n", + emu->port1, size); + emu->mem_size = size; emu->dram_checked = 1; } -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: rockchip: i2s: add rockchip_dmaengine_pcm_config
by Jianqun Xu 09 Jan '15

09 Jan '15

This patch makes snd_dmaengine_pcm_register with rockchip_dmaengine_pcm_config, which configure the parameters of period and buffer match to rockchip DMAC. ======================= without rockchip_dmaengine_pcm_config, and test with command - aplay -D hw:0,0 /tmp/a, there are the error dump: [ 134.899396] dma-pl330 ffb20000.dma-controller: fill_queue:2251 Bad Desc(7) [ 134.906270] dma-pl330 ffb20000.dma-controller: fill_queue:2251 Bad Desc(8) [ 134.913141] dma-pl330 ffb20000.dma-controller: fill_queue:2251 Bad Desc(9) And bellow sound from DMA debugger: "I debugged it a little and it looks like what is happening is that requests which aren't a multiple of burst size * burst length are coming in. Right now the i2s block is setting a burst size of 4 and burst length of 4, but the dmaengine code has no idea about this restriction. I was able to eliminate the messages by changing burst length to 1 in the i2s driver. This fix would always work as long as we're sending a multiple of 4 bytes (which so far seems to be the case)" This patch can make the length of dma buffer is aligned to a multiple of burst size and burst length. Signed-off-by: Jianqun Xu <jay.xu(a)rock-chips.com> --- sound/soc/rockchip/rockchip_i2s.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index c74ba37..bdd1021 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -408,6 +408,24 @@ static const struct regmap_config rockchip_i2s_regmap_config = { .cache_type = REGCACHE_FLAT, }; +static const struct snd_pcm_hardware rockchip_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED, + .period_bytes_min = 1024, + .period_bytes_max = PAGE_SIZE, + .periods_min = 3, + .periods_max = 8, + .buffer_bytes_max = 8 * PAGE_SIZE, + .fifo_size = 16, +}; + +static const struct snd_dmaengine_pcm_config rockchip_dmaengine_pcm_config = { + .pcm_hardware = &rockchip_pcm_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .prealloc_buffer_size = PAGE_SIZE * 8, +}; + static int rockchip_i2s_probe(struct platform_device *pdev) { struct rk_i2s_dev *i2s; @@ -478,7 +496,8 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = snd_dmaengine_pcm_register(&pdev->dev, + &rockchip_dmaengine_pcm_config, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); goto err_pcm_register; -- 1.9.1

2 1

[alsa-devel] [PATCH v3 REPOST] ASoC: add xtensa xtfpga I2S interface and platform
by Max Filippov 08 Jan '15

08 Jan '15

XTFPGA boards provides an audio subsystem that consists of TI CDCE706 clock synthesizer, I2S transmitter and TLV320AIC23 audio codec. I2S transmitter has MMIO-based interface that resembles that of the OpenCores I2S transmitter. I2S transmitter is always a master on I2S bus. There's no specialized audio DMA, sample data are transferred to I2S transmitter FIFO by CPU through memory-mapped queue interface. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- Changes v2->v3: - drop .owner = THIS_MODULE from the struct platform_driver initialization; - rebase to 3.18-rc7. Changes v1->v2: - fix module name in the sound/soc/xtensa/Makefile; - split PCM data transfer code and I2S interface setup code; - move clock enabling/disabling to runtime PM code; - document I2S interface RATIO field; - rewrite I2S interface LEVEL field calculation to avoid overflows; - drop select statements from Kconfig; - document RCU usage for tx_substream; - clean up code running in RCU read-side critical section; - use threaded IRQ handler instead of IRQ + tasklet; - remove empty PCM callback handlers; - add const to static structures where applicable; - use platform_set_drvdata instead of dev_set_drvdata; - drop redundant error reporting code; - use platform_get_irq instead of platform_get_resource; - clean up xtfpga_i2s_remove; - add comments. .../devicetree/bindings/sound/cdns,xtfpga-i2s.txt | 18 + MAINTAINERS | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/xtensa/Kconfig | 7 + sound/soc/xtensa/Makefile | 3 + sound/soc/xtensa/xtfpga-i2s.c | 675 +++++++++++++++++++++ 7 files changed, 706 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt create mode 100644 sound/soc/xtensa/Kconfig create mode 100644 sound/soc/xtensa/Makefile create mode 100644 sound/soc/xtensa/xtfpga-i2s.c diff --git a/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt new file mode 100644 index 0000000..befd125 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt @@ -0,0 +1,18 @@ +Bindings for I2S controller built into xtfpga Xtensa bitstreams. + +Required properties: +- compatible: shall be "cdns,xtfpga-i2s". +- reg: memory region (address and length) with device registers. +- interrupts: interrupt for the device. +- clocks: phandle to the clk used as master clock. I2S bus clock + is derived from it. + +Examples: + + i2s0: xtfpga-i2s@0d080000 { + #sound-dai-cells = <0>; + compatible = "cdns,xtfpga-i2s"; + reg = <0x0d080000 0x40>; + interrupts = <2 1>; + clocks = <&cdce706 4>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index ddb9ac8..f1eb40f 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -10628,6 +10628,7 @@ M: Max Filippov <jcmvbkbc(a)gmail.com> L: linux-xtensa(a)linux-xtensa.org S: Maintained F: drivers/spi/spi-xtensa-xtfpga.c +F: sound/soc/xtensa/xtfpga-i2s.c YAM DRIVER FOR AX.25 M: Jean-Paul Roubelat <jpr(a)f6fbb.org> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 7d5d644..dcc79aa 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -55,6 +55,7 @@ source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" +source "sound/soc/xtensa/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 865e090..5b3c8f6 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -36,3 +36,4 @@ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ +obj-$(CONFIG_SND_SOC) += xtensa/ diff --git a/sound/soc/xtensa/Kconfig b/sound/soc/xtensa/Kconfig new file mode 100644 index 0000000..c201beb --- /dev/null +++ b/sound/soc/xtensa/Kconfig @@ -0,0 +1,7 @@ +config SND_SOC_XTFPGA_I2S + tristate "XTFPGA I2S master" + select REGMAP_MMIO + help + Say Y or M if you want to add support for codecs attached to the + I2S interface on XTFPGA daughter board. You will also need to select + the drivers for the rest of XTFPGA audio subsystem. diff --git a/sound/soc/xtensa/Makefile b/sound/soc/xtensa/Makefile new file mode 100644 index 0000000..15efbf9 --- /dev/null +++ b/sound/soc/xtensa/Makefile @@ -0,0 +1,3 @@ +snd-soc-xtfpga-i2s-objs := xtfpga-i2s.o + +obj-$(CONFIG_SND_SOC_XTFPGA_I2S) += snd-soc-xtfpga-i2s.o diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c new file mode 100644 index 0000000..1cfb19e --- /dev/null +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -0,0 +1,675 @@ +/* + * Xtfpga I2S controller driver + * + * Copyright (c) 2014 Cadence Design Systems Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define DRV_NAME "xtfpga-i2s" + +#define XTFPGA_I2S_VERSION 0x00 +#define XTFPGA_I2S_CONFIG 0x04 +#define XTFPGA_I2S_INT_MASK 0x08 +#define XTFPGA_I2S_INT_STATUS 0x0c +#define XTFPGA_I2S_CHAN0_DATA 0x10 +#define XTFPGA_I2S_CHAN1_DATA 0x14 +#define XTFPGA_I2S_CHAN2_DATA 0x18 +#define XTFPGA_I2S_CHAN3_DATA 0x1c + +#define XTFPGA_I2S_CONFIG_TX_ENABLE 0x1 +#define XTFPGA_I2S_CONFIG_INT_ENABLE 0x2 +#define XTFPGA_I2S_CONFIG_LEFT 0x4 +#define XTFPGA_I2S_CONFIG_RATIO_BASE 8 +#define XTFPGA_I2S_CONFIG_RATIO_MASK 0x0000ff00 +#define XTFPGA_I2S_CONFIG_RES_BASE 16 +#define XTFPGA_I2S_CONFIG_RES_MASK 0x003f0000 +#define XTFPGA_I2S_CONFIG_LEVEL_BASE 24 +#define XTFPGA_I2S_CONFIG_LEVEL_MASK 0x0f000000 +#define XTFPGA_I2S_CONFIG_CHANNEL_BASE 28 + +#define XTFPGA_I2S_INT_UNDERRUN 0x1 +#define XTFPGA_I2S_INT_LEVEL 0x2 +#define XTFPGA_I2S_INT_VALID 0x3 + +#define XTFPGA_I2S_FIFO_SIZE 8192 + +/* + * I2S controller operation: + * + * Enabling TX: output 1 period of zeros (starting with left channel) + * and then queued data. + * + * Level status and interrupt: whenever FIFO level is below FIFO trigger, + * level status is 1 and an IRQ is asserted (if enabled). + * + * Underrun status and interrupt: whenever FIFO is empty, underrun status + * is 1 and an IRQ is asserted (if enabled). + */ +struct xtfpga_i2s { + struct device *dev; + struct clk *clk; + struct regmap *regmap; + void __iomem *regs; + + /* current playback substream. NULL if not playing. + * + * Access to that field is synchronized between the interrupt handler + * and userspace through RCU. + * + * Interrupt handler (threaded part) does PIO on substream data in RCU + * read-side critical section. Trigger callback sets and clears the + * pointer when the playback is started and stopped with + * rcu_assign_pointer. When userspace is about to free the playback + * stream in the pcm_close callback it synchronizes with the interrupt + * handler by means of synchronize_rcu call. + */ + struct snd_pcm_substream *tx_substream; + unsigned (*tx_fn)(struct xtfpga_i2s *i2s, + struct snd_pcm_runtime *runtime, + unsigned tx_ptr); + unsigned tx_ptr; /* next frame index in the sample buffer */ + + /* current fifo level estimate. + * Doesn't have to be perfectly accurate, but must be not less than + * the actual FIFO level in order to avoid stall on push attempt. + */ + unsigned tx_fifo_level; + + /* FIFO level at which level interrupt occurs */ + unsigned tx_fifo_low; + + /* maximal FIFO level */ + unsigned tx_fifo_high; +}; + +static bool xtfpga_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + return reg >= XTFPGA_I2S_CONFIG; +} + +static bool xtfpga_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + return reg < XTFPGA_I2S_CHAN0_DATA; +} + +static bool xtfpga_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == XTFPGA_I2S_INT_STATUS; +} + +static const struct regmap_config xtfpga_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = XTFPGA_I2S_CHAN3_DATA, + .writeable_reg = xtfpga_i2s_wr_reg, + .readable_reg = xtfpga_i2s_rd_reg, + .volatile_reg = xtfpga_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +/* Generate functions that do PIO from TX DMA area to FIFO for all supported + * stream formats. + * Functions will be called xtfpga_pcm_tx_<channels>x<sample bits>, e.g. + * xtfpga_pcm_tx_2x16 for 16-bit stereo. + * + * FIFO consists of 32-bit words, one word per channel, always 2 channels. + * If I2S interface is configured with smaller sample resolution, only + * the LSB of each word is used. + */ +#define xtfpga_pcm_tx_fn(channels, sample_bits) \ +static unsigned xtfpga_pcm_tx_##channels##x##sample_bits( \ + struct xtfpga_i2s *i2s, struct snd_pcm_runtime *runtime, \ + unsigned tx_ptr) \ +{ \ + const u##sample_bits (*p)[channels] = \ + (void *)runtime->dma_area; \ +\ + for (; i2s->tx_fifo_level < i2s->tx_fifo_high; \ + i2s->tx_fifo_level += 2) { \ + iowrite32(p[tx_ptr][0], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + iowrite32(p[tx_ptr][channels - 1], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + if (++tx_ptr >= runtime->buffer_size) \ + tx_ptr = 0; \ + } \ + return tx_ptr; \ +} + +xtfpga_pcm_tx_fn(1, 16) +xtfpga_pcm_tx_fn(2, 16) +xtfpga_pcm_tx_fn(1, 32) +xtfpga_pcm_tx_fn(2, 32) + +#undef xtfpga_pcm_tx_fn + +static bool xtfpga_pcm_push_tx(struct xtfpga_i2s *i2s) +{ + struct snd_pcm_substream *tx_substream; + bool tx_active; + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + tx_active = tx_substream && snd_pcm_running(tx_substream); + if (tx_active) { + unsigned tx_ptr = ACCESS_ONCE(i2s->tx_ptr); + unsigned new_tx_ptr = i2s->tx_fn(i2s, tx_substream->runtime, + tx_ptr); + + cmpxchg(&i2s->tx_ptr, tx_ptr, new_tx_ptr); + } + rcu_read_unlock(); + + return tx_active; +} + +static void xtfpga_pcm_refill_fifo(struct xtfpga_i2s *i2s) +{ + unsigned int_status; + unsigned i; + + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + for (i = 0; i < 2; ++i) { + bool tx_active = xtfpga_pcm_push_tx(i2s); + + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + if (tx_active) + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + if (!tx_active || + !(int_status & XTFPGA_I2S_INT_LEVEL)) + break; + + /* After the push the level IRQ is still asserted, + * means FIFO level is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + if (!(int_status & XTFPGA_I2S_INT_LEVEL)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_VALID); + else if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_UNDERRUN); + + if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE); + else + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); +} + +static irqreturn_t xtfpga_i2s_threaded_irq_handler(int irq, void *dev_id) +{ + struct xtfpga_i2s *i2s = dev_id; + struct snd_pcm_substream *tx_substream; + unsigned config, int_status, int_mask; + + regmap_read(i2s->regmap, XTFPGA_I2S_CONFIG, &config); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_MASK, &int_mask); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, &int_status); + + if (!(config & XTFPGA_I2S_CONFIG_INT_ENABLE) || + !(int_status & int_mask & XTFPGA_I2S_INT_VALID)) + return IRQ_NONE; + + /* Update FIFO level estimate in accordance with interrupt status + * register. + */ + if (int_status & XTFPGA_I2S_INT_UNDERRUN) { + i2s->tx_fifo_level = 0; + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); + } else { + /* The FIFO isn't empty, but is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + + if (tx_substream && snd_pcm_running(tx_substream)) { + snd_pcm_period_elapsed(tx_substream); + if (int_status & XTFPGA_I2S_INT_UNDERRUN) + dev_dbg_ratelimited(i2s->dev, "%s: underrun\n", + __func__); + } + rcu_read_unlock(); + + /* Refill FIFO, update allowed IRQ reasons, enable IRQ if FIFO is + * not empty. + */ + xtfpga_pcm_refill_fifo(i2s); + + return IRQ_HANDLED; +} + +static int xtfpga_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, i2s); + return 0; +} + +static int xtfpga_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned srate = params_rate(params); + unsigned channels = params_channels(params); + unsigned period_size = params_period_size(params); + unsigned sample_size = snd_pcm_format_width(params_format(params)); + unsigned freq, ratio, level; + int err; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RES_MASK, + sample_size << XTFPGA_I2S_CONFIG_RES_BASE); + + freq = 256 * srate; + err = clk_set_rate(i2s->clk, freq); + if (err < 0) + return err; + + /* ratio field of the config register controls MCLK->I2S clock + * derivation: I2S clock = MCLK / (2 * (ratio + 2)). + * + * So with MCLK = 256 * sample rate ratio is 0 for 32 bit stereo + * and 2 for 16 bit stereo. + */ + ratio = (freq - (srate * sample_size * 8)) / + (srate * sample_size * 4); + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RATIO_MASK, + ratio << XTFPGA_I2S_CONFIG_RATIO_BASE); + + i2s->tx_fifo_low = XTFPGA_I2S_FIFO_SIZE / 2; + + /* period_size * 2: FIFO always gets 2 samples per frame */ + for (level = 1; + i2s->tx_fifo_low / 2 >= period_size * 2 && + level < (XTFPGA_I2S_CONFIG_LEVEL_MASK >> + XTFPGA_I2S_CONFIG_LEVEL_BASE); ++level) + i2s->tx_fifo_low /= 2; + + i2s->tx_fifo_high = 2 * i2s->tx_fifo_low; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_LEVEL_MASK, + level << XTFPGA_I2S_CONFIG_LEVEL_BASE); + + dev_dbg(i2s->dev, + "%s srate: %u, channels: %u, sample_size: %u, period_size: %u\n", + __func__, srate, channels, sample_size, period_size); + dev_dbg(i2s->dev, "%s freq: %u, ratio: %u, level: %u\n", + __func__, freq, ratio, level); + + return 0; +} + +static int xtfpga_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) + return -EINVAL; + + return 0; +} + +/* PCM */ + +static const struct snd_pcm_hardware xtfpga_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .period_bytes_min = 2, + .period_bytes_max = XTFPGA_I2S_FIFO_SIZE / 2 * 8, + .periods_min = 2, + .periods_max = XTFPGA_I2S_FIFO_SIZE * 8 / 2, + .buffer_bytes_max = XTFPGA_I2S_FIFO_SIZE * 8, + .fifo_size = 16, +}; + +static int xtfpga_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + void *p; + + snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware); + p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + runtime->private_data = p; + + return 0; +} + +static int xtfpga_pcm_close(struct snd_pcm_substream *substream) +{ + synchronize_rcu(); + return 0; +} + +static int xtfpga_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int ret; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + unsigned channels = params_channels(hw_params); + + switch (channels) { + case 1: + case 2: + break; + + default: + return -EINVAL; + + } + + switch (params_format(hw_params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x16 : + xtfpga_pcm_tx_2x16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x32 : + xtfpga_pcm_tx_2x32; + break; + + default: + return -EINVAL; + } + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + return ret; +} + +static int xtfpga_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ACCESS_ONCE(i2s->tx_ptr) = 0; + rcu_assign_pointer(i2s->tx_substream, substream); + xtfpga_pcm_refill_fifo(i2s); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + rcu_assign_pointer(i2s->tx_substream, NULL); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static snd_pcm_uframes_t xtfpga_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + snd_pcm_uframes_t pos = ACCESS_ONCE(i2s->tx_ptr); + + return pos < runtime->buffer_size ? pos : 0; +} + +static int xtfpga_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = xtfpga_pcm_hardware.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, size, size); +} + +static void xtfpga_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static const struct snd_pcm_ops xtfpga_pcm_ops = { + .open = xtfpga_pcm_open, + .close = xtfpga_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = xtfpga_pcm_hw_params, + .trigger = xtfpga_pcm_trigger, + .pointer = xtfpga_pcm_pointer, +}; + +static const struct snd_soc_platform_driver xtfpga_soc_platform = { + .pcm_new = xtfpga_pcm_new, + .pcm_free = xtfpga_pcm_free, + .ops = &xtfpga_pcm_ops, +}; + +static const struct snd_soc_component_driver xtfpga_i2s_component = { + .name = DRV_NAME, +}; + +static const struct snd_soc_dai_ops xtfpga_i2s_dai_ops = { + .startup = xtfpga_i2s_startup, + .hw_params = xtfpga_i2s_hw_params, + .set_fmt = xtfpga_i2s_set_fmt, +}; + +static struct snd_soc_dai_driver xtfpga_i2s_dai[] = { + { + .name = "xtfpga-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &xtfpga_i2s_dai_ops, + }, +}; + +static int xtfpga_i2s_runtime_suspend(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk); + return 0; +} + +static int xtfpga_i2s_runtime_resume(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(i2s->clk); + if (ret) { + dev_err(dev, "clk_prepare_enable failed: %d\n", ret); + return ret; + } + return 0; +} + +static int xtfpga_i2s_probe(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s; + struct resource *mem; + int err, irq; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + err = -ENOMEM; + goto err; + } + platform_set_drvdata(pdev, i2s); + i2s->dev = &pdev->dev; + dev_dbg(&pdev->dev, "dev: %p, i2s: %p\n", &pdev->dev, i2s); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->regs)) { + err = PTR_ERR(i2s->regs); + goto err; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->regs, + &xtfpga_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + err = PTR_ERR(i2s->regmap); + goto err; + } + + i2s->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk)) { + dev_err(&pdev->dev, "couldn't get clock\n"); + err = PTR_ERR(i2s->clk); + goto err; + } + + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, + (0x1 << XTFPGA_I2S_CONFIG_CHANNEL_BASE)); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, XTFPGA_I2S_INT_VALID); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, XTFPGA_I2S_INT_UNDERRUN); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "No IRQ resource\n"); + err = irq; + goto err; + } + err = devm_request_threaded_irq(&pdev->dev, irq, NULL, + xtfpga_i2s_threaded_irq_handler, + IRQF_SHARED | IRQF_ONESHOT, + pdev->name, i2s); + if (err < 0) { + dev_err(&pdev->dev, "request_irq failed\n"); + goto err; + } + + err = snd_soc_register_platform(&pdev->dev, &xtfpga_soc_platform); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register platform\n"); + goto err; + } + err = devm_snd_soc_register_component(&pdev->dev, + &xtfpga_i2s_component, + xtfpga_i2s_dai, + ARRAY_SIZE(xtfpga_i2s_dai)); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register component\n"); + goto err_unregister_platform; + } + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + err = xtfpga_i2s_runtime_resume(&pdev->dev); + if (err) + goto err_pm_disable; + } + return 0; + +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_unregister_platform: + snd_soc_unregister_platform(&pdev->dev); +err: + dev_err(&pdev->dev, "%s: err = %d\n", __func__, err); + return err; +} + +static int xtfpga_i2s_remove(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + if (i2s->regmap && !IS_ERR(i2s->regmap)) { + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + } + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + xtfpga_i2s_runtime_suspend(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id xtfpga_i2s_of_match[] = { + { .compatible = "cdns,xtfpga-i2s", }, + {}, +}; +MODULE_DEVICE_TABLE(of, xtfpga_i2s_of_match); +#endif + +static const struct dev_pm_ops xtfpga_i2s_pm_ops = { + SET_RUNTIME_PM_OPS(xtfpga_i2s_runtime_suspend, + xtfpga_i2s_runtime_resume, NULL) +}; + +static struct platform_driver xtfpga_i2s_driver = { + .probe = xtfpga_i2s_probe, + .remove = xtfpga_i2s_remove, + .driver = { + .name = "xtfpga-i2s", + .of_match_table = of_match_ptr(xtfpga_i2s_of_match), + .pm = &xtfpga_i2s_pm_ops, + }, +}; + +module_platform_driver(xtfpga_i2s_driver); + +MODULE_AUTHOR("Max Filippov <jcmvbkbc(a)gmail.com>"); +MODULE_DESCRIPTION("xtfpga I2S controller driver"); +MODULE_LICENSE("GPL v2"); -- 1.8.1.4

2 1

[alsa-devel] [PATCH 1/5] ASoC: fsl_ssi: Make error message concise
by Fabio Estevam 08 Jan '15

08 Jan '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently the error message uses 'np->full_name' which leads to a very verbose log as: fsl-ssi-dai 202c000.ssi: no irq for node /soc/aips-bus@02000000/spba-bus@02000000/ssi@0202c000 We can have a concise log by using pdev->name instead: fsl-ssi-dai 202c000.ssi: no irq for node 202c000.ssi Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a65f17d..fa19507 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1363,7 +1363,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi_private->irq = platform_get_irq(pdev, 0); if (!ssi_private->irq) { - dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); + dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); return -ENXIO; } -- 1.9.1

3 6

[alsa-devel] [PATCH 1/2] ASoC: Intel: Add stream direction for pcm-module map
by Jie Yang 08 Jan '15

08 Jan '15

A DAI may have 2 streams(playback/capture) and different modules may be needed for them respectively, so we need add a stream direction here, the combination(dai_id + stream) can tell us which module we really need here. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 04393e3..f4cdf48 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -83,7 +83,6 @@ static const u32 volume_map[] = { #define HSW_PCM_DAI_ID_OFFLOAD0 1 #define HSW_PCM_DAI_ID_OFFLOAD1 2 #define HSW_PCM_DAI_ID_LOOPBACK 3 -#define HSW_PCM_DAI_ID_CAPTURE 4 static const struct snd_pcm_hardware hsw_pcm_hardware = { @@ -104,6 +103,7 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = { struct hsw_pcm_module_map { int dai_id; + int stream; enum sst_hsw_module_id mod_id; }; @@ -751,11 +751,11 @@ static struct snd_pcm_ops hsw_pcm_ops = { /* static mappings between PCMs and modules - may be dynamic in future */ static struct hsw_pcm_module_map mod_map[] = { - {HSW_PCM_DAI_ID_SYSTEM, SST_HSW_MODULE_PCM_SYSTEM}, - {HSW_PCM_DAI_ID_OFFLOAD0, SST_HSW_MODULE_PCM}, - {HSW_PCM_DAI_ID_OFFLOAD1, SST_HSW_MODULE_PCM}, - {HSW_PCM_DAI_ID_LOOPBACK, SST_HSW_MODULE_PCM_REFERENCE}, - {HSW_PCM_DAI_ID_CAPTURE, SST_HSW_MODULE_PCM_CAPTURE}, + {HSW_PCM_DAI_ID_SYSTEM, 0, SST_HSW_MODULE_PCM_SYSTEM}, + {HSW_PCM_DAI_ID_OFFLOAD0, 0, SST_HSW_MODULE_PCM}, + {HSW_PCM_DAI_ID_OFFLOAD1, 0, SST_HSW_MODULE_PCM}, + {HSW_PCM_DAI_ID_LOOPBACK, 1, SST_HSW_MODULE_PCM_REFERENCE}, + {HSW_PCM_DAI_ID_SYSTEM, 1, SST_HSW_MODULE_PCM_CAPTURE}, }; static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) @@ -1139,7 +1139,7 @@ static void hsw_pcm_complete(struct device *dev) return; } - for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) { + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { pcm_data = &pdata->pcm[i]; if (!pcm_data->substream) @@ -1173,7 +1173,7 @@ static int hsw_pcm_prepare(struct device *dev) if (pdata->pm_state == HSW_PM_STATE_D3) return 0; /* suspend all active streams */ - for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) { + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { pcm_data = &pdata->pcm[i]; if (!pcm_data->substream) @@ -1192,7 +1192,7 @@ static int hsw_pcm_prepare(struct device *dev) sst_hsw_dsp_runtime_suspend(hsw); /* preserve persistent memory */ - for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) { + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { pcm_data = &pdata->pcm[i]; if (!pcm_data->substream) -- 1.9.1

2 3

[alsa-devel] [PATCH] ASoC: Intel: Always enable DRAM block for FW dump
by Jie Yang 08 Jan '15

08 Jan '15

The first 512 bytes of data DRAM memory is used for FW dump, and this first data SRAM block should be never power gated (always on), here always enable the block(DSRAM[0]) for D0 stage. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-dsp.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 57039b0..c42ffae 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -306,7 +306,7 @@ static void hsw_reset(struct sst_dsp *sst) static int hsw_set_dsp_D0(struct sst_dsp *sst) { int tries = 10; - u32 reg; + u32 reg, fw_dump_bit; /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); @@ -368,7 +368,9 @@ finish: can't be accessed, please enable each block before accessing. */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + /* for D0, always enable the block(DSRAM[0]) used for FW dump */ + fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT; + writel(reg & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0); /* disable DMA finish function for SSP0 & SSP1 */ @@ -491,6 +493,7 @@ static const struct sst_sram_shift sram_shift[] = { {SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */ {SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */ }; + static u32 hsw_block_get_bit(struct sst_mem_block *block) { u32 bit = 0, shift = 0, index; @@ -587,7 +590,9 @@ static int hsw_block_disable(struct sst_mem_block *block) val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); bit = hsw_block_get_bit(block); - writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0); + /* don't disable DSRAM[0], keep it always enable for FW dump*/ + if (bit != (1 << SST_VDRTCL0_DSRAMPGE_SHIFT)) + writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0); /* wait 18 DSP clock ticks */ udelay(10); @@ -612,7 +617,7 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) const struct sst_adsp_memregion *region; struct device *dev; int ret = -ENODEV, i, j, region_count; - u32 offset, size; + u32 offset, size, fw_dump_bit; dev = sst->dma_dev; @@ -669,9 +674,11 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) } } + /* always enable the block(DSRAM[0]) used for FW dump */ + fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT; /* set default power gating control, enable power gating control for all blocks. that is, can't be accessed, please enable each block before accessing. */ - writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0); + writel(0xffffffff & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0); return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix irq error check
by Fabio Estevam 08 Jan '15

08 Jan '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Commit 2ffa531078037a0 ("ASoC: fsl_ssi: Fix module unbound") changed the way to retrieve the irq number from irq_of_parse_and_map() to platform_get_irq(), but missed to updated the irq error check accordingly. We should test for negative irq number and propagate it in the case of error. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a65f17d..059496e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1362,9 +1362,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) } ssi_private->irq = platform_get_irq(pdev, 0); - if (!ssi_private->irq) { + if (ssi_private->irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); - return -ENXIO; + return ssi_private->irq; } /* Are the RX and the TX clocks locked? */ -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Export snd_soc_runtime_set_dai_fmt()
by Lars-Peter Clausen 08 Jan '15

08 Jan '15

Export snd_soc_runtime_set_dai_fmt() so it can be used in modules. Fixes: ce64c8b9cf5b ("ASoC: Add helper function for changing the DAI link format") Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4b8dc5f..2c02e9c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1491,6 +1491,7 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, return 0; } +EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt); static int snd_soc_instantiate_card(struct snd_soc_card *card) { -- 1.8.0

2 1

[alsa-devel] [PATCH 0/5] sanitize hda/i915 interface using the component fw
by Imre Deak 08 Jan '15

08 Jan '15

The current hda/i915 interface to enable/disable power wells and query the CD clock rate is based on looking up the relevant i915 module symbols from the hda driver. By using the component framework we can get rid of some global state tracking in the i915 driver and pave the way to fully decouple the two drivers: once support is added to enable/disable the HDMI functionality dynamically in the hda driver, it can bind/unbind itself from the i915 component master, without the need to keep a reference on the i915 module. This also gets rid of the problem that currently the i915 driver exposes the interface only on HSW and BDW, while it's also needed at least on VLV/CHV. Imre Deak (5): drm/i915: add dev_to_i915_priv helper drm/i915: add component support ALSA: hda: pass chip to all i915 interface functions ALSA: hda: add component support drm/i915: remove unused power_well/get_cdclk_freq api drivers/gpu/drm/i915/i915_dma.c | 80 ++++++++++++++++++++ drivers/gpu/drm/i915/i915_drv.c | 15 ++-- drivers/gpu/drm/i915/intel_drv.h | 8 ++ drivers/gpu/drm/i915/intel_runtime_pm.c | 56 -------------- include/drm/i915_component.h | 38 ++++++++++ include/drm/i915_powerwell.h | 37 ---------- sound/pci/hda/hda_i915.c | 126 +++++++++++++++++++++----------- sound/pci/hda/hda_i915.h | 12 +-- sound/pci/hda/hda_intel.c | 16 ++-- sound/pci/hda/hda_priv.h | 7 ++ 10 files changed, 238 insertions(+), 157 deletions(-) create mode 100644 include/drm/i915_component.h delete mode 100644 include/drm/i915_powerwell.h -- 1.8.4

6 39

[alsa-devel] AES/SPDIF channel status bits in ALSA
by Petr Kulhavy 08 Jan '15

08 Jan '15

Hi, is there a way to get the channel status bits information from an AES or SPDIF input interface through ALSA and read it in userspace? I'm trying to solve some issues with the channel status bits sent from a custom hardware and finding a way to analyse the channel status info. Regards Petr -- Petr Kulhavy, MSc System Architect *BARIX* petr(a)barix.com <mailto:petr@barix.com> | Skype: brain.barix Barix AG, Seefeldstrasse 303 | 8008 Zurich, Switzerland T +41 43 43322 11 | www.barix.com <http://www.barix.com> You have received this email because of your relationship Barix AG and its affiliated companies. Barix AG and its affiliated companies do not sell or exchange email addresses, or any other personal contact information provided by you with any third parties. All email distributions are managed and controlled by Barix AG and its affiliated companies. Barix AG, Seefeldstr. 303, 8008 Zürich, Switzerland. Company Reg. No: CH-020.3.023.869-8, VAT Reg. No: CHE-105.687.663.

2 1

[alsa-devel] [PATCH] ASoC: rockchip: i2s: applys rate symmetry for CPU DAI
by Jianqun Xu 08 Jan '15

08 Jan '15

This patch applys rate symmetry for rockchip i2s DAI. Signed-off-by: Jianqun Xu <jay.xu(a)rock-chips.com> --- sound/soc/rockchip/rockchip_i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 59aeec4..abd7cdd 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -339,6 +339,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = { SNDRV_PCM_FMTBIT_S24_LE), }, .ops = &rockchip_i2s_dai_ops, + .symmetric_rates = 1, }; static const struct snd_soc_component_driver rockchip_i2s_component = { -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ALSA: fm801: PCI core handles power state for us
by Andy Shevchenko 07 Jan '15

07 Jan '15

There is no need to repeat the work that is already done in the PCI driver core. The patch removes excerpts from suspend and resume callbacks. Note that there is no more calls performed to enable or disable a PCI device during suspend-resume cycle. Nowadays they seems to be superflous. Someone can read more in [1]. [1] https://www.kernel.org/doc/ols/2009/ols2009-pages-319-330.pdf Signed-off-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> --- sound/pci/fm801.c | 15 --------------- 1 file changed, 15 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index c383eec..557c19c 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1369,7 +1369,6 @@ static unsigned char saved_regs[] = { static int snd_fm801_suspend(struct device *dev) { - struct pci_dev *pci = to_pci_dev(dev); struct snd_card *card = dev_get_drvdata(dev); struct fm801 *chip = card->private_data; int i; @@ -1381,29 +1380,15 @@ static int snd_fm801_suspend(struct device *dev) for (i = 0; i < ARRAY_SIZE(saved_regs); i++) chip->saved_regs[i] = inw(chip->port + saved_regs[i]); /* FIXME: tea575x suspend */ - - pci_disable_device(pci); - pci_save_state(pci); - pci_set_power_state(pci, PCI_D3hot); return 0; } static int snd_fm801_resume(struct device *dev) { - struct pci_dev *pci = to_pci_dev(dev); struct snd_card *card = dev_get_drvdata(dev); struct fm801 *chip = card->private_data; int i; - pci_set_power_state(pci, PCI_D0); - pci_restore_state(pci); - if (pci_enable_device(pci) < 0) { - dev_err(dev, "pci_enable_device failed, disabling device\n"); - snd_card_disconnect(card); - return -EIO; - } - pci_set_master(pci); - snd_fm801_chip_init(chip, 1); snd_ac97_resume(chip->ac97); snd_ac97_resume(chip->ac97_sec); -- 1.8.3.101.g727a46b

1 0

[alsa-devel] Persisting background noise on HDA Intel PCH (ALC3232)
by Jens John 07 Jan '15

07 Jan '15

Hello all, On my Thinkpad x240, I am experiencing exactly the same bug as someone on the x230 [0], that is I get clearly audible background noise on headphone and speaker outputs unless I mute the Headphone/Speaker control or Master. In both cases, a Realtek ALC on a Thinkpad is involved. I have attached the alsa-info.sh output. The problems occur with plain ALSA output, Jack or Pulse are not involved. I have tried all suggestions mentioned in [0], and also tested all available ALC-related model= parameters to snd_hda_intel, but the issue persists. Muting all mixers on all cards didn't change anything either. Other Thinkpad laptops with the ALC3232 codec are also affected by this bug, for example [1]. The bug makes the audio output unuseable for any serious purpose. I would be very grateful if somebody could look into this (again?), and also assist as far as I can. Thank you & Best regards -Jens. -- [0]: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-August/080039.html [1]: http://askubuntu.com/questions/541538/headphone-hiss-after-upgrade-to-14-10 PS. I'm sorry if this is not the correct place to report this, but the bug tracker linked on the ALSA project home page seems to be down.

3 3

[alsa-devel] [PATCH] ASoC: tegra: Add platform driver for rt5677 audio codec
by Anatol Pomozov 07 Jan '15

07 Jan '15

The driver supports NVIDIA Tegra Ryu board Sponsored: Google ChromeOS Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../bindings/sound/nvidia,tegra-audio-rt5677.txt | 67 ++++ sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_rt5677.c | 347 +++++++++++++++++++++ 4 files changed, 426 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt create mode 100644 sound/soc/tegra/tegra_rt5677.c diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt new file mode 100644 index 0000000..a4589cd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt @@ -0,0 +1,67 @@ +NVIDIA Tegra audio complex, with RT5677 CODEC + +Required properties: +- compatible : "nvidia,tegra-audio-rt5677" +- clocks : Must contain an entry for each entry in clock-names. + See ../clocks/clock-bindings.txt for details. +- clock-names : Must include the following entries: + - pll_a + - pll_a_out0 + - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) +- nvidia,model : The user-visible name of this sound complex. +- nvidia,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the RT5677's pins (as documented in its binding), and the jacks + on the board: + + * Headphone + * Speaker + * Headset Mic + * Internal Mic 1 + * Internal Mic 2 + +- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's + connected to the CODEC. +- nvidia,audio-codec : The phandle of the RT5677 audio codec. This binding + assumes that AIF1 on the CODEC is connected to Tegra. + +Optional properties: +- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in +- nvidia,hp-en-gpios : The GPIO that enables headphone amplifier +- nvidia,mic-present-gpios: The GPIO that mic jack is plugged in +- nvidia,dmic-clk-en-gpios : The GPIO that gates DMIC clock signal + +Example: + +sound { + compatible = "nvidia,tegra-audio-rt5677-ryu", + "nvidia,tegra-audio-rt5677"; + nvidia,model = "NVIDIA Tegra Ryu"; + + nvidia,audio-routing = + "Headphone", "LOUT2", + "Headphone", "LOUT1", + "Headset Mic", "MICBIAS1", + "IN1P", "Headset Mic", + "IN1N", "Headset Mic", + "DMIC L1", "Internal Mic 1", + "DMIC R1", "Internal Mic 1", + "DMIC L2", "Internal Mic 2", + "DMIC R2", "Internal Mic 2", + "Speaker", "PDM1L", + "Speaker", "PDM1R"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5677>; + + nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; + nvidia,mic-present-gpios = <&gpio TEGRA_GPIO(O, 5) GPIO_ACTIVE_LOW>; + nvidia,hp-en-gpios = <&rt5677 1 GPIO_ACTIVE_HIGH>; + nvidia,dmic-clk-en-gpios = <&rt5677 2 GPIO_ACTIVE_HIGH>; + + clocks = <&tegra_car TEGRA124_CLK_PLL_A>, + <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA124_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; +}; diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 31198cf7..a6768f8 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -128,3 +128,13 @@ config SND_SOC_TEGRA_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra boards using the MAX98090 codec, such as Venice2. + +config SND_SOC_TEGRA_RT5677 + tristate "SoC Audio support for Tegra boards using a RT5677 codec" + depends on SND_SOC_TEGRA && I2C && GPIOLIB + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_RT5677 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the RT5677 codec, such as Ryu. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 5ae588c..9171655 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -19,6 +19,7 @@ obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support snd-soc-tegra-rt5640-objs := tegra_rt5640.o +snd-soc-tegra-rt5677-objs := tegra_rt5677.o snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-wm9712-objs := tegra_wm9712.o @@ -27,6 +28,7 @@ snd-soc-tegra-alc5632-objs := tegra_alc5632.o snd-soc-tegra-max98090-objs := tegra_max98090.o obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o +obj-$(CONFIG_SND_SOC_TEGRA_RT5677) += snd-soc-tegra-rt5677.o obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c new file mode 100644 index 0000000..e4cf978 --- /dev/null +++ b/sound/soc/tegra/tegra_rt5677.c @@ -0,0 +1,347 @@ +/* +* tegra_rt5677.c - Tegra machine ASoC driver for boards using RT5677 codec. + * + * Copyright (c) 2014, The Chromium OS Authors. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + * Based on code copyright/by: + * + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * Copyright 2007 Wolfson Microelectronics PLC. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/rt5677.h" + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-rt5677" + +struct tegra_rt5677 { + struct tegra_asoc_utils_data util_data; + int gpio_hp_det; + int gpio_hp_en; + int gpio_mic_present; + int gpio_dmic_clk_en; +}; + +static int tegra_rt5677_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk, err; + + srate = params_rate(params); + mclk = 256 * srate; + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static int tegra_rt5677_event_hp(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (!gpio_is_valid(machine->gpio_hp_en)) + return 0; + + gpio_set_value_cansleep(machine->gpio_hp_en, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static struct snd_soc_ops tegra_rt5677_ops = { + .hw_params = tegra_rt5677_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_rt5677_hp_jack; + +static struct snd_soc_jack_pin tegra_rt5677_hp_jack_pins = { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, +}; +static struct snd_soc_jack_gpio tegra_rt5677_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, +}; + +static struct snd_soc_jack tegra_rt5677_mic_jack; + +static struct snd_soc_jack_pin tegra_rt5677_mic_jack_pins = { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, +}; + +static struct snd_soc_jack_gpio tegra_rt5677_mic_jack_gpio = { + .name = "Headset Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, + .invert = 1 +}; + +static const struct snd_soc_dapm_widget tegra_rt5677_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_HP("Headphone", tegra_rt5677_event_hp), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Internal Mic 1", NULL), + SND_SOC_DAPM_MIC("Internal Mic 2", NULL), +}; + +static const struct snd_kcontrol_new tegra_rt5677_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic 1"), + SOC_DAPM_PIN_SWITCH("Internal Mic 2"), +}; + +static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_rt5677_hp_jack); + snd_soc_jack_add_pins(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_pins); + + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_rt5677_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + + snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_rt5677_mic_jack); + snd_soc_jack_add_pins(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_pins); + + if (gpio_is_valid(machine->gpio_mic_present)) { + tegra_rt5677_mic_jack_gpio.gpio = machine->gpio_mic_present; + snd_soc_jack_add_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + + return 0; +} + +static int tegra_rt5677_card_remove(struct snd_soc_card *card) +{ + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + if (gpio_is_valid(machine->gpio_mic_present)) { + snd_soc_jack_free_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + return 0; +} + +static struct snd_soc_dai_link tegra_rt5677_dai = { + .name = "RT5677", + .stream_name = "RT5677 PCM", + .codec_dai_name = "rt5677-aif1", + .init = tegra_rt5677_asoc_init, + .ops = &tegra_rt5677_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_rt5677 = { + .name = "tegra-rt5677", + .owner = THIS_MODULE, + .remove = tegra_rt5677_card_remove, + .dai_link = &tegra_rt5677_dai, + .num_links = 1, + .controls = tegra_rt5677_controls, + .num_controls = ARRAY_SIZE(tegra_rt5677_controls), + .dapm_widgets = tegra_rt5677_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_rt5677_dapm_widgets), + .fully_routed = true, +}; + +static int tegra_rt5677_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_tegra_rt5677; + struct tegra_rt5677 *machine; + int ret; + + machine = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_rt5677), GFP_KERNEL); + if (!machine) + return -ENOMEM; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (machine->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_mic_present = of_get_named_gpio(np, + "nvidia,mic-present-gpios", 0); + if (machine->gpio_mic_present == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_hp_en = of_get_named_gpio(np, "nvidia,hp-en-gpios", 0); + if (machine->gpio_hp_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_hp_en)) { + ret = devm_gpio_request_one(&pdev->dev, machine->gpio_hp_en, + GPIOF_OUT_INIT_LOW, "hp_en"); + if (ret) { + dev_err(card->dev, "cannot get hp_en gpio\n"); + return ret; + } + } + + machine->gpio_dmic_clk_en = of_get_named_gpio(np, + "nvidia,dmic-clk-en-gpios", 0); + if (machine->gpio_dmic_clk_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_dmic_clk_en)) { + ret = devm_gpio_request_one(&pdev->dev, + machine->gpio_dmic_clk_en, + GPIOF_OUT_INIT_HIGH, "dmic_clk_en"); + if (ret) { + dev_err(card->dev, "cannot get dmic_clk_en gpio\n"); + return ret; + } + } + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_rt5677_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); + if (!tegra_rt5677_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_rt5677_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!tegra_rt5677_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + tegra_rt5677_dai.platform_of_node = tegra_rt5677_dai.cpu_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&machine->util_data); +err: + return ret; +} + +static int tegra_rt5677_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_rt5677_of_match[] = { + { .compatible = "nvidia,tegra-audio-rt5677", }, + {}, +}; + +static struct platform_driver tegra_rt5677_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_rt5677_of_match, + }, + .probe = tegra_rt5677_probe, + .remove = tegra_rt5677_remove, +}; +module_platform_driver(tegra_rt5677_driver); + +MODULE_AUTHOR("Anatol Pomozov <anatol(a)google.com>"); +MODULE_DESCRIPTION("Tegra+RT5677 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_rt5677_of_match); -- 2.2.0.rc0.207.ga3a616c

2 1

[alsa-devel] [PATCH 1/3] ASoC: Add helper function for changing the DAI link format
by Lars-Peter Clausen 07 Jan '15

07 Jan '15

For some setups it is necessary to change the DAI link format at runtime. This patch factors out the code that does the initial static DAI link format configuration into a separate helper function which can be used board drivers as well. This allows board drivers that have to change the DAI link format at runtime to reuse it instead of having to manually change the format on all DAIs. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 3 ++ sound/soc/soc-core.c | 125 ++++++++++++++++++++++++++++----------------------- 2 files changed, 72 insertions(+), 56 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index b4fca9a..edd4a0a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -429,6 +429,9 @@ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd); void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream); void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream); +int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, + unsigned int dai_fmt); + /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c024962..4b8dc5f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1427,11 +1427,75 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) return 0; } +/** + * snd_soc_runtime_set_dai_fmt() - Change DAI link format for a ASoC runtime + * @rtd: The runtime for which the DAI link format should be changed + * @dai_fmt: The new DAI link format + * + * This function updates the DAI link format for all DAIs connected to the DAI + * link for the specified runtime. + * + * Note: For setups with a static format set the dai_fmt field in the + * corresponding snd_dai_link struct instead of using this function. + * + * Returns 0 on success, otherwise a negative error code. + */ +int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, + unsigned int dai_fmt) +{ + struct snd_soc_dai **codec_dais = rtd->codec_dais; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int i; + int ret; + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = codec_dais[i]; + + ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); + if (ret != 0 && ret != -ENOTSUPP) { + dev_warn(codec_dai->dev, + "ASoC: Failed to set DAI format: %d\n", ret); + return ret; + } + } + + /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */ + if (cpu_dai->codec) { + unsigned int inv_dai_fmt; + + inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK; + switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + break; + case SND_SOC_DAIFMT_CBM_CFS: + inv_dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + break; + case SND_SOC_DAIFMT_CBS_CFM: + inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; + break; + case SND_SOC_DAIFMT_CBS_CFS: + inv_dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + break; + } + + dai_fmt = inv_dai_fmt; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, dai_fmt); + if (ret != 0 && ret != -ENOTSUPP) { + dev_warn(cpu_dai->dev, + "ASoC: Failed to set DAI format: %d\n", ret); + return ret; + } + + return 0; +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; - struct snd_soc_dai_link *dai_link; - int ret, i, order, dai_fmt; + int ret, i, order; mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); @@ -1542,60 +1606,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->num_dapm_routes); for (i = 0; i < card->num_links; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; - dai_link = &card->dai_link[i]; - dai_fmt = dai_link->dai_fmt; - - if (dai_fmt) { - struct snd_soc_dai **codec_dais = rtd->codec_dais; - int j; - - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = codec_dais[j]; - - ret = snd_soc_dai_set_fmt(codec_dai, dai_fmt); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(codec_dai->dev, - "ASoC: Failed to set DAI format: %d\n", - ret); - } - } - - /* If this is a regular CPU link there will be a platform */ - if (dai_fmt && - (dai_link->platform_name || dai_link->platform_of_node)) { - ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, - dai_fmt); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(card->rtd[i].cpu_dai->dev, - "ASoC: Failed to set DAI format: %d\n", - ret); - } else if (dai_fmt) { - /* Flip the polarity for the "CPU" end */ - dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; - switch (dai_link->dai_fmt & - SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; - break; - case SND_SOC_DAIFMT_CBM_CFS: - dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; - break; - case SND_SOC_DAIFMT_CBS_CFM: - dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; - break; - case SND_SOC_DAIFMT_CBS_CFS: - dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - break; - } - - ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, - dai_fmt); - if (ret != 0 && ret != -ENOTSUPP) - dev_warn(card->rtd[i].cpu_dai->dev, - "ASoC: Failed to set DAI format: %d\n", - ret); - } + if (card->dai_link[i].dai_fmt) + snd_soc_runtime_set_dai_fmt(&card->rtd[i], + card->dai_link[i].dai_fmt); } snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), -- 1.8.0

3 4

[alsa-devel] [PATCH 1/2] ASoC: fsl_dma: Remove 'path' field from dma_object struct
by Fabio Estevam 07 Jan '15

07 Jan '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> The 'path' field has not a real purpose, so let's get rid of it. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Buit tested only. sound/soc/fsl/fsl_dma.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 93d7e56..026da0a 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -63,7 +63,6 @@ struct dma_object { struct ccsr_dma_channel __iomem *channel; unsigned int irq; bool assigned; - char path[1]; }; /* @@ -910,7 +909,6 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) return -ENOMEM; } - strcpy(dma->path, np->full_name); dma->dai.ops = &fsl_dma_ops; dma->dai.pcm_new = fsl_dma_new; dma->dai.pcm_free = fsl_dma_free_dma_buffers; -- 1.9.1

3 7

[alsa-devel] [PATCH] ASoC: wm5102: Use put_unaligned_be16
by Vaishali Thakkar 07 Jan '15

07 Jan '15

This patch introduces the use of function put_unaligned_be16. This is done using Coccinelle and semantic patch used is as follows: @a@ typedef u16, __be16, uint16_t; {u16,__be16,uint16_t} e16; identifier tmp; expression ptr; expression y,e; type T; @@ - tmp = cpu_to_be16(y); <+... when != tmp ( - memcpy(ptr, (T)&tmp, $2\|sizeof(u16)\|sizeof(__be16)\|sizeof(uint16_t)\|sizeof(e16)$); + put_unaligned_be16(y,ptr); | - memcpy(ptr, (T)&tmp, ...); + put_unaligned_be16(y,ptr); ) ...+> ? tmp = e @@ type T; identifier a.tmp; @@ - T tmp; ...when != tmp Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/codecs/wm5102.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f602349..fec21f9 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -28,6 +28,7 @@ #include <linux/mfd/arizona/core.h> #include <linux/mfd/arizona/registers.h> +#include <asm/unaligned.h> #include "arizona.h" #include "wm5102.h" @@ -617,11 +618,11 @@ static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - uint16_t data; mutex_lock(&codec->mutex); - data = cpu_to_be16(arizona->dac_comp_coeff); - memcpy(ucontrol->value.bytes.data, &data, sizeof(data)); + + put_unaligned_be16(arizona->dac_comp_coeff, + ucontrol->value.bytes.data); mutex_unlock(&codec->mutex); return 0; -- 1.9.1

3 9

[alsa-devel] [PATCH 1/3] ASoC: rt5670: fix bit definition for ASRC control
by mengdong.lin＠intel.com 07 Jan '15

07 Jan '15

From: Bard Liao <bardliao(a)realtek.com> This patch fixes bit definitions for two ASRC control registers 0x84 and 0x85. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 54933f3..7c37b00 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1023,50 +1023,33 @@ #define RT5670_DMIC_2_M_NOR (0x0 << 8) #define RT5670_DMIC_2_M_ASYN (0x1 << 8) +/* ASRC clock source selection (0x84, 0x85) */ +#define RT5670_CLK_SEL_SYS (0x0) +#define RT5670_CLK_SEL_I2S1_ASRC (0x1) +#define RT5670_CLK_SEL_I2S2_ASRC (0x2) +#define RT5670_CLK_SEL_I2S3_ASRC (0x3) +#define RT5670_CLK_SEL_SYS2 (0x5) +#define RT5670_CLK_SEL_SYS3 (0x6) + /* ASRC Control 2 (0x84) */ -#define RT5670_MDA_L_M_MASK (0x1 << 15) -#define RT5670_MDA_L_M_SFT 15 -#define RT5670_MDA_L_M_NOR (0x0 << 15) -#define RT5670_MDA_L_M_ASYN (0x1 << 15) -#define RT5670_MDA_R_M_MASK (0x1 << 14) -#define RT5670_MDA_R_M_SFT 14 -#define RT5670_MDA_R_M_NOR (0x0 << 14) -#define RT5670_MDA_R_M_ASYN (0x1 << 14) -#define RT5670_MAD_L_M_MASK (0x1 << 13) -#define RT5670_MAD_L_M_SFT 13 -#define RT5670_MAD_L_M_NOR (0x0 << 13) -#define RT5670_MAD_L_M_ASYN (0x1 << 13) -#define RT5670_MAD_R_M_MASK (0x1 << 12) -#define RT5670_MAD_R_M_SFT 12 -#define RT5670_MAD_R_M_NOR (0x0 << 12) -#define RT5670_MAD_R_M_ASYN (0x1 << 12) -#define RT5670_ADC_M_MASK (0x1 << 11) -#define RT5670_ADC_M_SFT 11 -#define RT5670_ADC_M_NOR (0x0 << 11) -#define RT5670_ADC_M_ASYN (0x1 << 11) -#define RT5670_STO_DAC_M_MASK (0x1 << 5) -#define RT5670_STO_DAC_M_SFT 5 -#define RT5670_STO_DAC_M_NOR (0x0 << 5) -#define RT5670_STO_DAC_M_ASYN (0x1 << 5) -#define RT5670_I2S1_R_D_MASK (0x1 << 4) -#define RT5670_I2S1_R_D_SFT 4 -#define RT5670_I2S1_R_D_DIS (0x0 << 4) -#define RT5670_I2S1_R_D_EN (0x1 << 4) -#define RT5670_I2S2_R_D_MASK (0x1 << 3) -#define RT5670_I2S2_R_D_SFT 3 -#define RT5670_I2S2_R_D_DIS (0x0 << 3) -#define RT5670_I2S2_R_D_EN (0x1 << 3) -#define RT5670_PRE_SCLK_MASK (0x3) -#define RT5670_PRE_SCLK_SFT 0 -#define RT5670_PRE_SCLK_512 (0x0) -#define RT5670_PRE_SCLK_1024 (0x1) -#define RT5670_PRE_SCLK_2048 (0x2) +#define RT5670_DA_STO_CLK_SEL_MASK (0xf << 12) +#define RT5670_DA_STO_CLK_SEL_SFT 12 +#define RT5670_DA_MONOL_CLK_SEL_MASK (0xf << 8) +#define RT5670_DA_MONOL_CLK_SEL_SFT 8 +#define RT5670_DA_MONOR_CLK_SEL_MASK (0xf << 4) +#define RT5670_DA_MONOR_CLK_SEL_SFT 4 +#define RT5670_AD_STO1_CLK_SEL_MASK (0xf << 0) +#define RT5670_AD_STO1_CLK_SEL_SFT 0 /* ASRC Control 3 (0x85) */ -#define RT5670_I2S1_RATE_MASK (0xf << 12) -#define RT5670_I2S1_RATE_SFT 12 -#define RT5670_I2S2_RATE_MASK (0xf << 8) -#define RT5670_I2S2_RATE_SFT 8 +#define RT5670_UP_CLK_SEL_MASK (0xf << 12) +#define RT5670_UP_CLK_SEL_SFT 12 +#define RT5670_DOWN_CLK_SEL_MASK (0xf << 8) +#define RT5670_DOWN_CLK_SEL_SFT 8 +#define RT5670_AD_MONOL_CLK_SEL_MASK (0xf << 4) +#define RT5670_AD_MONOL_CLK_SEL_SFT 4 +#define RT5670_AD_MONOR_CLK_SEL_MASK (0xf << 0) +#define RT5670_AD_MONOR_CLK_SEL_SFT 0 /* ASRC Control 4 (0x89) */ #define RT5670_I2S1_PD_MASK (0x7 << 12) -- 1.9.1

4 10

[alsa-devel] [PATCH v2] ASoC: Intel: Add NULL checks for the stream pointer
by Jie Yang 07 Jan '15

07 Jan '15

We should not send IPC stream commands to FW when the stream is NULL, dereference the NULL pointer may also occur without precheck. Here add NULL pointer checks for these stream APIs. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 57d6e89..51fb15c 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1233,6 +1233,11 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) struct sst_dsp *sst = hsw->dsp; unsigned long flags; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to free, ignore it.\n"); + return 0; + } + /* dont free DSP streams that are not commited */ if (!stream->commited) goto out; @@ -1420,6 +1425,16 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) u32 header; int ret; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to commit, ignore it.\n"); + return 0; + } + + if (stream->commited) { + dev_warn(hsw->dev, "warning: stream is already committed, ignore it.\n"); + return 0; + } + trace_ipc_request("stream alloc", stream->host_id); header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM); @@ -1548,6 +1563,11 @@ int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream, { int ret; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to pause, ignore it.\n"); + return 0; + } + trace_ipc_request("stream pause", stream->reply.stream_hw_id); ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE, @@ -1564,6 +1584,11 @@ int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, { int ret; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to resume, ignore it.\n"); + return 0; + } + trace_ipc_request("stream resume", stream->reply.stream_hw_id); ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME, @@ -1579,6 +1604,11 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) { int ret, tries = 10; + if (!stream) { + dev_warn(hsw->dev, "warning: stream is NULL, no stream to reset, ignore it.\n"); + return 0; + } + /* dont reset streams that are not commited */ if (!stream->commited) return 0; -- 1.9.1

2 1

[alsa-devel] [PATCH, RFC] MIPS: jz4740: use dma filter function
by Arnd Bergmann 07 Jan '15

07 Jan '15

As discussed on the topic of shmobile DMA today, jz4740 is the only user of the slave_id field in dma_slave_config besides shmobile. This use is really incompatible with the way that other drivers use the dmaengine API, so we should get rid of it. This adds a trivial filter function that uses the filter param to pass the dma type, and uses that in both drivers. With this patch and the respective shmobile change, we can remove the slave_id field from dma_slave_config. To make the drivers more generic, another patch would be required that passes the filter function and the argument through platform_data, but that can be done later. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/mips/include/asm/mach-jz4740/dma.h | 7 +++++++ drivers/dma/dma-jz4740.c | 11 +++++++++-- drivers/mmc/host/jz4740_mmc.c | 8 ++++---- sound/soc/jz4740/jz4740-i2s.c | 18 ++++++++++++------ 4 files changed, 32 insertions(+), 12 deletions(-) completely untested diff --git a/arch/mips/include/asm/mach-jz4740/dma.h b/arch/mips/include/asm/mach-jz4740/dma.h index 14ecc5313d2d..ece7e39cdacb 100644 --- a/arch/mips/include/asm/mach-jz4740/dma.h +++ b/arch/mips/include/asm/mach-jz4740/dma.h @@ -16,6 +16,13 @@ #ifndef __ASM_MACH_JZ4740_DMA_H__ #define __ASM_MACH_JZ4740_DMA_H__ +#include <linux/dmaengine.h> + +static inline bool jz4740_dma_filter_fn(struct dma_chan *chan, void *filter_param) +{ + chan->private = filter_param; +} + enum jz4740_dma_request_type { JZ4740_DMA_TYPE_AUTO_REQUEST = 8, JZ4740_DMA_TYPE_UART_TRANSMIT = 20, diff --git a/drivers/dma/dma-jz4740.c b/drivers/dma/dma-jz4740.c index bdeafeefa5f6..f02e60e8e5cc 100644 --- a/drivers/dma/dma-jz4740.c +++ b/drivers/dma/dma-jz4740.c @@ -119,6 +119,7 @@ struct jz4740_dma_desc { struct jz4740_dmaengine_chan { struct virt_dma_chan vchan; unsigned int id; + unsigned int slave; dma_addr_t fifo_addr; unsigned int transfer_shift; @@ -220,6 +221,11 @@ static int jz4740_dma_slave_config(struct dma_chan *c, enum jz4740_dma_flags flags; uint32_t cmd; + /* deprecated: all drivers should request the channel with + * the correct slave argument */ + if (!chan->slave) + chan->slave = config->slave_id; + switch (config->direction) { case DMA_MEM_TO_DEV: flags = JZ4740_DMA_SRC_AUTOINC; @@ -265,8 +271,7 @@ static int jz4740_dma_slave_config(struct dma_chan *c, jz4740_dma_write(dmadev, JZ_REG_DMA_CMD(chan->id), cmd); jz4740_dma_write(dmadev, JZ_REG_DMA_STATUS_CTRL(chan->id), 0); - jz4740_dma_write(dmadev, JZ_REG_DMA_REQ_TYPE(chan->id), - config->slave_id); + jz4740_dma_write(dmadev, JZ_REG_DMA_REQ_TYPE(chan->id), chan->slave); return 0; } @@ -513,6 +518,8 @@ static enum dma_status jz4740_dma_tx_status(struct dma_chan *c, static int jz4740_dma_alloc_chan_resources(struct dma_chan *c) { + struct jz4740_dmaengine_chan *chan = to_jz4740_dma_chan(c); + chan->slave = (unsigned long)c->private; return 0; } diff --git a/drivers/mmc/host/jz4740_mmc.c b/drivers/mmc/host/jz4740_mmc.c index 76e8bce6f46e..a6680959ce83 100644 --- a/drivers/mmc/host/jz4740_mmc.c +++ b/drivers/mmc/host/jz4740_mmc.c @@ -178,13 +178,15 @@ static int jz4740_mmc_acquire_dma_channels(struct jz4740_mmc_host *host) dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); - host->dma_tx = dma_request_channel(mask, NULL, host); + host->dma_tx = dma_request_channel(mask, jz4740_dma_filter_fn, + (void*)JZ4740_DMA_TYPE_MMC_TRANSMIT); if (!host->dma_tx) { dev_err(mmc_dev(host->mmc), "Failed to get dma_tx channel\n"); return -ENODEV; } - host->dma_rx = dma_request_channel(mask, NULL, host); + host->dma_rx = dma_request_channel(mask, jz4740_dma_filter_fn, + (void*)JZ4740_DMA_TYPE_MMC_RECEIVE); if (!host->dma_rx) { dev_err(mmc_dev(host->mmc), "Failed to get dma_rx channel\n"); goto free_master_write; @@ -283,12 +285,10 @@ static int jz4740_mmc_start_dma_transfer(struct jz4740_mmc_host *host, if (data->flags & MMC_DATA_WRITE) { conf.direction = DMA_MEM_TO_DEV; conf.dst_addr = host->mem_res->start + JZ_REG_MMC_TXFIFO; - conf.slave_id = JZ4740_DMA_TYPE_MMC_TRANSMIT; chan = host->dma_tx; } else { conf.direction = DMA_DEV_TO_MEM; conf.src_addr = host->mem_res->start + JZ_REG_MMC_RXFIFO; - conf.slave_id = JZ4740_DMA_TYPE_MMC_RECEIVE; chan = host->dma_rx; } diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index d3d45c6f064f..7a60c075bd81 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -31,10 +31,12 @@ #include <sound/initval.h> #include <sound/dmaengine_pcm.h> +#include <asm/mach-jz4740/dma.h> + #include "jz4740-i2s.h" -#define JZ4740_DMA_TYPE_AIC_TRANSMIT 24 -#define JZ4740_DMA_TYPE_AIC_RECEIVE 25 +#define JZ4740_DMA_TYPE_AIC_TRANSMIT (void *)24ul +#define JZ4740_DMA_TYPE_AIC_RECEIVE (void *)25ul #define JZ_REG_AIC_CONF 0x00 #define JZ_REG_AIC_CTRL 0x04 @@ -337,13 +339,13 @@ static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) /* Playback */ dma_data = &i2s->playback_dma_data; dma_data->maxburst = 16; - dma_data->slave_id = JZ4740_DMA_TYPE_AIC_TRANSMIT; + dma_data->filter_data = JZ4740_DMA_TYPE_AIC_TRANSMIT; dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; /* Capture */ dma_data = &i2s->capture_dma_data; dma_data->maxburst = 16; - dma_data->slave_id = JZ4740_DMA_TYPE_AIC_RECEIVE; + dma_data->filter_data = JZ4740_DMA_TYPE_AIC_RECEIVE; dma_data->addr = i2s->phys_base + JZ_REG_AIC_FIFO; } @@ -415,6 +417,10 @@ static const struct snd_soc_component_driver jz4740_i2s_component = { .name = "jz4740-i2s", }; +static struct snd_dmaengine_pcm_config jz4740_i2s_pcm_config = { + .filter_fn = jz4740_dma_filter_fn, +}; + static int jz4740_i2s_dev_probe(struct platform_device *pdev) { struct jz4740_i2s *i2s; @@ -447,8 +453,8 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) if (ret) return ret; - return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, - SND_DMAENGINE_PCM_FLAG_COMPAT); + return devm_snd_dmaengine_pcm_register(&pdev->dev, + jz4740_i2s_pcm_config, SND_DMAENGINE_PCM_FLAG_COMPAT); } static struct platform_driver jz4740_i2s_driver = {

3 7

[alsa-devel] endianness problems in fireworks/bebob_maudio
by Clemens Ladisch 07 Jan '15

07 Jan '15

Hi, sparse complains: sound/firewire/fireworks/fireworks_transaction.c:127:18: warning: restricted __be32 degrades to integer t = (struct snd_efw_transaction *)data; length = min_t(size_t, t->length * sizeof(t->length), length); 't->length' is still a big-endian value. This means that the driver ends up always using 'length'. sound/firewire/bebob/bebob_maudio.c:100:17: warning: incorrect type in initializer (different base types) sound/firewire/bebob/bebob_maudio.c:100:17: expected restricted __be32 sound/firewire/bebob/bebob_maudio.c:100:17: got int __be32 cues[3] = { MAUDIO_BOOTLOADER_CUE1, MAUDIO_BOOTLOADER_CUE2, MAUDIO_BOOTLOADER_CUE3 }; rcode = fw_run_transaction(device->card, TCODE_WRITE_BLOCK_REQUEST, device->node_id, device->generation, device->max_speed, BEBOB_ADDR_REG_REQ, cues, sizeof(cues)); The three MAUDIO_BOOTLOADER_CUEx values will end up as a different byte sequence on big-endian machines. The simplest way to have these twelve bytes unchanged on the bus is to have a twelve-byte array in the driver. Regards, Clemens

3 5

[alsa-devel] [PATCH v2] ALSA: hda - Enable mic mute hotkey and LEDs for an HP machine
by David Henningsson 07 Jan '15

07 Jan '15

On this machine, the mic mute hotkey is connected to GPIO2. We therefore create an extra input device for this key. Also enable LEDs connected to GPIO3 and GPIO4. (Note: if this patch is backported to older kernels, where KEY_MIC_MUTE is not available, change the KEY_MIC_MUTE to KEY_F20, which was previously used for mic mute keys.) BugLink: https://bugs.launchpad.net/bugs/1408295 Tested-by: Keng-Yu Lin <keng-yu.lin(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Changes since v1 (which was by mistake not posted to alsa-devel) - fixes according to Takashi's review comments sound/pci/hda/patch_realtek.c | 88 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 88 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 65f1f4e..09d2131 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -29,6 +29,7 @@ #include <linux/pci.h> #include <linux/dmi.h> #include <linux/module.h> +#include <linux/input.h> #include <sound/core.h> #include <sound/jack.h> #include "hda_codec.h" @@ -120,6 +121,9 @@ struct alc_spec { hda_nid_t pll_nid; unsigned int pll_coef_idx, pll_coef_bit; unsigned int coef0; +#if IS_ENABLED(CONFIG_INPUT) + struct input_dev *kb_dev; +#endif }; /* @@ -3472,6 +3476,84 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec, } } +#if IS_ENABLED(CONFIG_INPUT) +static void gpio2_mic_hotkey_event(struct hda_codec *codec, + struct hda_jack_callback *event) +{ + struct alc_spec *spec = codec->spec; + + /* GPIO2 just toggles on a keypress/keyrelease cycle. Therefore + send both key on and key off event for every interrupt. */ + input_report_key(spec->kb_dev, KEY_MICMUTE, 1); + input_sync(spec->kb_dev); + input_report_key(spec->kb_dev, KEY_MICMUTE, 0); + input_sync(spec->kb_dev); +} +#endif + +static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ +#if IS_ENABLED(CONFIG_INPUT) + /* GPIO1 = set according to SKU external amp + GPIO2 = mic mute hotkey + GPIO3 = mute LED + GPIO4 = mic mute LED */ + static const struct hda_verb gpio_init[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x1e }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x1a }, + { 0x01, AC_VERB_SET_GPIO_DATA, 0x02 }, + {} + }; + + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->kb_dev = input_allocate_device(); + if (!spec->kb_dev) { + codec_err(codec, "Out of memory (input_allocate_device)\n"); + return; + } + spec->kb_dev->name = "Microphone Mute Button"; + spec->kb_dev->evbit[0] = BIT_MASK(EV_KEY); + spec->kb_dev->keybit[BIT_WORD(KEY_MICMUTE)] = BIT_MASK(KEY_MICMUTE); + if (input_register_device(spec->kb_dev)) { + codec_err(codec, "input_register_device failed\n"); + input_free_device(spec->kb_dev); + spec->kb_dev = NULL; + return; + } + + snd_hda_add_verbs(codec, gpio_init); + snd_hda_codec_write_cache(codec, codec->afg, 0, + AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x04); + snd_hda_jack_detect_enable_callback(codec, codec->afg, + gpio2_mic_hotkey_event); + + spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; + spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; + spec->gpio_led = 0; + spec->mute_led_polarity = 0; + spec->gpio_mute_led_mask = 0x08; + spec->gpio_mic_led_mask = 0x10; + return; + } + + if (!spec->kb_dev) + return; + + switch (action) { + case HDA_FIXUP_ACT_PROBE: + spec->init_amp = ALC_INIT_DEFAULT; + break; + case HDA_FIXUP_ACT_FREE: + input_unregister_device(spec->kb_dev); + input_free_device(spec->kb_dev); + spec->kb_dev = NULL; + } +#endif +} + static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4341,6 +4423,7 @@ enum { ALC282_FIXUP_ASPIRE_V5_PINS, ALC280_FIXUP_HP_GPIO4, ALC286_FIXUP_HP_GPIO_LED, + ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, }; static const struct hda_fixup alc269_fixups[] = { @@ -4814,6 +4897,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc286_fixup_hp_gpio_led, }, + [ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc280_fixup_hp_gpio2_mic_hotkey, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -4843,6 +4930,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x225f, "HP", ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY), /* ALC282 */ SND_PCI_QUIRK(0x103c, 0x2210, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2214, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Print codec->chip_name in autoconfig
by David Henningsson 07 Jan '15

07 Jan '15

This will help end users figure out what HDA codec they have. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 1ede822..6ec3775 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -409,10 +409,10 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, /* * debug prints of the parsed results */ - codec_info(codec, "autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n", - cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1], - cfg->line_out_pins[2], cfg->line_out_pins[3], - cfg->line_out_pins[4], + codec_info(codec, "autoconfig for %s: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n", + codec->chip_name, cfg->line_outs, cfg->line_out_pins[0], + cfg->line_out_pins[1], cfg->line_out_pins[2], + cfg->line_out_pins[3], cfg->line_out_pins[4], cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" : (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ? "speaker" : "line")); -- 1.9.1

2 3

[alsa-devel] [PATCH] ALSA: fm801: PCI core handles power state for us
by Andy Shevchenko 07 Jan '15

07 Jan '15

There is no need to duplicate the work that is already done in the PCI driver core. The patch removes excerpts from suspend and resume callbacks. While here amend a definition of the snd_fm801_pm. The PM core has the stubs for non-PM_SLEEP cases, thus we can always define the variable. Signed-off-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> --- sound/pci/fm801.c | 22 ++-------------------- 1 file changed, 2 insertions(+), 20 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 9a2122f..dcda3c1 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1384,7 +1384,6 @@ static unsigned char saved_regs[] = { static int snd_fm801_suspend(struct device *dev) { - struct pci_dev *pci = to_pci_dev(dev); struct snd_card *card = dev_get_drvdata(dev); struct fm801 *chip = card->private_data; int i; @@ -1396,29 +1395,15 @@ static int snd_fm801_suspend(struct device *dev) for (i = 0; i < ARRAY_SIZE(saved_regs); i++) chip->saved_regs[i] = inw(chip->port + saved_regs[i]); /* FIXME: tea575x suspend */ - - pci_disable_device(pci); - pci_save_state(pci); - pci_set_power_state(pci, PCI_D3hot); return 0; } static int snd_fm801_resume(struct device *dev) { - struct pci_dev *pci = to_pci_dev(dev); struct snd_card *card = dev_get_drvdata(dev); struct fm801 *chip = card->private_data; int i; - pci_set_power_state(pci, PCI_D0); - pci_restore_state(pci); - if (pci_enable_device(pci) < 0) { - dev_err(dev, "pci_enable_device failed, disabling device\n"); - snd_card_disconnect(card); - return -EIO; - } - pci_set_master(pci); - snd_fm801_chip_init(chip, 1); snd_ac97_resume(chip->ac97); snd_ac97_resume(chip->ac97_sec); @@ -1428,12 +1413,9 @@ static int snd_fm801_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } +#endif /* CONFIG_PM_SLEEP */ static SIMPLE_DEV_PM_OPS(snd_fm801_pm, snd_fm801_suspend, snd_fm801_resume); -#define SND_FM801_PM_OPS &snd_fm801_pm -#else -#define SND_FM801_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP */ static struct pci_driver fm801_driver = { .name = KBUILD_MODNAME, @@ -1441,7 +1423,7 @@ static struct pci_driver fm801_driver = { .probe = snd_card_fm801_probe, .remove = snd_card_fm801_remove, .driver = { - .pm = SND_FM801_PM_OPS, + .pm = &snd_fm801_pm, }, }; -- 1.8.3.101.g727a46b

2 5

[alsa-devel] [PATCH] ASoC: Intel: Add NULL checks for the stream pointer
by Jie Yang 07 Jan '15

07 Jan '15

We should not send IPC stream commands to FW when the stream is NULL, dereference the NULL pointer may also occur without precheck. Here add NULL pointer checks for these stream APIs. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 57d6e89..3fd9838 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1233,6 +1233,9 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) struct sst_dsp *sst = hsw->dsp; unsigned long flags; + if (!stream) + return 0; + /* dont free DSP streams that are not commited */ if (!stream->commited) goto out; @@ -1420,6 +1423,9 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) u32 header; int ret; + if (stream->commited) + return 0; + trace_ipc_request("stream alloc", stream->host_id); header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM); @@ -1548,6 +1554,9 @@ int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream, { int ret; + if (!stream) + return 0; + trace_ipc_request("stream pause", stream->reply.stream_hw_id); ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE, @@ -1564,6 +1573,9 @@ int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, { int ret; + if (!stream) + return 0; + trace_ipc_request("stream resume", stream->reply.stream_hw_id); ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME, @@ -1579,6 +1591,9 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) { int ret, tries = 10; + if (!stream) + return 0; + /* dont reset streams that are not commited */ if (!stream->commited) return 0; -- 1.9.1

3 3

[alsa-devel] [PATCH] ALSA: hda - Add new GPU codec ID 0x10de0072 to snd-hda
by Aaron Plattner 07 Jan '15

07 Jan '15

Vendor ID 0x10de0072 is used by a yet-to-be-named GPU chip. Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> --- Do you think it would make sense to add some kind of wildcard match for 0x10de* -> patch_nvhdmi? sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5f13d2d18079..b422e406a9cb 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3353,6 +3353,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de0070, .name = "GPU 70 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0071, .name = "GPU 71 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0072, .name = "GPU 72 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, { .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -3413,6 +3414,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0060"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de0070"); MODULE_ALIAS("snd-hda-codec-id:10de0071"); +MODULE_ALIAS("snd-hda-codec-id:10de0072"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); MODULE_ALIAS("snd-hda-codec-id:11069f81"); -- 2.2.1

2 1

[alsa-devel] Intel HDA / ca0132: Multichannel + S/PDIF support
by Stefan Oltmanns 06 Jan '15

06 Jan '15

Hi, I own a Gigabyte G1.Sniper M3 Z77 mainboard with ca0132 onboard audio. Currently only stereo out and headphone out (at the back, always same signal) work by default. I was able to fix S/PDIF out by changing the headphone pin in function ca0132_config to something below 0x0d, like: spec->out_pins[1] = 0x0a; No idea why this fixes S/PDIF, but no other effects noticed. No idea if it´s going to break something on other devices. Original value is 0x10. In this function also 3 DACs are listed. The board has a total of 5 stereo outputs: -Frontpanel (HDA pin connector on mainboard) 4x 3,5mm output at backpanel: -Stereo out -Headphone out -Rear out -Center/LFE out I tried to increase the number of channels and create additional outputs and connected to the other DACs. The maximum I´m able to is to get different signals out of stereo out and front headphone (not useful, just some weird mixes) or same signal from all 3 outputs. Rear and c/lfe out always no signal. During initialization snd_hda_parse_pin_def_config is called and two outputs are found. I tried to override that and change spec->autocfg after that function is called, no success. Also no matter what I change there, a little later I get this kernel messages: kernel: [ 2.642972] input: HDA Intel PCH Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 kernel: [ 2.643142] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input7 kernel: [ 2.644932] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1b.0/sound/card0/input6 kernel: [ 2.645050] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input5 I could not find the code where this messages are generated, where is that? Any ideas what I could/should try to get multichannel support working? Is there any kind of documentation on the ca0132 that could be helpful? Best regards Stefan

3 8

[alsa-devel] [PATCH] ASoC: tegra: Add platform driver for rt5677 audio codec
by Anatol Pomozov 06 Jan '15

06 Jan '15

The driver supports NVIDIA Tegra Ryu board Sponsored: Google ChromeOS Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../bindings/sound/nvidia,tegra-audio-rt5677.txt | 67 ++++ sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_rt5677.c | 351 +++++++++++++++++++++ 4 files changed, 430 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt create mode 100644 sound/soc/tegra/tegra_rt5677.c diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt new file mode 100644 index 0000000..a4589cd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt @@ -0,0 +1,67 @@ +NVIDIA Tegra audio complex, with RT5677 CODEC + +Required properties: +- compatible : "nvidia,tegra-audio-rt5677" +- clocks : Must contain an entry for each entry in clock-names. + See ../clocks/clock-bindings.txt for details. +- clock-names : Must include the following entries: + - pll_a + - pll_a_out0 + - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) +- nvidia,model : The user-visible name of this sound complex. +- nvidia,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the RT5677's pins (as documented in its binding), and the jacks + on the board: + + * Headphone + * Speaker + * Headset Mic + * Internal Mic 1 + * Internal Mic 2 + +- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's + connected to the CODEC. +- nvidia,audio-codec : The phandle of the RT5677 audio codec. This binding + assumes that AIF1 on the CODEC is connected to Tegra. + +Optional properties: +- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in +- nvidia,hp-en-gpios : The GPIO that enables headphone amplifier +- nvidia,mic-present-gpios: The GPIO that mic jack is plugged in +- nvidia,dmic-clk-en-gpios : The GPIO that gates DMIC clock signal + +Example: + +sound { + compatible = "nvidia,tegra-audio-rt5677-ryu", + "nvidia,tegra-audio-rt5677"; + nvidia,model = "NVIDIA Tegra Ryu"; + + nvidia,audio-routing = + "Headphone", "LOUT2", + "Headphone", "LOUT1", + "Headset Mic", "MICBIAS1", + "IN1P", "Headset Mic", + "IN1N", "Headset Mic", + "DMIC L1", "Internal Mic 1", + "DMIC R1", "Internal Mic 1", + "DMIC L2", "Internal Mic 2", + "DMIC R2", "Internal Mic 2", + "Speaker", "PDM1L", + "Speaker", "PDM1R"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5677>; + + nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; + nvidia,mic-present-gpios = <&gpio TEGRA_GPIO(O, 5) GPIO_ACTIVE_LOW>; + nvidia,hp-en-gpios = <&rt5677 1 GPIO_ACTIVE_HIGH>; + nvidia,dmic-clk-en-gpios = <&rt5677 2 GPIO_ACTIVE_HIGH>; + + clocks = <&tegra_car TEGRA124_CLK_PLL_A>, + <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA124_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; +}; diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 31198cf7..a6768f8 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -128,3 +128,13 @@ config SND_SOC_TEGRA_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra boards using the MAX98090 codec, such as Venice2. + +config SND_SOC_TEGRA_RT5677 + tristate "SoC Audio support for Tegra boards using a RT5677 codec" + depends on SND_SOC_TEGRA && I2C && GPIOLIB + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_RT5677 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the RT5677 codec, such as Ryu. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 5ae588c..9171655 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -19,6 +19,7 @@ obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support snd-soc-tegra-rt5640-objs := tegra_rt5640.o +snd-soc-tegra-rt5677-objs := tegra_rt5677.o snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-wm9712-objs := tegra_wm9712.o @@ -27,6 +28,7 @@ snd-soc-tegra-alc5632-objs := tegra_alc5632.o snd-soc-tegra-max98090-objs := tegra_max98090.o obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o +obj-$(CONFIG_SND_SOC_TEGRA_RT5677) += snd-soc-tegra-rt5677.o obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c new file mode 100644 index 0000000..6e98d10 --- /dev/null +++ b/sound/soc/tegra/tegra_rt5677.c @@ -0,0 +1,351 @@ +/* +* tegra_rt5677.c - Tegra machine ASoC driver for boards using RT5677 codec. + * + * Copyright (c) 2014, The Chromium OS Authors. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + * Based on code copyright/by: + * + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * Copyright 2007 Wolfson Microelectronics PLC. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/rt5677.h" + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-rt5677" + +struct tegra_rt5677 { + struct tegra_asoc_utils_data util_data; + int gpio_hp_det; + int gpio_hp_en; + int gpio_mic_present; + int gpio_dmic_clk_en; +}; + +static int tegra_rt5677_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk, err; + + srate = params_rate(params); + mclk = 256 * srate; + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static int tegra_rt5677_event_hp(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (!gpio_is_valid(machine->gpio_hp_en)) + return 0; + + gpio_set_value_cansleep(machine->gpio_hp_en, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static struct snd_soc_ops tegra_rt5677_ops = { + .hw_params = tegra_rt5677_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_rt5677_hp_jack; + +static struct snd_soc_jack_pin tegra_rt5677_hp_jack_pins = { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, +}; +static struct snd_soc_jack_gpio tegra_rt5677_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, +}; + +static struct snd_soc_jack tegra_rt5677_mic_jack; + +static struct snd_soc_jack_pin tegra_rt5677_mic_jack_pins = { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, +}; + +static struct snd_soc_jack_gpio tegra_rt5677_mic_jack_gpio = { + .name = "Headset Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, + .invert = 1 +}; + +static const struct snd_soc_dapm_widget tegra_rt5677_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_HP("Headphone", tegra_rt5677_event_hp), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Internal Mic 1", NULL), + SND_SOC_DAPM_MIC("Internal Mic 2", NULL), +}; + +static const struct snd_kcontrol_new tegra_rt5677_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic 1"), + SOC_DAPM_PIN_SWITCH("Internal Mic 2"), +}; + +static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_rt5677_hp_jack); + snd_soc_jack_add_pins(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_pins); + + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_rt5677_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + + snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_rt5677_mic_jack); + snd_soc_jack_add_pins(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_pins); + + if (gpio_is_valid(machine->gpio_mic_present)) { + tegra_rt5677_mic_jack_gpio.gpio = machine->gpio_mic_present; + snd_soc_jack_add_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + + return 0; +} + +static int tegra_rt5677_card_remove(struct snd_soc_card *card) +{ + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + if (gpio_is_valid(machine->gpio_mic_present)) { + snd_soc_jack_free_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + return 0; +} + +static struct snd_soc_dai_link tegra_rt5677_dai = { + .name = "RT5677", + .stream_name = "RT5677 PCM", + .codec_dai_name = "rt5677-aif1", + .init = tegra_rt5677_asoc_init, + .ops = &tegra_rt5677_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_rt5677 = { + .name = "tegra-rt5677", + .owner = THIS_MODULE, + .remove = tegra_rt5677_card_remove, + .dai_link = &tegra_rt5677_dai, + .num_links = 1, + .controls = tegra_rt5677_controls, + .num_controls = ARRAY_SIZE(tegra_rt5677_controls), + .dapm_widgets = tegra_rt5677_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_rt5677_dapm_widgets), + .fully_routed = true, +}; + +static int tegra_rt5677_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_tegra_rt5677; + struct tegra_rt5677 *machine; + int ret; + +#if defined(CONFIG_MODULES) && defined(MODULE) + request_module("snd_soc_tegra30_i2s"); +#endif + + machine = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_rt5677), GFP_KERNEL); + if (!machine) + return -ENOMEM; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (machine->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_mic_present = of_get_named_gpio(np, + "nvidia,mic-present-gpios", 0); + if (machine->gpio_mic_present == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_hp_en = of_get_named_gpio(np, "nvidia,hp-en-gpios", 0); + if (machine->gpio_hp_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_hp_en)) { + ret = devm_gpio_request_one(&pdev->dev, machine->gpio_hp_en, + GPIOF_OUT_INIT_LOW, "hp_en"); + if (ret) { + dev_err(card->dev, "cannot get hp_en gpio\n"); + return ret; + } + } + + machine->gpio_dmic_clk_en = of_get_named_gpio(np, + "nvidia,dmic-clk-en-gpios", 0); + if (machine->gpio_dmic_clk_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_dmic_clk_en)) { + ret = devm_gpio_request_one(&pdev->dev, + machine->gpio_dmic_clk_en, + GPIOF_OUT_INIT_HIGH, "dmic_clk_en"); + if (ret) { + dev_err(card->dev, "cannot get dmic_clk_en gpio\n"); + return ret; + } + } + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_rt5677_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); + if (!tegra_rt5677_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_rt5677_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!tegra_rt5677_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + tegra_rt5677_dai.platform_of_node = tegra_rt5677_dai.cpu_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&machine->util_data); +err: + return ret; +} + +static int tegra_rt5677_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_rt5677_of_match[] = { + { .compatible = "nvidia,tegra-audio-rt5677", }, + {}, +}; + +static struct platform_driver tegra_rt5677_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_rt5677_of_match, + }, + .probe = tegra_rt5677_probe, + .remove = tegra_rt5677_remove, +}; +module_platform_driver(tegra_rt5677_driver); + +MODULE_AUTHOR("Anatol Pomozov <anatol(a)google.com>"); +MODULE_DESCRIPTION("Tegra+RT5677 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_rt5677_of_match); -- 2.2.0.rc0.207.ga3a616c

2 5

[alsa-devel] [PATCH] sound: soc: atmel sam9g20_wm8731: remove useless include
by Alexandre Belloni 06 Jan '15

06 Jan '15

A mach/ header is included but never used. Simply remove it. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/sam9g20_wm8731.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 66b66d0e7514..531728975bbb 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -47,7 +47,6 @@ #include <sound/soc.h> #include <asm/mach-types.h> -#include <mach/hardware.h> #include "../codecs/wm8731.h" #include "atmel-pcm.h" -- 2.1.0

3 3

[alsa-devel] [PATCH 1/4] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 06 Jan '15

06 Jan '15

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK. When current bias_level is SND_SOC_BIAS_ON, it is preparing for a transition away from ON. In this case, disable the codec mclk. When current bias_level is not SND_SOC_BIAS_ON, it preparing for a transition to ON. In this case, enable the codec mclk. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 40 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 39 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae..1a5f47b 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -117,6 +118,7 @@ static bool wm8960_volatile(struct device *dev, unsigned int reg) } struct wm8960_priv { + struct clk *mclk; struct regmap *regmap; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); @@ -618,6 +620,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -626,6 +629,20 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* Set VMID to 2x50k */ snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); + + if (!IS_ERR(wm8960->mclk)) { + if (codec->dapm.bias_level == SND_SOC_BIAS_ON) + clk_disable_unprepare(wm8960->mclk); + else { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", + ret); + return ret; + } + } + } break; case SND_SOC_BIAS_STANDBY: @@ -661,6 +678,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, /* Disable VMID and VREF, let them discharge */ snd_soc_write(codec, WM8960_POWER1, 0); + msleep(600); break; } @@ -674,7 +692,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int reg; + int reg, ret; switch (level) { case SND_SOC_BIAS_ON: @@ -715,9 +733,22 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, WM8960_VREF, WM8960_VREF); msleep(100); + + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", + ret); + return ret; + } + } break; case SND_SOC_BIAS_ON: + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); + /* Enable anti-pop mode */ snd_soc_update_bits(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | @@ -728,6 +759,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, /* Disable VMID and VREF */ snd_soc_update_bits(codec, WM8960_POWER1, WM8960_VREF | WM8960_VMID_MASK, 0); + break; case SND_SOC_BIAS_OFF: @@ -1002,6 +1034,12 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (wm8960 == NULL) return -ENOMEM; + wm8960->mclk = devm_clk_get(&i2c->dev, "codec_mclk"); + if (IS_ERR(wm8960->mclk)) { + if (PTR_ERR(wm8960->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap); if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); -- 1.9.1

3 7

[alsa-devel] [PATCH 00/21] ASoC: Use static DAI format setup
by Lars-Peter Clausen 06 Jan '15

06 Jan '15

There is still a rather large amount of board driver which manually call snd_soc_dai_set_fmt() for both the CPU and the CODEC DAI, while pretty much all of them can use static DAI setup by setting the dai_fmt field in the snd_soc_dai_link struct. Using the later method results in shorter and cleaner code and will also make it easier to eventually switch to a component type independent way to specify the clock master/slave settings (Right now the meaning of the value depends on whether a DAI belongs to a CODEC or not). This series converts all the trivial cases of snd_soc_dai_set_fmt() users to use the dai_fmt field instead. - Lars Lars-Peter Clausen (21): ASoC: sam9g20_wm8731: Use static DAI format setup ASoC: db1200: Use static DAI format setup ASoC: eureka-tlv320: Use static DAI format setup ASoC: mx27vis-aci32x4: Use static DAI format setup ASoC: wm1133-ev1: Use static DAI format setup ASoC: mxs-sgtl5000: Use static DAI format setup ASoC: ams-delta: Use static DAI format setup ASoC: raumfeld: Use static DAI format setup ASoC: zylonite: Use static DAI format setup ASoC: goni_wm8994: Use static DAI format setup ASoC: h1940_uda1380: Use static DAI format setup ASoC: jive_wm8750: Use static DAI format setup ASoC: neo1973_wm8753: Use static DAI format setup ASoC: rx1950_uda1380: Use static DAI format setup ASoC: s3c24xx_simtec: Use static DAI format setup ASoC: s3c24xx_uda134x: Use static DAI format setup ASoC: smartq_wm8987: Use static DAI format setup ASoC: smdk_wm8580: Use static DAI format setup ASoC: smdk_wm8570pcm: Use static DAI format setup ASoC: smdk_wm8994pcm: Use static DAI format setup ASoC: migor: Use static DAI format setup sound/soc/atmel/sam9g20_wm8731.c | 30 ++---------------------------- sound/soc/au1x/db1200.c | 19 +++---------------- sound/soc/fsl/eukrea-tlv320.c | 21 ++------------------- sound/soc/fsl/mx27vis-aic32x4.c | 12 ++---------- sound/soc/fsl/wm1133-ev1.c | 12 ++---------- sound/soc/mxs/mxs-sgtl5000.c | 27 ++++++--------------------- sound/soc/omap/ams-delta.c | 18 +++--------------- sound/soc/pxa/raumfeld.c | 35 ++++++++--------------------------- sound/soc/pxa/zylonite.c | 12 ++---------- sound/soc/samsung/goni_wm8994.c | 23 ++++------------------- sound/soc/samsung/h1940_uda1380.c | 15 ++------------- sound/soc/samsung/jive_wm8750.c | 16 ++-------------- sound/soc/samsung/neo1973_wm8753.c | 25 ++++--------------------- sound/soc/samsung/rx1950_uda1380.c | 15 ++------------- sound/soc/samsung/s3c24xx_simtec.c | 20 ++------------------ sound/soc/samsung/s3c24xx_uda134x.c | 12 ++---------- sound/soc/samsung/smartq_wm8987.c | 16 ++-------------- sound/soc/samsung/smdk_wm8580.c | 21 ++++++--------------- sound/soc/samsung/smdk_wm8580pcm.c | 19 +++++-------------- sound/soc/samsung/smdk_wm8994pcm.c | 16 ++-------------- sound/soc/sh/migor.c | 12 ++---------- 21 files changed, 65 insertions(+), 331 deletions(-) -- 1.8.0

3 23

[alsa-devel] [PATCH v3] ALSA: sound/atmel/ac97c.c: Add device tree support
by Alexander Stein 06 Jan '15

06 Jan '15

This adds device tree support for the AC97 controller. It uses the soc-ac97link bindings, but actually only ac97-reset is used. Signed-off-by: Alexander Stein <alexanders83(a)web.de> Acked-By Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- Changes in v3: * Renamed the compatible string to atmel,at91sam9263-ac97c * Added 'Acked-By Alexandre Belloni <alexandre.belloni(a)free-electrons.com>' Changes in v2: * It now uses the soc-ac97link bindings (partly) instead of defining its own. This should ease the transition to ASoC once a new driver has been written. sound/atmel/ac97c.c | 45 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 42 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 4f6b14d..a0df1b1 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -22,6 +22,9 @@ #include <linux/gpio.h> #include <linux/types.h> #include <linux/io.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/initval.h> @@ -902,6 +905,40 @@ static void atmel_ac97c_reset(struct atmel_ac97c *chip) } } +#ifdef CONFIG_OF +static const struct of_device_id atmel_ac97c_dt_ids[] = { + { .compatible = "atmel,at91sam9263-ac97c", }, + { } +}; +MODULE_DEVICE_TABLE(of, atmel_ac97c_dt_ids); + +static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) +{ + struct ac97c_platform_data *pdata; + struct device_node *node = dev->of_node; + const struct of_device_id *match; + + if (!node) { + dev_err(dev, "Device does not have associated DT data\n"); + return ERR_PTR(-EINVAL); + } + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return ERR_PTR(-ENOMEM); + + pdata->reset_pin = of_get_named_gpio(dev->of_node, "ac97-gpios", 2); + + return pdata; +} +#else +static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) +{ + dev_err(dev, "no platform data defined\n"); + return ERR_PTR(-ENXIO); +} +#endif + static int atmel_ac97c_probe(struct platform_device *pdev) { struct snd_card *card; @@ -922,10 +959,11 @@ static int atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } - pdata = pdev->dev.platform_data; + pdata = dev_get_platdata(&pdev->dev); if (!pdata) { - dev_dbg(&pdev->dev, "no platform data\n"); - return -ENXIO; + pdata = atmel_ac97c_probe_dt(&pdev->dev); + if (IS_ERR(pdata)) + return PTR_ERR(pdata); } irq = platform_get_irq(pdev, 0); @@ -1204,6 +1242,7 @@ static struct platform_driver atmel_ac97c_driver = { .driver = { .name = "atmel_ac97c", .pm = ATMEL_AC97C_PM_OPS, + .of_match_table = of_match_ptr(atmel_ac97c_dt_ids), }, }; module_platform_driver(atmel_ac97c_driver); -- 2.2.1

2 1

[alsa-devel] [PATCH 1/8] ASoC: alc5632: Constify struct regmap_config and snd_soc_codec_driver
by Krzysztof Kozlowski 06 Jan '15

06 Jan '15

The regmap_config struct may be const because it is not modified by the driver and regmap_init() accepts pointer to const. Make struct snd_soc_codec_driver const as well (snd_soc_register_codec() accepts pointer to const). Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- sound/soc/codecs/alc5632.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index d1fdbc266631..07630260e1ba 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1066,7 +1066,7 @@ static int alc5632_probe(struct snd_soc_codec *codec) return 0; } -static struct snd_soc_codec_driver soc_codec_device_alc5632 = { +static const struct snd_soc_codec_driver soc_codec_device_alc5632 = { .probe = alc5632_probe, .resume = alc5632_resume, .set_bias_level = alc5632_set_bias_level, @@ -1080,7 +1080,7 @@ static struct snd_soc_codec_driver soc_codec_device_alc5632 = { .num_dapm_routes = ARRAY_SIZE(alc5632_dapm_routes), }; -static struct regmap_config alc5632_regmap = { +static const struct regmap_config alc5632_regmap = { .reg_bits = 8, .val_bits = 16, -- 1.9.1

4 12

Re: [alsa-devel] DSD over HDMI: audio data packetization
by Anssi Hannula 06 Jan '15

06 Jan '15

06.01.2015, 06:33, Andrej Falout kirjoitti: > Hi Anssi, Hi, > Just wondering how Microsoft documentation fits into this discussion: > > "Representing Formats for IEC 61937 Transmissions" > > http://msdn.microsoft.com/en-us/library/windows/desktop/dd316761%28v=vs.85%… > > > "Microsoft-provided HD Audio class driver supports PCM, AC3, DTS, AAC, > Dolby Digital Plus, WMA Pro, MAT(MLP) formats. The GUIDs for the > compressed audio formats that are not supported by the HD audio class > driver and can be implemented by third-party solutions are listed in the > following table. " > > CEA 861 Type SubFormat GUID Description > 0x08 > > 00000008-0cea-0010-8000-00aa00389b71 > > KSDATAFORMAT_SUBTYPE_IEC61937_ATRAC > > Adaptive Transform Acoustic Coding (ATRAC) > 0x09 > > 00000009-0cea-0010-8000-00aa00389b71 > > KSDATAFORMAT_SUBTYPE_IEC61937_ONE_BIT_AUDIO > > One-Bit Audio > 0x0d > > 0000000d-0cea-0010-8000-00aa00389b71 > > KSDATAFORMAT_SUBTYPE_IEC61937_DST > > Direct Stream Transport (DST)—lossless compressed DSD (Direct Stream > Digital). > > > And knowing that IEC 61937 from HDMI 1.3 also supports above 6.144 Mbps... > > Is there a doubt that above information is correct, and that DST/DST can > be carried over 61937 in Microsoft HDA class driver, in this example, by > "implementing a third-party solution" as described in this table? I think the One-Bit Audio and DST formats are just misnamed here and can't be carried over IEC 61937 (ATRAC can be) - the IEC 61937 specification does not cover One-Bit Audio or DST. I think the "third-party solution" refers to a driver implementing support for a non-HDA-spec HW feature which allows One-Bit audio / DST / raw packets to be transmitted (in a similar way "EPT" (Encoded Packet Type field in Pin Widget Control 7.3.3.13) allows setting HBR or ASP packet transmission). > Thanks, > Andrej Falout > > On Mon, Jan 5, 2015 at 4:46 PM, Anssi Hannula <anssi.hannula(a)iki.fi > <mailto:anssi.hannula@iki.fi>> wrote: > > 04.01.2015, 05:10, Xuebing Wang kirjoitti: > > Anssi, Libin, > > > > Thanks for your help. > > > > After a little bit of research, I do think it is possible to stream DSD > > over HDMI (the mechanism is as Anssi mentioned, similar way as HBR). > > > > Please see HDA spec: > > 3.7.1 Stream Format Structure > > (If the TYPE is set to Non-PCM, the controller just pushes data over the > > link and is not concerned > > with formatting. The base rate, data type, and number of Words (MULT) to > > send each valid frame > > are specified to control the rate at which the non-PCM data is sent.) > > > > 7.3.3.8 Converter Format > > > > What do you think? In order to implement this similiar mechanism > > (similar to HBR), I still have some detailed questions that I will write > > in another email. Thanks. > > That setting will still not allow you to send One-bit audio or DST > packets. The stream will still be encapsulated in either ASP or HBR > packets, depending on the value of the EPT field (Encoded Packet Type, > 7.3.3.13). > > > > > > On Thu, 2015-01-01 at 06:37 +0200, Anssi Hannula wrote: > >> 01.01.2015, 04:07, Yang, Libin kirjoitti: > >>> Hi, > >>> > >>>> -----Original Message----- > >>>> From: Anssi Hannula [mailto:anssi.hannula@iki.fi > <mailto:anssi.hannula@iki.fi>] > >>>> Sent: Wednesday, December 31, 2014 11:45 PM > >>>> To: Xuebing Wang > >>>> Cc: Yang, Libin; alsa-devel(a)alsa-project.org > <mailto:alsa-devel@alsa-project.org>; Lin, Mengdong; ; Takashi Iwai; ; > >>>> Wu Fengguang > >>>> Subject: Re: [alsa-devel] DSD over HDMI: audio data packetization > >>>> > >>>> 31.12.2014, 17:38, Xuebing Wang kirjoitti: > >>>>> Anssi, Libin, > >>>>> > >>>>> Thank you very much for your help. > >>>>> > >>>>> If L-PCM audio data packetization is done by hardware, is it > possible > >>>>> to implement DSD over HDMI on Intel integrated graphics? > Thanks again. > >>>> > >>>> I don't think so, unless: > >>>> > >>>> (a) It is possible to somehow encapsulate the stream in Audio > Sample > >>>> Packets and be accepted by the receiver (like e.g. AC3, DTS). > I'm not aware of > >>>> such a way. > >>>> > >>>> or: > >>>> (b) There is some undocumented feature in the Intel HDMI codecs > allowing > >>>> the driver to switch to DSD or driver-constructed packets > (instead of > >>>> ASP) in similar way as there is a way to switch to HBR packets. > >>>> > >>> > >>> Yes, I agree. And I'm not sure whether HDMI spec supports DSD. > >> > >> It does, via the "One Bit Audio" and "DST" formats (spec 5.3.9, > 5.3.10 > >> in 1.3a/1.4a). > > > > > > > -- > Anssi Hannula > > -- Anssi Hannula

2 2

Re: [alsa-devel] DSD over HDMI: audio data packetization
by Xuebing Wang 06 Jan '15

06 Jan '15

Andrej, I have a slight doubt about "KSDATAFORMAT_SUBTYPE_IEC61937_ONE_BIT_AUDIO". To be concise "IEC61937_ONE_BIT_AUDIO" I do not have all IEC 61937 specs, I can not find "IEC61937_ONE_BIT_AUDIO" in the list below. Part 1: General Part 2: Burst-info Part 3: Non-linear PCM bitstreams according to the AC-3 format Part 4: Non-linear PCM bitstreams according to the MPEG audio format Part 5: Non-linear PCM bitstreams according to the DTS (Digital Theater Systems) format(s) Part 6: Non-linear PCM bitstreams according to the MPEG-2 AAC and MPEG-4 AAC audio formats Part 7: Non-linear PCM bitstreams according to the ATRAC, ATRAC2/3 and ATRAC-X formats Part 8: Non-linear PCM bitstreams according to the Windows Media Audio (WMA) Professional format Part 9: Non-linear PCM bitstreams according to the MAT format Part 10: Non-linear PCM bitstreams according to the MPEG-4 audio lossless coding (ALS) format - Edition 1.0 Part 11: MPEG-4 AAC and its extensions in LATM/LOAS Part 12: Non-linear PCM bitstreams according to the DRA formats Thanks. On Tue, 2015-01-06 at 17:33 +1300, Andrej Falout wrote: > Hi Anssi, > > > Just wondering how Microsoft documentation fits into this discussion: > > "Representing Formats for IEC 61937 Transmissions" > > http://msdn.microsoft.com/en-us/library/windows/desktop/dd316761% > 28v=vs.85%29.aspx > > > "Microsoft-provided HD Audio class driver supports PCM, AC3, DTS, AAC, > Dolby Digital Plus, WMA Pro, MAT(MLP) formats. The GUIDs for the > compressed audio formats that are not supported by the HD audio class > driver and can be implemented by third-party solutions are listed in > the following table. " > > CEA 861 Type > SubFormat GUID > Description > 0x08 > 00000008-0cea-0010-8000-00aa00389b71 > > KSDATAFORMAT_SUBTYPE_IEC61937_ATRAC > > > Adaptive Transform > Acoustic Coding (ATRAC) > 0x09 > 00000009-0cea-0010-8000-00aa00389b71 > > KSDATAFORMAT_SUBTYPE_IEC61937_ONE_BIT_AUDIO > > > One-Bit Audio > 0x0d > 0000000d-0cea-0010-8000-00aa00389b71 > > KSDATAFORMAT_SUBTYPE_IEC61937_DST > > > Direct Stream Transport > (DST)—lossless > compressed DSD (Direct > Stream Digital). > > > And knowing that IEC 61937 from HDMI 1.3 also supports above 6.144 > Mbps... > > > Is there a doubt that above information is correct, and that DST/DST > can be carried over 61937 in Microsoft HDA class driver, in this > example, by "implementing a third-party solution" as described in this > table? > > > Thanks, > > Andrej Falout > > > On Mon, Jan 5, 2015 at 4:46 PM, Anssi Hannula <anssi.hannula(a)iki.fi> > wrote: > 04.01.2015, 05:10, Xuebing Wang kirjoitti: > > Anssi, Libin, > > > > Thanks for your help. > > > > After a little bit of research, I do think it is possible to > stream DSD > > over HDMI (the mechanism is as Anssi mentioned, similar way > as HBR). > > > > Please see HDA spec: > > 3.7.1 Stream Format Structure > > (If the TYPE is set to Non-PCM, the controller just pushes > data over the > > link and is not concerned > > with formatting. The base rate, data type, and number of > Words (MULT) to > > send each valid frame > > are specified to control the rate at which the non-PCM data > is sent.) > > > > 7.3.3.8 Converter Format > > > > What do you think? In order to implement this similiar > mechanism > > (similar to HBR), I still have some detailed questions that > I will write > > in another email. Thanks. > > That setting will still not allow you to send One-bit audio or > DST > packets. The stream will still be encapsulated in either ASP > or HBR > packets, depending on the value of the EPT field (Encoded > Packet Type, > 7.3.3.13). > > > > > > On Thu, 2015-01-01 at 06:37 +0200, Anssi Hannula wrote: > >> 01.01.2015, 04:07, Yang, Libin kirjoitti: > >>> Hi, > >>> > >>>> -----Original Message----- > >>>> From: Anssi Hannula [mailto:anssi.hannula@iki.fi] > >>>> Sent: Wednesday, December 31, 2014 11:45 PM > >>>> To: Xuebing Wang > >>>> Cc: Yang, Libin; alsa-devel(a)alsa-project.org; Lin, > Mengdong; ; Takashi Iwai; ; > >>>> Wu Fengguang > >>>> Subject: Re: [alsa-devel] DSD over HDMI: audio data > packetization > >>>> > >>>> 31.12.2014, 17:38, Xuebing Wang kirjoitti: > >>>>> Anssi, Libin, > >>>>> > >>>>> Thank you very much for your help. > >>>>> > >>>>> If L-PCM audio data packetization is done by hardware, > is it possible > >>>>> to implement DSD over HDMI on Intel integrated graphics? > Thanks again. > >>>> > >>>> I don't think so, unless: > >>>> > >>>> (a) It is possible to somehow encapsulate the stream in > Audio Sample > >>>> Packets and be accepted by the receiver (like e.g. AC3, > DTS). I'm not aware of > >>>> such a way. > >>>> > >>>> or: > >>>> (b) There is some undocumented feature in the Intel HDMI > codecs allowing > >>>> the driver to switch to DSD or driver-constructed packets > (instead of > >>>> ASP) in similar way as there is a way to switch to HBR > packets. > >>>> > >>> > >>> Yes, I agree. And I'm not sure whether HDMI spec supports > DSD. > >> > >> It does, via the "One Bit Audio" and "DST" formats (spec > 5.3.9, 5.3.10 > >> in 1.3a/1.4a). > > > > > > > > -- > Anssi Hannula > >

1 0

[alsa-devel] [PATCH 1/7] ASoC: dmaengine: Remove unnecessary snd_pcm_lib_preallocate_free_for_all()
by Lars-Peter Clausen 05 Jan '15

05 Jan '15

The ALSA core takes care that all preallocated memory is freed when the PCM itself is freed. There is no need to do this manually in the driver. Similarly there is also no need to do it on the pcm_new() error path as the PCM will be freed if a error is returned. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-generic-dmaengine-pcm.c | 15 ++------------- 1 file changed, 2 insertions(+), 13 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index b329b84..4864392 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -200,11 +200,6 @@ static int dmaengine_pcm_open(struct snd_pcm_substream *substream) return snd_dmaengine_pcm_open(substream, chan); } -static void dmaengine_pcm_free(struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} - static struct dma_chan *dmaengine_pcm_compat_request_channel( struct snd_soc_pcm_runtime *rtd, struct snd_pcm_substream *substream) @@ -283,8 +278,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!pcm->chan[i]) { dev_err(rtd->platform->dev, "Missing dma channel for stream: %d\n", i); - ret = -EINVAL; - goto err_free; + return -EINVAL; } ret = snd_pcm_lib_preallocate_pages(substream, @@ -293,7 +287,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) prealloc_buffer_size, max_buffer_size); if (ret) - goto err_free; + return ret; /* * This will only return false if we know for sure that at least @@ -307,10 +301,6 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) } return 0; - -err_free: - dmaengine_pcm_free(rtd->pcm); - return ret; } static snd_pcm_uframes_t dmaengine_pcm_pointer( @@ -341,7 +331,6 @@ static const struct snd_soc_platform_driver dmaengine_pcm_platform = { }, .ops = &dmaengine_pcm_ops, .pcm_new = dmaengine_pcm_new, - .pcm_free = dmaengine_pcm_free, }; static const char * const dmaengine_pcm_dma_channel_names[] = { -- 1.8.0

3 8

[alsa-devel] [PATCH v2 00/10] audio timestamping evolutions
by Pierre-Louis Bossart 05 Jan '15

05 Jan '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields Pierre-Louis Bossart (10): ALSA: core: don't override timestamp unconditionally ALSA: core: allow for trigger_tstamp snapshot in .trigger ALSA: hda: read trigger_timestamp immediately after starting DMA ALSA: usb: update trigger timestamp on first non-zero URB submitted ALSA: core: add info_ext field in hw_params and pcm_hardware ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: replace .wall_clock by .get_time_info ALSA: hda: replace .wallclock by .get_time_info Documentation/sound/alsa/timestamping.txt | 176 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 71 +++++++++++- include/uapi/sound/asound.h | 31 +++++- sound/core/pcm_compat.c | 15 ++- sound/core/pcm_lib.c | 81 +++++++++----- sound/core/pcm_native.c | 41 ++++++- sound/pci/hda/hda_controller.c | 44 ++++++-- sound/usb/pcm.c | 9 ++ 8 files changed, 417 insertions(+), 51 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

4 17

[alsa-devel] [PATCH] ASoC: adi: Add missing return statement.
by Andrew Jackson 05 Jan '15

05 Jan '15

From: Andrew Jackson <Andrew.Jackson(a)arm.com> The probe routine was disabling the clock even if the system was configured successfully. Add a return statement to leave clocks enabled. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- Spotted while reviewing clock preparation sound/soc/adi/axi-i2s.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 7752860..4c23381 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -240,6 +240,8 @@ static int axi_i2s_probe(struct platform_device *pdev) if (ret) goto err_clk_disable; + return 0; + err_clk_disable: clk_disable_unprepare(i2s->clk); return ret; -- 1.7.1

3 2

[alsa-devel] [PATCH 1/2] ASoC: byt-rt5640: Fix snd_soc_dapm_ignore_suspend() calls
by Lars-Peter Clausen 05 Jan '15

05 Jan '15

To work properly snd_soc_dapm_ignore_suspend() needs to be called on endpoint widgets. In this case those are the board level Speaker and Headphone widgets and not the CODEC output widgets that are connected to them. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/intel/byt-rt5640.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 0cba783..a51856e 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -171,13 +171,8 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) return ret; } - snd_soc_dapm_ignore_suspend(dapm, "HPOL"); - snd_soc_dapm_ignore_suspend(dapm, "HPOR"); - - snd_soc_dapm_ignore_suspend(dapm, "SPOLP"); - snd_soc_dapm_ignore_suspend(dapm, "SPOLN"); - snd_soc_dapm_ignore_suspend(dapm, "SPORP"); - snd_soc_dapm_ignore_suspend(dapm, "SPORN"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); return ret; } -- 1.8.0

3 3

[alsa-devel] [PATCH] ASoC: boardwell: Drop unnecessary snd_soc_dapm_enable() calls
by Lars-Peter Clausen 05 Jan '15

05 Jan '15

DAPM widgets are enabled by default, there is no need to enable them unless they have previously been explicitly disabled. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/intel/broadwell.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 7cf95d5..9cf7d01 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -140,8 +140,6 @@ static struct snd_soc_ops broadwell_rt286_ops = { static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); struct sst_hsw *broadwell = pdata->dsp; int ret; @@ -155,14 +153,6 @@ static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) return ret; } - /* always connected - check HP for jack detect */ - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Speaker"); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_enable_pin(dapm, "Line Jack"); - snd_soc_dapm_enable_pin(dapm, "DMIC1"); - snd_soc_dapm_enable_pin(dapm, "DMIC2"); - return 0; } -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: ts3a227e: Check and report jack status at probe
by Cheng-Yi Chiang 05 Jan '15

05 Jan '15

ts3a227e does not trigger interrupt to report jack status when system boots from warm reset because ts3a227e's power remains on during warm reset. Read jack status at probe to get current jack status. Note that if system boots from EC reset, then this issue will not happen. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/codecs/ts3a227e.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 1d12057..9f2dced 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -254,6 +254,7 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c, struct ts3a227e *ts3a227e; struct device *dev = &i2c->dev; int ret; + unsigned int acc_reg; ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL); if (ts3a227e == NULL) @@ -283,6 +284,11 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c, INTB_DISABLE | ADC_COMPLETE_INT_DISABLE, ADC_COMPLETE_INT_DISABLE); + /* Read jack status because chip might not trigger interrupt at boot. */ + regmap_read(ts3a227e->regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg); + ts3a227e_new_jack_state(ts3a227e, acc_reg); + ts3a227e_jack_report(ts3a227e); + return 0; } -- 2.1.2

2 1

[alsa-devel] [PATCH] ASoC: Intel: Clean data after SST fw fetch
by libin.yang＠intel.com 05 Jan '15

05 Jan '15

From: Libin Yang <libin.yang(a)intel.com> The BDW audio firmware DSP manages the DMA and the DMA cannot be stopped exactly at the end of the playback stream. This means stale samples may be played at PCM stop unless the driver copies silence to the subsequent periods. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 29 ++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 9 ++++++ sound/soc/intel/sst-haswell-pcm.c | 64 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 102 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 3f8c482..57d6e89 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -31,6 +31,7 @@ #include <linux/dma-mapping.h> #include <linux/debugfs.h> #include <linux/pm_runtime.h> +#include <sound/asound.h> #include "sst-haswell-ipc.h" #include "sst-dsp.h" @@ -240,6 +241,10 @@ struct sst_hsw_stream { u32 (*notify_position)(struct sst_hsw_stream *stream, void *data); void *pdata; + /* record the fw read position when playback */ + snd_pcm_uframes_t old_position; + bool play_silence; + struct list_head node; }; @@ -1476,6 +1481,30 @@ u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, return stream->reply.volume_register_address[channel]; } +snd_pcm_uframes_t sst_hsw_stream_get_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->old_position; +} + +void sst_hsw_stream_set_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, snd_pcm_uframes_t val) +{ + stream->old_position = val; +} + +bool sst_hsw_stream_get_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->play_silence; +} + +void sst_hsw_stream_set_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, bool val) +{ + stream->play_silence = val; +} + int sst_hsw_mixer_get_info(struct sst_hsw *hsw) { struct sst_hsw_ipc_stream_info_reply *reply; diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 138e894..5c72e49 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -20,6 +20,7 @@ #include <linux/types.h> #include <linux/kernel.h> #include <linux/platform_device.h> +#include <sound/asound.h> #define SST_HSW_NO_CHANNELS 4 #define SST_HSW_MAX_DX_REGIONS 14 @@ -452,6 +453,14 @@ u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 channel); u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 channel); +snd_pcm_uframes_t sst_hsw_stream_get_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +void sst_hsw_stream_set_old_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, snd_pcm_uframes_t val); +bool sst_hsw_stream_get_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +void sst_hsw_stream_set_silence_start(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, bool val); int sst_hsw_mixer_get_info(struct sst_hsw *hsw); /* Stream ALSA trigger operations */ diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 6195252..4b70bdf 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -36,6 +36,11 @@ #define HSW_PCM_COUNT 6 #define HSW_VOLUME_MAX 0x7FFFFFFF /* 0dB */ +#define SST_OLD_POSITION(d, r, o) ((d) + \ + frames_to_bytes(r, o)) +#define SST_SAMPLES(r, x) (bytes_to_samples(r, \ + frames_to_bytes(r, (x)))) + /* simple volume table */ static const u32 volume_map[] = { HSW_VOLUME_MAX >> 30, @@ -553,19 +558,29 @@ static int hsw_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(rtd->platform); struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_hsw_stream *sst_stream = pcm_data->stream; struct sst_hsw *hsw = pdata->hsw; + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t pos; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sst_hsw_stream_set_silence_start(hsw, sst_stream, false); sst_hsw_stream_resume(hsw, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sst_hsw_stream_set_silence_start(hsw, sst_stream, false); sst_hsw_stream_pause(hsw, pcm_data->stream, 0); break; + case SNDRV_PCM_TRIGGER_DRAIN: + pos = runtime->control->appl_ptr % runtime->buffer_size; + sst_hsw_stream_set_old_position(hsw, pcm_data->stream, pos); + sst_hsw_stream_set_silence_start(hsw, sst_stream, true); + break; default: break; } @@ -579,13 +594,62 @@ static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data) struct snd_pcm_substream *substream = pcm_data->substream; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_hsw *hsw = pdata->hsw; u32 pos; + snd_pcm_uframes_t position = bytes_to_frames(runtime, + sst_hsw_get_dsp_position(hsw, pcm_data->stream)); + unsigned char *dma_area = runtime->dma_area; + snd_pcm_uframes_t dma_frames = + bytes_to_frames(runtime, runtime->dma_bytes); + snd_pcm_uframes_t old_position; + ssize_t samples; pos = frames_to_bytes(runtime, (runtime->control->appl_ptr % runtime->buffer_size)); dev_vdbg(rtd->dev, "PCM: App pointer %d bytes\n", pos); + /* SST fw don't know where to stop dma + * So, SST driver need to clean the data which has been consumed + */ + if (dma_area == NULL || dma_frames <= 0 + || (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + || !sst_hsw_stream_get_silence_start(hsw, stream)) { + snd_pcm_period_elapsed(substream); + return pos; + } + + old_position = sst_hsw_stream_get_old_position(hsw, stream); + if (position > old_position) { + if (position < dma_frames) { + samples = SST_SAMPLES(runtime, position - old_position); + snd_pcm_format_set_silence(runtime->format, + SST_OLD_POSITION(dma_area, + runtime, old_position), + samples); + } else + dev_err(rtd->dev, "PCM: position is wrong\n"); + } else { + if (old_position < dma_frames) { + samples = SST_SAMPLES(runtime, + dma_frames - old_position); + snd_pcm_format_set_silence(runtime->format, + SST_OLD_POSITION(dma_area, + runtime, old_position), + samples); + } else + dev_err(rtd->dev, "PCM: dma_bytes is wrong\n"); + if (position < dma_frames) { + samples = SST_SAMPLES(runtime, position); + snd_pcm_format_set_silence(runtime->format, + dma_area, samples); + } else + dev_err(rtd->dev, "PCM: position is wrong\n"); + } + sst_hsw_stream_set_old_position(hsw, stream, position); + /* let alsa know we have play a period */ snd_pcm_period_elapsed(substream); return pos; -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix wrong gpio_dir & gpio_mask hint setups for IDT/STAC codecs
by Takashi Iwai 05 Jan '15

05 Jan '15

stac_store_hints() does utterly wrong for masking the values for gpio_dir and gpio_data, likely due to copy&paste errors. Fortunately, this feature is used very rarely, so the impact must be really small. Reported-by: Rasmus Villemoes <linux(a)rasmusvillemoes.dk> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4f6413e01c13..605d14003d25 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -568,9 +568,9 @@ static void stac_store_hints(struct hda_codec *codec) spec->gpio_mask; } if (get_int_hint(codec, "gpio_dir", &spec->gpio_dir)) - spec->gpio_mask &= spec->gpio_mask; - if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) spec->gpio_dir &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) + spec->gpio_data &= spec->gpio_mask; if (get_int_hint(codec, "eapd_mask", &spec->eapd_mask)) spec->eapd_mask &= spec->gpio_mask; if (get_int_hint(codec, "gpio_mute", &spec->gpio_mute)) -- 2.2.1

1 0

[alsa-devel] Odd no-op in sound/pci/hda/patch_sigmatel.c
by Rasmus Villemoes 05 Jan '15

05 Jan '15

Hi Takashi The "spec->gpio_mask &= spec->gpio_mask" on line 571 of sound/pci/hda/patch_sigmatel.c is obviously a no-op. Is that on purpose, or was it supposed to be something else? Rasmus

2 1

[alsa-devel] [patch] ALSA: asihpi: fix an information leak in asihpi_hpi_ioctl()
by Dan Carpenter 05 Jan '15

05 Jan '15

So far as I can see "hr->h.size" is set to "res_max_size" which is a user controlled value between 12 and USHRT_MAX. If it's larger than sizeof(*hr), then that leads to an information leak. I am not very familiar with this code, my other question here is that on lines before we set "hr->h.size = sizeof(hr->h)". It think this is a bug. I also think this particular code is never executed and I added a comment to that effect. But we do it in earlier in the function as well: copy_to_user(puhr, hr, sizeof(hr->h)); It doesn't make sense to me. Anyway, I think my patch is safe and it seems to fix a real information leak. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 6aa677e..f88109a 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -283,6 +283,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) goto out; } + /* FIXME: isn't this a no-op? */ if (hr->h.size > res_max_size) { HPI_DEBUG_LOG(ERROR, "response too big %d %d\n", hr->h.size, res_max_size); @@ -290,6 +291,9 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) hr->h.specific_error = hr->h.size; hr->h.size = sizeof(hr->h); } + /* prevent an information leak */ + if (hr->h.size > sizeof(*hr)) + hr->h.size = sizeof(*hr); uncopied_bytes = copy_to_user(puhr, hr, hr->h.size); if (uncopied_bytes) {

3 6

[alsa-devel] [PATCH] ALSA: snd-usb-caiaq: fix stream count check
by Daniel Mack 05 Jan '15

05 Jan '15

Commit 897c329bc ("ALSA: usb: caiaq: check for cdev->n_streams > 1") introduced a safety check to protect against bogus data provided by devices. However, the n_streams variable is already divided by CHANNELS_PER_STREAM, so the correct check is 'n_streams > 0'. Fix this to un-break support for stereo devices. Signed-off-by: Daniel Mack <daniel(a)zonque.org> Cc: stable(a)kernel.org [v3.18+] --- sound/usb/caiaq/audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 2728447..327f864 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -816,7 +816,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) return -EINVAL; } - if (cdev->n_streams < 2) { + if (cdev->n_streams < 1) { dev_err(dev, "bogus number of streams: %d\n", cdev->n_streams); return -EINVAL; } -- 2.2.0

2 1

[alsa-devel] DSD over HDMI: audio data packetization
by Xuebing Wang 05 Jan '15

05 Jan '15

Hi Community, I am trying to implement DSD (one bit audio) over HDMI, for Intel integrated graphics. I am referencing HDMI specification 1.3a. Regarding audio data packetization, I am referencing spec: 5.3.4 Audio Sample Packet 7.6 Audio Data Packetization Q1) Would you please point to me where is the source code to handle L-PCM audio sample packetization (to implement 5.3.4 Audio Sample Packet)? Q2) I admit that I did not carefully read HD-Audio specification. Is this possible that this L-PCM audio data packetization is done by *hardware* (e.g HDMI related widgets/pins)? Q3) Did we implement "7.2 Audio Sample Clock Capture and Regeneration"? Thanks. Xuebing Wang

3 8

[alsa-devel] [PATCH - ALSA: snd-virtuoso 1/1] Xonar DSX support
by sgiurgiu11＠gmail.com 04 Jan '15

04 Jan '15

From: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> This patch adds support for ASUS - Xonar DSX sound cards. Tested on openSUSE 12.2 with kernel: Linux 3.4.6-2.10-desktop #1 SMP PREEMPT Thu Jul 26 09:36:26 UTC 2012 (641c197) x86_64 x86_64 x86_64 GNU/Linux Works: - play sounds - adjust volume on master channel. - mute . Since Xonar DS uses the same chip, everything that works for DS should work for DSX as well. Thank you, Sergiu Giurgiu. Signed-off-by: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> diff --git a/mirror/sound/pci/oxygen/virtuoso.c b/mirror/sound/pci/oxygen/virtuoso.c index 3d71423..8104eab 100644 --- a/mirror/sound/pci/oxygen/virtuoso.c +++ b/mirror/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; diff --git a/mirror/sound/pci/oxygen/xonar_wm87x6.c b/mirror/sound/pci/oxygen/xonar_wm87x6.c index 63cff90..b555b62 100644 --- a/mirror/sound/pci/oxygen/xonar_wm87x6.c +++ b/mirror/sound/pci/oxygen/xonar_wm87x6.c @@ -1255,7 +1255,6 @@ static void dump_wm87x6_registers(struct oxygen *chip, } static const struct oxygen_model model_xonar_ds = { - .shortname = "Xonar DS", .longname = "Asus Virtuoso 66", .chip = "AV200", .init = xonar_ds_init, @@ -1327,6 +1326,11 @@ int __devinit get_xonar_wm87x6_model(struct oxygen *chip, switch (id->subdevice) { case 0x838e: chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DS"; + break; + case 0x8522: + chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DSX"; break; case 0x835e: chip->model = model_xonar_hdav_slim; -- 1.7.10.4

4 5

[alsa-devel] [PATCH] ASoC: Intel: Don't change offset of block allocator during fixed allocate
by Jie Yang 04 Jan '15

04 Jan '15

The offset of block allocator, ba->offset, should not be changed during fixed address allocating, for the caller may treat it as the offset of allocated memory and use it. In the case that we allocate more than 1 blocks, we should make sure this offset is correct. Here introduces a temp allocator for the later continuous allocating. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-firmware.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 50d6925..a2ae2c5 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -707,6 +707,7 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba struct list_head *block_list) { struct sst_mem_block *block, *tmp; + struct sst_block_allocator ba_tmp = *ba; u32 end = ba->offset + ba->size, block_end; int err; @@ -731,9 +732,9 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba if (ba->offset >= block->offset && ba->offset < block_end) { /* align ba to block boundary */ - ba->size -= block_end - ba->offset; - ba->offset = block_end; - err = block_alloc_contiguous(dsp, ba, block_list); + ba_tmp.size -= block_end - ba->offset; + ba_tmp.offset = block_end; + err = block_alloc_contiguous(dsp, &ba_tmp, block_list); if (err < 0) return -ENOMEM; @@ -768,10 +769,10 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba list_move(&block->list, &dsp->used_block_list); list_add(&block->module_list, block_list); /* align ba to block boundary */ - ba->size -= block_end - ba->offset; - ba->offset = block_end; + ba_tmp.size -= block_end - ba->offset; + ba_tmp.offset = block_end; - err = block_alloc_contiguous(dsp, ba, block_list); + err = block_alloc_contiguous(dsp, &ba_tmp, block_list); if (err < 0) return -ENOMEM; -- 1.9.1

1 0

[alsa-devel] [PATCH] ALSA: pcm: Remove unused functions declaration
by Lars-Peter Clausen 03 Jan '15

03 Jan '15

Remove function declarations for functions that don't have a matching implementation. For snd_pcm_build_linear_format the implementation was removed in 64d27f96cb719cf8 ("[ALSA] Support 3-bytes 24bit format in PCM OSS emulation"). All the others never had one (as far as git history goes). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/pcm.h | 7 ------- 1 file changed, 7 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index bf32cea..cd09c1b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -518,7 +518,6 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream); int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, struct file *file, struct snd_pcm_substream **rsubstream); void snd_pcm_detach_substream(struct snd_pcm_substream *substream); -void snd_pcm_vma_notify_data(void *client, void *data); int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, struct vm_area_struct *area); @@ -984,21 +983,15 @@ int snd_pcm_format_physical_width(snd_pcm_format_t format); /* in bits */ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples); const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format); int snd_pcm_format_set_silence(snd_pcm_format_t format, void *buf, unsigned int frames); -snd_pcm_format_t snd_pcm_build_linear_format(int width, int unsigned, int big_endian); void snd_pcm_set_ops(struct snd_pcm * pcm, int direction, const struct snd_pcm_ops *ops); void snd_pcm_set_sync(struct snd_pcm_substream *substream); -int snd_pcm_lib_interleave_len(struct snd_pcm_substream *substream); int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg); int snd_pcm_update_state(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime); int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream); -int snd_pcm_playback_xrun_check(struct snd_pcm_substream *substream); -int snd_pcm_capture_xrun_check(struct snd_pcm_substream *substream); -int snd_pcm_playback_xrun_asap(struct snd_pcm_substream *substream); -int snd_pcm_capture_xrun_asap(struct snd_pcm_substream *substream); void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_uframes_t new_hw_ptr); void snd_pcm_period_elapsed(struct snd_pcm_substream *substream); snd_pcm_sframes_t snd_pcm_lib_write(struct snd_pcm_substream *substream, -- 1.8.0

2 1

[alsa-devel] [PATCH] sound: soc: soc-pcm: Remove useless varialbe 'i'
by Chen Gang 03 Jan '15

03 Jan '15

'i' is useless in soc_pcm_set_msb(), so remove it, the related warning (with allmodconfig under s390): CC [M] sound/soc/soc-pcm.o sound/soc/soc-pcm.c: In function 'soc_pcm_set_msb': sound/soc/soc-pcm.c:307:11: warning: unused variable 'i' [-Wunused-variable] int ret, i; ^ Signed-off-by: Chen Gang <gang.chen.5i5j(a)gmail.com> --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d6e8d6b..6b0136e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -304,7 +304,7 @@ static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) static void soc_pcm_set_msb(struct snd_pcm_substream *substream, int bits) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - int ret, i; + int ret; if (!bits) return; -- 1.7.9.5

3 2

[alsa-devel] [PATCH] sound: pci: hda: patch_analog: Remove some unused functions
by Rickard Strandqvist 03 Jan '15

03 Jan '15

Removes some functions that are not used anywhere: ad198x_ch_mode_put() ad198x_ch_mode_info() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/hda/patch_analog.c | 21 --------------------- 1 file changed, 21 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 06275f8..ed0b71c 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -698,15 +698,6 @@ static int patch_ad1981(struct hda_codec *codec) */ #ifdef ENABLE_AD_STATIC_QUIRKS -static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode, - spec->num_channel_mode); -} - static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -716,18 +707,6 @@ static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol, spec->num_channel_mode, spec->multiout.max_channels); } -static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - &spec->multiout.max_channels); - if (err >= 0 && spec->need_dac_fix) - spec->multiout.num_dacs = spec->multiout.max_channels / 2; - return err; -} #endif /* ENABLE_AD_STATIC_QUIRKS */ static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol, -- 1.7.10.4

2 2

[alsa-devel] [PATCH 1/2] ALSA: oxygen: Remove unnecessary snd_pcm_lib_preallocate_free_for_all()
by Lars-Peter Clausen 02 Jan '15

02 Jan '15

The ALSA core takes care that all preallocated memory is freed when the PCM itself is freed. There is no need to do this manually in the driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/pci/oxygen/oxygen_pcm.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 0282824..af22a74 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -676,11 +676,6 @@ static struct snd_pcm_ops oxygen_ac97_ops = { .pointer = oxygen_pointer, }; -static void oxygen_pcm_free(struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_free_for_all(pcm); -} - int oxygen_pcm_init(struct oxygen *chip) { struct snd_pcm *pcm; @@ -705,7 +700,6 @@ int oxygen_pcm_init(struct oxygen *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &oxygen_rec_b_ops); pcm->private_data = chip; - pcm->private_free = oxygen_pcm_free; strcpy(pcm->name, "Multichannel"); if (outs) snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream, @@ -734,7 +728,6 @@ int oxygen_pcm_init(struct oxygen *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &oxygen_rec_c_ops); pcm->private_data = chip; - pcm->private_free = oxygen_pcm_free; strcpy(pcm->name, "Digital"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), @@ -765,7 +758,6 @@ int oxygen_pcm_init(struct oxygen *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &oxygen_rec_b_ops); pcm->private_data = chip; - pcm->private_free = oxygen_pcm_free; strcpy(pcm->name, outs ? "Front Panel" : "Analog 2"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), -- 1.8.0

2 2

[alsa-devel] [PATCH 00/21] ALSA: Remove always NULL parameters
by Lars-Peter Clausen 02 Jan '15

02 Jan '15

There is a bit of a anti-pattern in some of the (mostly older) driver. In these drivers functions which allocate PCMs, timers, etc. take a parameter which is a pointer to a pointer of the allocated object and if the pointer is provided it is set to the newly allocate object. The code essentially looks like this. int foobar_allocate(..., struct snd_pcm **rpcm) { struct snd_pcm *pcm; if (rpcm) *rpcm = NULL; ... /* Create and initialize object */ if (rpcm) *rpcm = pcm; return 0; } But then all callers to these functions always pass NULL for the pointer. So the code that checks and initializes the pointer becomes dead code. While it can be argued that such a pattern makes sense for library functions where there might be new users which want to provide a valid pointer (at least in the absence of PTR_ERR and friends), it makes less sense for functions that are only used internally in a driver which hasn't seen any new hardware support in the last 10 years. This series goes through these drivers and removes the pointer parameter from functions which follow this pattern. There is one exception which is the wss driver, which is kind of a library driver, where some of the users actually pass a valid pointer. But the same PCM object that is stored in that pointer can also be accessed through the wss chip state struct. So that driver is update as well. - Lars Lars-Peter Clausen (21): ALSA: ml403-ac97cr: Remove always NULL parameter ALSA: ad1816a: Remove always NULL parameters ALSA: es1688: Remove almost always NULL parameter ALSA: es18xx: Remove always NULL parameter ALSA: gus: Remove always NULL parameters ALSA: msnd: Remove always NULL parameter ALSA: sb8: Remove always NULL parameters ASoC: sb16: Simplify snd_sb16dsp_pcm() ALSA: wss: Remove (almost) always NULL parameters ASLA: ad1889: Remove always NULL parameter ALSA: asihpi: Remove always NULL parameter ALSA: cs4281: Remove always NULL parameters ALSA: cs46xx: Remove always NULL parameters ALSA: ens1370: Remove always NULL parameters ALSA: emu10k1: Remove always NULL parameters ALSA: fm801: Remove always NULL parameter ALSA: ice1712: Remove always NULL parameters ALSA: riptide: Remove always NULL parameter ALSA: sonicvibes: Remove always NULL parameter ALSA: trident: Remove always NULL parameters ALSA: ymfpci: Remove always NULL parameters include/sound/ad1816a.h | 5 ++--- include/sound/emu10k1.h | 14 ++++++------- include/sound/es1688.h | 3 +-- include/sound/gus.h | 4 ++-- include/sound/sb.h | 6 +++--- include/sound/wss.h | 6 +++--- sound/drivers/ml403-ac97cr.c | 9 ++------- sound/isa/ad1816a/ad1816a.c | 5 ++--- sound/isa/ad1816a/ad1816a_lib.c | 9 ++------- sound/isa/ad1848/ad1848.c | 7 +++---- sound/isa/als100.c | 2 +- sound/isa/azt2320.c | 4 ++-- sound/isa/cmi8328.c | 4 ++-- sound/isa/cs423x/cs4231.c | 9 ++++----- sound/isa/cs423x/cs4236.c | 13 ++++++------- sound/isa/cs423x/cs4236_lib.c | 9 +++------ sound/isa/es1688/es1688.c | 7 +++---- sound/isa/es1688/es1688_lib.c | 6 +----- sound/isa/es18xx.c | 10 ++-------- sound/isa/galaxy/galaxy.c | 4 ++-- sound/isa/gus/gus_pcm.c | 6 +----- sound/isa/gus/gus_uart.c | 6 +----- sound/isa/gus/gusclassic.c | 4 ++-- sound/isa/gus/gusextreme.c | 4 ++-- sound/isa/gus/gusmax.c | 8 ++++---- sound/isa/gus/interwave.c | 14 ++++++------- sound/isa/msnd/msnd.c | 6 +----- sound/isa/msnd/msnd.h | 2 +- sound/isa/msnd/msnd_pinnacle.c | 2 +- sound/isa/opl3sa2.c | 4 ++-- sound/isa/opti9xx/miro.c | 12 +++++------- sound/isa/opti9xx/opti92x-ad1848.c | 13 +++++-------- sound/isa/sb/jazz16.c | 2 +- sound/isa/sb/sb16.c | 2 +- sound/isa/sb/sb16_main.c | 8 ++------ sound/isa/sb/sb8.c | 4 ++-- sound/isa/sb/sb8_main.c | 6 +----- sound/isa/sb/sb8_midi.c | 6 +----- sound/isa/sc6000.c | 2 +- sound/isa/sscape.c | 5 ++--- sound/isa/wavefront/wavefront.c | 4 ++-- sound/isa/wss/wss_lib.c | 8 ++------ sound/pci/ad1889.c | 10 ++-------- sound/pci/asihpi/asihpi.c | 9 ++------- sound/pci/cs4281.c | 18 ++++------------- sound/pci/cs46xx/cs46xx.c | 10 +++++----- sound/pci/cs46xx/cs46xx.h | 10 +++++----- sound/pci/cs46xx/cs46xx_lib.c | 40 +++++--------------------------------- sound/pci/cs5530.c | 2 +- sound/pci/emu10k1/emu10k1.c | 12 ++++++------ sound/pci/emu10k1/emu10k1x.c | 19 +++++------------- sound/pci/emu10k1/emufx.c | 7 +------ sound/pci/emu10k1/emupcm.c | 33 ++++--------------------------- sound/pci/emu10k1/p16v.c | 7 +------ sound/pci/ens1370.c | 37 ++++++++--------------------------- sound/pci/fm801.c | 14 +++---------- sound/pci/ice1712/ice1712.c | 26 ++++++------------------- sound/pci/riptide/riptide.c | 9 ++------- sound/pci/sonicvibes.c | 7 ++----- sound/pci/trident/trident.c | 6 +++--- sound/pci/trident/trident.h | 6 +++--- sound/pci/trident/trident_main.c | 21 +++----------------- sound/pci/ymfpci/ymfpci.c | 8 ++++---- sound/pci/ymfpci/ymfpci.h | 8 ++++---- sound/pci/ymfpci/ymfpci_main.c | 38 ++++++------------------------------ 65 files changed, 185 insertions(+), 436 deletions(-) -- 1.8.0

2 22

[alsa-devel] [PATCH v8 0/2] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 02 Jan '15

02 Jan '15

The NXP TDA998x HDMI transmitter may transmit audio to the HDMI link from 2 different sources, I2S and S/PDIF. This patch set first adds an interface between a HDMI transmitter and the HDMI CODEC. The interface is then used by the TDA998x driver to describe its audio capabilities (DAIs), to give the audio constraints of the HDMI device (EDID) to the audio subsystem, and to connect the chosen audio source to the HDMI link. v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (2): ASoC: hdmi: Add a transmitter interface to the HDMI CODEC drm/i2c: tda998x: Use the HDMI audio CODEC interface .../devicetree/bindings/drm/i2c/tda998x.txt | 18 ++ drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 264 ++++++++++++++++++++- include/sound/hdmi.h | 20 ++ sound/soc/codecs/hdmi.c | 176 +++++++++++++- 5 files changed, 460 insertions(+), 19 deletions(-) create mode 100644 include/sound/hdmi.h -- 2.1.1

3 7

[alsa-devel] [PATCH] ALSA: ASoC: soc-compress.c: fix NULL dereference
by Qais Yousef 02 Jan '15

02 Jan '15

In soc_new_compress() when rtd->dai_link->daynmic is set, we create the pcm substreams with this call: ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, 1, 0, &be_pcm); which passes 0 as capture_count leading to be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream being NULL, hence when trying to set rtd a few lines below we get an oops. Fix by removing this line of code since CAPTURE substream will always be NULL. Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- Not sure if this is the correct fix but that's what I could come up with my limited knowledge. I think the more correct solution would be to use the value of rtd->dai_link->dpcm_playback and rtd->dai_link->dpcm_capture in the args of snd_pcm_new_internal() for playback_count and capture_count. sound/soc/soc-compress.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 590a82f01d0b..7ab39f65384c 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -669,7 +669,6 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = be_pcm; rtd->fe_compr = 1; be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); -- 2.1.0

2 2

[alsa-devel] [PATCH] sound: drivers: portman2x4.c: Remove some unused functions
by Rickard Strandqvist 02 Jan '15

02 Jan '15

Removes some functions that are not used anywhere: portman_read_data() portman_read_command() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/drivers/portman2x4.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 78ccfa4..b5b6b4e 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -198,21 +198,11 @@ static inline void portman_write_command(struct portman *pm, u8 value) parport_write_control(pm->pardev->port, value); } -static inline u8 portman_read_command(struct portman *pm) -{ - return parport_read_control(pm->pardev->port); -} - static inline u8 portman_read_status(struct portman *pm) { return parport_read_status(pm->pardev->port); } -static inline u8 portman_read_data(struct portman *pm) -{ - return parport_read_data(pm->pardev->port); -} - static inline void portman_write_data(struct portman *pm, u8 value) { parport_write_data(pm->pardev->port, value); -- 1.7.10.4

2 1

[alsa-devel] [PATCH] Removed unused variable i from sound/soc/soc-pcm.c Signed-off-by: Shakil A Khan <shakilk1729@gmail.com>
by Shakil A Khan 02 Jan '15

02 Jan '15

--- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d6e8d6b..6b0136e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -304,7 +304,7 @@ static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) static void soc_pcm_set_msb(struct snd_pcm_substream *substream, int bits) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - int ret, i; + int ret; if (!bits) return; -- 1.9.1

2 1

[alsa-devel] [PATCH v2] sound: pci: asihpi: hpifunc.c: Remove some unused functions
by Rickard Strandqvist 01 Jan '15

01 Jan '15

Removes some functions that are not used anywhere: hpi_stream_response_to_legacy() hpi_outstream_host_buffer_free() hpi_outstream_host_buffer_get_info() hpi_outstream_host_buffer_allocate() hpi_outstream_set_time_scale() hpi_outstream_ancillary_read() hpi_outstream_ancillary_get_info() hpi_outstream_ancillary_reset() hpi_outstream_set_punch_in_out() hpi_outstream_set_velocity() hpi_outstream_set_format() hpi_outstream_sinegen() hpi_outstream_wait_start() hpi_stream_estimate_buffer_size() hpi_vox_get_threshold() hpi_vox_set_threshold() hpi_volume_query_auto_fade_profile() hpi_volume_auto_fade() hpi_adapter_enumerate_property() hpi_volume_query_channels() hpi_pad_get_rdsPI() hpi_pad_get_program_type() hpi_pad_get_comment() hpi_pad_get_title() hpi_pad_get_artist() hpi_pad_get_channel_name() hpi_tuner_get_rds() hpi_tuner_set_hd_radio_signal_blend() hpi_tuner_get_hd_radio_signal_blend() hpi_tuner_get_hd_radio_signal_quality() hpi_tuner_get_mode() hpi_tuner_set_mode() hpi_tuner_get_status() hpi_tuner_get_hd_radio_sdk_version() hpi_tuner_get_hd_radio_dsp_version() hpi_tuner_get_program() hpi_tuner_set_program() hpi_tuner_query_program() hpi_tuner_get_deemphasis() hpi_tuner_set_deemphasis() hpi_tuner_query_deemphasis() hpi_tuner_get_raw_rf_level() hpi_tuner_get_rf_level() hpi_silence_detector_get_threshold() hpi_silence_detector_set_threshold() hpi_silence_detector_get_delay() hpi_silence_detector_set_delay() hpi_silence_detector_get_event_enable() hpi_silence_detector_set_event_enable() hpi_silence_detector_get_enable() hpi_silence_detector_set_enable() hpi_silence_detector_get_state() hpi_tone_detector_get_threshold() hpi_tone_detector_set_threshold() hpi_tone_detector_get_event_enable() hpi_tone_detector_set_event_enable() hpi_tone_detector_get_enable() hpi_volume_auto_fade_profile() hpi_tone_detector_set_enable() hpi_tone_detector_get_state() hpi_tone_detector_get_frequency() hpi_sample_clock_get_local_rate_lock() hpi_sample_clock_set_local_rate_lock() hpi_sample_clock_get_auto() hpi_sample_clock_set_auto() hpi_parametric_eq_get_coeffs() hpi_parametric_eq_set_band() hpi_parametric_eq_get_band() hpi_parametric_eq_set_state() hpi_parametric_eq_get_info() hpi_adapter_get_module_by_index() hpi_microphone_get_phantom_power() hpi_microphone_set_phantom_power() hpi_meter_get_peak_ballistics() hpi_meter_set_peak_ballistics() hpi_meter_get_rms_ballistics() hpi_meter_set_rms_ballistics() hpi_meter_get_rms() hpi_compander_get_ratio() hpi_compander_set_ratio() hpi_compander_get_threshold() hpi_compander_set_threshold() hpi_compander_get_decay_time_constant() hpi_compander_set_decay_time_constant() hpi_compander_get_attack_time_constant() hpi_compander_set_attack_time_constant() hpi_compander_get_makeup_gain() hpi_compander_set_makeup_gain() hpi_compander_get_enable() hpi_compander_set_enable() hpi_adapter_get_mode() hpi_cobranet_get_macaddress() hpi_cobranet_set_static_ip_address() hpi_cobranet_get_static_ip_address() hpi_cobranet_set_ip_address() hpi_cobranet_get_ip_address() hpi_cobranet_hmi_get_status() hpi_adapter_set_mode() hpi_bitstream_get_activity() hpi_bitstream_set_data_polarity() hpi_bitstream_set_clock_edge() hpi_aesebu_transmitter_query_format() hpi_aesebu_transmitter_get_channel_status() hpi_aesebu_transmitter_set_channel_status() hpi_aesebu_transmitter_set_user_data() hpi_aesebu_transmitter_set_sample_rate() hpi_aesebu_receiver_get_channel_status() hpi_aesebu_receiver_get_user_data() hpi_aesebu_receiver_get_sample_rate() hpi_adapter_close() hpi_aesebu_receiver_query_format() hpi_adapter_open() hpi_mixer_store() hpi_mixer_close() hpi_subsys_get_adapter() hpi_instream_host_buffer_get_info() hpi_subsys_get_num_adapters() hpi_instream_host_buffer_allocate() hpi_instream_ancillary_write() hpi_instream_ancillary_get_info() hpi_instream_ancillary_reset() hpi_adapter_set_mode_ex() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/asihpi/hpi.h | 304 ------- sound/pci/asihpi/hpi_internal.h | 7 - sound/pci/asihpi/hpifunc.c | 1693 ++++----------------------------------- 3 files changed, 136 insertions(+), 1868 deletions(-) diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 2088724..fa9c607 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -1152,13 +1152,6 @@ struct hpi_async_event { #pragma pack(pop) #endif -/*****************/ -/* HPI FUNCTIONS */ -/*****************/ - -/* Stream */ -u16 hpi_stream_estimate_buffer_size(struct hpi_format *pF, - u32 host_polling_rate_in_milli_seconds, u32 *recommended_buffer_size); /*************/ /* SubSystem */ @@ -1166,34 +1159,14 @@ u16 hpi_stream_estimate_buffer_size(struct hpi_format *pF, u16 hpi_subsys_get_version_ex(u32 *pversion_ex); -u16 hpi_subsys_get_num_adapters(int *pn_num_adapters); - -u16 hpi_subsys_get_adapter(int iterator, u32 *padapter_index, - u16 *pw_adapter_type); - /***********/ /* Adapter */ /***********/ -u16 hpi_adapter_open(u16 adapter_index); - -u16 hpi_adapter_close(u16 adapter_index); - u16 hpi_adapter_get_info(u16 adapter_index, u16 *pw_num_outstreams, u16 *pw_num_instreams, u16 *pw_version, u32 *pserial_number, u16 *pw_adapter_type); -u16 hpi_adapter_get_module_by_index(u16 adapter_index, u16 module_index, - u16 *pw_num_outputs, u16 *pw_num_inputs, u16 *pw_version, - u32 *pserial_number, u16 *pw_module_type, u32 *ph_module); - -u16 hpi_adapter_set_mode(u16 adapter_index, u32 adapter_mode); - -u16 hpi_adapter_set_mode_ex(u16 adapter_index, u32 adapter_mode, - u16 query_or_set); - -u16 hpi_adapter_get_mode(u16 adapter_index, u32 *padapter_mode); - u16 hpi_adapter_get_assert2(u16 adapter_index, u16 *p_assert_count, char *psz_assert, u32 *p_param1, u32 *p_param2, u32 *p_dsp_string_addr, u16 *p_processor_id); @@ -1213,8 +1186,6 @@ u16 hpi_adapter_set_property(u16 adapter_index, u16 property, u16 paramter1, u16 hpi_adapter_get_property(u16 adapter_index, u16 property, u16 *pw_paramter1, u16 *pw_paramter2); -u16 hpi_adapter_enumerate_property(u16 adapter_index, u16 index, - u16 what_to_enumerate, u16 property_index, u32 *psetting); /*************/ /* OutStream */ /*************/ @@ -1232,38 +1203,12 @@ u16 hpi_outstream_write_buf(u32 h_outstream, const u8 *pb_write_buf, u16 hpi_outstream_start(u32 h_outstream); -u16 hpi_outstream_wait_start(u32 h_outstream); - u16 hpi_outstream_stop(u32 h_outstream); -u16 hpi_outstream_sinegen(u32 h_outstream); - u16 hpi_outstream_reset(u32 h_outstream); u16 hpi_outstream_query_format(u32 h_outstream, struct hpi_format *p_format); -u16 hpi_outstream_set_format(u32 h_outstream, struct hpi_format *p_format); - -u16 hpi_outstream_set_punch_in_out(u32 h_outstream, u32 punch_in_sample, - u32 punch_out_sample); - -u16 hpi_outstream_set_velocity(u32 h_outstream, short velocity); - -u16 hpi_outstream_ancillary_reset(u32 h_outstream, u16 mode); - -u16 hpi_outstream_ancillary_get_info(u32 h_outstream, u32 *pframes_available); - -u16 hpi_outstream_ancillary_read(u32 h_outstream, - struct hpi_anc_frame *p_anc_frame_buffer, - u32 anc_frame_buffer_size_in_bytes, - u32 number_of_ancillary_frames_to_read); - -u16 hpi_outstream_set_time_scale(u32 h_outstream, u32 time_scaleX10000); - -u16 hpi_outstream_host_buffer_allocate(u32 h_outstream, u32 size_in_bytes); - -u16 hpi_outstream_host_buffer_free(u32 h_outstream); - u16 hpi_outstream_group_add(u32 h_outstream, u32 h_stream); u16 hpi_outstream_group_get_map(u32 h_outstream, u32 *poutstream_map, @@ -1299,18 +1244,6 @@ u16 hpi_instream_get_info_ex(u32 h_instream, u16 *pw_state, u32 *pbuffer_size, u32 *pdata_recorded, u32 *psamples_recorded, u32 *pauxiliary_data_recorded); -u16 hpi_instream_ancillary_reset(u32 h_instream, u16 bytes_per_frame, - u16 mode, u16 alignment, u16 idle_bit); - -u16 hpi_instream_ancillary_get_info(u32 h_instream, u32 *pframe_space); - -u16 hpi_instream_ancillary_write(u32 h_instream, - const struct hpi_anc_frame *p_anc_frame_buffer, - u32 anc_frame_buffer_size_in_bytes, - u32 number_of_ancillary_frames_to_write); - -u16 hpi_instream_host_buffer_allocate(u32 h_instream, u32 size_in_bytes); - u16 hpi_instream_host_buffer_free(u32 h_instream); u16 hpi_instream_group_add(u32 h_instream, u32 h_stream); @@ -1325,8 +1258,6 @@ u16 hpi_instream_group_reset(u32 h_instream); /*********/ u16 hpi_mixer_open(u16 adapter_index, u32 *ph_mixer); -u16 hpi_mixer_close(u32 h_mixer); - u16 hpi_mixer_get_control(u32 h_mixer, u16 src_node_type, u16 src_node_type_index, u16 dst_node_type, u16 dst_node_type_index, u16 control_type, u32 *ph_control); @@ -1335,8 +1266,6 @@ u16 hpi_mixer_get_control_by_index(u32 h_mixer, u16 control_index, u16 *pw_src_node_type, u16 *pw_src_node_index, u16 *pw_dst_node_type, u16 *pw_dst_node_index, u16 *pw_control_type, u32 *ph_control); -u16 hpi_mixer_store(u32 h_mixer, enum HPI_MIXER_STORE_COMMAND command, - u16 index); /************/ /* Controls */ /************/ @@ -1358,18 +1287,6 @@ u16 hpi_volume_get_mute(u32 h_control, u32 *mute); u16 hpi_volume_query_range(u32 h_control, short *min_gain_01dB, short *max_gain_01dB, short *step_gain_01dB); -u16 hpi_volume_query_channels(const u32 h_control, u32 *p_channels); - -u16 hpi_volume_auto_fade(u32 h_control, - short an_stop_gain0_01dB[HPI_MAX_CHANNELS], u32 duration_ms); - -u16 hpi_volume_auto_fade_profile(u32 h_control, - short an_stop_gain0_01dB[HPI_MAX_CHANNELS], u32 duration_ms, - u16 profile); - -u16 hpi_volume_query_auto_fade_profile(const u32 h_control, const u32 i, - u16 *profile); - /*****************/ /* Level control */ /*****************/ @@ -1392,17 +1309,6 @@ u16 hpi_meter_get_peak(u32 h_control, short an_peak0_01dB_out[HPI_MAX_CHANNELS] ); -u16 hpi_meter_get_rms(u32 h_control, short an_peak0_01dB_out[HPI_MAX_CHANNELS] - ); - -u16 hpi_meter_set_peak_ballistics(u32 h_control, u16 attack, u16 decay); - -u16 hpi_meter_set_rms_ballistics(u32 h_control, u16 attack, u16 decay); - -u16 hpi_meter_get_peak_ballistics(u32 h_control, u16 *attack, u16 *decay); - -u16 hpi_meter_get_rms_ballistics(u32 h_control, u16 *attack, u16 *decay); - /************************/ /* ChannelMode control */ /************************/ @@ -1429,106 +1335,31 @@ u16 hpi_tuner_set_frequency(u32 h_control, u32 freq_ink_hz); u16 hpi_tuner_get_frequency(u32 h_control, u32 *pw_freq_ink_hz); -u16 hpi_tuner_get_rf_level(u32 h_control, short *pw_level); - -u16 hpi_tuner_get_raw_rf_level(u32 h_control, short *pw_level); - u16 hpi_tuner_query_gain(const u32 h_tuner, const u32 index, u16 *pw_gain); u16 hpi_tuner_set_gain(u32 h_control, short gain); u16 hpi_tuner_get_gain(u32 h_control, short *pn_gain); -u16 hpi_tuner_get_status(u32 h_control, u16 *pw_status_mask, u16 *pw_status); - -u16 hpi_tuner_set_mode(u32 h_control, u32 mode, u32 value); - -u16 hpi_tuner_get_mode(u32 h_control, u32 mode, u32 *pn_value); - -u16 hpi_tuner_get_rds(u32 h_control, char *p_rds_data); - -u16 hpi_tuner_query_deemphasis(const u32 h_tuner, const u32 index, - const u16 band, u32 *pdeemphasis); - -u16 hpi_tuner_set_deemphasis(u32 h_control, u32 deemphasis); -u16 hpi_tuner_get_deemphasis(u32 h_control, u32 *pdeemphasis); - -u16 hpi_tuner_query_program(const u32 h_tuner, u32 *pbitmap_program); - -u16 hpi_tuner_set_program(u32 h_control, u32 program); - -u16 hpi_tuner_get_program(u32 h_control, u32 *pprogram); - -u16 hpi_tuner_get_hd_radio_dsp_version(u32 h_control, char *psz_dsp_version, - const u32 string_size); - -u16 hpi_tuner_get_hd_radio_sdk_version(u32 h_control, char *psz_sdk_version, - const u32 string_size); - -u16 hpi_tuner_get_hd_radio_signal_quality(u32 h_control, u32 *pquality); - -u16 hpi_tuner_get_hd_radio_signal_blend(u32 h_control, u32 *pblend); - -u16 hpi_tuner_set_hd_radio_signal_blend(u32 h_control, const u32 blend); - /***************/ /* PAD control */ /***************/ -u16 hpi_pad_get_channel_name(u32 h_control, char *psz_string, - const u32 string_length); - -u16 hpi_pad_get_artist(u32 h_control, char *psz_string, - const u32 string_length); - -u16 hpi_pad_get_title(u32 h_control, char *psz_string, - const u32 string_length); - -u16 hpi_pad_get_comment(u32 h_control, char *psz_string, - const u32 string_length); - -u16 hpi_pad_get_program_type(u32 h_control, u32 *ppTY); - -u16 hpi_pad_get_rdsPI(u32 h_control, u32 *ppI); - u16 hpi_pad_get_program_type_string(u32 h_control, const u32 data_type, const u32 pTY, char *psz_string, const u32 string_length); /****************************/ /* AES/EBU Receiver control */ /****************************/ -u16 hpi_aesebu_receiver_query_format(const u32 h_aes_rx, const u32 index, - u16 *pw_format); - u16 hpi_aesebu_receiver_set_format(u32 h_control, u16 source); u16 hpi_aesebu_receiver_get_format(u32 h_control, u16 *pw_source); -u16 hpi_aesebu_receiver_get_sample_rate(u32 h_control, u32 *psample_rate); - -u16 hpi_aesebu_receiver_get_user_data(u32 h_control, u16 index, u16 *pw_data); - -u16 hpi_aesebu_receiver_get_channel_status(u32 h_control, u16 index, - u16 *pw_data); - u16 hpi_aesebu_receiver_get_error_status(u32 h_control, u16 *pw_error_data); /*******************************/ /* AES/EBU Transmitter control */ /*******************************/ -u16 hpi_aesebu_transmitter_set_sample_rate(u32 h_control, u32 sample_rate); - -u16 hpi_aesebu_transmitter_set_user_data(u32 h_control, u16 index, u16 data); - -u16 hpi_aesebu_transmitter_set_channel_status(u32 h_control, u16 index, - u16 data); - -u16 hpi_aesebu_transmitter_get_channel_status(u32 h_control, u16 index, - u16 *pw_data); - -u16 hpi_aesebu_transmitter_query_format(const u32 h_aes_tx, const u32 index, - u16 *pw_format); - u16 hpi_aesebu_transmitter_set_format(u32 h_control, u16 output_format); u16 hpi_aesebu_transmitter_get_format(u32 h_control, u16 *pw_output_format); @@ -1545,22 +1376,6 @@ u16 hpi_multiplexer_get_source(u32 h_control, u16 *source_node_type, u16 hpi_multiplexer_query_source(u32 h_control, u16 index, u16 *source_node_type, u16 *source_node_index); -/***************/ -/* Vox control */ -/***************/ -u16 hpi_vox_set_threshold(u32 h_control, short an_gain0_01dB); - -u16 hpi_vox_get_threshold(u32 h_control, short *an_gain0_01dB); - -/*********************/ -/* Bitstream control */ -/*********************/ -u16 hpi_bitstream_set_clock_edge(u32 h_control, u16 edge_type); - -u16 hpi_bitstream_set_data_polarity(u32 h_control, u16 polarity); - -u16 hpi_bitstream_get_activity(u32 h_control, u16 *pw_clk_activity, - u16 *pw_data_activity); /***********************/ /* SampleClock control */ @@ -1589,72 +1404,6 @@ u16 hpi_sample_clock_set_local_rate(u32 h_control, u32 sample_rate); u16 hpi_sample_clock_get_local_rate(u32 h_control, u32 *psample_rate); -u16 hpi_sample_clock_set_auto(u32 h_control, u32 enable); - -u16 hpi_sample_clock_get_auto(u32 h_control, u32 *penable); - -u16 hpi_sample_clock_set_local_rate_lock(u32 h_control, u32 lock); - -u16 hpi_sample_clock_get_local_rate_lock(u32 h_control, u32 *plock); - -/***********************/ -/* Microphone control */ -/***********************/ -u16 hpi_microphone_set_phantom_power(u32 h_control, u16 on_off); - -u16 hpi_microphone_get_phantom_power(u32 h_control, u16 *pw_on_off); - -/********************************/ -/* Parametric Equalizer control */ -/********************************/ -u16 hpi_parametric_eq_get_info(u32 h_control, u16 *pw_number_of_bands, - u16 *pw_enabled); - -u16 hpi_parametric_eq_set_state(u32 h_control, u16 on_off); - -u16 hpi_parametric_eq_set_band(u32 h_control, u16 index, u16 type, - u32 frequency_hz, short q100, short gain0_01dB); - -u16 hpi_parametric_eq_get_band(u32 h_control, u16 index, u16 *pn_type, - u32 *pfrequency_hz, short *pnQ100, short *pn_gain0_01dB); - -u16 hpi_parametric_eq_get_coeffs(u32 h_control, u16 index, short coeffs[5] - ); - -/*******************************/ -/* Compressor Expander control */ -/*******************************/ - -u16 hpi_compander_set_enable(u32 h_control, u32 on); - -u16 hpi_compander_get_enable(u32 h_control, u32 *pon); - -u16 hpi_compander_set_makeup_gain(u32 h_control, short makeup_gain0_01dB); - -u16 hpi_compander_get_makeup_gain(u32 h_control, short *pn_makeup_gain0_01dB); - -u16 hpi_compander_set_attack_time_constant(u32 h_control, u32 index, - u32 attack); - -u16 hpi_compander_get_attack_time_constant(u32 h_control, u32 index, - u32 *pw_attack); - -u16 hpi_compander_set_decay_time_constant(u32 h_control, u32 index, - u32 decay); - -u16 hpi_compander_get_decay_time_constant(u32 h_control, u32 index, - u32 *pw_decay); - -u16 hpi_compander_set_threshold(u32 h_control, u32 index, - short threshold0_01dB); - -u16 hpi_compander_get_threshold(u32 h_control, u32 index, - short *pn_threshold0_01dB); - -u16 hpi_compander_set_ratio(u32 h_control, u32 index, u32 ratio100); - -u16 hpi_compander_get_ratio(u32 h_control, u32 index, u32 *pw_ratio100); - /********************/ /* Cobranet control */ /********************/ @@ -1664,59 +1413,6 @@ u16 hpi_cobranet_hmi_write(u32 h_control, u32 hmi_address, u32 byte_count, u16 hpi_cobranet_hmi_read(u32 h_control, u32 hmi_address, u32 max_byte_count, u32 *pbyte_count, u8 *pb_data); -u16 hpi_cobranet_hmi_get_status(u32 h_control, u32 *pstatus, - u32 *preadable_size, u32 *pwriteable_size); - -u16 hpi_cobranet_get_ip_address(u32 h_control, u32 *pdw_ip_address); - -u16 hpi_cobranet_set_ip_address(u32 h_control, u32 dw_ip_address); - -u16 hpi_cobranet_get_static_ip_address(u32 h_control, u32 *pdw_ip_address); - -u16 hpi_cobranet_set_static_ip_address(u32 h_control, u32 dw_ip_address); - -u16 hpi_cobranet_get_macaddress(u32 h_control, u32 *p_mac_msbs, - u32 *p_mac_lsbs); - -/*************************/ -/* Tone Detector control */ -/*************************/ -u16 hpi_tone_detector_get_state(u32 hC, u32 *state); - -u16 hpi_tone_detector_set_enable(u32 hC, u32 enable); - -u16 hpi_tone_detector_get_enable(u32 hC, u32 *enable); - -u16 hpi_tone_detector_set_event_enable(u32 hC, u32 event_enable); - -u16 hpi_tone_detector_get_event_enable(u32 hC, u32 *event_enable); - -u16 hpi_tone_detector_set_threshold(u32 hC, int threshold); - -u16 hpi_tone_detector_get_threshold(u32 hC, int *threshold); - -u16 hpi_tone_detector_get_frequency(u32 hC, u32 index, u32 *frequency); - -/****************************/ -/* Silence Detector control */ -/****************************/ -u16 hpi_silence_detector_get_state(u32 hC, u32 *state); - -u16 hpi_silence_detector_set_enable(u32 hC, u32 enable); - -u16 hpi_silence_detector_get_enable(u32 hC, u32 *enable); - -u16 hpi_silence_detector_set_event_enable(u32 hC, u32 event_enable); - -u16 hpi_silence_detector_get_event_enable(u32 hC, u32 *event_enable); - -u16 hpi_silence_detector_set_delay(u32 hC, u32 delay); - -u16 hpi_silence_detector_get_delay(u32 hC, u32 *delay); - -u16 hpi_silence_detector_set_threshold(u32 hC, int threshold); - -u16 hpi_silence_detector_get_threshold(u32 hC, int *threshold); /*********************/ /* Utility functions */ /*********************/ diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index bc86cb7..7c11994 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -1404,12 +1404,6 @@ void hpi_send_recv(struct hpi_message *phm, struct hpi_response *phr); u16 hpi_subsys_create_adapter(const struct hpi_resource *p_resource, u16 *pw_adapter_index); -u16 hpi_outstream_host_buffer_get_info(u32 h_outstream, u8 **pp_buffer, - struct hpi_hostbuffer_status **pp_status); - -u16 hpi_instream_host_buffer_get_info(u32 h_instream, u8 **pp_buffer, - struct hpi_hostbuffer_status **pp_status); - u16 hpi_adapter_restart(u16 adapter_index); /* @@ -1421,7 +1415,6 @@ compatibility. void hpi_format_to_msg(struct hpi_msg_format *pMF, const struct hpi_format *pF); -void hpi_stream_response_to_legacy(struct hpi_stream_res *pSR); /*////////////////////////////////////////////////////////////////////////// */ /* declarations for individual HPI entry points */ diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 510e56c..5d4d2f6 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -80,13 +80,6 @@ static void hpi_msg_to_format(struct hpi_format *pF, pF->unused = 0; } -void hpi_stream_response_to_legacy(struct hpi_stream_res *pSR) -{ - pSR->u.legacy_stream_info.auxiliary_data_available = - pSR->u.stream_info.auxiliary_data_available; - pSR->u.legacy_stream_info.state = pSR->u.stream_info.state; -} - static inline void hpi_send_recvV1(struct hpi_message_header *m, struct hpi_response_header *r) { @@ -105,93 +98,6 @@ u16 hpi_subsys_get_version_ex(u32 *pversion_ex) return hr.error; } -u16 hpi_subsys_get_num_adapters(int *pn_num_adapters) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM, - HPI_SUBSYS_GET_NUM_ADAPTERS); - hpi_send_recv(&hm, &hr); - *pn_num_adapters = (int)hr.u.s.num_adapters; - return hr.error; -} - -u16 hpi_subsys_get_adapter(int iterator, u32 *padapter_index, - u16 *pw_adapter_type) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM, - HPI_SUBSYS_GET_ADAPTER); - hm.obj_index = (u16)iterator; - hpi_send_recv(&hm, &hr); - *padapter_index = (int)hr.u.s.adapter_index; - *pw_adapter_type = hr.u.s.adapter_type; - - return hr.error; -} - -u16 hpi_adapter_open(u16 adapter_index) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, - HPI_ADAPTER_OPEN); - hm.adapter_index = adapter_index; - - hpi_send_recv(&hm, &hr); - - return hr.error; - -} - -u16 hpi_adapter_close(u16 adapter_index) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, - HPI_ADAPTER_CLOSE); - hm.adapter_index = adapter_index; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_adapter_set_mode(u16 adapter_index, u32 adapter_mode) -{ - return hpi_adapter_set_mode_ex(adapter_index, adapter_mode, - HPI_ADAPTER_MODE_SET); -} - -u16 hpi_adapter_set_mode_ex(u16 adapter_index, u32 adapter_mode, - u16 query_or_set) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, - HPI_ADAPTER_SET_MODE); - hm.adapter_index = adapter_index; - hm.u.ax.mode.adapter_mode = adapter_mode; - hm.u.ax.mode.query_or_set = query_or_set; - hpi_send_recv(&hm, &hr); - return hr.error; -} - -u16 hpi_adapter_get_mode(u16 adapter_index, u32 *padapter_mode) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, - HPI_ADAPTER_GET_MODE); - hm.adapter_index = adapter_index; - hpi_send_recv(&hm, &hr); - if (padapter_mode) - *padapter_mode = hr.u.ax.mode.adapter_mode; - return hr.error; -} - u16 hpi_adapter_get_info(u16 adapter_index, u16 *pw_num_outstreams, u16 *pw_num_instreams, u16 *pw_version, u32 *pserial_number, u16 *pw_adapter_type) @@ -212,30 +118,6 @@ u16 hpi_adapter_get_info(u16 adapter_index, u16 *pw_num_outstreams, return hr.error; } -u16 hpi_adapter_get_module_by_index(u16 adapter_index, u16 module_index, - u16 *pw_num_outputs, u16 *pw_num_inputs, u16 *pw_version, - u32 *pserial_number, u16 *pw_module_type, u32 *ph_module) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, - HPI_ADAPTER_MODULE_INFO); - hm.adapter_index = adapter_index; - hm.u.ax.module_info.index = module_index; - - hpi_send_recv(&hm, &hr); - - *pw_module_type = hr.u.ax.info.adapter_type; - *pw_num_outputs = hr.u.ax.info.num_outstreams; - *pw_num_inputs = hr.u.ax.info.num_instreams; - *pw_version = hr.u.ax.info.version; - *pserial_number = hr.u.ax.info.serial_number; - *ph_module = 0; - - return hr.error; -} - u16 hpi_adapter_set_property(u16 adapter_index, u16 property, u16 parameter1, u16 parameter2) { @@ -274,12 +156,6 @@ u16 hpi_adapter_get_property(u16 adapter_index, u16 property, return hr.error; } -u16 hpi_adapter_enumerate_property(u16 adapter_index, u16 index, - u16 what_to_enumerate, u16 property_index, u32 *psetting) -{ - return 0; -} - u16 hpi_format_create(struct hpi_format *p_format, u16 channels, u16 format, u32 sample_rate, u32 bit_rate, u32 attributes) { @@ -375,52 +251,6 @@ u16 hpi_format_create(struct hpi_format *p_format, u16 channels, u16 format, return err; } -u16 hpi_stream_estimate_buffer_size(struct hpi_format *p_format, - u32 host_polling_rate_in_milli_seconds, u32 *recommended_buffer_size) -{ - - u32 bytes_per_second; - u32 size; - u16 channels; - struct hpi_format *pF = p_format; - - channels = pF->channels; - - switch (pF->format) { - case HPI_FORMAT_PCM16_BIGENDIAN: - case HPI_FORMAT_PCM16_SIGNED: - bytes_per_second = pF->sample_rate * 2L * channels; - break; - case HPI_FORMAT_PCM24_SIGNED: - bytes_per_second = pF->sample_rate * 3L * channels; - break; - case HPI_FORMAT_PCM32_SIGNED: - case HPI_FORMAT_PCM32_FLOAT: - bytes_per_second = pF->sample_rate * 4L * channels; - break; - case HPI_FORMAT_PCM8_UNSIGNED: - bytes_per_second = pF->sample_rate * 1L * channels; - break; - case HPI_FORMAT_MPEG_L1: - case HPI_FORMAT_MPEG_L2: - case HPI_FORMAT_MPEG_L3: - bytes_per_second = pF->bit_rate / 8L; - break; - case HPI_FORMAT_DOLBY_AC2: - - bytes_per_second = 256000L / 8L; - break; - default: - return HPI_ERROR_INVALID_FORMAT; - } - size = (bytes_per_second * host_polling_rate_in_milli_seconds * 2) / - 1000L; - - *recommended_buffer_size = - roundup_pow_of_two(((size + 4095L) & ~4095L)); - return 0; -} - u16 hpi_outstream_open(u16 adapter_index, u16 outstream_index, u32 *ph_outstream) { @@ -527,12 +357,12 @@ u16 hpi_outstream_start(u32 h_outstream) return hr.error; } -u16 hpi_outstream_wait_start(u32 h_outstream) +u16 hpi_outstream_stop(u32 h_outstream) { struct hpi_message hm; struct hpi_response hr; hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_WAIT_START); + HPI_OSTREAM_STOP); if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; @@ -541,12 +371,12 @@ u16 hpi_outstream_wait_start(u32 h_outstream) return hr.error; } -u16 hpi_outstream_stop(u32 h_outstream) +u16 hpi_outstream_reset(u32 h_outstream) { struct hpi_message hm; struct hpi_response hr; hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_STOP); + HPI_OSTREAM_RESET); if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; @@ -555,432 +385,231 @@ u16 hpi_outstream_stop(u32 h_outstream) return hr.error; } -u16 hpi_outstream_sinegen(u32 h_outstream) +u16 hpi_outstream_query_format(u32 h_outstream, struct hpi_format *p_format) { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_SINEGEN); + HPI_OSTREAM_QUERY_FORMAT); if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; + hpi_format_to_msg(&hm.u.d.u.data.format, p_format); + hpi_send_recv(&hm, &hr); return hr.error; } -u16 hpi_outstream_reset(u32 h_outstream) +u16 hpi_outstream_group_add(u32 h_outstream, u32 h_stream) { struct hpi_message hm; struct hpi_response hr; + u16 adapter; + char c_obj_type; + hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_RESET); + HPI_OSTREAM_GROUP_ADD); + if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); + if (hpi_handle_indexes(h_stream, &adapter, + &hm.u.d.u.stream.stream_index)) + return HPI_ERROR_INVALID_HANDLE; + + c_obj_type = hpi_handle_object(h_stream); + switch (c_obj_type) { + case HPI_OBJ_OSTREAM: + case HPI_OBJ_ISTREAM: + hm.u.d.u.stream.object_type = c_obj_type; + break; + default: + return HPI_ERROR_INVALID_OBJ; + } + if (adapter != hm.adapter_index) + return HPI_ERROR_NO_INTERADAPTER_GROUPS; + hpi_send_recv(&hm, &hr); return hr.error; } -u16 hpi_outstream_query_format(u32 h_outstream, struct hpi_format *p_format) +u16 hpi_outstream_group_get_map(u32 h_outstream, u32 *poutstream_map, + u32 *pinstream_map) { struct hpi_message hm; struct hpi_response hr; hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_QUERY_FORMAT); + HPI_OSTREAM_GROUP_GETMAP); if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - - hpi_format_to_msg(&hm.u.d.u.data.format, p_format); - hpi_send_recv(&hm, &hr); + if (poutstream_map) + *poutstream_map = hr.u.d.u.group_info.outstream_group_map; + if (pinstream_map) + *pinstream_map = hr.u.d.u.group_info.instream_group_map; + return hr.error; } -u16 hpi_outstream_set_format(u32 h_outstream, struct hpi_format *p_format) +u16 hpi_outstream_group_reset(u32 h_outstream) { struct hpi_message hm; struct hpi_response hr; hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_SET_FORMAT); + HPI_OSTREAM_GROUP_RESET); if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - - hpi_format_to_msg(&hm.u.d.u.data.format, p_format); - hpi_send_recv(&hm, &hr); - return hr.error; } -u16 hpi_outstream_set_velocity(u32 h_outstream, short velocity) +u16 hpi_instream_open(u16 adapter_index, u16 instream_index, u32 *ph_instream) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_SET_VELOCITY); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.velocity = velocity; + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_OPEN); + hm.adapter_index = adapter_index; + hm.obj_index = instream_index; hpi_send_recv(&hm, &hr); + if (hr.error == 0) + *ph_instream = + hpi_indexes_to_handle(HPI_OBJ_ISTREAM, adapter_index, + instream_index); + else + *ph_instream = 0; + return hr.error; } -u16 hpi_outstream_set_punch_in_out(u32 h_outstream, u32 punch_in_sample, - u32 punch_out_sample) +u16 hpi_instream_close(u32 h_instream) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_SET_PUNCHINOUT); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_HOSTBUFFER_FREE); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; + hpi_send_recv(&hm, &hr); - hm.u.d.u.pio.punch_in_sample = punch_in_sample; - hm.u.d.u.pio.punch_out_sample = punch_out_sample; + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_GROUP_RESET); + hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index); + hpi_send_recv(&hm, &hr); + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_CLOSE); + hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index); hpi_send_recv(&hm, &hr); return hr.error; } -u16 hpi_outstream_ancillary_reset(u32 h_outstream, u16 mode) +u16 hpi_instream_query_format(u32 h_instream, + const struct hpi_format *p_format) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_ANC_RESET); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_QUERY_FORMAT); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.format.channels = mode; + hpi_format_to_msg(&hm.u.d.u.data.format, p_format); + hpi_send_recv(&hm, &hr); + return hr.error; } -u16 hpi_outstream_ancillary_get_info(u32 h_outstream, u32 *pframes_available) +u16 hpi_instream_set_format(u32 h_instream, const struct hpi_format *p_format) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_ANC_GET_INFO); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_SET_FORMAT); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; + hpi_format_to_msg(&hm.u.d.u.data.format, p_format); + hpi_send_recv(&hm, &hr); - if (hr.error == 0) { - if (pframes_available) - *pframes_available = - hr.u.d.u.stream_info.data_available / - sizeof(struct hpi_anc_frame); - } + return hr.error; } -u16 hpi_outstream_ancillary_read(u32 h_outstream, - struct hpi_anc_frame *p_anc_frame_buffer, - u32 anc_frame_buffer_size_in_bytes, - u32 number_of_ancillary_frames_to_read) +u16 hpi_instream_read_buf(u32 h_instream, u8 *pb_data, u32 bytes_to_read) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_ANC_READ); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_READ); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.pb_data = (u8 *)p_anc_frame_buffer; - hm.u.d.u.data.data_size = - number_of_ancillary_frames_to_read * - sizeof(struct hpi_anc_frame); - if (hm.u.d.u.data.data_size <= anc_frame_buffer_size_in_bytes) - hpi_send_recv(&hm, &hr); - else - hr.error = HPI_ERROR_INVALID_DATASIZE; + hm.u.d.u.data.data_size = bytes_to_read; + hm.u.d.u.data.pb_data = pb_data; + + hpi_send_recv(&hm, &hr); + return hr.error; } -u16 hpi_outstream_set_time_scale(u32 h_outstream, u32 time_scale) +u16 hpi_instream_start(u32 h_instream) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_SET_TIMESCALE); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_START); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.time_scale = time_scale; - hpi_send_recv(&hm, &hr); return hr.error; } -u16 hpi_outstream_host_buffer_allocate(u32 h_outstream, u32 size_in_bytes) +u16 hpi_instream_wait_start(u32 h_instream) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_HOSTBUFFER_ALLOC); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_WAIT_START); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.data_size = size_in_bytes; + hpi_send_recv(&hm, &hr); + return hr.error; } -u16 hpi_outstream_host_buffer_get_info(u32 h_outstream, u8 **pp_buffer, - struct hpi_hostbuffer_status **pp_status) +u16 hpi_instream_stop(u32 h_instream) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_HOSTBUFFER_GET_INFO); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_STOP); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; + hpi_send_recv(&hm, &hr); - if (hr.error == 0) { - if (pp_buffer) - *pp_buffer = hr.u.d.u.hostbuffer_info.p_buffer; - if (pp_status) - *pp_status = hr.u.d.u.hostbuffer_info.p_status; - } return hr.error; } -u16 hpi_outstream_host_buffer_free(u32 h_outstream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_HOSTBUFFER_FREE); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); - return hr.error; -} - -u16 hpi_outstream_group_add(u32 h_outstream, u32 h_stream) -{ - struct hpi_message hm; - struct hpi_response hr; - u16 adapter; - char c_obj_type; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_GROUP_ADD); - - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - if (hpi_handle_indexes(h_stream, &adapter, - &hm.u.d.u.stream.stream_index)) - return HPI_ERROR_INVALID_HANDLE; - - c_obj_type = hpi_handle_object(h_stream); - switch (c_obj_type) { - case HPI_OBJ_OSTREAM: - case HPI_OBJ_ISTREAM: - hm.u.d.u.stream.object_type = c_obj_type; - break; - default: - return HPI_ERROR_INVALID_OBJ; - } - if (adapter != hm.adapter_index) - return HPI_ERROR_NO_INTERADAPTER_GROUPS; - - hpi_send_recv(&hm, &hr); - return hr.error; -} - -u16 hpi_outstream_group_get_map(u32 h_outstream, u32 *poutstream_map, - u32 *pinstream_map) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_GROUP_GETMAP); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); - - if (poutstream_map) - *poutstream_map = hr.u.d.u.group_info.outstream_group_map; - if (pinstream_map) - *pinstream_map = hr.u.d.u.group_info.instream_group_map; - - return hr.error; -} - -u16 hpi_outstream_group_reset(u32 h_outstream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_GROUP_RESET); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); - return hr.error; -} - -u16 hpi_instream_open(u16 adapter_index, u16 instream_index, u32 *ph_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_OPEN); - hm.adapter_index = adapter_index; - hm.obj_index = instream_index; - - hpi_send_recv(&hm, &hr); - - if (hr.error == 0) - *ph_instream = - hpi_indexes_to_handle(HPI_OBJ_ISTREAM, adapter_index, - instream_index); - else - *ph_instream = 0; - - return hr.error; -} - -u16 hpi_instream_close(u32 h_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_HOSTBUFFER_FREE); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_GROUP_RESET); - hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index); - hpi_send_recv(&hm, &hr); - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_CLOSE); - hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index); - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_query_format(u32 h_instream, - const struct hpi_format *p_format) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_QUERY_FORMAT); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_format_to_msg(&hm.u.d.u.data.format, p_format); - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_set_format(u32 h_instream, const struct hpi_format *p_format) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_SET_FORMAT); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_format_to_msg(&hm.u.d.u.data.format, p_format); - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_read_buf(u32 h_instream, u8 *pb_data, u32 bytes_to_read) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_READ); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.data_size = bytes_to_read; - hm.u.d.u.data.pb_data = pb_data; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_start(u32 h_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_START); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_wait_start(u32 h_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_WAIT_START); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_stop(u32 h_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_STOP); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_reset(u32 h_instream) +u16 hpi_instream_reset(u32 h_instream) { struct hpi_message hm; struct hpi_response hr; @@ -1022,98 +651,6 @@ u16 hpi_instream_get_info_ex(u32 h_instream, u16 *pw_state, u32 *pbuffer_size, return hr.error; } -u16 hpi_instream_ancillary_reset(u32 h_instream, u16 bytes_per_frame, - u16 mode, u16 alignment, u16 idle_bit) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_ANC_RESET); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.format.attributes = bytes_per_frame; - hm.u.d.u.data.format.format = (mode << 8) | (alignment & 0xff); - hm.u.d.u.data.format.channels = idle_bit; - hpi_send_recv(&hm, &hr); - return hr.error; -} - -u16 hpi_instream_ancillary_get_info(u32 h_instream, u32 *pframe_space) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_ANC_GET_INFO); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); - if (pframe_space) - *pframe_space = - (hr.u.d.u.stream_info.buffer_size - - hr.u.d.u.stream_info.data_available) / - sizeof(struct hpi_anc_frame); - return hr.error; -} - -u16 hpi_instream_ancillary_write(u32 h_instream, - const struct hpi_anc_frame *p_anc_frame_buffer, - u32 anc_frame_buffer_size_in_bytes, - u32 number_of_ancillary_frames_to_write) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_ANC_WRITE); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.pb_data = (u8 *)p_anc_frame_buffer; - hm.u.d.u.data.data_size = - number_of_ancillary_frames_to_write * - sizeof(struct hpi_anc_frame); - if (hm.u.d.u.data.data_size <= anc_frame_buffer_size_in_bytes) - hpi_send_recv(&hm, &hr); - else - hr.error = HPI_ERROR_INVALID_DATASIZE; - return hr.error; -} - -u16 hpi_instream_host_buffer_allocate(u32 h_instream, u32 size_in_bytes) -{ - - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_HOSTBUFFER_ALLOC); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.data_size = size_in_bytes; - hpi_send_recv(&hm, &hr); - return hr.error; -} - -u16 hpi_instream_host_buffer_get_info(u32 h_instream, u8 **pp_buffer, - struct hpi_hostbuffer_status **pp_status) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_HOSTBUFFER_GET_INFO); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); - - if (hr.error == 0) { - if (pp_buffer) - *pp_buffer = hr.u.d.u.hostbuffer_info.p_buffer; - if (pp_status) - *pp_status = hr.u.d.u.hostbuffer_info.p_status; - } - return hr.error; -} - u16 hpi_instream_host_buffer_free(u32 h_instream) { @@ -1215,19 +752,6 @@ u16 hpi_mixer_open(u16 adapter_index, u32 *ph_mixer) return hr.error; } -u16 hpi_mixer_close(u32 h_mixer) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_MIXER, HPI_MIXER_CLOSE); - if (hpi_handle_indexes(h_mixer, &hm.adapter_index, NULL)) - return HPI_ERROR_INVALID_HANDLE; - - hpi_send_recv(&hm, &hr); - return hr.error; -} - u16 hpi_mixer_get_control(u32 h_mixer, u16 src_node_type, u16 src_node_type_index, u16 dst_node_type, u16 dst_node_type_index, u16 control_type, u32 *ph_control) @@ -1289,20 +813,6 @@ u16 hpi_mixer_get_control_by_index(u32 h_mixer, u16 control_index, return hr.error; } -u16 hpi_mixer_store(u32 h_mixer, enum HPI_MIXER_STORE_COMMAND command, - u16 index) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_MIXER, HPI_MIXER_STORE); - if (hpi_handle_indexes(h_mixer, &hm.adapter_index, NULL)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.mx.store.command = command; - hm.u.mx.store.index = index; - hpi_send_recv(&hm, &hr); - return hr.error; -} - static u16 hpi_control_param_set(const u32 h_control, const u16 attrib, const u32 param1, const u32 param2) @@ -1465,17 +975,6 @@ static u16 hpi_control_get_string(const u32 h_control, const u16 attribute, return err; } -u16 hpi_aesebu_receiver_query_format(const u32 h_aes_rx, const u32 index, - u16 *pw_format) -{ - u32 qr; - u16 err; - - err = hpi_control_query(h_aes_rx, HPI_AESEBURX_FORMAT, index, 0, &qr); - *pw_format = (u16)qr; - return err; -} - u16 hpi_aesebu_receiver_set_format(u32 h_control, u16 format) { return hpi_control_param_set(h_control, HPI_AESEBURX_FORMAT, format, @@ -1494,49 +993,6 @@ u16 hpi_aesebu_receiver_get_format(u32 h_control, u16 *pw_format) return err; } -u16 hpi_aesebu_receiver_get_sample_rate(u32 h_control, u32 *psample_rate) -{ - return hpi_control_param1_get(h_control, HPI_AESEBURX_SAMPLERATE, - psample_rate); -} - -u16 hpi_aesebu_receiver_get_user_data(u32 h_control, u16 index, u16 *pw_data) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_AESEBURX_USERDATA; - hm.u.c.param1 = index; - - hpi_send_recv(&hm, &hr); - - if (pw_data) - *pw_data = (u16)hr.u.c.param2; - return hr.error; -} - -u16 hpi_aesebu_receiver_get_channel_status(u32 h_control, u16 index, - u16 *pw_data) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_AESEBURX_CHANNELSTATUS; - hm.u.c.param1 = index; - - hpi_send_recv(&hm, &hr); - - if (pw_data) - *pw_data = (u16)hr.u.c.param2; - return hr.error; -} - u16 hpi_aesebu_receiver_get_error_status(u32 h_control, u16 *pw_error_data) { u32 error_data = 0; @@ -1549,42 +1005,6 @@ u16 hpi_aesebu_receiver_get_error_status(u32 h_control, u16 *pw_error_data) return err; } -u16 hpi_aesebu_transmitter_set_sample_rate(u32 h_control, u32 sample_rate) -{ - return hpi_control_param_set(h_control, HPI_AESEBUTX_SAMPLERATE, - sample_rate, 0); -} - -u16 hpi_aesebu_transmitter_set_user_data(u32 h_control, u16 index, u16 data) -{ - return hpi_control_param_set(h_control, HPI_AESEBUTX_USERDATA, index, - data); -} - -u16 hpi_aesebu_transmitter_set_channel_status(u32 h_control, u16 index, - u16 data) -{ - return hpi_control_param_set(h_control, HPI_AESEBUTX_CHANNELSTATUS, - index, data); -} - -u16 hpi_aesebu_transmitter_get_channel_status(u32 h_control, u16 index, - u16 *pw_data) -{ - return HPI_ERROR_INVALID_OPERATION; -} - -u16 hpi_aesebu_transmitter_query_format(const u32 h_aes_tx, const u32 index, - u16 *pw_format) -{ - u32 qr; - u16 err; - - err = hpi_control_query(h_aes_tx, HPI_AESEBUTX_FORMAT, index, 0, &qr); - *pw_format = (u16)qr; - return err; -} - u16 hpi_aesebu_transmitter_set_format(u32 h_control, u16 output_format) { return hpi_control_param_set(h_control, HPI_AESEBUTX_FORMAT, @@ -1598,39 +1018,9 @@ u16 hpi_aesebu_transmitter_get_format(u32 h_control, u16 *pw_output_format) err = hpi_control_param1_get(h_control, HPI_AESEBUTX_FORMAT, &param); if (!err && pw_output_format) - *pw_output_format = (u16)param; - - return err; -} - -u16 hpi_bitstream_set_clock_edge(u32 h_control, u16 edge_type) -{ - return hpi_control_param_set(h_control, HPI_BITSTREAM_CLOCK_EDGE, - edge_type, 0); -} - -u16 hpi_bitstream_set_data_polarity(u32 h_control, u16 polarity) -{ - return hpi_control_param_set(h_control, HPI_BITSTREAM_DATA_POLARITY, - polarity, 0); -} - -u16 hpi_bitstream_get_activity(u32 h_control, u16 *pw_clk_activity, - u16 *pw_data_activity) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_BITSTREAM_ACTIVITY; - hpi_send_recv(&hm, &hr); - if (pw_clk_activity) - *pw_clk_activity = (u16)hr.u.c.param1; - if (pw_data_activity) - *pw_data_activity = (u16)hr.u.c.param2; - return hr.error; + *pw_output_format = (u16)param; + + return err; } u16 hpi_channel_mode_query_mode(const u32 h_mode, const u32 index, @@ -1723,243 +1113,6 @@ u16 hpi_cobranet_hmi_read(u32 h_control, u32 hmi_address, u32 max_byte_count, return hr.h.error; } -u16 hpi_cobranet_hmi_get_status(u32 h_control, u32 *pstatus, - u32 *preadable_size, u32 *pwriteable_size) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hm.u.c.attribute = HPI_COBRANET_GET_STATUS; - - hpi_send_recv(&hm, &hr); - if (!hr.error) { - if (pstatus) - *pstatus = hr.u.cu.cobranet.status.status; - if (preadable_size) - *preadable_size = - hr.u.cu.cobranet.status.readable_size; - if (pwriteable_size) - *pwriteable_size = - hr.u.cu.cobranet.status.writeable_size; - } - return hr.error; -} - -u16 hpi_cobranet_get_ip_address(u32 h_control, u32 *pdw_ip_address) -{ - u32 byte_count; - u32 iP; - u16 err; - - err = hpi_cobranet_hmi_read(h_control, - HPI_COBRANET_HMI_cobra_ip_mon_currentIP, 4, &byte_count, - (u8 *)&iP); - - *pdw_ip_address = - ((iP & 0xff000000) >> 8) | ((iP & 0x00ff0000) << 8) | ((iP & - 0x0000ff00) >> 8) | ((iP & 0x000000ff) << 8); - - if (err) - *pdw_ip_address = 0; - - return err; - -} - -u16 hpi_cobranet_set_ip_address(u32 h_control, u32 dw_ip_address) -{ - u32 iP; - u16 err; - - iP = ((dw_ip_address & 0xff000000) >> 8) | ((dw_ip_address & - 0x00ff0000) << 8) | ((dw_ip_address & 0x0000ff00) >> - 8) | ((dw_ip_address & 0x000000ff) << 8); - - err = hpi_cobranet_hmi_write(h_control, - HPI_COBRANET_HMI_cobra_ip_mon_currentIP, 4, (u8 *)&iP); - - return err; - -} - -u16 hpi_cobranet_get_static_ip_address(u32 h_control, u32 *pdw_ip_address) -{ - u32 byte_count; - u32 iP; - u16 err; - err = hpi_cobranet_hmi_read(h_control, - HPI_COBRANET_HMI_cobra_ip_mon_staticIP, 4, &byte_count, - (u8 *)&iP); - - *pdw_ip_address = - ((iP & 0xff000000) >> 8) | ((iP & 0x00ff0000) << 8) | ((iP & - 0x0000ff00) >> 8) | ((iP & 0x000000ff) << 8); - - if (err) - *pdw_ip_address = 0; - - return err; - -} - -u16 hpi_cobranet_set_static_ip_address(u32 h_control, u32 dw_ip_address) -{ - u32 iP; - u16 err; - - iP = ((dw_ip_address & 0xff000000) >> 8) | ((dw_ip_address & - 0x00ff0000) << 8) | ((dw_ip_address & 0x0000ff00) >> - 8) | ((dw_ip_address & 0x000000ff) << 8); - - err = hpi_cobranet_hmi_write(h_control, - HPI_COBRANET_HMI_cobra_ip_mon_staticIP, 4, (u8 *)&iP); - - return err; - -} - -u16 hpi_cobranet_get_macaddress(u32 h_control, u32 *p_mac_msbs, - u32 *p_mac_lsbs) -{ - u32 byte_count; - u16 err; - u32 mac; - - err = hpi_cobranet_hmi_read(h_control, - HPI_COBRANET_HMI_cobra_if_phy_address, 4, &byte_count, - (u8 *)&mac); - - if (!err) { - *p_mac_msbs = - ((mac & 0xff000000) >> 8) | ((mac & 0x00ff0000) << 8) - | ((mac & 0x0000ff00) >> 8) | ((mac & 0x000000ff) << - 8); - - err = hpi_cobranet_hmi_read(h_control, - HPI_COBRANET_HMI_cobra_if_phy_address + 1, 4, - &byte_count, (u8 *)&mac); - } - - if (!err) { - *p_mac_lsbs = - ((mac & 0xff000000) >> 8) | ((mac & 0x00ff0000) << 8) - | ((mac & 0x0000ff00) >> 8) | ((mac & 0x000000ff) << - 8); - } else { - *p_mac_msbs = 0; - *p_mac_lsbs = 0; - } - - return err; -} - -u16 hpi_compander_set_enable(u32 h_control, u32 enable) -{ - return hpi_control_param_set(h_control, HPI_GENERIC_ENABLE, enable, - 0); -} - -u16 hpi_compander_get_enable(u32 h_control, u32 *enable) -{ - return hpi_control_param1_get(h_control, HPI_GENERIC_ENABLE, enable); -} - -u16 hpi_compander_set_makeup_gain(u32 h_control, short makeup_gain0_01dB) -{ - return hpi_control_log_set2(h_control, HPI_COMPANDER_MAKEUPGAIN, - makeup_gain0_01dB, 0); -} - -u16 hpi_compander_get_makeup_gain(u32 h_control, short *makeup_gain0_01dB) -{ - return hpi_control_log_get2(h_control, HPI_COMPANDER_MAKEUPGAIN, - makeup_gain0_01dB, NULL); -} - -u16 hpi_compander_set_attack_time_constant(u32 h_control, unsigned int index, - u32 attack) -{ - return hpi_control_param_set(h_control, HPI_COMPANDER_ATTACK, attack, - index); -} - -u16 hpi_compander_get_attack_time_constant(u32 h_control, unsigned int index, - u32 *attack) -{ - return hpi_control_param_get(h_control, HPI_COMPANDER_ATTACK, 0, - index, attack, NULL); -} - -u16 hpi_compander_set_decay_time_constant(u32 h_control, unsigned int index, - u32 decay) -{ - return hpi_control_param_set(h_control, HPI_COMPANDER_DECAY, decay, - index); -} - -u16 hpi_compander_get_decay_time_constant(u32 h_control, unsigned int index, - u32 *decay) -{ - return hpi_control_param_get(h_control, HPI_COMPANDER_DECAY, 0, index, - decay, NULL); - -} - -u16 hpi_compander_set_threshold(u32 h_control, unsigned int index, - short threshold0_01dB) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_SET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_COMPANDER_THRESHOLD; - hm.u.c.param2 = index; - hm.u.c.an_log_value[0] = threshold0_01dB; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_compander_get_threshold(u32 h_control, unsigned int index, - short *threshold0_01dB) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_COMPANDER_THRESHOLD; - hm.u.c.param2 = index; - - hpi_send_recv(&hm, &hr); - *threshold0_01dB = hr.u.c.an_log_value[0]; - - return hr.error; -} - -u16 hpi_compander_set_ratio(u32 h_control, u32 index, u32 ratio100) -{ - return hpi_control_param_set(h_control, HPI_COMPANDER_RATIO, ratio100, - index); -} - -u16 hpi_compander_get_ratio(u32 h_control, u32 index, u32 *ratio100) -{ - return hpi_control_param_get(h_control, HPI_COMPANDER_RATIO, 0, index, - ratio100, NULL); -} - u16 hpi_level_query_range(u32 h_control, short *min_gain_01dB, short *max_gain_01dB, short *step_gain_01dB) { @@ -2030,233 +1183,48 @@ u16 hpi_meter_get_peak(u32 h_control, short an_peakdB[HPI_MAX_CHANNELS] else for (i = 0; i < HPI_MAX_CHANNELS; i++) an_peakdB[i] = HPI_METER_MINIMUM; - return hr.error; -} - -u16 hpi_meter_get_rms(u32 h_control, short an_rmsdB[HPI_MAX_CHANNELS] - ) -{ - short i = 0; - - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_METER_RMS; - - hpi_send_recv(&hm, &hr); - - if (!hr.error) - memcpy(an_rmsdB, hr.u.c.an_log_value, - sizeof(short) * HPI_MAX_CHANNELS); - else - for (i = 0; i < HPI_MAX_CHANNELS; i++) - an_rmsdB[i] = HPI_METER_MINIMUM; - - return hr.error; -} - -u16 hpi_meter_set_rms_ballistics(u32 h_control, u16 attack, u16 decay) -{ - return hpi_control_param_set(h_control, HPI_METER_RMS_BALLISTICS, - attack, decay); -} - -u16 hpi_meter_get_rms_ballistics(u32 h_control, u16 *pn_attack, u16 *pn_decay) -{ - u32 attack; - u32 decay; - u16 error; - - error = hpi_control_param2_get(h_control, HPI_METER_RMS_BALLISTICS, - &attack, &decay); - - if (pn_attack) - *pn_attack = (unsigned short)attack; - if (pn_decay) - *pn_decay = (unsigned short)decay; - - return error; -} - -u16 hpi_meter_set_peak_ballistics(u32 h_control, u16 attack, u16 decay) -{ - return hpi_control_param_set(h_control, HPI_METER_PEAK_BALLISTICS, - attack, decay); -} - -u16 hpi_meter_get_peak_ballistics(u32 h_control, u16 *pn_attack, - u16 *pn_decay) -{ - u32 attack; - u32 decay; - u16 error; - - error = hpi_control_param2_get(h_control, HPI_METER_PEAK_BALLISTICS, - &attack, &decay); - - if (pn_attack) - *pn_attack = (short)attack; - if (pn_decay) - *pn_decay = (short)decay; - - return error; -} - -u16 hpi_microphone_set_phantom_power(u32 h_control, u16 on_off) -{ - return hpi_control_param_set(h_control, HPI_MICROPHONE_PHANTOM_POWER, - (u32)on_off, 0); -} - -u16 hpi_microphone_get_phantom_power(u32 h_control, u16 *pw_on_off) -{ - u16 error = 0; - u32 on_off = 0; - error = hpi_control_param1_get(h_control, - HPI_MICROPHONE_PHANTOM_POWER, &on_off); - if (pw_on_off) - *pw_on_off = (u16)on_off; - return error; -} - -u16 hpi_multiplexer_set_source(u32 h_control, u16 source_node_type, - u16 source_node_index) -{ - return hpi_control_param_set(h_control, HPI_MULTIPLEXER_SOURCE, - source_node_type, source_node_index); -} - -u16 hpi_multiplexer_get_source(u32 h_control, u16 *source_node_type, - u16 *source_node_index) -{ - u32 node, index; - u16 err = hpi_control_param2_get(h_control, - HPI_MULTIPLEXER_SOURCE, &node, - &index); - if (source_node_type) - *source_node_type = (u16)node; - if (source_node_index) - *source_node_index = (u16)index; - return err; -} - -u16 hpi_multiplexer_query_source(u32 h_control, u16 index, - u16 *source_node_type, u16 *source_node_index) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_MULTIPLEXER_QUERYSOURCE; - hm.u.c.param1 = index; - - hpi_send_recv(&hm, &hr); - - if (source_node_type) - *source_node_type = (u16)hr.u.c.param1; - if (source_node_index) - *source_node_index = (u16)hr.u.c.param2; - return hr.error; -} - -u16 hpi_parametric_eq_get_info(u32 h_control, u16 *pw_number_of_bands, - u16 *pw_on_off) -{ - u32 oB = 0; - u32 oO = 0; - u16 error = 0; - - error = hpi_control_param2_get(h_control, HPI_EQUALIZER_NUM_FILTERS, - &oO, &oB); - if (pw_number_of_bands) - *pw_number_of_bands = (u16)oB; - if (pw_on_off) - *pw_on_off = (u16)oO; - return error; -} - -u16 hpi_parametric_eq_set_state(u32 h_control, u16 on_off) -{ - return hpi_control_param_set(h_control, HPI_EQUALIZER_NUM_FILTERS, - on_off, 0); -} - -u16 hpi_parametric_eq_get_band(u32 h_control, u16 index, u16 *pn_type, - u32 *pfrequency_hz, short *pnQ100, short *pn_gain0_01dB) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_EQUALIZER_FILTER; - hm.u.c.param2 = index; - - hpi_send_recv(&hm, &hr); - - if (pfrequency_hz) - *pfrequency_hz = hr.u.c.param1; - if (pn_type) - *pn_type = (u16)(hr.u.c.param2 >> 16); - if (pnQ100) - *pnQ100 = hr.u.c.an_log_value[1]; - if (pn_gain0_01dB) - *pn_gain0_01dB = hr.u.c.an_log_value[0]; - - return hr.error; -} - -u16 hpi_parametric_eq_set_band(u32 h_control, u16 index, u16 type, - u32 frequency_hz, short q100, short gain0_01dB) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_SET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hm.u.c.param1 = frequency_hz; - hm.u.c.param2 = (index & 0xFFFFL) + ((u32)type << 16); - hm.u.c.an_log_value[0] = gain0_01dB; - hm.u.c.an_log_value[1] = q100; - hm.u.c.attribute = HPI_EQUALIZER_FILTER; + return hr.error; +} - hpi_send_recv(&hm, &hr); +u16 hpi_multiplexer_set_source(u32 h_control, u16 source_node_type, + u16 source_node_index) +{ + return hpi_control_param_set(h_control, HPI_MULTIPLEXER_SOURCE, + source_node_type, source_node_index); +} - return hr.error; +u16 hpi_multiplexer_get_source(u32 h_control, u16 *source_node_type, + u16 *source_node_index) +{ + u32 node, index; + u16 err = hpi_control_param2_get(h_control, + HPI_MULTIPLEXER_SOURCE, &node, + &index); + if (source_node_type) + *source_node_type = (u16)node; + if (source_node_index) + *source_node_index = (u16)index; + return err; } -u16 hpi_parametric_eq_get_coeffs(u32 h_control, u16 index, short coeffs[5] - ) +u16 hpi_multiplexer_query_source(u32 h_control, u16 index, + u16 *source_node_type, u16 *source_node_index) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE); if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_EQUALIZER_COEFFICIENTS; - hm.u.c.param2 = index; + hm.u.c.attribute = HPI_MULTIPLEXER_QUERYSOURCE; + hm.u.c.param1 = index; hpi_send_recv(&hm, &hr); - coeffs[0] = (short)hr.u.c.an_log_value[0]; - coeffs[1] = (short)hr.u.c.an_log_value[1]; - coeffs[2] = (short)hr.u.c.param1; - coeffs[3] = (short)(hr.u.c.param1 >> 16); - coeffs[4] = (short)hr.u.c.param2; - + if (source_node_type) + *source_node_type = (u16)hr.u.c.param1; + if (source_node_index) + *source_node_index = (u16)hr.u.c.param2; return hr.error; } @@ -2360,130 +1328,6 @@ u16 hpi_sample_clock_get_sample_rate(u32 h_control, u32 *psample_rate) return err; } -u16 hpi_sample_clock_set_auto(u32 h_control, u32 enable) -{ - return hpi_control_param_set(h_control, HPI_SAMPLECLOCK_AUTO, enable, - 0); -} - -u16 hpi_sample_clock_get_auto(u32 h_control, u32 *penable) -{ - return hpi_control_param1_get(h_control, HPI_SAMPLECLOCK_AUTO, - penable); -} - -u16 hpi_sample_clock_set_local_rate_lock(u32 h_control, u32 lock) -{ - return hpi_control_param_set(h_control, HPI_SAMPLECLOCK_LOCAL_LOCK, - lock, 0); -} - -u16 hpi_sample_clock_get_local_rate_lock(u32 h_control, u32 *plock) -{ - return hpi_control_param1_get(h_control, HPI_SAMPLECLOCK_LOCAL_LOCK, - plock); -} - -u16 hpi_tone_detector_get_frequency(u32 h_control, u32 index, u32 *frequency) -{ - return hpi_control_param_get(h_control, HPI_TONEDETECTOR_FREQUENCY, - index, 0, frequency, NULL); -} - -u16 hpi_tone_detector_get_state(u32 h_control, u32 *state) -{ - return hpi_control_param1_get(h_control, HPI_TONEDETECTOR_STATE, - state); -} - -u16 hpi_tone_detector_set_enable(u32 h_control, u32 enable) -{ - return hpi_control_param_set(h_control, HPI_GENERIC_ENABLE, enable, - 0); -} - -u16 hpi_tone_detector_get_enable(u32 h_control, u32 *enable) -{ - return hpi_control_param1_get(h_control, HPI_GENERIC_ENABLE, enable); -} - -u16 hpi_tone_detector_set_event_enable(u32 h_control, u32 event_enable) -{ - return hpi_control_param_set(h_control, HPI_GENERIC_EVENT_ENABLE, - (u32)event_enable, 0); -} - -u16 hpi_tone_detector_get_event_enable(u32 h_control, u32 *event_enable) -{ - return hpi_control_param1_get(h_control, HPI_GENERIC_EVENT_ENABLE, - event_enable); -} - -u16 hpi_tone_detector_set_threshold(u32 h_control, int threshold) -{ - return hpi_control_param_set(h_control, HPI_TONEDETECTOR_THRESHOLD, - (u32)threshold, 0); -} - -u16 hpi_tone_detector_get_threshold(u32 h_control, int *threshold) -{ - return hpi_control_param1_get(h_control, HPI_TONEDETECTOR_THRESHOLD, - (u32 *)threshold); -} - -u16 hpi_silence_detector_get_state(u32 h_control, u32 *state) -{ - return hpi_control_param1_get(h_control, HPI_SILENCEDETECTOR_STATE, - state); -} - -u16 hpi_silence_detector_set_enable(u32 h_control, u32 enable) -{ - return hpi_control_param_set(h_control, HPI_GENERIC_ENABLE, enable, - 0); -} - -u16 hpi_silence_detector_get_enable(u32 h_control, u32 *enable) -{ - return hpi_control_param1_get(h_control, HPI_GENERIC_ENABLE, enable); -} - -u16 hpi_silence_detector_set_event_enable(u32 h_control, u32 event_enable) -{ - return hpi_control_param_set(h_control, HPI_GENERIC_EVENT_ENABLE, - event_enable, 0); -} - -u16 hpi_silence_detector_get_event_enable(u32 h_control, u32 *event_enable) -{ - return hpi_control_param1_get(h_control, HPI_GENERIC_EVENT_ENABLE, - event_enable); -} - -u16 hpi_silence_detector_set_delay(u32 h_control, u32 delay) -{ - return hpi_control_param_set(h_control, HPI_SILENCEDETECTOR_DELAY, - delay, 0); -} - -u16 hpi_silence_detector_get_delay(u32 h_control, u32 *delay) -{ - return hpi_control_param1_get(h_control, HPI_SILENCEDETECTOR_DELAY, - delay); -} - -u16 hpi_silence_detector_set_threshold(u32 h_control, int threshold) -{ - return hpi_control_param_set(h_control, HPI_SILENCEDETECTOR_THRESHOLD, - threshold, 0); -} - -u16 hpi_silence_detector_get_threshold(u32 h_control, int *threshold) -{ - return hpi_control_param1_get(h_control, - HPI_SILENCEDETECTOR_THRESHOLD, (u32 *)threshold); -} - u16 hpi_tuner_query_band(const u32 h_tuner, const u32 index, u16 *pw_band) { u32 qr; @@ -2554,196 +1398,6 @@ u16 hpi_tuner_get_gain(u32 h_control, short *pn_gain) return error; } -u16 hpi_tuner_get_rf_level(u32 h_control, short *pw_level) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.cu.attribute = HPI_TUNER_LEVEL_AVG; - hpi_send_recv(&hm, &hr); - if (pw_level) - *pw_level = hr.u.cu.tuner.s_level; - return hr.error; -} - -u16 hpi_tuner_get_raw_rf_level(u32 h_control, short *pw_level) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.cu.attribute = HPI_TUNER_LEVEL_RAW; - hpi_send_recv(&hm, &hr); - if (pw_level) - *pw_level = hr.u.cu.tuner.s_level; - return hr.error; -} - -u16 hpi_tuner_query_deemphasis(const u32 h_tuner, const u32 index, - const u16 band, u32 *pdeemphasis) -{ - return hpi_control_query(h_tuner, HPI_TUNER_DEEMPHASIS, index, band, - pdeemphasis); -} - -u16 hpi_tuner_set_deemphasis(u32 h_control, u32 deemphasis) -{ - return hpi_control_param_set(h_control, HPI_TUNER_DEEMPHASIS, - deemphasis, 0); -} - -u16 hpi_tuner_get_deemphasis(u32 h_control, u32 *pdeemphasis) -{ - return hpi_control_param1_get(h_control, HPI_TUNER_DEEMPHASIS, - pdeemphasis); -} - -u16 hpi_tuner_query_program(const u32 h_tuner, u32 *pbitmap_program) -{ - return hpi_control_query(h_tuner, HPI_TUNER_PROGRAM, 0, 0, - pbitmap_program); -} - -u16 hpi_tuner_set_program(u32 h_control, u32 program) -{ - return hpi_control_param_set(h_control, HPI_TUNER_PROGRAM, program, - 0); -} - -u16 hpi_tuner_get_program(u32 h_control, u32 *pprogram) -{ - return hpi_control_param1_get(h_control, HPI_TUNER_PROGRAM, pprogram); -} - -u16 hpi_tuner_get_hd_radio_dsp_version(u32 h_control, char *psz_dsp_version, - const u32 string_size) -{ - return hpi_control_get_string(h_control, - HPI_TUNER_HDRADIO_DSP_VERSION, psz_dsp_version, string_size); -} - -u16 hpi_tuner_get_hd_radio_sdk_version(u32 h_control, char *psz_sdk_version, - const u32 string_size) -{ - return hpi_control_get_string(h_control, - HPI_TUNER_HDRADIO_SDK_VERSION, psz_sdk_version, string_size); -} - -u16 hpi_tuner_get_status(u32 h_control, u16 *pw_status_mask, u16 *pw_status) -{ - u32 status = 0; - u16 error = 0; - - error = hpi_control_param1_get(h_control, HPI_TUNER_STATUS, &status); - if (pw_status) { - if (!error) { - *pw_status_mask = (u16)(status >> 16); - *pw_status = (u16)(status & 0xFFFF); - } else { - *pw_status_mask = 0; - *pw_status = 0; - } - } - return error; -} - -u16 hpi_tuner_set_mode(u32 h_control, u32 mode, u32 value) -{ - return hpi_control_param_set(h_control, HPI_TUNER_MODE, mode, value); -} - -u16 hpi_tuner_get_mode(u32 h_control, u32 mode, u32 *pn_value) -{ - return hpi_control_param_get(h_control, HPI_TUNER_MODE, mode, 0, - pn_value, NULL); -} - -u16 hpi_tuner_get_hd_radio_signal_quality(u32 h_control, u32 *pquality) -{ - return hpi_control_param1_get(h_control, - HPI_TUNER_HDRADIO_SIGNAL_QUALITY, pquality); -} - -u16 hpi_tuner_get_hd_radio_signal_blend(u32 h_control, u32 *pblend) -{ - return hpi_control_param1_get(h_control, HPI_TUNER_HDRADIO_BLEND, - pblend); -} - -u16 hpi_tuner_set_hd_radio_signal_blend(u32 h_control, const u32 blend) -{ - return hpi_control_param_set(h_control, HPI_TUNER_HDRADIO_BLEND, - blend, 0); -} - -u16 hpi_tuner_get_rds(u32 h_control, char *p_data) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_TUNER_RDS; - hpi_send_recv(&hm, &hr); - if (p_data) { - *(u32 *)&p_data[0] = hr.u.cu.tuner.rds.data[0]; - *(u32 *)&p_data[4] = hr.u.cu.tuner.rds.data[1]; - *(u32 *)&p_data[8] = hr.u.cu.tuner.rds.bLER; - } - return hr.error; -} - -u16 hpi_pad_get_channel_name(u32 h_control, char *psz_string, - const u32 data_length) -{ - return hpi_control_get_string(h_control, HPI_PAD_CHANNEL_NAME, - psz_string, data_length); -} - -u16 hpi_pad_get_artist(u32 h_control, char *psz_string, const u32 data_length) -{ - return hpi_control_get_string(h_control, HPI_PAD_ARTIST, psz_string, - data_length); -} - -u16 hpi_pad_get_title(u32 h_control, char *psz_string, const u32 data_length) -{ - return hpi_control_get_string(h_control, HPI_PAD_TITLE, psz_string, - data_length); -} - -u16 hpi_pad_get_comment(u32 h_control, char *psz_string, - const u32 data_length) -{ - return hpi_control_get_string(h_control, HPI_PAD_COMMENT, psz_string, - data_length); -} - -u16 hpi_pad_get_program_type(u32 h_control, u32 *ppTY) -{ - return hpi_control_param1_get(h_control, HPI_PAD_PROGRAM_TYPE, ppTY); -} - -u16 hpi_pad_get_rdsPI(u32 h_control, u32 *ppI) -{ - return hpi_control_param1_get(h_control, HPI_PAD_PROGRAM_ID, ppI); -} - -u16 hpi_volume_query_channels(const u32 h_volume, u32 *p_channels) -{ - return hpi_control_query(h_volume, HPI_VOLUME_NUM_CHANNELS, 0, 0, - p_channels); -} - u16 hpi_volume_set_gain(u32 h_control, short an_log_gain[HPI_MAX_CHANNELS] ) { @@ -2794,78 +1448,3 @@ u16 hpi_volume_query_range(u32 h_control, short *min_gain_01dB, *step_gain_01dB = (short)hr.u.c.param1; return hr.error; } - -u16 hpi_volume_auto_fade_profile(u32 h_control, - short an_stop_gain0_01dB[HPI_MAX_CHANNELS], u32 duration_ms, - u16 profile) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_SET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - memcpy(hm.u.c.an_log_value, an_stop_gain0_01dB, - sizeof(short) * HPI_MAX_CHANNELS); - - hm.u.c.attribute = HPI_VOLUME_AUTOFADE; - hm.u.c.param1 = duration_ms; - hm.u.c.param2 = profile; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_volume_auto_fade(u32 h_control, - short an_stop_gain0_01dB[HPI_MAX_CHANNELS], u32 duration_ms) -{ - return hpi_volume_auto_fade_profile(h_control, an_stop_gain0_01dB, - duration_ms, HPI_VOLUME_AUTOFADE_LOG); -} - -u16 hpi_volume_query_auto_fade_profile(const u32 h_volume, const u32 i, - u16 *profile) -{ - u16 e; - u32 u; - e = hpi_control_query(h_volume, HPI_VOLUME_AUTOFADE, i, 0, &u); - *profile = (u16)u; - return e; -} - -u16 hpi_vox_set_threshold(u32 h_control, short an_gain0_01dB) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_SET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_VOX_THRESHOLD; - - hm.u.c.an_log_value[0] = an_gain0_01dB; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_vox_get_threshold(u32 h_control, short *an_gain0_01dB) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_VOX_THRESHOLD; - - hpi_send_recv(&hm, &hr); - - *an_gain0_01dB = hr.u.c.an_log_value[0]; - - return hr.error; -} -- 1.7.10.4

3 9

[alsa-devel] Regression for Lenovo X240 sound in 3.18
by Jan Engelhardt 01 Jan '15

01 Jan '15

I have a X240 (without any dock), and in Linux 3.17.4, I used to have - "Speaker" with switch and volume bar (in alsamixer), - "PCM" with switch and volume bar, - the headphone output volume would be controlled by both Master and PCM. With 3.18.1, bisected down to below commit, this all went bonkers. - "Speaker" has only a switch but no volume bar, - speakers do not output _any_ sound whatsoever anymore - changing the PCM volume does not affect either Headphone or Speaker. commit 7c21539c45eace48e9f6fbdd59347d8416960af5 Author: Lukas Bossard <mr.bobukas(a)gmail.com> Date: Wed Oct 29 18:31:07 2014 +0100 ALSA: hda - Add ultra dock support for Thinkpad X240. Adding ultra doch support for Lenovo Thinkpad X240 (17aa:2214). [Actually replaced the entry with ALC292_FIXUP_TPT440_DOCK -- tiwai] Reverting this commit on top of v3.18.1 restores the sought functionality.

3 7

[alsa-devel] [PATCH v6 0/3] ASoC: simple-card: Add multi-CODEC support
by Jean-Francois Moine 01 Jan '15

01 Jan '15

This patch set adds multi-CODEC support to the simple card. v6: (Mark Brown) - don't stop unreferencing the codec nodes on null node - split the patch about simple-card.c - add example 3 in the DT binding v5: accept only "sound-dai" as the dai link property (Arnd Bergmann) v4: (Mark Brown) - document the new core function - calculation of the number of codecs moved to the core function - split the patch about simple-card.c - less code changes - remove the cast of pointers - less changes in the DT binding example 2 v3: rebase on broonie git 2014/11/03 v2: accept list of CODECs in 'sound-dais' (Benoit Cousson) Happy New Year! Jean-Francois Moine (3): ASoC: simple-card: prepare multi-CODECs ASoC: simple-card: Add multi-CODECs in DT ASoC: simple-card: Remove useless check .../devicetree/bindings/sound/simple-card.txt | 19 ++++- sound/soc/generic/simple-card.c | 81 +++++++++++----------- 2 files changed, 60 insertions(+), 40 deletions(-) -- 2.1.4

1 3

[alsa-devel] [PATCH] [ALSA] add SNDRV_PCM_TRIGGER_DRAIN trigger
by libin.yang＠intel.com 01 Jan '15

01 Jan '15

From: Libin Yang <libin.yang(a)intel.com> Add SNDRV_PCM_TRIGGER_DRAIN trigger for pcm drain. Some audio devices require notification of drain events in order to properly drain and shutdown an audio stream. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/sound/pcm.h | 1 + include/uapi/sound/asound.h | 1 + sound/core/pcm_native.c | 10 +++++++++- 3 files changed, 11 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 3652c94..bf32cea 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -106,6 +106,7 @@ struct snd_pcm_ops { #define SNDRV_PCM_TRIGGER_PAUSE_RELEASE 4 #define SNDRV_PCM_TRIGGER_SUSPEND 5 #define SNDRV_PCM_TRIGGER_RESUME 6 +#define SNDRV_PCM_TRIGGER_DRAIN 7 #define SNDRV_PCM_POS_XRUN ((snd_pcm_uframes_t)-1) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1f23cd6..0e88e7a 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -268,6 +268,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ #define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ typedef int __bitwise snd_pcm_state_t; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 095d957..a5fe336 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -420,7 +420,8 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, hw = &substream->runtime->hw; if (!params->info) { - params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES; + params->info = hw->info & ~(SNDRV_PCM_INFO_FIFO_IN_FRAMES | + SNDRV_PCM_INFO_DRAIN_TRIGGER); if (!hw_support_mmap(substream)) params->info &= ~(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID); @@ -1566,6 +1567,13 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) snd_pcm_post_stop(substream, new_state); } } + + if (runtime->status->state == SNDRV_PCM_STATE_DRAINING && + substream->runtime->trigger_master == substream && + (substream->runtime->hw.info & SNDRV_PCM_INFO_DRAIN_TRIGGER)) + return substream->ops->trigger(substream, + SNDRV_PCM_TRIGGER_DRAIN); + return 0; } -- 1.9.1

3 2

[alsa-devel] [PATCH 1/3] ALSA: Fix handling of multiple msbits constraints on the same runtime
by Lars-Peter Clausen 31 Dec '14

31 Dec '14

If the sound card is made up of discrete components, each with their own driver (e.g. like in the ASoC case), we might end up with multiple msbits constraint rules installed. Currently this will result in msbits being set to whatever the last rule set it to. This patch updates the behavior of the rule to choose the minimum (other than zero) of all the installed rules. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ec9e786..b0c1535 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1300,7 +1300,7 @@ static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params, unsigned int msbits = l >> 16; struct snd_interval *i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); if (snd_interval_single(i) && snd_interval_value(i) == width) - params->msbits = msbits; + params->msbits = min_not_zero(params->msbits, msbits); return 0; } -- 1.8.0

3 6

[alsa-devel] [PATCH] ASoC: dwc: Remove unnecessary conditional compilation
by Andrew Jackson 31 Dec '14

31 Dec '14

From: Andrew Jackson <Andrew.Jackson(a)arm.com> of_match_ptr is already conditionally compiled based on CONFIG_OF so further conditional compilation is not required. Remove conditional compilation surrounding of_match_ptr. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 99cf64b..a3e97b4 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -624,9 +624,7 @@ static struct platform_driver dw_i2s_driver = { .remove = dw_i2s_remove, .driver = { .name = "designware-i2s", -#ifdef CONFIG_OF .of_match_table = of_match_ptr(dw_i2s_of_match), -#endif }, }; -- 1.7.1

1 0

Re: [alsa-devel] [patch v2] ALSA: asihpi: fix an information leak in asihpi_hpi_ioctl()
by Takashi Iwai 31 Dec '14

31 Dec '14

At Wed, 31 Dec 2014 23:48:32 +1300, Eliot Blennerhassett wrote: > > Add missing limits to keep copied data within allocated buffer. > > Signed-off-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> > --- > Regenerated, this should apply cleanly to for-next OK, applied now. Thanks. Takashi > > sound/pci/asihpi/hpi6000.c | 7 +++++-- > sound/pci/asihpi/hpioctl.c | 2 ++ > 2 files changed, 7 insertions(+), 2 deletions(-) > > diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c > index 2414d7a..2d63648 100644 > --- a/sound/pci/asihpi/hpi6000.c > +++ b/sound/pci/asihpi/hpi6000.c > @@ -47,7 +47,7 @@ > > /* operational/messaging errors */ > #define HPI6000_ERROR_MSG_RESP_IDLE_TIMEOUT 901 > - > +#define HPI6000_ERROR_RESP_GET_LEN 902 > #define HPI6000_ERROR_MSG_RESP_GET_RESP_ACK 903 > #define HPI6000_ERROR_MSG_GET_ADR 904 > #define HPI6000_ERROR_RESP_GET_ADR 905 > @@ -1365,7 +1365,10 @@ static short hpi6000_message_response_sequence(struct hpi_adapter_obj *pao, > length = hpi_read_word(pdo, HPI_HIF_ADDR(length)); > } while (hpi6000_check_PCI2040_error_flag(pao, H6READ) && --timeout); > if (!timeout) > - length = sizeof(struct hpi_response); > + return HPI6000_ERROR_RESP_GET_LEN; > + > + if (length > phr->size) > + return HPI_ERROR_RESPONSE_BUFFER_TOO_SMALL; > > /* get the response */ > p_data = (u32 *)phr; > diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c > index 6aa677e..72af66b 100644 > --- a/sound/pci/asihpi/hpioctl.c > +++ b/sound/pci/asihpi/hpioctl.c > @@ -153,6 +153,8 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) > goto out; > } > > + res_max_size = min_t(size_t, res_max_size, sizeof(*hr)); > + > switch (hm->h.function) { > case HPI_SUBSYS_CREATE_ADAPTER: > case HPI_ADAPTER_DELETE: > -- > 1.9.1 > >

1 0

[alsa-devel] [PATCH] pcm: fix buffer overflow in snd_pcm_chmap_print()
by Anssi Hannula 31 Dec '14

31 Dec '14

The size argument is wrong for one of the snprintf() calls in snd_pcm_chmap_print(), allowing an overflow to happen (the user-provided buffer may be written data up to 2x its actual size). Seen in an user report here: http://trac.kodi.tv/ticket/15641 Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- src/pcm/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index baa47c7..e74e02f 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -7621,7 +7621,7 @@ int snd_pcm_chmap_print(const snd_pcm_chmap_t *map, size_t maxlen, char *buf) return -ENOMEM; } if (map->pos[i] & SND_CHMAP_DRIVER_SPEC) - len += snprintf(buf + len, maxlen, "%d", p); + len += snprintf(buf + len, maxlen - len, "%d", p); else { const char *name = chmap_names[p]; if (name) -- 1.8.4.5

2 1

[alsa-devel] [PATCH] [ALSA] add SNDRV_PCM_TRIGGER_DRAIN trigger
by libin.yang＠intel.com 31 Dec '14

31 Dec '14

From: Libin Yang <libin.yang(a)intel.com> Add SNDRV_PCM_TRIGGER_DRAIN trigger for pcm drain. Some audio devices require notification of drain events in order to properly drain and shutdown an audio stream. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/sound/pcm.h | 1 + include/uapi/sound/asound.h | 1 + sound/core/pcm_native.c | 10 +++++++++- 3 files changed, 11 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 3652c94..bf32cea 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -106,6 +106,7 @@ struct snd_pcm_ops { #define SNDRV_PCM_TRIGGER_PAUSE_RELEASE 4 #define SNDRV_PCM_TRIGGER_SUSPEND 5 #define SNDRV_PCM_TRIGGER_RESUME 6 +#define SNDRV_PCM_TRIGGER_DRAIN 7 #define SNDRV_PCM_POS_XRUN ((snd_pcm_uframes_t)-1) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1f23cd6..0e88e7a 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -268,6 +268,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ #define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ typedef int __bitwise snd_pcm_state_t; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 095d957..bd57f6e 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -420,7 +420,8 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, hw = &substream->runtime->hw; if (!params->info) { - params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES; + params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES + & ~SNDRV_PCM_INFO_DRAIN_TRIGGER; if (!hw_support_mmap(substream)) params->info &= ~(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID); @@ -1566,6 +1567,13 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) snd_pcm_post_stop(substream, new_state); } } + + if (runtime->status->state == SNDRV_PCM_STATE_DRAINING && + substream->runtime->trigger_master == substream && + (substream->runtime->hw.info & SNDRV_PCM_INFO_DRAIN_TRIGGER)) + return substream->ops->trigger(substream, + SNDRV_PCM_TRIGGER_DRAIN); + return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH v4] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 30 Dec '14

30 Dec '14

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK to save power. Enable runtime power management, and auto enable/disable MCLK in pm_runtime resume and suspend. When wm8960 codec is being used, it will triger resume() to enable MCLK. When codec is not being used, it will triger suspend() to disable MCLK. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae..aff24a7 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,8 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -117,6 +119,7 @@ static bool wm8960_volatile(struct device *dev, unsigned int reg) } struct wm8960_priv { + struct clk *mclk; struct regmap *regmap; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); @@ -1002,6 +1005,13 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (wm8960 == NULL) return -ENOMEM; + wm8960->mclk = devm_clk_get(&i2c->dev, "codec_mclk"); + + if (IS_ERR(wm8960->mclk)) { + if (PTR_ERR(wm8960->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap); if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); @@ -1041,6 +1051,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); @@ -1053,6 +1066,38 @@ static int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm8960_runtime_resume(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int wm8960_runtime_suspend(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8960_pm = { + SET_RUNTIME_PM_OPS(wm8960_runtime_suspend, wm8960_runtime_resume, NULL) +}; + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -1070,6 +1115,7 @@ static struct i2c_driver wm8960_i2c_driver = { .name = "wm8960", .owner = THIS_MODULE, .of_match_table = wm8960_of_match, + .pm = &wm8960_pm, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

3 6

[alsa-devel] [PATCH v5 0/6] ASoC: dwc: Add device tree support to designware I2S
by Andrew Jackson 30 Dec '14

30 Dec '14

From: Andrew Jackson <Andrew.Jackson(a)arm.com> This patch set extends the DesignWare I2S driver to provide device tree support and fixes a couple of small faults. Changes v4->v5 + Remove all clk_put calls [Andrew Jackson] + Call clk_disable_unprepare [Andrew Jackson] Changes v3->v4 + Drop applied patches + Use combined function to prepare clock [Mark Brown] + Use managed clock resources to avoid clk_put [Mark Brown] + Read configuration prameters from hardware for both platform data and device tree [Mark Brown] + Re-order patch sequence [Mark Brown] + Change union name to avoid future collisions [Mark Brown] + Check return code from clk_set_rate [Mark Brown] + Check parametsrs read from hardware agaist array limits and add further comments [Mark Brown] + Add of_match_ptr [Mark Brown] Changes v2->v3 + Drop applied patch + Flush FIFOs in prepare rather than hw_params [Lars-Peter Clausen] Changes v1->v2 + Drop negative use count patch [Mark Brown] + Remove unnecessary debug print messages [Lars-Peter Clausen] + Rewrite iteration as for loop rather than do...while [Mark Brown] + Reorder patches to send fixes first [Mark Brown] + Simplify device tree code [Mark Brown] + Split device tree patch in two [Mark Brown] + Expand explanatory comment on channel configuration [Rajeev Kumar] Arnd: I've not forgotten about updating the spear entries and will submit as a separate patch. Andrew Jackson (6): ASoC: dwc: Switch to managed clock resource ASoC: dwc: Prepare clock before use ASoC: dwc: Read I2S block configuration from registers ASoC: dwc: Register components with managed interface ASoC: dwc: Add documentation for I2S DT ASoC: dwc: Add devicetree support for Designware I2S .../devicetree/bindings/sound/designware-i2s.txt | 31 +++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 283 ++++++++++++++++---- 3 files changed, 263 insertions(+), 52 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt

3 11

[alsa-devel] [PATCH] ASoC: mxs-saif: Do not store the irq number in the private structure
by Fabio Estevam 30 Dec '14

30 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> The irq number is only used inside the probe function, so there is really no need to store it in the private structure. Use a local 'irq' variable to hold the the irq number instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 10 +++++----- sound/soc/mxs/mxs-saif.h | 1 - 2 files changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index d986508..c866ade 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -710,7 +710,7 @@ static int mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores; struct mxs_saif *saif; - int ret = 0; + int irq, ret = 0; struct device_node *master; if (!np) @@ -763,16 +763,16 @@ static int mxs_saif_probe(struct platform_device *pdev) if (IS_ERR(saif->base)) return PTR_ERR(saif->base); - saif->irq = platform_get_irq(pdev, 0); - if (saif->irq < 0) { - ret = saif->irq; + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + ret = irq; dev_err(&pdev->dev, "failed to get irq resource: %d\n", ret); return ret; } saif->dev = &pdev->dev; - ret = devm_request_irq(&pdev->dev, saif->irq, mxs_saif_irq, 0, + ret = devm_request_irq(&pdev->dev, irq, mxs_saif_irq, 0, dev_name(&pdev->dev), saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index fbaf7ba..9a4c0b2 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -116,7 +116,6 @@ struct mxs_saif { unsigned int mclk; unsigned int mclk_in_use; void __iomem *base; - int irq; unsigned int id; unsigned int master_id; unsigned int cur_rate; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Support for combined tx/rx interrupt line
by Peter Ujfalusi 30 Dec '14

30 Dec '14

Some SoC, like da850/OMAP-L138 uses one common interrupt request for TX/RX events. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 30b94d4f9c5d..95eef582331b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -364,6 +364,20 @@ static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) return IRQ_RETVAL(handled_mask); } +static irqreturn_t davinci_mcasp_common_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + irqreturn_t ret = IRQ_NONE; + + if (mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]) + ret = davinci_mcasp_tx_irq_handler(irq, data); + + if (mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]) + ret |= davinci_mcasp_rx_irq_handler(irq, data); + + return ret; +} + static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -1441,6 +1455,22 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->dev = &pdev->dev; + irq = platform_get_irq_byname(pdev, "common"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_common_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "common IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN; + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN; + } + irq = platform_get_irq_byname(pdev, "rx"); if (irq >= 0) { irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", -- 2.2.1

2 1

[alsa-devel] [RFC] ASoC: wm8962: Allow codec to work when CONFIG_PM=n
by Fabio Estevam 30 Dec '14

30 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently the codec does not work in the CONFIG_PM=n case because the regulators and clock are kept disabled. Allow the codec to be functional by calling wm8962_runtime_resume() when CONFIG_PM=n. Tested on a imx6sl-evk in the CONFIG_PM=y and CONFIG_PM=n cases. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Probably there is a better solution, hence I am sending it as RFC. Thanks sound/soc/codecs/wm8962.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d32d554..0b041c9 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3546,6 +3546,9 @@ static int wm8962_set_pdata_from_of(struct i2c_client *i2c, return 0; } +static int wm8962_runtime_resume(struct device *dev); +static int wm8962_runtime_suspend(struct device *dev); + static int wm8962_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3771,6 +3774,9 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, /* The drivers should power up as needed */ regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); + if (!IS_ENABLED(CONFIG_PM)) + wm8962_runtime_resume(&i2c->dev); + return 0; err_enable: @@ -3781,11 +3787,12 @@ err: static int wm8962_i2c_remove(struct i2c_client *client) { + if (!IS_ENABLED(CONFIG_PM)) + wm8962_runtime_suspend(&client->dev); snd_soc_unregister_codec(&client->dev); return 0; } -#ifdef CONFIG_PM static int wm8962_runtime_resume(struct device *dev) { struct wm8962_priv *wm8962 = dev_get_drvdata(dev); @@ -3860,7 +3867,6 @@ static int wm8962_runtime_suspend(struct device *dev) return 0; } -#endif static struct dev_pm_ops wm8962_pm = { SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) -- 1.9.1

2 1

[alsa-devel] [PATCH 1/9] ALSA: pcm_params: Remove unused add/sub functions
by Lars-Peter Clausen 30 Dec '14

30 Dec '14

Those two functions are not used anywhere and also their name is a bit to generic to be in a global header, so remove them. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/pcm_params.h | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 6b1c78f..15760f5 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -325,20 +325,6 @@ static inline int snd_interval_eq(const struct snd_interval *i1, const struct sn i1->max == i2->max && i1->openmax == i2->openmax; } -static inline unsigned int add(unsigned int a, unsigned int b) -{ - if (a >= UINT_MAX - b) - return UINT_MAX; - return a + b; -} - -static inline unsigned int sub(unsigned int a, unsigned int b) -{ - if (a > b) - return a - b; - return 0; -} - #define params_access(p) ((__force snd_pcm_access_t)\ snd_mask_min(hw_param_mask_c((p), SNDRV_PCM_HW_PARAM_ACCESS))) #define params_format(p) ((__force snd_pcm_format_t)\ -- 1.8.0

2 9

[alsa-devel] [GIT PULL] ASoC updates for v3.20
by Mark Brown 30 Dec '14

30 Dec '14

The following changes since commit 97bf6af1f928216fd6c5a66e8a57bfa95a659672: Linux 3.19-rc1 (2014-12-20 17:08:50 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.19-rc2 for you to fetch changes up to 1b9f1ae8907cb13d3d2538d98385b2744c9490d5: Merge remote-tracking branches 'asoc/topic/samsung' and 'asoc/topic/wm8750' into asoc-next (2014-12-24 12:56:48 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.20 Nothing too exciting here yet, a small optimization for DAPM from Lars-Peter and a few small bits and pieces for drivers but nothing that really stands out. ---------------------------------------------------------------- Andrew Jackson (4): ASoC: dwc: Remove unnecessary debug messages and tests ASoC: dwc: Reorder code in preparation for DT support ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap ASoC: dwc: Iterate over all channels Bard Liao (3): ASoC: rt5670: set platform data by dmi ASoC: rt5670: Add runtime PM support ASoC: rt5670: Keep sysclk on if JD func is used Fang, Yang A (1): ASoC: rt5677: fixed rt5677_dsp_vad_put rt5677_dsp_vad_get panic Jarkko Nikula (4): ASoC: Intel: Add I2C dependency to two new machines ASoC: Intel: Fix BYTCR firmware name ASoC: Intel: Fix BYTCR machine driver MODULE_ALIAS ASoC: pcm: Fix vague codec and cpu DAI prepare error messages Jianqun Xu (3): ASoC: rockchip: i2s: fix error defination of transmit data level ASoC: rockchip: i2s: fix maxburst of dma data to 4 ASoC: rockchip: i2s: set TDL and RDL to 16 samples Jie Yang (2): ASoC: Intel: correct the fixed free block allocation ASoC: Intel: Store the entry_point read from FW file Lars-Peter Clausen (4): ASoC: wm8750: Make VREF a VMID instead of OUTPUT widget ASoC: wm8750: Cleanup manual bias level transitions ASoC: dapm: Don't mark MICBIAS widgets as auto non-connected ASoC: dapm: Simplify fully route card handling Mark Brown (9): ASoC: dapm: Remove snd_soc_of_parse_audio_routing() due to deferred probe Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/dwc', 'asoc/fix/intel', 'asoc/fix/rockchip' and 'asoc/fix/rt5677' into asoc-linus Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next Merge remote-tracking branches 'asoc/topic/dwc', 'asoc/topic/intel' and 'asoc/topic/omap' into asoc-next Merge remote-tracking branches 'asoc/topic/pxa', 'asoc/topic/rockchip', 'asoc/topic/rt5670' and 'asoc/topic/rt5677' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung' and 'asoc/topic/wm8750' into asoc-next Oder Chiou (3): ASoC: rt5677: Revise the filter powers ASoC: rt5677: Add the ASRC support ASoC: rt5677: Adjust the routing of "PLL1" Peter Rosin (1): ASoC: pcm512x: Also support PCM514x devices Takashi Iwai (1): ASoC: intel: Remove superfluous backslash in Kconfig Wei Yongjun (1): ASoC: Intel: fix sparse non static symbol warnings Wolfram Sang (4): ASoC: intel: sst: drop owner assignment from platform_drivers ASoC: omap: drop owner assignment from platform_drivers ASoC: pxa: drop owner assignment from platform_drivers ASoC: samsung: drop owner assignment from platform_drivers kbuild test robot (2): ASoC: Intel: fix platform_no_drv_owner.cocci warnings ASoC: Intel: fix platform_no_drv_owner.cocci warnings .../devicetree/bindings/sound/pcm512x.txt | 3 +- include/sound/soc-dapm.h | 1 - sound/soc/codecs/pcm512x-i2c.c | 4 + sound/soc/codecs/pcm512x-spi.c | 4 + sound/soc/codecs/rt5670.c | 38 ++++ sound/soc/codecs/rt5677.c | 204 ++++++++++++++++++--- sound/soc/codecs/wm8750.c | 2 +- sound/soc/dwc/designware_i2s.c | 135 +++++++------- sound/soc/intel/Kconfig | 8 +- sound/soc/intel/bytcr_dpcm_rt5640.c | 3 +- sound/soc/intel/cht_bsw_rt5672.c | 1 - sound/soc/intel/sst-firmware.c | 7 +- sound/soc/intel/sst/sst_acpi.c | 7 +- sound/soc/omap/omap-hdmi-audio.c | 1 - sound/soc/pxa/spitz.c | 1 - sound/soc/rockchip/rockchip_i2s.c | 8 +- sound/soc/rockchip/rockchip_i2s.h | 2 +- sound/soc/samsung/arndale_rt5631.c | 1 - sound/soc/soc-core.c | 17 +- sound/soc/soc-dapm.c | 105 ++--------- sound/soc/soc-pcm.c | 7 +- 21 files changed, 349 insertions(+), 210 deletions(-)

2 1

[alsa-devel] ASoC fixes for v3.19
by Mark Brown 30 Dec '14

30 Dec '14

The following changes since commit 97bf6af1f928216fd6c5a66e8a57bfa95a659672: Linux 3.19-rc1 (2014-12-20 17:08:50 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v3.19-rc2 for you to fetch changes up to f2ec1d3b25776051a9205289dab12e9401d33f8a: Merge remote-tracking branches 'asoc/fix/dwc', 'asoc/fix/intel', 'asoc/fix/rockchip' and 'asoc/fix/rt5677' into asoc-linus (2014-12-24 12:56:32 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.19 A few fixes for v3.19, a few driver specifics and one core fix which fixes a boot crash on OMAP if deferred probing kicks in due to attempting to modify static data. ---------------------------------------------------------------- Andrew Jackson (2): ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap ASoC: dwc: Iterate over all channels Fang, Yang A (1): ASoC: rt5677: fixed rt5677_dsp_vad_put rt5677_dsp_vad_get panic Jarkko Nikula (3): ASoC: Intel: Add I2C dependency to two new machines ASoC: Intel: Fix BYTCR firmware name ASoC: Intel: Fix BYTCR machine driver MODULE_ALIAS Jianqun Xu (2): ASoC: rockchip: i2s: fix error defination of transmit data level ASoC: rockchip: i2s: fix maxburst of dma data to 4 Jie Yang (1): ASoC: Intel: correct the fixed free block allocation Mark Brown (3): ASoC: dapm: Remove snd_soc_of_parse_audio_routing() due to deferred probe Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/dwc', 'asoc/fix/intel', 'asoc/fix/rockchip' and 'asoc/fix/rt5677' into asoc-linus sound/soc/codecs/rt5677.c | 9 ++++--- sound/soc/dwc/designware_i2s.c | 49 +++++++++++++++++++++++-------------- sound/soc/intel/Kconfig | 4 +-- sound/soc/intel/bytcr_dpcm_rt5640.c | 2 +- sound/soc/intel/sst-firmware.c | 6 ++++- sound/soc/intel/sst/sst_acpi.c | 2 +- sound/soc/rockchip/rockchip_i2s.c | 4 +-- sound/soc/rockchip/rockchip_i2s.h | 2 +- sound/soc/soc-core.c | 14 ++++------- 9 files changed, 52 insertions(+), 40 deletions(-)

2 1

[alsa-devel] [PATCH 0/2 v7] Digidesign Mbox 1 enhancements
by Damien Zammit 30 Dec '14

30 Dec '14

Revision 7. Damien Zammit (2): ALSA: usb-audio: Add mixer control for Digidesign Mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/mixer_maps.c | 9 ++++ sound/usb/mixer_quirks.c | 129 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 41 +++++++++++---- 3 files changed, 170 insertions(+), 9 deletions(-) -- 1.9.1

3 5

[alsa-devel] [PATCH 1/2] [ALSA] add SNDRV_PCM_TRIGGER_DRAIN trigger
by libin.yang＠intel.com 30 Dec '14

30 Dec '14

From: Libin Yang <libin.yang(a)intel.com> Add SNDRV_PCM_TRIGGER_DRAIN trigger for pcm drain. Some audio devices require notification of drain events in order to properly drain and shutdown an audio stream. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/sound/pcm.h | 1 + sound/core/pcm_native.c | 5 +++++ 2 files changed, 6 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 8bb00a2..bb77aa3 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -109,6 +109,7 @@ struct snd_pcm_ops { #define SNDRV_PCM_TRIGGER_PAUSE_RELEASE 4 #define SNDRV_PCM_TRIGGER_SUSPEND 5 #define SNDRV_PCM_TRIGGER_RESUME 6 +#define SNDRV_PCM_TRIGGER_DRAIN 7 #define SNDRV_PCM_POS_XRUN ((snd_pcm_uframes_t)-1) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 166d59c..15fe682 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1517,6 +1517,11 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + + if (substream->runtime->trigger_master == substream) + substream->ops->trigger(substream, + SNDRV_PCM_TRIGGER_DRAIN); + switch (runtime->status->state) { case SNDRV_PCM_STATE_PREPARED: /* start playback stream if possible */ -- 1.9.1

3 8

[alsa-devel] [PATCH] mfd: wm8994: set mfd id-base for regulator devs creation to avoid conflicts
by Inha Song 30 Dec '14

30 Dec '14

After commit: 6e3f62f0793e ("mfd: core: Fix platform-device id generation") We must set the id base when register a duplicate name of mfd_cell. If not, duplicate filename error was reported. - sysfs: cannot create duplicate filename '/devices/.../wm8994-ldo' Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/wm8994-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index e6fab94..47cd3cc 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -36,12 +36,12 @@ static const struct mfd_cell wm8994_regulator_devs[] = { { .name = "wm8994-ldo", - .id = 1, + .id = 0, .pm_runtime_no_callbacks = true, }, { .name = "wm8994-ldo", - .id = 2, + .id = 1, .pm_runtime_no_callbacks = true, }, }; @@ -344,7 +344,7 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) dev_set_drvdata(wm8994->dev, wm8994); /* Add the on-chip regulators first for bootstrapping */ - ret = mfd_add_devices(wm8994->dev, -1, + ret = mfd_add_devices(wm8994->dev, 0, wm8994_regulator_devs, ARRAY_SIZE(wm8994_regulator_devs), NULL, 0, NULL); -- 2.0.0.390.gcb682f8

1 0

[alsa-devel] [PATCH] mfd: wm8994: use PLATFORM_DEVID_AUTO for regulator devs creation to avoid conflicts
by Inha Song 30 Dec '14

30 Dec '14

After commit: 6e3f62f0793e ("mfd: core: Fix platform-device id generation") We must set the id base when register a duplicate name of mfd_cell. but, if we use PLATFORM_DEVID_AUTO flag, cell ids are automatically allocated and managed without cell id setting. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/wm8994-core.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index e6fab94..0505e45 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -36,12 +36,10 @@ static const struct mfd_cell wm8994_regulator_devs[] = { { .name = "wm8994-ldo", - .id = 1, .pm_runtime_no_callbacks = true, }, { .name = "wm8994-ldo", - .id = 2, .pm_runtime_no_callbacks = true, }, }; @@ -344,7 +342,7 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) dev_set_drvdata(wm8994->dev, wm8994); /* Add the on-chip regulators first for bootstrapping */ - ret = mfd_add_devices(wm8994->dev, -1, + ret = mfd_add_devices(wm8994->dev, PLATFORM_DEVID_AUTO, wm8994_regulator_devs, ARRAY_SIZE(wm8994_regulator_devs), NULL, 0, NULL); -- 2.0.0.390.gcb682f8

2 2

[alsa-devel] [PATCH] ASoC: fsl_esai: Fix incorrect xDC field width of xCCR registers
by Nicolin Chen 30 Dec '14

30 Dec '14

The xDC field should have 5 bit width according to Reference Manual. Thus this patch fixes it. Signed-off-by: Aurelien BOUIN <a_bouin(a)yahoo.fr> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- The patch was originally submitted by Aurelien BOUIN while in an informal way. And he hasn't finished any re-submitting during the past two weeks. Thus I create a new patch with his signed-off included so as to fix the probelm as soon as possible, not sure whether it's decent or not though. --Nicolin sound/soc/fsl/fsl_esai.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 91a550f..5e793bb 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -302,7 +302,7 @@ #define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) #define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) #define ESAI_xCCR_xDC_SHIFT 9 -#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_WIDTH 5 #define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) #define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) #define ESAI_xCCR_xPSR_SHIFT 8 -- 1.9.1

2 2

[alsa-devel] [PATCH 1/2] ASoC: fsl_spdif: Use dev_name() for registering the irq
by Fabio Estevam 30 Dec '14

30 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> The 'name' array is currently stored inside the fsl_spdif_priv private structure only for registering the interrupt name. This can be simplified by registering it with dev_name() instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index af042942..73da1f0 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -90,7 +90,6 @@ struct spdif_mixer_control { * @sysclk: system clock for rx clock rate measurement * @dma_params_tx: DMA parameters for transmit channel * @dma_params_rx: DMA parameters for receive channel - * @name: driver name */ struct fsl_spdif_priv { struct spdif_mixer_control fsl_spdif_control; @@ -109,12 +108,8 @@ struct fsl_spdif_priv { struct clk *sysclk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; - - /* The name space will be allocated dynamically */ - char name[0]; }; - /* DPLL locked and lock loss interrupt handler */ static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv) { @@ -1169,19 +1164,15 @@ static int fsl_spdif_probe(struct platform_device *pdev) if (!np) return -ENODEV; - spdif_priv = devm_kzalloc(&pdev->dev, - sizeof(struct fsl_spdif_priv) + strlen(np->name) + 1, - GFP_KERNEL); + spdif_priv = devm_kzalloc(&pdev->dev, sizeof(*spdif_priv), GFP_KERNEL); if (!spdif_priv) return -ENOMEM; - strcpy(spdif_priv->name, np->name); - spdif_priv->pdev = pdev; /* Initialize this copy of the CPU DAI driver structure */ memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); - spdif_priv->cpu_dai_drv.name = spdif_priv->name; + spdif_priv->cpu_dai_drv.name = dev_name(&pdev->dev); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -1203,7 +1194,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) } ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0, - spdif_priv->name, spdif_priv); + dev_name(&pdev->dev), spdif_priv); if (ret) { dev_err(&pdev->dev, "could not claim irq %u\n", irq); return ret; -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: fsl_esai: Use dev_name() for registering the irq
by Fabio Estevam 30 Dec '14

30 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> The 'name' array is currently stored inside the fsl_esai private structure only for registering the interrupt name. This can be simplified by registering it with dev_name() instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_esai.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 1c08ab1..dbe9204 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -44,7 +44,6 @@ * @sck_div: if using PSR/PM dividers for SCKx clock * @slave_mode: if fully using DAI slave mode * @synchronous: if using tx/rx synchronous mode - * @name: driver name */ struct fsl_esai { struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -63,7 +62,6 @@ struct fsl_esai { bool sck_div[2]; bool slave_mode; bool synchronous; - char name[32]; }; static irqreturn_t esai_isr(int irq, void *devid) @@ -739,7 +737,6 @@ static int fsl_esai_probe(struct platform_device *pdev) return -ENOMEM; esai_priv->pdev = pdev; - strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -779,7 +776,7 @@ static int fsl_esai_probe(struct platform_device *pdev) } ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0, - esai_priv->name, esai_priv); + dev_name(&pdev->dev), esai_priv); if (ret) { dev_err(&pdev->dev, "failed to claim irq %u\n", irq); return ret; -- 1.9.1

3 3

[alsa-devel] [PATCH v5 0/4] ASoC: simple-card: Add multi-CODEC support
by Jean-Francois Moine 29 Dec '14

29 Dec '14

This patch set adds multi-CODEC support to the simple card. v5: accept only "sound-dai" as the dai link property (Arnd Bergmann) v4: (Mark Brown) - document the new core function - calculation of the number of codecs moved to the core function - split the patch about simple-card.c - less code changes - remove the cast of pointers - less changes in the DT binding example 2 v3: rebase on broonie git 2014/11/03 v2: accept list of CODECs in 'sound-dais' (Benoit Cousson) Jean-Francois Moine (4): ASoC: core: add multi-codec support in DT ASoC: simple-card: Remove useless function argument ASoC: simple-card: add multi-CODECs in DT ASoC: simple-card: Remove useless check .../devicetree/bindings/sound/simple-card.txt | 18 +--- include/sound/soc.h | 3 + sound/soc/generic/simple-card.c | 87 ++++++++-------- sound/soc/soc-core.c | 110 ++++++++++++++++++--- 4 files changed, 149 insertions(+), 69 deletions(-) -- 2.1.3

2 7

[alsa-devel] [Patch V3 00/10] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 29 Dec '14

29 Dec '14

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This set of patches adds support for audio on the Qualcomm Technologies ipq806x SOC. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This encapsulates the MI2S port, which is what these drivers are configured to use. The I2S pins are connected to an external DAC/amp chip. In addition, a single GPIO is connected to the same DAC/amp, which gives the SOC enable/disable control. The specific drivers added are: - a Codec DAI driver that controls the SOC external pins - a CPU DAI driver for controlling the LPASS-LPAIF block - a PCM MI2S platform driver These drivers, together, are used by simple-audio-card to complete the audio implementation. Corresponding additions to the device tree for the ipq806x and its documentation has also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well. == Updates from my previous post: [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC http://thread.gmane.org/gmane.linux.ports.arm.msm/10989 - Removed the PCM platform driver from the DTS platform and tied it to the CPU DAI driver. - Changed I2S pinctrl to use generic naming convention and moved control to CPU DAI driver. It should be controlled now by soc-core's pinctrl_pm_* functionality. - Added stub DAPM support in codec driver. As the DAC GPIO needs to be enabled last when starting playback, and disabled first when stopping playback, it seems as though the trigger function may be the place for this. Suggestions are welcome for a better place to put this. - Removed machine driver and tied DAI drivers to simple-audio-card. - Packaged the build files and Maxim codec files together in one change. - Removed QCOM as vendor from Maxim code and documentation. - Separated the SOC and board definitions into the correct DTS files. - Update device tree documentation to reflect changes. - General code cleanup. == Additional patch series upon which this patch series has device-tree dependencies (lcc) [PATCH v2 0/8] qcom audio clock control drivers http://thread.gmane.org/gmane.linux.ports.arm.msm/10793 Kenneth Westfield (10): MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: qcom: Document MAX98357A bindings ASoC: qcom: Document LPASS CPU bindings ASoC: codec: Add MAX98357A codec driver ASoC: ipq806x: add LPASS header files ASoC: ipq806x: Add LPASS CPU DAI driver ASoC: ipq806x: Add I2S PCM platform driver ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware .../devicetree/bindings/sound/max98357a.txt | 14 + .../bindings/sound/qcom,lpass-cpu-mi2s.txt | 42 ++ MAINTAINERS | 7 + arch/arm/boot/dts/qcom-ipq8064.dtsi | 23 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98357a.c | 136 ++++++ sound/soc/qcom/Kconfig | 27 ++ sound/soc/qcom/Makefile | 6 + sound/soc/qcom/lpass-cpu-mi2s.c | 378 ++++++++++++++++ sound/soc/qcom/lpass-lpaif-reg.h | 155 +++++++ sound/soc/qcom/lpass-mi2s.h | 34 ++ sound/soc/qcom/lpass-pcm-mi2s.c | 486 +++++++++++++++++++++ 15 files changed, 1316 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98357a.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu-mi2s.txt create mode 100644 sound/soc/codecs/max98357a.c create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/lpass-cpu-mi2s.c create mode 100644 sound/soc/qcom/lpass-lpaif-reg.h create mode 100644 sound/soc/qcom/lpass-mi2s.h create mode 100644 sound/soc/qcom/lpass-pcm-mi2s.c -- 1.8.2.1

2 15

[alsa-devel] minos-one live DVD with Advanced Gtk+ Sequencer
by Joël Krähemann 28 Dec '14

28 Dec '14

Hi out there, I just built a http://linuxfromscratch.org based live DVD called minos-one http://sourceforge.net/projects/ags/files/minos-one.iso Cheers Joël

1 0

[alsa-devel] [PATCH v3] ALSA: atmel: fix building the ac97 driver for at91-multiplatform
by Alexandre Belloni 27 Dec '14

27 Dec '14

From: Arnd Bergmann <arnd(a)arndb.de> at91 will no longer export the mach/cpu.h and mach/hardware.h header files in the future, which would break building the atmel ac97c driver. Since the cpu_is_* check is only used to find out whether we are running on avr32 or arm/at91, we can hardcode that check in the ARM case. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Link: http://www.spinics.net/lists/arm-kernel/msg382068.html Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- Changes in v3: - added my SoB sound/atmel/ac97c.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b59427d5a697..e458957c5f52 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -34,10 +34,10 @@ #include <linux/platform_data/dma-dw.h> #include <linux/dma/dw.h> +#ifdef CONFIG_AVR32 #include <mach/cpu.h> - -#ifdef CONFIG_ARCH_AT91 -#include <mach/hardware.h> +#else +#define cpu_is_at32ap7000() 0 #endif #include "ac97c.h" -- 2.1.0

2 1

[alsa-devel] [PATCH v2] ALSA: atmel: fix building the ac97 driver for at91-multiplatform
by Alexandre Belloni 26 Dec '14

26 Dec '14

From: Arnd Bergmann <arnd(a)arndb.de> at91 will no longer export the mach/cpu.h and mach/hardware.h header files in the future, which would break building the atmel ac97c driver. Since the cpu_is_* check is only used to find out whether we are running on avr32 or arm/at91, we can hardcode that check in the ARM case. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Link: http://www.spinics.net/lists/arm-kernel/msg382068.html --- sound/atmel/ac97c.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b59427d5a697..83975f8d5947 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -34,10 +34,10 @@ #include <linux/platform_data/dma-dw.h> #include <linux/dma/dw.h> +#ifdef CONFIG_AVR32 #include <mach/cpu.h> - -#ifdef CONFIG_ARCH_AT91 -#include <mach/hardware.h> +#else +#define cpu_is_at32ap7000() 0 #endif #include "ac97c.h" -- 2.1.0

2 3

[alsa-devel] [PATCH] ASoC: wm5102: Use put_unaligned_be16
by Vaishali Thakkar 26 Dec '14

26 Dec '14

This patch introduces the use of function put_unaligned_le16. This is done using Coccinelle and semantic patch used is as follows: @a@ typedef u16, __be16, uint16_t; {u16,__be16,uint16_t} e16; identifier tmp; expression ptr; expression y,e; type T; @@ - tmp = cpu_to_be16(y); <+... when != tmp ( - memcpy(ptr, (T)&tmp, $2\|sizeof(u16)\|sizeof(__be16)\|sizeof(uint16_t)\|sizeof(e16)$); + put_unaligned_be16(y,ptr); | - memcpy(ptr, (T)&tmp, ...); + put_unaligned_be16(y,ptr); ) ...+> ? tmp = e @@ type T; identifier a.tmp; @@ - T tmp; ...when != tmp Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/codecs/wm5102.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f602349..fec21f9 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -28,6 +28,7 @@ #include <linux/mfd/arizona/core.h> #include <linux/mfd/arizona/registers.h> +#include <asm/unaligned.h> #include "arizona.h" #include "wm5102.h" @@ -617,11 +618,11 @@ static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - uint16_t data; mutex_lock(&codec->mutex); - data = cpu_to_be16(arizona->dac_comp_coeff); - memcpy(ucontrol->value.bytes.data, &data, sizeof(data)); + + put_unaligned_be16(arizona->dac_comp_coeff, + ucontrol->value.bytes.data); mutex_unlock(&codec->mutex); return 0; -- 1.9.1

1 0

[alsa-devel] [PATCH] sound: pci: hda: patch_analog.c: Remove some unused functions
by Rickard Strandqvist 25 Dec '14

25 Dec '14

Removes some functions that are not used anywhere: ad198x_ch_mode_get() ad198x_ch_mode_info() ad198x_ch_mode_info() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/hda/patch_analog.c | 33 --------------------------------- 1 file changed, 33 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 06275f8..0e76dcc 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -697,39 +697,6 @@ static int patch_ad1981(struct hda_codec *codec) * E/F quad mic array */ -#ifdef ENABLE_AD_STATIC_QUIRKS -static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode, - spec->num_channel_mode); -} - -static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, spec->multiout.max_channels); -} - -static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - &spec->multiout.max_channels); - if (err >= 0 && spec->need_dac_fix) - spec->multiout.num_dacs = spec->multiout.max_channels / 2; - return err; -} -#endif /* ENABLE_AD_STATIC_QUIRKS */ - static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { -- 1.7.10.4

2 1

[alsa-devel] [PATCH] sound: pci: hda: patch_analog.c: Remove unused function
by Rickard Strandqvist 25 Dec '14

25 Dec '14

Remove the function ad198x_ch_mode_get() that is not used anywhere. This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/hda/patch_analog.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 06275f8..9d5257b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -707,15 +707,6 @@ static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, spec->num_channel_mode); } -static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, spec->multiout.max_channels); -} - static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { -- 1.7.10.4

3 3

[alsa-devel] [PATCH] sound: isa: gus: gus_instr.c: Remove unused function
by Rickard Strandqvist 25 Dec '14

25 Dec '14

Remove the function snd_gus_simple_remove_sample() that is not used anywhere. This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/isa/gus/gus_instr.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/isa/gus/gus_instr.c b/sound/isa/gus/gus_instr.c index 4dc9caf..2d21408 100644 --- a/sound/isa/gus/gus_instr.c +++ b/sound/isa/gus/gus_instr.c @@ -162,11 +162,3 @@ int snd_gus_simple_get_sample(void *private_data, struct simple_instrument *inst return snd_gus_dram_read(gus, data, instr->address.memory, instr->size, 0); } - -int snd_gus_simple_remove_sample(void *private_data, struct simple_instrument *instr, - int atomic) -{ - struct snd_gus_card *gus = private_data; - - return snd_gf1_mem_free(&gus->gf1.mem_alloc, instr->address.memory); -} -- 1.7.10.4

2 1

[alsa-devel] [PATCH] sound: pci: ice1712: wm8776.c: Remove some unused functions
by Rickard Strandqvist 25 Dec '14

25 Dec '14

Removes some functions that are not used anywhere: snd_wm8776_set_master_mode() snd_wm8776_set_adc_if() snd_wm8776_set_dac_if() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/ice1712/wm8776.c | 15 --------------- sound/pci/ice1712/wm8776.h | 3 --- 2 files changed, 18 deletions(-) diff --git a/sound/pci/ice1712/wm8776.c b/sound/pci/ice1712/wm8776.c index e66c0da..ebd2fe4 100644 --- a/sound/pci/ice1712/wm8776.c +++ b/sound/pci/ice1712/wm8776.c @@ -452,21 +452,6 @@ void snd_wm8776_resume(struct snd_wm8776 *wm) snd_wm8776_write(wm, i, wm->regs[i]); } -void snd_wm8776_set_dac_if(struct snd_wm8776 *wm, u16 dac) -{ - snd_wm8776_write(wm, WM8776_REG_DACIFCTRL, dac); -} - -void snd_wm8776_set_adc_if(struct snd_wm8776 *wm, u16 adc) -{ - snd_wm8776_write(wm, WM8776_REG_ADCIFCTRL, adc); -} - -void snd_wm8776_set_master_mode(struct snd_wm8776 *wm, u16 mode) -{ - snd_wm8776_write(wm, WM8776_REG_MSTRCTRL, mode); -} - void snd_wm8776_set_power(struct snd_wm8776 *wm, u16 power) { snd_wm8776_write(wm, WM8776_REG_PWRDOWN, power); diff --git a/sound/pci/ice1712/wm8776.h b/sound/pci/ice1712/wm8776.h index 93a2d69..42acef0 100644 --- a/sound/pci/ice1712/wm8776.h +++ b/sound/pci/ice1712/wm8776.h @@ -216,9 +216,6 @@ struct snd_wm8776 { void snd_wm8776_init(struct snd_wm8776 *wm); void snd_wm8776_resume(struct snd_wm8776 *wm); -void snd_wm8776_set_dac_if(struct snd_wm8776 *wm, u16 dac); -void snd_wm8776_set_adc_if(struct snd_wm8776 *wm, u16 adc); -void snd_wm8776_set_master_mode(struct snd_wm8776 *wm, u16 mode); void snd_wm8776_set_power(struct snd_wm8776 *wm, u16 power); void snd_wm8776_volume_restore(struct snd_wm8776 *wm); int snd_wm8776_build_controls(struct snd_wm8776 *wm); -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ALSA: pcm_dmaengine: correct the error handler of dmaengine_prep_dma_cyclic
by Barry Song 25 Dec '14

25 Dec '14

From: Barry Song <Baohua.Song(a)csr.com> preparing cyclic DMA description can fail either due to DMA desc list is full(-ENOMEM), or due to the coming DMA configuration is illegal or not supported by the acting DMA hardware(other ERR codes). Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/core/pcm_dmaengine.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 6542c40..5cac7e4 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -163,6 +163,8 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) if (!desc) return -ENOMEM; + if (IS_ERR(desc)) + return PTR_ERR(desc); desc->callback = dmaengine_pcm_dma_complete; desc->callback_param = substream; -- 2.2.0

2 5

[alsa-devel] Can not rmmod snd_hda_codec_hdmi
by Xuebing Wang 24 Dec '14

24 Dec '14

Hi community, I am trying to implement the function of streaming SACD music over HDMI, my PC is Intel Haswell based CPU (Intel i7 4770). When I am trying to 'rmmod snd_hda_codec_hdmi', I get the prompt that this module is in use (detailed message as below) rmmod: ERROR: Module snd_hda_codec_hdmi is in use Is it possible that workqueue [hd-audio] is using this kernel module (snd_hda_codec_hdmi)? Can I use rmmod / insmod as the way to debug this module? Thanks. Xuebing Wang

5 9

[alsa-devel] [PATCH 0/2] ASoC: rockchip: i2s: fix TDL and RDL to 16 bits
by Jianqun Xu 24 Dec '14

24 Dec '14

To make Bus DMA works more efficiency, DMA may work in burst mode, we set the watermark of transmit and receive reach to 16 bits. Jianqun Xu (2): ASoC: rockchip: i2s: fix error defination of transmit data level ASoC: rockchip: i2s: set TDL and RDL to 16 bits sound/soc/rockchip/rockchip_i2s.c | 4 ++++ sound/soc/rockchip/rockchip_i2s.h | 2 +- 2 files changed, 5 insertions(+), 1 deletion(-) -- 1.9.1

4 9

[alsa-devel] [PATCH 1/3 v2] ASoC: rt5677: Revise the filter powers
by Oder Chiou 24 Dec '14

24 Dec '14

Add the filter powers of the dac mono3 and mono4, and remove the connection of dac stereo1 filter that connect to DAC1 MIX. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 22 ++++++++++++++++------ 1 file changed, 16 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 81fe146..a15e39d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -2645,10 +2645,18 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* DAC Mixer */ SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_S1F_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("dac mono left filter", RT5677_PWR_DIG2, + SND_SOC_DAPM_SUPPLY("dac mono2 left filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_M2F_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("dac mono right filter", RT5677_PWR_DIG2, + SND_SOC_DAPM_SUPPLY("dac mono2 right filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_M2F_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono3 left filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M3F_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono3 right filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M3F_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono4 left filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M4F_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono4 right filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M4F_R_BIT, 0, NULL, 0), SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, rt5677_sto1_dac_l_mix, ARRAY_SIZE(rt5677_sto1_dac_l_mix)), @@ -3455,10 +3463,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC1 MIXL", "Stereo ADC Switch", "ADDA1 Mux" }, { "DAC1 MIXL", "DAC1 Switch", "DAC1 Mux" }, - { "DAC1 MIXL", NULL, "dac stereo1 filter" }, { "DAC1 MIXR", "Stereo ADC Switch", "ADDA1 Mux" }, { "DAC1 MIXR", "DAC1 Switch", "DAC1 Mux" }, - { "DAC1 MIXR", NULL, "dac stereo1 filter" }, { "DAC1 FS", NULL, "DAC1 MIXL" }, { "DAC1 FS", NULL, "DAC1 MIXR" }, @@ -3530,30 +3536,34 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "Mono DAC MIXL", "DAC1 L Switch", "DAC1 MIXL" }, { "Mono DAC MIXL", "DAC2 L Switch", "DAC2 L Mux" }, { "Mono DAC MIXL", "DAC2 R Switch", "DAC2 R Mux" }, - { "Mono DAC MIXL", NULL, "dac mono left filter" }, + { "Mono DAC MIXL", NULL, "dac mono2 left filter" }, { "Mono DAC MIXR", "ST R Switch", "Sidetone Mux" }, { "Mono DAC MIXR", "DAC1 R Switch", "DAC1 MIXR" }, { "Mono DAC MIXR", "DAC2 R Switch", "DAC2 R Mux" }, { "Mono DAC MIXR", "DAC2 L Switch", "DAC2 L Mux" }, - { "Mono DAC MIXR", NULL, "dac mono right filter" }, + { "Mono DAC MIXR", NULL, "dac mono2 right filter" }, { "DD1 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, { "DD1 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" }, { "DD1 MIXL", "DAC3 L Switch", "DAC3 L Mux" }, { "DD1 MIXL", "DAC3 R Switch", "DAC3 R Mux" }, + { "DD1 MIXL", NULL, "dac mono3 left filter" }, { "DD1 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, { "DD1 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" }, { "DD1 MIXR", "DAC3 L Switch", "DAC3 L Mux" }, { "DD1 MIXR", "DAC3 R Switch", "DAC3 R Mux" }, + { "DD1 MIXR", NULL, "dac mono3 right filter" }, { "DD2 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, { "DD2 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" }, { "DD2 MIXL", "DAC4 L Switch", "DAC4 L Mux" }, { "DD2 MIXL", "DAC4 R Switch", "DAC4 R Mux" }, + { "DD2 MIXL", NULL, "dac mono4 left filter" }, { "DD2 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, { "DD2 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" }, { "DD2 MIXR", "DAC4 L Switch", "DAC4 L Mux" }, { "DD2 MIXR", "DAC4 R Switch", "DAC4 R Mux" }, + { "DD2 MIXR", NULL, "dac mono4 right filter" }, { "Stereo DAC MIX", NULL, "Stereo DAC MIXL" }, { "Stereo DAC MIX", NULL, "Stereo DAC MIXR" }, -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH] ASoC: Intel: Store the entry_point read from FW file
by Jie Yang 24 Dec '14

24 Dec '14

To enable some modules from other than base FW, according to FW interface spec, we need pass the correct entry point param to FW, so here store the entry_point read from FW file for later usage. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-firmware.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index ef2e8b5..50d6925 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -497,6 +497,7 @@ struct sst_module *sst_module_new(struct sst_fw *sst_fw, sst_module->sst_fw = sst_fw; sst_module->scratch_size = template->scratch_size; sst_module->persistent_size = template->persistent_size; + sst_module->entry = template->entry; INIT_LIST_HEAD(&sst_module->block_list); INIT_LIST_HEAD(&sst_module->runtime_list); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: correct the fixed free block allocation
by Jie Yang 24 Dec '14

24 Dec '14

For block span more than 1 section, when allocate it from a free block, we need allocate the remain buffers within the block, and then continue alloc the rest of needed size buffer. Here also make sure this free block is moved from free list to used list, and add it to block_list which may be used for power gating disabling later. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-firmware.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 4a5bde9..ef2e8b5 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -763,8 +763,12 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba /* does block span more than 1 section */ if (ba->offset >= block->offset && ba->offset < block_end) { + /* add block */ + list_move(&block->list, &dsp->used_block_list); + list_add(&block->module_list, block_list); /* align ba to block boundary */ - ba->offset = block->offset; + ba->size -= block_end - ba->offset; + ba->offset = block_end; err = block_alloc_contiguous(dsp, ba, block_list); if (err < 0) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: pcm: Fix vague codec and cpu DAI prepare error messages
by Jarkko Nikula 24 Dec '14

24 Dec '14

Both codec and cpu DAI prepare print the same error message making it a bit more difficult to grep quickly from sources. Fix this by telling it explicitly. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/soc-pcm.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index eb87d96e2cf0..0ae0e2a9eed7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -746,7 +746,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) codec_dai); if (ret < 0) { dev_err(codec_dai->dev, - "ASoC: DAI prepare error: %d\n", ret); + "ASoC: codec DAI prepare error: %d\n", + ret); goto out; } } @@ -755,8 +756,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops && cpu_dai->driver->ops->prepare) { ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); if (ret < 0) { - dev_err(cpu_dai->dev, "ASoC: DAI prepare error: %d\n", - ret); + dev_err(cpu_dai->dev, + "ASoC: cpu DAI prepare error: %d\n", ret); goto out; } } -- 2.1.3

2 1

[alsa-devel] [PATCH] ASoC: rt5677: fixed rt5677_dsp_vad_put rt5677_dsp_vad_get panic
by yang.a.fang＠intel.com 24 Dec '14

24 Dec '14

From: "Fang, Yang A" <yang.a.fang(a)intel.com> snd_kcontrol_chip should return snd_soc_component instead of snd_soc_codec Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/rt5677.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 81fe146..c0fbe18 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -784,8 +784,8 @@ static unsigned int bst_tlv[] = { static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); ucontrol->value.integer.value[0] = rt5677->dsp_vad_en; @@ -795,8 +795,9 @@ static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol, static int rt5677_dsp_vad_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); rt5677->dsp_vad_en = !!ucontrol->value.integer.value[0]; -- 1.7.9.5

2 1

[alsa-devel] [PATCH v4 0/5] ASoC: dwc: Add device tree support to designware I2S
by Andrew Jackson 24 Dec '14

24 Dec '14

From: Andrew Jackson <Andrew.Jackson(a)arm.com> This patch set extends the DesignWare I2S driver to provide device tree support and fixes a couple of small faults. Changes v3->v4 + Drop applied patches + Use combined function to prepare clock [Mark Brown] + Use managed clock resources to avoid clk_put [Mark Brown] + Read configuration prameters from hardware for both platform data and device tree [Mark Brown] + Re-order patch sequence [Mark Brown] + Change union name to avoid future collisions [Mark Brown] + Check return code from clk_set_rate [Mark Brown] + Check parametsrs read from hardware agaist array limits and add further comments [Mark Brown] + Add of_match_ptr [Mark Brown] Changes v2->v3 + Drop applied patch + Flush FIFOs in prepare rather than hw_params [Lars-Peter Clausen] Changes v1->v2 + Drop negative use count patch [Mark Brown] + Remove unnecessary debug print messages [Lars-Peter Clausen] + Rewrite iteration as for loop rather than do...while [Mark Brown] + Reorder patches to send fixes first [Mark Brown] + Simplify device tree code [Mark Brown] + Split device tree patch in two [Mark Brown] + Expand explanatory comment on channel configuration [Rajeev Kumar] Arnd: I've not forgotten about updating the spear entries and will submit as a separate patch. Andrew Jackson (5): ASoC: dwc: Prepare clock before use ASoC: dwc: Switch to managed clock resource ASoC: dwc: Read I2S block configuration from registers ASoC: dwc: Add documentation for I2S DT ASoC: dwc: Add devicetree support for Designware I2S .../devicetree/bindings/sound/designware-i2s.txt | 31 +++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 275 ++++++++++++++++---- 3 files changed, 260 insertions(+), 47 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt

1 7

Re: [alsa-devel] [PATCH] sound: pci: asihpi: hpifunc.c: Remove some unused functions
by Eliot Blennerhassett 23 Dec '14

23 Dec '14

On 22/12/14 05:50, Rickard Strandqvist wrote: > Removes some functions that are not used anywhere: NAK - because hpi_adapter_set_property() is now used in for-next branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Otherwise I have no problem with the patch.

2 2

[alsa-devel] [PATCH v2 1/2] ASoC: rt5670: Add runtime PM support
by Bard Liao 22 Dec '14

22 Dec '14

This patch adds runtime PM support on rt5670 codec. Signed-off-by: Lin Mengdong <mengdong.lin(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * Remove the unused runtime suspend/resume code. We will switch the sysclk source on another patch. --- sound/soc/codecs/rt5670.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 09cf100..a8a471b 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -14,6 +14,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/acpi.h> @@ -2734,18 +2735,26 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, } + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, rt5670_dai, ARRAY_SIZE(rt5670_dai)); if (ret < 0) goto err; + pm_runtime_put(&i2c->dev); + return 0; err: + pm_runtime_disable(&i2c->dev); + return ret; } static int rt5670_i2c_remove(struct i2c_client *i2c) { + pm_runtime_disable(&i2c->dev); snd_soc_unregister_codec(&i2c->dev); return 0; -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH 1/3] ASoC: rt5677: Revise the filter powers
by Oder Chiou 22 Dec '14

22 Dec '14

Revise the filter powers. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 22 ++++++++++++++++------ 1 file changed, 16 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 81fe146..a15e39d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -2645,10 +2645,18 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* DAC Mixer */ SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_S1F_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("dac mono left filter", RT5677_PWR_DIG2, + SND_SOC_DAPM_SUPPLY("dac mono2 left filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_M2F_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("dac mono right filter", RT5677_PWR_DIG2, + SND_SOC_DAPM_SUPPLY("dac mono2 right filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_M2F_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono3 left filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M3F_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono3 right filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M3F_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono4 left filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M4F_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono4 right filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M4F_R_BIT, 0, NULL, 0), SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, rt5677_sto1_dac_l_mix, ARRAY_SIZE(rt5677_sto1_dac_l_mix)), @@ -3455,10 +3463,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC1 MIXL", "Stereo ADC Switch", "ADDA1 Mux" }, { "DAC1 MIXL", "DAC1 Switch", "DAC1 Mux" }, - { "DAC1 MIXL", NULL, "dac stereo1 filter" }, { "DAC1 MIXR", "Stereo ADC Switch", "ADDA1 Mux" }, { "DAC1 MIXR", "DAC1 Switch", "DAC1 Mux" }, - { "DAC1 MIXR", NULL, "dac stereo1 filter" }, { "DAC1 FS", NULL, "DAC1 MIXL" }, { "DAC1 FS", NULL, "DAC1 MIXR" }, @@ -3530,30 +3536,34 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "Mono DAC MIXL", "DAC1 L Switch", "DAC1 MIXL" }, { "Mono DAC MIXL", "DAC2 L Switch", "DAC2 L Mux" }, { "Mono DAC MIXL", "DAC2 R Switch", "DAC2 R Mux" }, - { "Mono DAC MIXL", NULL, "dac mono left filter" }, + { "Mono DAC MIXL", NULL, "dac mono2 left filter" }, { "Mono DAC MIXR", "ST R Switch", "Sidetone Mux" }, { "Mono DAC MIXR", "DAC1 R Switch", "DAC1 MIXR" }, { "Mono DAC MIXR", "DAC2 R Switch", "DAC2 R Mux" }, { "Mono DAC MIXR", "DAC2 L Switch", "DAC2 L Mux" }, - { "Mono DAC MIXR", NULL, "dac mono right filter" }, + { "Mono DAC MIXR", NULL, "dac mono2 right filter" }, { "DD1 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, { "DD1 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" }, { "DD1 MIXL", "DAC3 L Switch", "DAC3 L Mux" }, { "DD1 MIXL", "DAC3 R Switch", "DAC3 R Mux" }, + { "DD1 MIXL", NULL, "dac mono3 left filter" }, { "DD1 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, { "DD1 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" }, { "DD1 MIXR", "DAC3 L Switch", "DAC3 L Mux" }, { "DD1 MIXR", "DAC3 R Switch", "DAC3 R Mux" }, + { "DD1 MIXR", NULL, "dac mono3 right filter" }, { "DD2 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, { "DD2 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" }, { "DD2 MIXL", "DAC4 L Switch", "DAC4 L Mux" }, { "DD2 MIXL", "DAC4 R Switch", "DAC4 R Mux" }, + { "DD2 MIXL", NULL, "dac mono4 left filter" }, { "DD2 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, { "DD2 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" }, { "DD2 MIXR", "DAC4 L Switch", "DAC4 L Mux" }, { "DD2 MIXR", "DAC4 R Switch", "DAC4 R Mux" }, + { "DD2 MIXR", NULL, "dac mono4 right filter" }, { "Stereo DAC MIX", NULL, "Stereo DAC MIXL" }, { "Stereo DAC MIX", NULL, "Stereo DAC MIXR" }, -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] [PATCH -next] ASoC: Intel: fix sparse non static symbol warnings
by weiyj_lk＠163.com 22 Dec '14

22 Dec '14

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fixes the following sparse warnings: sound/soc/intel/sst/sst_acpi.c:248:5: warning: symbol 'sst_acpi_probe' was not declared. Should it be static? sound/soc/intel/sst/sst_acpi.c:335:5: warning: symbol 'sst_acpi_remove' was not declared. Should it be static? Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/intel/sst/sst_acpi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 31124aa..98a8f66 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -245,7 +245,7 @@ static struct sst_machines *sst_acpi_find_machine( return NULL; } -int sst_acpi_probe(struct platform_device *pdev) +static int sst_acpi_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; int ret = 0; @@ -332,7 +332,7 @@ do_sst_cleanup: * This function is called by OS when a device is unloaded * This frees the interrupt etc */ -int sst_acpi_remove(struct platform_device *pdev) +static int sst_acpi_remove(struct platform_device *pdev) { struct intel_sst_drv *ctx;

2 1

[alsa-devel] [PATCH] ASoC: pcm512x: Also support PCM514x devices
by Peter Rosin 22 Dec '14

22 Dec '14

From: Peter Rosin <peda(a)axentia.se> Signed-off-by: Peter Rosin <peda(a)axentia.se> --- Documentation/devicetree/bindings/sound/pcm512x.txt | 3 ++- sound/soc/codecs/pcm512x-i2c.c | 4 ++++ sound/soc/codecs/pcm512x-spi.c | 4 ++++ 3 files changed, 10 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt index faff75e64573..98e0d34915e8 100644 --- a/Documentation/devicetree/bindings/sound/pcm512x.txt +++ b/Documentation/devicetree/bindings/sound/pcm512x.txt @@ -5,7 +5,8 @@ on the board). Required properties: - - compatible : One of "ti,pcm5121" or "ti,pcm5122" + - compatible : One of "ti,pcm5121", "ti,pcm5122", "ti,pcm5141" or + "ti,pcm5142" - reg : the I2C address of the device for I2C, the chip select number for SPI. diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c index 4d62230bd378..08047ac952f4 100644 --- a/sound/soc/codecs/pcm512x-i2c.c +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -41,6 +41,8 @@ static int pcm512x_i2c_remove(struct i2c_client *i2c) static const struct i2c_device_id pcm512x_i2c_id[] = { { "pcm5121", }, { "pcm5122", }, + { "pcm5141", }, + { "pcm5142", }, { } }; MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); @@ -48,6 +50,8 @@ MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); static const struct of_device_id pcm512x_of_match[] = { { .compatible = "ti,pcm5121", }, { .compatible = "ti,pcm5122", }, + { .compatible = "ti,pcm5141", }, + { .compatible = "ti,pcm5142", }, { } }; MODULE_DEVICE_TABLE(of, pcm512x_of_match); diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c index f297058c0038..7b64a9cef704 100644 --- a/sound/soc/codecs/pcm512x-spi.c +++ b/sound/soc/codecs/pcm512x-spi.c @@ -43,6 +43,8 @@ static int pcm512x_spi_remove(struct spi_device *spi) static const struct spi_device_id pcm512x_spi_id[] = { { "pcm5121", }, { "pcm5122", }, + { "pcm5141", }, + { "pcm5142", }, { }, }; MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); @@ -50,6 +52,8 @@ MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); static const struct of_device_id pcm512x_of_match[] = { { .compatible = "ti,pcm5121", }, { .compatible = "ti,pcm5122", }, + { .compatible = "ti,pcm5141", }, + { .compatible = "ti,pcm5142", }, { } }; MODULE_DEVICE_TABLE(of, pcm512x_of_match); -- 1.7.10.4

2 1

[alsa-devel] [PATCH v3 0/5] ASoC: dwc: Add device tree support to designware I2S
by Andrew Jackson 22 Dec '14

22 Dec '14

From: Andrew Jackson <Andrew.Jackson(a)arm.com> This patch set extends the DesignWare I2S driver to provide device tree support and fixes a couple of small faults. Changes v2->v3 + Drop applied patch + Flush FIFOs in prepare rather than hw_params [Lars-Peter Clausen] Changes v1->v2 + Drop negative use count patch [Mark Brown] + Remove unnecessary debug print messages [Lars-Peter Clausen] + Rewrite iteration as for loop rather than do...while [Mark Brown] + Reorder patches to send fixes first [Mark Brown] + Simplify device tree code [Mark Brown] + Split device tree patch in two [Mark Brown] + Expand explanatory comment on channel configuration [Rajeev Kumar] Arnd: I've not forgotten about updating the spear entries and will submit as a separate patch. Andrew Jackson (5): ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap ASoC: dwc: Iterate over all channels ASoC: dwc: Reorder code in preparation for DT support ASoC: dwc: Add devicetree support for Designware I2S ASoC: dwc: Add documentation for I2S DT .../devicetree/bindings/sound/designware-i2s.txt | 32 +++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 293 +++++++++++++++----- 3 files changed, 258 insertions(+), 68 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt

2 16

[alsa-devel] [PATCH 1/2] ASoC: Intel: Fix BYTCR firmware name
by Jarkko Nikula 22 Dec '14

22 Dec '14

BYTCR DSP firmware is in intel/ subdirectory. See linux-firmware.git commit d562a3b63632 ("linux-firmware: add sst audio firmware for baytrail platforms"). Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 3abc29e8a928..2ac72eb5e75d 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -343,7 +343,7 @@ int sst_acpi_remove(struct platform_device *pdev) } static struct sst_machines sst_acpi_bytcr[] = { - {"10EC5640", "T100", "bytt100_rt5640", NULL, "fw_sst_0f28.bin", + {"10EC5640", "T100", "bytt100_rt5640", NULL, "intel/fw_sst_0f28.bin", &byt_rvp_platform_data }, {}, }; -- 2.1.3

2 2

[alsa-devel] X-Fi XtremeGamer front header ( headphones ) audio
by Marius Cirsta 22 Dec '14

22 Dec '14

Hi, I've got a X-Fi XtremeGamer ( SB0770 ) and I can't get sound output from the headphone plug in my case which is plugged into the front panel connector on the SB0770. This does work in Windows so I'm assuming that it's unimplemented in ALSO for the X-Fi. I can see no mention of headphones in the alsamixer either. Also looking at the code I can see nothing regarding the headphones part of this. If the implementation is indeed missing how difficult would it be to implement this ? I'd like try.

1 0

[alsa-devel] [PATCH v3] ASoC: add xtensa xtfpga I2S interface and platform
by Max Filippov 22 Dec '14

22 Dec '14

XTFPGA boards provides an audio subsystem that consists of TI CDCE706 clock synthesizer, I2S transmitter and TLV320AIC23 audio codec. I2S transmitter has MMIO-based interface that resembles that of the OpenCores I2S transmitter. I2S transmitter is always a master on I2S bus. There's no specialized audio DMA, sample data are transferred to I2S transmitter FIFO by CPU through memory-mapped queue interface. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- Changes v2->v3: - drop .owner = THIS_MODULE from the struct platform_driver initialization; - rebase to 3.18-rc7. Changes v1->v2: - fix module name in the sound/soc/xtensa/Makefile; - split PCM data transfer code and I2S interface setup code; - move clock enabling/disabling to runtime PM code; - document I2S interface RATIO field; - rewrite I2S interface LEVEL field calculation to avoid overflows; - drop select statements from Kconfig; - document RCU usage for tx_substream; - clean up code running in RCU read-side critical section; - use threaded IRQ handler instead of IRQ + tasklet; - remove empty PCM callback handlers; - add const to static structures where applicable; - use platform_set_drvdata instead of dev_set_drvdata; - drop redundant error reporting code; - use platform_get_irq instead of platform_get_resource; - clean up xtfpga_i2s_remove; - add comments. .../devicetree/bindings/sound/cdns,xtfpga-i2s.txt | 18 + MAINTAINERS | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/xtensa/Kconfig | 7 + sound/soc/xtensa/Makefile | 3 + sound/soc/xtensa/xtfpga-i2s.c | 675 +++++++++++++++++++++ 7 files changed, 706 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt create mode 100644 sound/soc/xtensa/Kconfig create mode 100644 sound/soc/xtensa/Makefile create mode 100644 sound/soc/xtensa/xtfpga-i2s.c diff --git a/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt new file mode 100644 index 0000000..befd125 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt @@ -0,0 +1,18 @@ +Bindings for I2S controller built into xtfpga Xtensa bitstreams. + +Required properties: +- compatible: shall be "cdns,xtfpga-i2s". +- reg: memory region (address and length) with device registers. +- interrupts: interrupt for the device. +- clocks: phandle to the clk used as master clock. I2S bus clock + is derived from it. + +Examples: + + i2s0: xtfpga-i2s@0d080000 { + #sound-dai-cells = <0>; + compatible = "cdns,xtfpga-i2s"; + reg = <0x0d080000 0x40>; + interrupts = <2 1>; + clocks = <&cdce706 4>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index 180d560..6a09926 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -10409,6 +10409,7 @@ M: Max Filippov <jcmvbkbc(a)gmail.com> L: linux-xtensa(a)linux-xtensa.org S: Maintained F: drivers/spi/spi-xtensa-xtfpga.c +F: sound/soc/xtensa/xtfpga-i2s.c YAM DRIVER FOR AX.25 M: Jean-Paul Roubelat <jpr(a)f6fbb.org> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 7d5d644..dcc79aa 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -55,6 +55,7 @@ source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" +source "sound/soc/xtensa/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d88edfc..6c24e6a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -32,3 +32,4 @@ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ +obj-$(CONFIG_SND_SOC) += xtensa/ diff --git a/sound/soc/xtensa/Kconfig b/sound/soc/xtensa/Kconfig new file mode 100644 index 0000000..c201beb --- /dev/null +++ b/sound/soc/xtensa/Kconfig @@ -0,0 +1,7 @@ +config SND_SOC_XTFPGA_I2S + tristate "XTFPGA I2S master" + select REGMAP_MMIO + help + Say Y or M if you want to add support for codecs attached to the + I2S interface on XTFPGA daughter board. You will also need to select + the drivers for the rest of XTFPGA audio subsystem. diff --git a/sound/soc/xtensa/Makefile b/sound/soc/xtensa/Makefile new file mode 100644 index 0000000..15efbf9 --- /dev/null +++ b/sound/soc/xtensa/Makefile @@ -0,0 +1,3 @@ +snd-soc-xtfpga-i2s-objs := xtfpga-i2s.o + +obj-$(CONFIG_SND_SOC_XTFPGA_I2S) += snd-soc-xtfpga-i2s.o diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c new file mode 100644 index 0000000..1cfb19e --- /dev/null +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -0,0 +1,675 @@ +/* + * Xtfpga I2S controller driver + * + * Copyright (c) 2014 Cadence Design Systems Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define DRV_NAME "xtfpga-i2s" + +#define XTFPGA_I2S_VERSION 0x00 +#define XTFPGA_I2S_CONFIG 0x04 +#define XTFPGA_I2S_INT_MASK 0x08 +#define XTFPGA_I2S_INT_STATUS 0x0c +#define XTFPGA_I2S_CHAN0_DATA 0x10 +#define XTFPGA_I2S_CHAN1_DATA 0x14 +#define XTFPGA_I2S_CHAN2_DATA 0x18 +#define XTFPGA_I2S_CHAN3_DATA 0x1c + +#define XTFPGA_I2S_CONFIG_TX_ENABLE 0x1 +#define XTFPGA_I2S_CONFIG_INT_ENABLE 0x2 +#define XTFPGA_I2S_CONFIG_LEFT 0x4 +#define XTFPGA_I2S_CONFIG_RATIO_BASE 8 +#define XTFPGA_I2S_CONFIG_RATIO_MASK 0x0000ff00 +#define XTFPGA_I2S_CONFIG_RES_BASE 16 +#define XTFPGA_I2S_CONFIG_RES_MASK 0x003f0000 +#define XTFPGA_I2S_CONFIG_LEVEL_BASE 24 +#define XTFPGA_I2S_CONFIG_LEVEL_MASK 0x0f000000 +#define XTFPGA_I2S_CONFIG_CHANNEL_BASE 28 + +#define XTFPGA_I2S_INT_UNDERRUN 0x1 +#define XTFPGA_I2S_INT_LEVEL 0x2 +#define XTFPGA_I2S_INT_VALID 0x3 + +#define XTFPGA_I2S_FIFO_SIZE 8192 + +/* + * I2S controller operation: + * + * Enabling TX: output 1 period of zeros (starting with left channel) + * and then queued data. + * + * Level status and interrupt: whenever FIFO level is below FIFO trigger, + * level status is 1 and an IRQ is asserted (if enabled). + * + * Underrun status and interrupt: whenever FIFO is empty, underrun status + * is 1 and an IRQ is asserted (if enabled). + */ +struct xtfpga_i2s { + struct device *dev; + struct clk *clk; + struct regmap *regmap; + void __iomem *regs; + + /* current playback substream. NULL if not playing. + * + * Access to that field is synchronized between the interrupt handler + * and userspace through RCU. + * + * Interrupt handler (threaded part) does PIO on substream data in RCU + * read-side critical section. Trigger callback sets and clears the + * pointer when the playback is started and stopped with + * rcu_assign_pointer. When userspace is about to free the playback + * stream in the pcm_close callback it synchronizes with the interrupt + * handler by means of synchronize_rcu call. + */ + struct snd_pcm_substream *tx_substream; + unsigned (*tx_fn)(struct xtfpga_i2s *i2s, + struct snd_pcm_runtime *runtime, + unsigned tx_ptr); + unsigned tx_ptr; /* next frame index in the sample buffer */ + + /* current fifo level estimate. + * Doesn't have to be perfectly accurate, but must be not less than + * the actual FIFO level in order to avoid stall on push attempt. + */ + unsigned tx_fifo_level; + + /* FIFO level at which level interrupt occurs */ + unsigned tx_fifo_low; + + /* maximal FIFO level */ + unsigned tx_fifo_high; +}; + +static bool xtfpga_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + return reg >= XTFPGA_I2S_CONFIG; +} + +static bool xtfpga_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + return reg < XTFPGA_I2S_CHAN0_DATA; +} + +static bool xtfpga_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == XTFPGA_I2S_INT_STATUS; +} + +static const struct regmap_config xtfpga_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = XTFPGA_I2S_CHAN3_DATA, + .writeable_reg = xtfpga_i2s_wr_reg, + .readable_reg = xtfpga_i2s_rd_reg, + .volatile_reg = xtfpga_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +/* Generate functions that do PIO from TX DMA area to FIFO for all supported + * stream formats. + * Functions will be called xtfpga_pcm_tx_<channels>x<sample bits>, e.g. + * xtfpga_pcm_tx_2x16 for 16-bit stereo. + * + * FIFO consists of 32-bit words, one word per channel, always 2 channels. + * If I2S interface is configured with smaller sample resolution, only + * the LSB of each word is used. + */ +#define xtfpga_pcm_tx_fn(channels, sample_bits) \ +static unsigned xtfpga_pcm_tx_##channels##x##sample_bits( \ + struct xtfpga_i2s *i2s, struct snd_pcm_runtime *runtime, \ + unsigned tx_ptr) \ +{ \ + const u##sample_bits (*p)[channels] = \ + (void *)runtime->dma_area; \ +\ + for (; i2s->tx_fifo_level < i2s->tx_fifo_high; \ + i2s->tx_fifo_level += 2) { \ + iowrite32(p[tx_ptr][0], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + iowrite32(p[tx_ptr][channels - 1], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + if (++tx_ptr >= runtime->buffer_size) \ + tx_ptr = 0; \ + } \ + return tx_ptr; \ +} + +xtfpga_pcm_tx_fn(1, 16) +xtfpga_pcm_tx_fn(2, 16) +xtfpga_pcm_tx_fn(1, 32) +xtfpga_pcm_tx_fn(2, 32) + +#undef xtfpga_pcm_tx_fn + +static bool xtfpga_pcm_push_tx(struct xtfpga_i2s *i2s) +{ + struct snd_pcm_substream *tx_substream; + bool tx_active; + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + tx_active = tx_substream && snd_pcm_running(tx_substream); + if (tx_active) { + unsigned tx_ptr = ACCESS_ONCE(i2s->tx_ptr); + unsigned new_tx_ptr = i2s->tx_fn(i2s, tx_substream->runtime, + tx_ptr); + + cmpxchg(&i2s->tx_ptr, tx_ptr, new_tx_ptr); + } + rcu_read_unlock(); + + return tx_active; +} + +static void xtfpga_pcm_refill_fifo(struct xtfpga_i2s *i2s) +{ + unsigned int_status; + unsigned i; + + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + for (i = 0; i < 2; ++i) { + bool tx_active = xtfpga_pcm_push_tx(i2s); + + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + if (tx_active) + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + if (!tx_active || + !(int_status & XTFPGA_I2S_INT_LEVEL)) + break; + + /* After the push the level IRQ is still asserted, + * means FIFO level is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + if (!(int_status & XTFPGA_I2S_INT_LEVEL)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_VALID); + else if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_UNDERRUN); + + if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE); + else + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); +} + +static irqreturn_t xtfpga_i2s_threaded_irq_handler(int irq, void *dev_id) +{ + struct xtfpga_i2s *i2s = dev_id; + struct snd_pcm_substream *tx_substream; + unsigned config, int_status, int_mask; + + regmap_read(i2s->regmap, XTFPGA_I2S_CONFIG, &config); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_MASK, &int_mask); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, &int_status); + + if (!(config & XTFPGA_I2S_CONFIG_INT_ENABLE) || + !(int_status & int_mask & XTFPGA_I2S_INT_VALID)) + return IRQ_NONE; + + /* Update FIFO level estimate in accordance with interrupt status + * register. + */ + if (int_status & XTFPGA_I2S_INT_UNDERRUN) { + i2s->tx_fifo_level = 0; + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); + } else { + /* The FIFO isn't empty, but is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + + if (tx_substream && snd_pcm_running(tx_substream)) { + snd_pcm_period_elapsed(tx_substream); + if (int_status & XTFPGA_I2S_INT_UNDERRUN) + dev_dbg_ratelimited(i2s->dev, "%s: underrun\n", + __func__); + } + rcu_read_unlock(); + + /* Refill FIFO, update allowed IRQ reasons, enable IRQ if FIFO is + * not empty. + */ + xtfpga_pcm_refill_fifo(i2s); + + return IRQ_HANDLED; +} + +static int xtfpga_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, i2s); + return 0; +} + +static int xtfpga_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned srate = params_rate(params); + unsigned channels = params_channels(params); + unsigned period_size = params_period_size(params); + unsigned sample_size = snd_pcm_format_width(params_format(params)); + unsigned freq, ratio, level; + int err; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RES_MASK, + sample_size << XTFPGA_I2S_CONFIG_RES_BASE); + + freq = 256 * srate; + err = clk_set_rate(i2s->clk, freq); + if (err < 0) + return err; + + /* ratio field of the config register controls MCLK->I2S clock + * derivation: I2S clock = MCLK / (2 * (ratio + 2)). + * + * So with MCLK = 256 * sample rate ratio is 0 for 32 bit stereo + * and 2 for 16 bit stereo. + */ + ratio = (freq - (srate * sample_size * 8)) / + (srate * sample_size * 4); + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RATIO_MASK, + ratio << XTFPGA_I2S_CONFIG_RATIO_BASE); + + i2s->tx_fifo_low = XTFPGA_I2S_FIFO_SIZE / 2; + + /* period_size * 2: FIFO always gets 2 samples per frame */ + for (level = 1; + i2s->tx_fifo_low / 2 >= period_size * 2 && + level < (XTFPGA_I2S_CONFIG_LEVEL_MASK >> + XTFPGA_I2S_CONFIG_LEVEL_BASE); ++level) + i2s->tx_fifo_low /= 2; + + i2s->tx_fifo_high = 2 * i2s->tx_fifo_low; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_LEVEL_MASK, + level << XTFPGA_I2S_CONFIG_LEVEL_BASE); + + dev_dbg(i2s->dev, + "%s srate: %u, channels: %u, sample_size: %u, period_size: %u\n", + __func__, srate, channels, sample_size, period_size); + dev_dbg(i2s->dev, "%s freq: %u, ratio: %u, level: %u\n", + __func__, freq, ratio, level); + + return 0; +} + +static int xtfpga_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) + return -EINVAL; + + return 0; +} + +/* PCM */ + +static const struct snd_pcm_hardware xtfpga_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .period_bytes_min = 2, + .period_bytes_max = XTFPGA_I2S_FIFO_SIZE / 2 * 8, + .periods_min = 2, + .periods_max = XTFPGA_I2S_FIFO_SIZE * 8 / 2, + .buffer_bytes_max = XTFPGA_I2S_FIFO_SIZE * 8, + .fifo_size = 16, +}; + +static int xtfpga_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + void *p; + + snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware); + p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + runtime->private_data = p; + + return 0; +} + +static int xtfpga_pcm_close(struct snd_pcm_substream *substream) +{ + synchronize_rcu(); + return 0; +} + +static int xtfpga_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int ret; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + unsigned channels = params_channels(hw_params); + + switch (channels) { + case 1: + case 2: + break; + + default: + return -EINVAL; + + } + + switch (params_format(hw_params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x16 : + xtfpga_pcm_tx_2x16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x32 : + xtfpga_pcm_tx_2x32; + break; + + default: + return -EINVAL; + } + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + return ret; +} + +static int xtfpga_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ACCESS_ONCE(i2s->tx_ptr) = 0; + rcu_assign_pointer(i2s->tx_substream, substream); + xtfpga_pcm_refill_fifo(i2s); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + rcu_assign_pointer(i2s->tx_substream, NULL); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static snd_pcm_uframes_t xtfpga_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + snd_pcm_uframes_t pos = ACCESS_ONCE(i2s->tx_ptr); + + return pos < runtime->buffer_size ? pos : 0; +} + +static int xtfpga_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = xtfpga_pcm_hardware.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, size, size); +} + +static void xtfpga_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static const struct snd_pcm_ops xtfpga_pcm_ops = { + .open = xtfpga_pcm_open, + .close = xtfpga_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = xtfpga_pcm_hw_params, + .trigger = xtfpga_pcm_trigger, + .pointer = xtfpga_pcm_pointer, +}; + +static const struct snd_soc_platform_driver xtfpga_soc_platform = { + .pcm_new = xtfpga_pcm_new, + .pcm_free = xtfpga_pcm_free, + .ops = &xtfpga_pcm_ops, +}; + +static const struct snd_soc_component_driver xtfpga_i2s_component = { + .name = DRV_NAME, +}; + +static const struct snd_soc_dai_ops xtfpga_i2s_dai_ops = { + .startup = xtfpga_i2s_startup, + .hw_params = xtfpga_i2s_hw_params, + .set_fmt = xtfpga_i2s_set_fmt, +}; + +static struct snd_soc_dai_driver xtfpga_i2s_dai[] = { + { + .name = "xtfpga-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &xtfpga_i2s_dai_ops, + }, +}; + +static int xtfpga_i2s_runtime_suspend(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk); + return 0; +} + +static int xtfpga_i2s_runtime_resume(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(i2s->clk); + if (ret) { + dev_err(dev, "clk_prepare_enable failed: %d\n", ret); + return ret; + } + return 0; +} + +static int xtfpga_i2s_probe(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s; + struct resource *mem; + int err, irq; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + err = -ENOMEM; + goto err; + } + platform_set_drvdata(pdev, i2s); + i2s->dev = &pdev->dev; + dev_dbg(&pdev->dev, "dev: %p, i2s: %p\n", &pdev->dev, i2s); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->regs)) { + err = PTR_ERR(i2s->regs); + goto err; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->regs, + &xtfpga_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + err = PTR_ERR(i2s->regmap); + goto err; + } + + i2s->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk)) { + dev_err(&pdev->dev, "couldn't get clock\n"); + err = PTR_ERR(i2s->clk); + goto err; + } + + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, + (0x1 << XTFPGA_I2S_CONFIG_CHANNEL_BASE)); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, XTFPGA_I2S_INT_VALID); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, XTFPGA_I2S_INT_UNDERRUN); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "No IRQ resource\n"); + err = irq; + goto err; + } + err = devm_request_threaded_irq(&pdev->dev, irq, NULL, + xtfpga_i2s_threaded_irq_handler, + IRQF_SHARED | IRQF_ONESHOT, + pdev->name, i2s); + if (err < 0) { + dev_err(&pdev->dev, "request_irq failed\n"); + goto err; + } + + err = snd_soc_register_platform(&pdev->dev, &xtfpga_soc_platform); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register platform\n"); + goto err; + } + err = devm_snd_soc_register_component(&pdev->dev, + &xtfpga_i2s_component, + xtfpga_i2s_dai, + ARRAY_SIZE(xtfpga_i2s_dai)); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register component\n"); + goto err_unregister_platform; + } + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + err = xtfpga_i2s_runtime_resume(&pdev->dev); + if (err) + goto err_pm_disable; + } + return 0; + +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_unregister_platform: + snd_soc_unregister_platform(&pdev->dev); +err: + dev_err(&pdev->dev, "%s: err = %d\n", __func__, err); + return err; +} + +static int xtfpga_i2s_remove(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + if (i2s->regmap && !IS_ERR(i2s->regmap)) { + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + } + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + xtfpga_i2s_runtime_suspend(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id xtfpga_i2s_of_match[] = { + { .compatible = "cdns,xtfpga-i2s", }, + {}, +}; +MODULE_DEVICE_TABLE(of, xtfpga_i2s_of_match); +#endif + +static const struct dev_pm_ops xtfpga_i2s_pm_ops = { + SET_RUNTIME_PM_OPS(xtfpga_i2s_runtime_suspend, + xtfpga_i2s_runtime_resume, NULL) +}; + +static struct platform_driver xtfpga_i2s_driver = { + .probe = xtfpga_i2s_probe, + .remove = xtfpga_i2s_remove, + .driver = { + .name = "xtfpga-i2s", + .of_match_table = of_match_ptr(xtfpga_i2s_of_match), + .pm = &xtfpga_i2s_pm_ops, + }, +}; + +module_platform_driver(xtfpga_i2s_driver); + +MODULE_AUTHOR("Max Filippov <jcmvbkbc(a)gmail.com>"); +MODULE_DESCRIPTION("xtfpga I2S controller driver"); +MODULE_LICENSE("GPL v2"); -- 1.8.1.4

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[alsa-devel] [PATCH v3] ASoC: rt5670: set platform data by dmi
by Bard Liao 22 Dec '14

22 Dec '14

This patch set specific data according to dmi data. Signed-off-by: Jin, Yao <yao.jin(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * Remove the __initconst in definition since the data of dmi_platform_intel_braswell[] could not be accessible by dmi_check_system(). --- sound/soc/codecs/rt5670.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 8bf3a56..d6895aa 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -18,6 +18,7 @@ #include <linux/platform_device.h> #include <linux/acpi.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -2549,6 +2550,17 @@ static struct acpi_device_id rt5670_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); #endif +static const struct dmi_system_id dmi_platform_intel_braswell[] = { + { + .ident = "Intel Braswell", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Braswell CRB"), + }, + }, + {} +}; + static int rt5670_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2568,6 +2580,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (pdata) rt5670->pdata = *pdata; + if (dmi_check_system(dmi_platform_intel_braswell)) { + rt5670->pdata.dmic_en = true; + rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.jd_mode = 1; + } + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); if (IS_ERR(rt5670->regmap)) { ret = PTR_ERR(rt5670->regmap); -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 1/2] ASoC: dapm: Don't mark MICBIAS widgets as auto non-connected
by Lars-Peter Clausen 22 Dec '14

22 Dec '14

The connected flag of a widget only affects widgets that are either a source or a sink. The MICBIAS widget is a simple pass-through widget though and hence its behavior is the same regardless of whether the connected flag is set or not. Hence there is not much point in trying to automatically mark MICBIAS widgets as non-connected, so just remove it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5136bb..b8eba93 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3880,7 +3880,6 @@ void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card) switch (w->id) { case snd_soc_dapm_input: case snd_soc_dapm_output: - case snd_soc_dapm_micbias: dev_dbg(card->dev, "ASoC: Auto NC: Checking widget %s\n", w->name); if (!snd_soc_dapm_widget_in_card_paths(card, w)) { -- 1.8.0

2 2

[alsa-devel] [PATCH] ASoC: intel: Remove superfluous backslash in Kconfig
by Takashi Iwai 22 Dec '14

22 Dec '14

The line continuation in Kconfig should be a '\' instead of '\\'. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index e989ecf046c9..f06fcf1e21a5 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -46,7 +46,7 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C && \\ + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C && \ I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 @@ -76,7 +76,7 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \\ + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \ I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL select SND_COMPRESS_OFFLOAD -- 2.2.0

2 2

[alsa-devel] [PATCH] ASoC: Intel: Add I2C dependency to two new machines
by Jarkko Nikula 22 Dec '14

22 Dec '14

Fix following build error when CONFIG_I2C is not enabled: sound/soc/codecs/rt5640.c:2252:1: warning: data definition has no type or storage class module_i2c_driver(rt5640_i2c_driver); ^ sound/soc/codecs/rt5640.c:2252:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int] sound/soc/codecs/rt5640.c:2252:1: warning: parameter names (without types) in function declaration sound/soc/codecs/rt5640.c:2241:26: warning: ‘rt5640_i2c_driver’ defined but not used [-Wunused-variable] static struct i2c_driver rt5640_i2c_driver = { ^ cc1: some warnings being treated as errors Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Cc: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/soc/intel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index e989ecf046c9..f86de1211b96 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -89,7 +89,7 @@ config SND_SOC_INTEL_BROADWELL_MACH config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "ASoC Audio DSP Support for MID BYT Platform" - depends on X86 + depends on X86 && I2C select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -101,7 +101,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" - depends on X86_INTEL_LPSS + depends on X86_INTEL_LPSS && I2C select SND_SOC_RT5670 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI -- 2.1.3

3 2

[alsa-devel] [PATCH 00/28] remove .owner for most platform_drivers: the missing bits
by Wolfram Sang 22 Dec '14

22 Dec '14

Generated with coccinelle. The big cleanup was pulled in this merge window. This series catches the bits fallen through. The patches shall go in via the subsystem trees. If possible for 3.19 to increase consistency I'd say, but you decide, of course. cocci-file used: @match1@ declarer name module_platform_driver; declarer name module_platform_driver_probe; declarer name for_each_node_by_type; identifier __driver; @@ ( module_platform_driver(__driver); | module_platform_driver_probe(__driver, ...); ) @fix1 depends on match1@ identifier match1.__driver; @@ static struct platform_driver __driver = { .driver = { - .owner = THIS_MODULE, } }; @match2@ identifier __driver; @@ ( platform_driver_register(&__driver) | platform_driver_probe(&__driver, ...) | platform_create_bundle(&__driver, ...) ) @fix2 depends on match2@ identifier match2.__driver; @@ static struct platform_driver __driver = { .driver = { - .owner = THIS_MODULE, } }; Thanks again to Julia Lawall for support. And hey, we fixed a coccinelle bug on the way :) Wolfram Sang (28): ARM: mach-exynos: drop owner assignment from platform_drivers mips: lantiq: xway: drop owner assignment from platform_drivers mips: pci: drop owner assignment from platform_drivers char: ipmi: drop owner assignment from platform_drivers cpufreq: drop owner assignment from platform_drivers dma: drop owner assignment from platform_drivers gpio: drop owner assignment from platform_drivers gpu: drm: rockchip: drop owner assignment from platform_drivers iommu: drop owner assignment from platform_drivers net: ethernet: stmicro: stmmac: drop owner assignment from platform_drivers net: wireless: ath: ath5k: drop owner assignment from platform_drivers of: drop owner assignment from platform_drivers pci: host: drop owner assignment from platform_drivers phy: drop owner assignment from platform_drivers pinctrl: intel: drop owner assignment from platform_drivers rtc: drop owner assignment from platform_drivers scsi: drop owner assignment from platform_drivers thermal: drop owner assignment from platform_drivers thermal: int340x_thermal: drop owner assignment from platform_drivers tty: serial: 8250: drop owner assignment from platform_drivers usb: gadget: udc: bdc: drop owner assignment from platform_drivers watchdog: drop owner assignment from platform_drivers ASoC: intel: drop owner assignment from platform_drivers ASoC: intel: sst: drop owner assignment from platform_drivers ASoC: omap: drop owner assignment from platform_drivers ASoC: pxa: drop owner assignment from platform_drivers ASoC: samsung: drop owner assignment from platform_drivers macintosh: drop owner assignment from platform_drivers arch/arm/mach-exynos/pmu.c | 1 - arch/mips/lantiq/xway/vmmc.c | 1 - arch/mips/pci/pci-ar2315.c | 1 - arch/mips/pci/pci-rt2880.c | 1 - drivers/char/ipmi/ipmi_powernv.c | 1 - drivers/cpufreq/ls1x-cpufreq.c | 1 - drivers/dma/at_xdmac.c | 1 - drivers/gpio/gpio-vf610.c | 1 - drivers/gpu/drm/rockchip/rockchip_drm_drv.c | 1 - drivers/iommu/rockchip-iommu.c | 1 - drivers/macintosh/windfarm_pm112.c | 1 - drivers/macintosh/windfarm_pm72.c | 1 - drivers/macintosh/windfarm_rm31.c | 1 - drivers/net/ethernet/stmicro/stmmac/stmmac_platform.c | 1 - drivers/net/wireless/ath/ath5k/ahb.c | 1 - drivers/of/unittest.c | 1 - drivers/pci/host/pci-layerscape.c | 1 - drivers/phy/phy-armada375-usb2.c | 1 - drivers/phy/phy-berlin-usb.c | 1 - drivers/phy/phy-miphy28lp.c | 1 - drivers/pinctrl/intel/pinctrl-cherryview.c | 1 - drivers/rtc/rtc-opal.c | 1 - drivers/scsi/atari_scsi.c | 1 - drivers/scsi/mac_scsi.c | 1 - drivers/scsi/sun3_scsi.c | 1 - drivers/thermal/int340x_thermal/int3400_thermal.c | 1 - drivers/thermal/int340x_thermal/int3402_thermal.c | 1 - drivers/thermal/rockchip_thermal.c | 1 - drivers/tty/serial/8250/8250_omap.c | 1 - drivers/usb/gadget/udc/bdc/bdc_core.c | 1 - drivers/watchdog/cadence_wdt.c | 1 - drivers/watchdog/meson_wdt.c | 1 - sound/soc/intel/bytcr_dpcm_rt5640.c | 1 - sound/soc/intel/cht_bsw_rt5672.c | 1 - sound/soc/intel/sst/sst_acpi.c | 1 - sound/soc/omap/omap-hdmi-audio.c | 1 - sound/soc/pxa/spitz.c | 1 - sound/soc/samsung/arndale_rt5631.c | 1 - 38 files changed, 38 deletions(-) -- 2.1.3

2 10

[alsa-devel] [PATCH] ASoC: Intel: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 22 Dec '14

22 Dec '14

sound/soc/intel/cht_bsw_rt5672.c:273:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci CC: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- cht_bsw_rt5672.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -270,7 +270,6 @@ static int snd_cht_mc_probe(struct platf static struct platform_driver snd_cht_mc_driver = { .driver = { - .owner = THIS_MODULE, .name = "cht-bsw-rt5672", .pm = &snd_soc_pm_ops, },

2 1

[alsa-devel] [PATCH] ALSA: atmel: fix building the ac97 driver for at91-multiplatform
by Alexandre Belloni 20 Dec '14

20 Dec '14

From: Arnd Bergmann <arnd(a)arndb.de> at91 will no longer export the mach/cpu.h and mach/hardware.h header files in the future, which would break building the atmel ac97c driver. Since the cpu_is_* check is only used to find out whether we are running on avr32 or arm/at91, we can hardcode that check in the ARM case. Unfortunately the driver is missing other work to be useful again on ARM, if anyone wants to actually use it, they need to add a DT binding, and the driver should really be converted to use the ASoC framework. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Link: http://www.spinics.net/lists/arm-kernel/msg382068.html --- sound/atmel/ac97c.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b59427d5a697..83975f8d5947 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -34,10 +34,10 @@ #include <linux/platform_data/dma-dw.h> #include <linux/dma/dw.h> +#ifdef CONFIG_AVR32 #include <mach/cpu.h> - -#ifdef CONFIG_ARCH_AT91 -#include <mach/hardware.h> +#else +#define cpu_is_at32ap7000() (0) #endif #include "ac97c.h" -- 2.1.0

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[alsa-devel] [PATCH v2] ALSA: sound/atmel/ac97c.c: Add device tree support
by Alexander Stein 19 Dec '14

19 Dec '14

This adds device tree support for the AC97 controller. It uses the soc-ac97link bindings, but actually only ac97-reset is used. Signed-off-by: Alexander Stein <alexanders83(a)web.de> --- Changes in v2: * It now uses the soc-ac97link bindings (partly) instead of defining its own. This should ease the transition to ASoC once a new driver has been written. sound/atmel/ac97c.c | 45 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 42 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index cb44c74..73b61b2 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -22,6 +22,9 @@ #include <linux/gpio.h> #include <linux/types.h> #include <linux/io.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/initval.h> @@ -902,6 +905,40 @@ static void atmel_ac97c_reset(struct atmel_ac97c *chip) } } +#ifdef CONFIG_OF +static const struct of_device_id atmel_ac97c_dt_ids[] = { + { .compatible = "atmel,atmel_ac97c", }, + { } +}; +MODULE_DEVICE_TABLE(of, atmel_ac97c_dt_ids); + +static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) +{ + struct ac97c_platform_data *pdata; + struct device_node *node = dev->of_node; + const struct of_device_id *match; + + if (!node) { + dev_err(dev, "Device does not have associated DT data\n"); + return ERR_PTR(-EINVAL); + } + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return ERR_PTR(-ENOMEM); + + pdata->reset_pin = of_get_named_gpio(dev->of_node, "ac97-gpios", 2); + + return pdata; +} +#else +static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) +{ + dev_err(dev, "no platform data defined\n"); + return ERR_PTR(-ENXIO); +} +#endif + static int atmel_ac97c_probe(struct platform_device *pdev) { struct snd_card *card; @@ -922,10 +959,11 @@ static int atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } - pdata = pdev->dev.platform_data; + pdata = dev_get_platdata(&pdev->dev); if (!pdata) { - dev_dbg(&pdev->dev, "no platform data\n"); - return -ENXIO; + pdata = atmel_ac97c_probe_dt(&pdev->dev); + if (IS_ERR(pdata)) + return PTR_ERR(pdata); } irq = platform_get_irq(pdev, 0); @@ -1205,6 +1243,7 @@ static struct platform_driver atmel_ac97c_driver = { .name = "atmel_ac97c", .owner = THIS_MODULE, .pm = ATMEL_AC97C_PM_OPS, + .of_match_table = of_match_ptr(atmel_ac97c_dt_ids), }, }; module_platform_driver(atmel_ac97c_driver); -- 2.2.0

3 8

[alsa-devel] iMX6 ESAI TDM with underrun and overrun support
by Aurelien Bouin 19 Dec '14

19 Dec '14

Hello, I am currently working on an iMX6 board, I work with the master branch of git://git.kernel.org/pub/scm/linux/kernel/git/stable/linux-stable.git I have seen some problems in the freescale ESAI implementation : 1.The registers TCCR and RCCR have TFP and RFP bits in register of a width of 5, not 4 ! With a value of 5 we are able to use the full range of slots (32) Patch : http://pastebin.com/Ai1RZ9mE 2.The channels number are limited in Playback to 12, but actually we can rise 128, in Capture it is limited to 8 but actually we can rise 128 also Patch : http://pastebin.com/SHk5eHPa I use the iMX6 ESAI bus to send audio datas to other processors ... but actually I don't have any codec behind ... I am wondering if there is any solution with the devicetree to implement such configuration with "simple-audio-card" Because for the moment I have not find another solution than doing a module : http://pastebin.com/cnGwnEjn I also see that the chip errata of the iMX6 ERR008000 is not handle ...Which make channel swapping on my ESAI bus ... Is there someone that succeed in implementing the underrun overrun catch ? Thank you for your return, Regards Aurelien BOUIN

3 9

[alsa-devel] [PATCH/RFC 00/14] ASoC: samsung: Add clk provider for I2S internal clocks
by Sylwester Nawrocki 19 Dec '14

19 Dec '14

This series is an attempt to resolve the CDCLK clock gating issue on Odroid X2/U3 reported by Daniel Drake [1], by exposing the CDCLK gate clock through clk API. The remaining clocks (mux and divider) are also exposed, so the clk API could be already used instead of the set_sysclk() calls in the machine drivers. I need some more thought about interaction between the clk API calls on the clocks being exposed and the ASoC calls into sound/soc/samsung/i2s.c. I'm sending teh patches for review though to avoid any waste of time should it turn out the direction taken is wrong. This whole series definitely needs more testing, so far I only tested it on Odroid X2, with the I2S working in slave mode. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081753.… Sylwester Nawrocki (14): ASoC: samsung: i2s: Remove unused gpios field from struct i2s ASoC: samsung: i2s: samsung_i2s_get_driver_data() cleanup ASoC: samsung: i2s: Add return value checks in probe() ASoC: samsung: i2s: Request memory region in driver probe() ASoC: samsung: i2s: Move clk_get() to platform driver probe() ASoC: samsung: i2s: Move clk enable to the platform driver probe() ASoC: samsung: i2s: Add get_other_dai helper function ASoC: samsung: i2s: Remove an unneeded goto usage ASoC: samsung: i2s: Add spinlock in place of local_irq_* calls ASoC: samsung: i2s: Protect access to more registers with a spinlock ASoC: samsung: odroidx2: Handle I2S CDCLK clock conditionally ASoC: samsung: i2s: Add clock provider for the I2S internal clocks ARM: dts: Exynos4 and Odroid X2/U3 sound device nodes update ARM: dts: Switch Odroid X2/U2 to simple-audio-card .../devicetree/bindings/sound/samsung-i2s.txt | 18 +- arch/arm/boot/dts/exynos4.dtsi | 9 + arch/arm/boot/dts/exynos4412-odroid-common.dtsi | 26 +- arch/arm/boot/dts/exynos4412-odroidu3.dts | 8 +- arch/arm/boot/dts/exynos4412-odroidx2.dts | 8 +- sound/soc/samsung/i2s.c | 365 ++++++++++++-------- sound/soc/samsung/odroidx2_max98090.c | 6 +- 7 files changed, 282 insertions(+), 158 deletions(-) -- 1.7.9.5

3 22

[alsa-devel] [PATCH v2 1/3] snd_pcm_start_at: Add enum for timestamp class, and helper struct for start_at ioctl.
by Tim Cussins 19 Dec '14

19 Dec '14

As per Pierre's suggestion, timestamp types are of two types: System, and Audio. Signed-off-by: Tim Cussins <timcussins(a)eml.cc> diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1f23cd6..bd1e9be 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -466,12 +466,30 @@ struct snd_xfern { }; enum { + SNDRV_PCM_TSTAMP_CLASS_SYSTEM = 0, + SNDRV_PCM_TSTAMP_CLASS_AUDIO, + SNDRV_PCM_TSTAMP_CLASS_LAST = SNDRV_PCM_TSTAMP_CLASS_AUDIO, +}; + +enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, }; +enum { + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 0, + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK, + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK, +}; + +struct snd_start_at { + int tstamp_class; + int tstamp_type; + struct timespec start_time; +}; + /* channel positions */ enum { SNDRV_CHMAP_UNKNOWN = 0, @@ -550,6 +568,8 @@ enum { #define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) #define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) #define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) +#define SNDRV_PCM_IOCTL_START_AT _IOW('A', 0x62, struct snd_start_at) + /***************************************************************************** * * -- 2.1.0

2 4

[alsa-devel] [PATCH] ASoC: dapm: Remove snd_soc_of_parse_audio_routing() due to deferred probe
by Mark Brown 19 Dec '14

19 Dec '14

This reverts commit f8781db8aeb18d (ASoC: dapm: Augment existing card DAPM routes in snd_soc_of_parse_audio_routing) since it is broken for deferred probing as it ends up storing data allocated with devm_ over multiple instantiations of the device. Reported-by: Russell King - ARM Linux <linux(a)arm.linux.org.uk> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 935721062c21..d7595465cfbc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3231,7 +3231,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname) { struct device_node *np = card->dev->of_node; - int num_routes, old_routes; + int num_routes; struct snd_soc_dapm_route *routes; int i, ret; @@ -3249,9 +3249,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - old_routes = card->num_dapm_routes; - routes = devm_kzalloc(card->dev, - (old_routes + num_routes) * sizeof(*routes), + routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes), GFP_KERNEL); if (!routes) { dev_err(card->dev, @@ -3259,11 +3257,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - memcpy(routes, card->dapm_routes, old_routes * sizeof(*routes)); - for (i = 0; i < num_routes; i++) { ret = of_property_read_string_index(np, propname, - 2 * i, &routes[old_routes + i].sink); + 2 * i, &routes[i].sink); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -3271,7 +3267,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } ret = of_property_read_string_index(np, propname, - (2 * i) + 1, &routes[old_routes + i].source); + (2 * i) + 1, &routes[i].source); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -3280,7 +3276,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes += num_routes; + card->num_dapm_routes = num_routes; card->dapm_routes = routes; return 0; -- 2.1.3

2 3

[alsa-devel] [PATCH 1/2] ALSA: hda_controller: Separate stream_tag for input and output streams.
by libin.yang＠intel.com 19 Dec '14

19 Dec '14

From: Rafal Redzimski <rafal.f.redzimski(a)intel.com> Implemented separate stream_tag assignment for input and output streams. According to hda specification stream tag must be unique throughout the input streams group, however an output stream might use a stream tag which is already in use by an input stream. This change is necessary to support HW which provides a total of more than 15 stream DMA engines which with legacy implementation causes an overflow on SDxCTL.STRM field (and the whole SDxCTL register) and as a result usage of Reserved value 0 in the SDxCTL.STRM field which confuses HDA controller. Signed-off-by: Rafal Redzimski <rafal.f.redzimski(a)intel.com> Signed-off-by: Jayachandran B <jayachandran.b(a)intel.com> Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/hda_controller.c | 24 ++++++++++++++++++++++-- sound/pci/hda/hda_priv.h | 1 + 2 files changed, 23 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 8337645..dcca1ef 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1922,10 +1922,18 @@ int azx_mixer_create(struct azx *chip) EXPORT_SYMBOL_GPL(azx_mixer_create); +static bool is_input_stream(struct azx *chip, unsigned char index) +{ + return (index >= chip->capture_index_offset && + index < chip->capture_index_offset + chip->capture_streams); +} + /* initialize SD streams */ int azx_init_stream(struct azx *chip) { int i; + int in_stream_tag = 0; + int out_stream_tag = 0; /* initialize each stream (aka device) * assign the starting bdl address to each stream (device) @@ -1938,9 +1946,21 @@ int azx_init_stream(struct azx *chip) azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */ azx_dev->sd_int_sta_mask = 1 << i; - /* stream tag: must be non-zero and unique */ azx_dev->index = i; - azx_dev->stream_tag = i + 1; + + /* stream tag must be unique throughout + * the stream direction group, + * valid values 1...15 + * use separate stream tag if the flag + * AZX_DCAPS_SEPARATE_STREAM_TAG is used + */ + if (chip->driver_caps & AZX_DCAPS_SEPARATE_STREAM_TAG) + azx_dev->stream_tag = + is_input_stream(chip, i) ? + ++in_stream_tag : + ++out_stream_tag; + else + azx_dev->stream_tag = i + 1; } return 0; diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index aa484fd..166e3e8 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -171,6 +171,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ +#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ enum { AZX_SNOOP_TYPE_NONE , -- 1.9.1

1 1

[alsa-devel] [PATCH] ALSA: usb-audio: extend KEF X300A FU 10 tweak to Arcam rPAC
by Jiri Jaburek 18 Dec '14

18 Dec '14

The Arcam rPAC seems to have the same problem - whenever anything (alsamixer, udevd, 3.9+ kernel from 60af3d037eb8c, ..) attempts to access mixer / control interface of the card, the firmware "locks up" the entire device, resulting in SNDRV_PCM_IOCTL_HW_PARAMS failed (-5): Input/output error from alsa-lib. Other operating systems can somehow read the mixer (there seems to be playback volume/mute), but any manipulation is ignored by the device (which has hardware volume controls). Cc: <stable(a)vger.kernel.org> Signed-off-by: Jiri Jaburek <jjaburek(a)redhat.com> --- sound/usb/mixer_maps.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 1994d41..b703cb3 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -333,8 +333,11 @@ static struct usbmix_name_map gamecom780_map[] = { {} }; -static const struct usbmix_name_map kef_x300a_map[] = { - { 10, NULL }, /* firmware locks up (?) when we try to access this FU */ +/* some (all?) SCMS USB3318 devices are affected by a firmware lock up + * when anything attempts to access FU 10 (control) + */ +static const struct usbmix_name_map scms_usb3318_map[] = { + { 10, NULL }, { 0 } }; @@ -434,8 +437,14 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = ebox44_map, }, { + /* KEF X300A */ .id = USB_ID(0x27ac, 0x1000), - .map = kef_x300a_map, + .map = scms_usb3318_map, + }, + { + /* Arcam rPAC */ + .id = USB_ID(0x25c4, 0x0003), + .map = scms_usb3318_map, }, { 0 } /* terminator */ }; -- 2.1.0

2 1

[alsa-devel] [PATCH v2] ASoC: fsl_esai: Fix ESAI_xCCR_xDC_WIDTH definition
by Aurelien Bouin 18 Dec '14

18 Dec '14

>From 9bcae891b8fe7a42cbc2cd88319f7bd4ab44d012 Mon Sep 17 00:00:00 2001 From: Aurelien BOUIN <a_bouin(a)yahoo.fr> Date: Thu, 18 Dec 2014 15:14:53 +0100 Subject: [PATCH 1/1] According to the mx6 reference manual the TDC and RDC field of registers ESAI_TCCR and ESAI_RCCR have a width of 5 bits, not 4 Signed-off-by: Aurelien BOUIN <a_bouin(a)yahoo.fr> diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 91a550f..5e793bb 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -302,7 +302,7 @@ #define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) #define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) #define ESAI_xCCR_xDC_SHIFT 9 -#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_WIDTH 5 #define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) #define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) #define ESAI_xCCR_xPSR_SHIFT 8 -- 1.9.1

2 1

[alsa-devel] [PATCH] sound/soc/adi/axi-spdif.c: Support programmable master clock
by Mike Looijmans 18 Dec '14

18 Dec '14

If the master clock supports programmable rates, program it to generate the desired frequency. Only apply constraints when the clock is fixed. This allows proper clock generation for both 44100 and 48000 Hz based sampling rates if the platform supports it. The clock frequency must be set before enabling it. Enabling the clock was done in "startup", but that occurs before "hw_params" where the rate is known. Move the clock start to the hw_params routine, and keep track of whether the clock has been started, because shutdown may be called without having called hw_params first. Starting the clock and enabling the SPDIF output AFTER programming the dividers is a more logical order anyway. Signed-off-by: Mike Looijmans <mike.looijmans(a)topic.nl> --- sound/soc/adi/axi-spdif.c | 56 ++++++++++++++++++++++++++++----------------- 1 file changed, 35 insertions(+), 21 deletions(-) diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c index 198e3a4..d67e010 100644 --- a/sound/soc/adi/axi-spdif.c +++ b/sound/soc/adi/axi-spdif.c @@ -4,7 +4,6 @@ * * Licensed under the GPL-2. */ - #include <linux/init.h> #include <linux/kernel.h> #include <linux/module.h> @@ -44,6 +43,8 @@ struct axi_spdif { struct snd_ratnum ratnum; struct snd_pcm_hw_constraint_ratnums rate_constraints; + + bool clk_ref_running; }; static int axi_spdif_trigger(struct snd_pcm_substream *substream, int cmd, @@ -79,6 +80,7 @@ static int axi_spdif_hw_params(struct snd_pcm_substream *substream, struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); unsigned int rate = params_rate(params); unsigned int clkdiv, stat; + int ret; switch (params_rate(params)) { case 32000: @@ -95,6 +97,9 @@ static int axi_spdif_hw_params(struct snd_pcm_substream *substream, break; } + /* Try to set the master clock */ + clk_set_rate(spdif->clk_ref, rate * 128); + clkdiv = DIV_ROUND_CLOSEST(clk_get_rate(spdif->clk_ref), rate * 64 * 2) - 1; clkdiv <<= AXI_SPDIF_CTRL_CLKDIV_OFFSET; @@ -103,6 +108,14 @@ static int axi_spdif_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, AXI_SPDIF_CTRL_CLKDIV_MASK, clkdiv); + ret = clk_prepare_enable(spdif->clk_ref); + if (ret) + return ret; + spdif->clk_ref_running = true; + + regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, + AXI_SPDIF_CTRL_TXEN, AXI_SPDIF_CTRL_TXEN); + return 0; } @@ -121,18 +134,13 @@ static int axi_spdif_startup(struct snd_pcm_substream *substream, struct axi_spdif *spdif = snd_soc_dai_get_drvdata(dai); int ret; - ret = snd_pcm_hw_constraint_ratnums(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &spdif->rate_constraints); - if (ret) - return ret; - - ret = clk_prepare_enable(spdif->clk_ref); - if (ret) - return ret; - - regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, - AXI_SPDIF_CTRL_TXEN, AXI_SPDIF_CTRL_TXEN); + if (spdif->rate_constraints.nrats) { + ret = snd_pcm_hw_constraint_ratnums(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &spdif->rate_constraints); + if (ret) + return ret; + } return 0; } @@ -145,7 +153,10 @@ static void axi_spdif_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(spdif->regmap, AXI_SPDIF_REG_CTRL, AXI_SPDIF_CTRL_TXEN, 0); - clk_disable_unprepare(spdif->clk_ref); + if (spdif->clk_ref_running) { + clk_disable_unprepare(spdif->clk_ref); + spdif->clk_ref_running = false; + } } static const struct snd_soc_dai_ops axi_spdif_dai_ops = { @@ -216,14 +227,17 @@ static int axi_spdif_probe(struct platform_device *pdev) spdif->dma_data.addr_width = 4; spdif->dma_data.maxburst = 1; - spdif->ratnum.num = clk_get_rate(spdif->clk_ref) / 128; - spdif->ratnum.den_step = 1; - spdif->ratnum.den_min = 1; - spdif->ratnum.den_max = 64; - - spdif->rate_constraints.rats = &spdif->ratnum; - spdif->rate_constraints.nrats = 1; + /* Determine if the clock rate is fixed. If it cannot change frequency, + * it returns an error here. */ + if (clk_round_rate(spdif->clk_ref, 128 * 44100) < 0) { + spdif->ratnum.num = clk_get_rate(spdif->clk_ref) / 128; + spdif->ratnum.den_step = 1; + spdif->ratnum.den_min = 1; + spdif->ratnum.den_max = 64; + spdif->rate_constraints.rats = &spdif->ratnum; + spdif->rate_constraints.nrats = 1; + } ret = devm_snd_soc_register_component(&pdev->dev, &axi_spdif_component, &axi_spdif_dai, 1); if (ret) -- 1.7.9.5

2 8

[alsa-devel] [[PATCH] ALSA: In Freescale ESAI driver xDxDC bits in xCCR registers is 5 bits width not 4
by Aurelien BOUIN 18 Dec '14

18 Dec '14

Signed-off-by: Aurelien BOUIN <a_bouin(a)yahoo.fr> diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 91a550f..5e793bb 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -302,7 +302,7 @@ #define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) #define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) #define ESAI_xCCR_xDC_SHIFT 9 -#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_WIDTH 5 #define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) #define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) #define ESAI_xCCR_xPSR_SHIFT 8 -- 1.9.1

2 1

[alsa-devel] Device Tree simple-audio-card without codec HOWTO ?
by Aurelien Bouin 18 Dec '14

18 Dec '14

Hello, I use the iMX6 ESAI bus to send audio datas to other processors ... but actually I don't have any codec behind ... I am wondering if there is any solution with the devicetree to implement such configuration with "simple-audio-card" That will configure tdm slots and width. I tried without success that : sound { compatible = "simple-audio-card"; simple-audio-card,name = "Generic-ESAI-Sound-Card"; simple-audio-card,format = "left_j"; simple-audio-card,cpu { sound-dai = <&esai 0>; dai-tdm-slot-num = <32>; dai-tdm-slot-width = <8>; }; simple-audio-card,codec { sound-dai = <0 0>; }; }; Because for the moment I have not find another solution than doing a module : http://pastebin.com/cnGwnEjn Thank you for your help, Regards, Aurelien BOUIN

1 0

[alsa-devel] [PATCH] ALSA: pcm: Don't leave PREPARED state after draining
by Takashi Iwai 18 Dec '14

18 Dec '14

When a PCM draining is performed to an empty stream that has been already in PREPARED state, the current code just ignores and leaves as it is, although the drain is supposed to set all such streams to SETUP state. This patch covers that overlooked case. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 095d9572ad2b..64d9863d6565 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1546,6 +1546,8 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) if (! snd_pcm_playback_empty(substream)) { snd_pcm_do_start(substream, SNDRV_PCM_STATE_DRAINING); snd_pcm_post_start(substream, SNDRV_PCM_STATE_DRAINING); + } else { + runtime->status->state = SNDRV_PCM_STATE_SETUP; } break; case SNDRV_PCM_STATE_RUNNING: -- 2.2.0

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda_controller: Separate stream_tag for input and output streams.
by libin.yang＠intel.com 18 Dec '14

18 Dec '14

From: rfredzim <rafal.f.redzimski(a)intel.com> Implemented separate stream_tag assignment for input and output streams. According to hda specification stream tag must be unique throughout the input streams group, however an output stream might use a stream tag which is already in use by an input stream. This change is necessary to support HW which provides a total of more than 15 stream DMA engines which with legacy implementation causes an overflow on SDxCTL.STRM field (and the whole SDxCTL register) and as a result usage of Reserved value 0 in the SDxCTL.STRM field which confuses HDA controller. Signed-off-by: rfredzim <rafal.f.redzimski(a)intel.com> Signed-off-by: Jayachandran B <jayachandran.b(a)intel.com> Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/hda_controller.c | 24 ++++++++++++++++++++++-- sound/pci/hda/hda_priv.h | 1 + 2 files changed, 23 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 8337645..dcca1ef 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1922,10 +1922,18 @@ int azx_mixer_create(struct azx *chip) EXPORT_SYMBOL_GPL(azx_mixer_create); +static bool is_input_stream(struct azx *chip, unsigned char index) +{ + return (index >= chip->capture_index_offset && + index < chip->capture_index_offset + chip->capture_streams); +} + /* initialize SD streams */ int azx_init_stream(struct azx *chip) { int i; + int in_stream_tag = 0; + int out_stream_tag = 0; /* initialize each stream (aka device) * assign the starting bdl address to each stream (device) @@ -1938,9 +1946,21 @@ int azx_init_stream(struct azx *chip) azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */ azx_dev->sd_int_sta_mask = 1 << i; - /* stream tag: must be non-zero and unique */ azx_dev->index = i; - azx_dev->stream_tag = i + 1; + + /* stream tag must be unique throughout + * the stream direction group, + * valid values 1...15 + * use separate stream tag if the flag + * AZX_DCAPS_SEPARATE_STREAM_TAG is used + */ + if (chip->driver_caps & AZX_DCAPS_SEPARATE_STREAM_TAG) + azx_dev->stream_tag = + is_input_stream(chip, i) ? + ++in_stream_tag : + ++out_stream_tag; + else + azx_dev->stream_tag = i + 1; } return 0; diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index aa484fd..166e3e8 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -171,6 +171,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ +#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ enum { AZX_SNOOP_TYPE_NONE , -- 1.9.1

3 3

[alsa-devel] [PATCH 1/3] ALSA: asihpi: relax firmware version check
by eliot＠blennerhassett.gen.nz 18 Dec '14

18 Dec '14

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Some products firmware is no longer being updated e.g. dsp5000, dsp8700 but it should continue to work with updated HPI versions. Avoid regression by allowing this firmware to be loaded as long as major version is the same. Warn about mismatching versions, as matching versions are preferred. Signed-off-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> --- sound/pci/asihpi/hpidspcd.c | 26 ++++++++++++-------------- 1 file changed, 12 insertions(+), 14 deletions(-) diff --git a/sound/pci/asihpi/hpidspcd.c b/sound/pci/asihpi/hpidspcd.c index ac91637..3603c24 100644 --- a/sound/pci/asihpi/hpidspcd.c +++ b/sound/pci/asihpi/hpidspcd.c @@ -1,8 +1,9 @@ -/***********************************************************************/ -/** +/*********************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. <support(a)audioscience.com> + Functions for reading DSP code using hotplug firmware loader + + Copyright (C) 1997-2014 AudioScience Inc. <support(a)audioscience.com> This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -17,11 +18,7 @@ along with this program; if not, write to the Free Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -\file -Functions for reading DSP code using -hotplug firmware loader from individual dsp code files -*/ -/***********************************************************************/ +***********************************************************************/ #define SOURCEFILE_NAME "hpidspcd.c" #include "hpidspcd.h" #include "hpidebug.h" @@ -68,17 +65,18 @@ short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code, goto error2; } - if ((header.version >> 9) != (HPI_VER >> 9)) { - /* Consider even and subsequent odd minor versions to be compatible */ - dev_err(&dev->dev, "Incompatible firmware version DSP image %X != Driver %X\n", + if (HPI_VER_MAJOR(header.version) != HPI_VER_MAJOR(HPI_VER)) { + /* Major version change probably means Host-DSP protocol change */ + dev_err(&dev->dev, + "Incompatible firmware version DSP image %X != Driver %X\n", header.version, HPI_VER); goto error2; } if (header.version != HPI_VER) { - dev_info(&dev->dev, - "Firmware: release version mismatch DSP image %X != Driver %X\n", - header.version, HPI_VER); + dev_warn(&dev->dev, + "Firmware version mismatch: DSP image %X != Driver %X\n", + header.version, HPI_VER); } HPI_DEBUG_LOG(DEBUG, "dsp code %s opened\n", fw_name); -- 1.9.1

3 5

[alsa-devel] [PATCH] ALSA: usb-audio: Fix Scarlett 6i6 initialization typo
by Chris J Arges 18 Dec '14

18 Dec '14

The num_controls field was incorrectly set to 0 causing 6i6 to not be initialized. Set this to 9. Reported-and-tested-by: Mark Roberts <sunifiram(a)gmail.com> Signed-off-by: Chris J Arges <chris.j.arges(a)canonical.com> --- sound/usb/mixer_scarlett.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 9109652..7438e7c 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -655,7 +655,7 @@ static struct scarlett_device_info s6i6_info = { .names = NULL }, - .num_controls = 0, + .num_controls = 9, .controls = { { .num = 0, .type = SCARLETT_OUTPUTS, .name = "Monitor" }, { .num = 1, .type = SCARLETT_OUTPUTS, .name = "Headphone" }, -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Create capture source ctls when stereo mix input is added
by Takashi Iwai 18 Dec '14

18 Dec '14

When the stereo mix input is explicitly enabled via a user hint, the driver should create always a capture source enum ctl and disable the auto-mic switch. Otherwise the behavior gets confused. For doing it, this patch just sets spec->suppress_auto_mic flag appropriately. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 63b69f750d8e..293af6b5ba3d 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3224,6 +3224,8 @@ static int create_input_ctls(struct hda_codec *codec) "Stereo Mix", 0); if (err < 0) return err; + else + spec->suppress_auto_mic = 1; } return 0; -- 2.2.0

2 3

[alsa-devel] [PATCH] ALSA: hda - Add quirk for Packard Bell EasyNote MX65
by Takashi Iwai 17 Dec '14

17 Dec '14

Packard Bell EasyNote MX65 with AD1986A codec needs a few fixups, namely, the pin config overrides to set only the known I/O pins and the EAPD has to be turned on. In addition, add stereo mix input forcibly for avoiding the weird KDE behavior by this update. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=88251 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 39 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 39 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index eb54da757407..a9d78e275138 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -257,6 +257,18 @@ static void ad1986a_fixup_eapd(struct hda_codec *codec, } } +/* enable stereo-mix input for avoiding regression on KDE (bko#88251) */ +static void ad1986a_fixup_eapd_mix_in(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct ad198x_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + ad1986a_fixup_eapd(codec, fix, action); + spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_ENABLE; + } +} + enum { AD1986A_FIXUP_INV_JACK_DETECT, AD1986A_FIXUP_ULTRA, @@ -265,6 +277,8 @@ enum { AD1986A_FIXUP_LAPTOP, AD1986A_FIXUP_LAPTOP_IMIC, AD1986A_FIXUP_EAPD, + AD1986A_FIXUP_EAPD_MIX_IN, + AD1986A_FIXUP_EASYNOTE, }; static const struct hda_fixup ad1986a_fixups[] = { @@ -329,6 +343,30 @@ static const struct hda_fixup ad1986a_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = ad1986a_fixup_eapd, }, + [AD1986A_FIXUP_EAPD_MIX_IN] = { + .type = HDA_FIXUP_FUNC, + .v.func = ad1986a_fixup_eapd_mix_in, + }, + [AD1986A_FIXUP_EASYNOTE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x0421402f }, /* headphone */ + { 0x1b, 0x90170110 }, /* speaker */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x90a70130 }, /* int mic */ + { 0x1e, 0x411111f0 }, /* N/A */ + { 0x1f, 0x04a19040 }, /* mic */ + { 0x20, 0x411111f0 }, /* N/A */ + { 0x21, 0x411111f0 }, /* N/A */ + { 0x22, 0x411111f0 }, /* N/A */ + { 0x23, 0x411111f0 }, /* N/A */ + { 0x24, 0x411111f0 }, /* N/A */ + { 0x25, 0x411111f0 }, /* N/A */ + {} + }, + .chained = true, + .chain_id = AD1986A_FIXUP_EAPD_MIX_IN, + }, }; static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { @@ -342,6 +380,7 @@ static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP), SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA), + SND_PCI_QUIRK(0x1631, 0xc022, "PackardBell EasyNote MX65", AD1986A_FIXUP_EASYNOTE), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_FIXUP_3STACK), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_FIXUP_3STACK), -- 2.2.0

1 0

[alsa-devel] [PATCH] usb: add native DSD support for Matrix Audio DACs
by Jurgen Kramer 17 Dec '14

17 Dec '14

This patch adds native DSD support for two XMOS based DACs from Matrix Audio: - X-Sabre - Mini-i Pro Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4dbfb3d..a739841 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1245,8 +1245,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* XMOS based USB DACs */ switch (chip->usb_id) { - /* iFi Audio micro/nano iDSD */ - case USB_ID(0x20b1, 0x3008): + case USB_ID(0x20b1, 0x3008): /* iFi Audio micro/nano iDSD */ + case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */ + case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 1.9.3

3 2

[alsa-devel] Again: Alsa Driver Scarlett
by Orm Finnendahl 17 Dec '14

17 Dec '14

Hi, this is relating to my post from September: There doesn't seem to be any progress concerning the integration of the Scarlett alsa-driver patch into Alsa. Can someone on the list tell me how to invoke a process to get it integrated? The patch sources are here: https://github.com/smilingthax/alsa-driver_scarlett At the moment most of our students regularly have to reapply the patch as soon as they upgrade their systems and it's pretty time consuming and painful for the studio technicians and me to guide them through the process of patching kernel modules, so it would be great to simplify this process by integrating it into the official alsa distribution. I can confirm that the patch still works well on our systems. -- Orm ------------------------------------------------------------------ Prof. Orm Finnendahl Komposition Hochschule für Musik und Darstellende Kunst Eschersheimer Landstr. 29-39 60322 Frankfurt am Main ----- Forwarded message from Orm Finnendahl <orm.finnendahl(a)selma.hfmdk-frankfurt.de> ----- Subject: Re: [alsa-devel] Alsa Driver Scarlett From: Orm Finnendahl <orm.finnendahl(a)selma.hfmdk-frankfurt.de> To: alsa-devel(a)alsa-project.org Date: Tue, 16 Sep 2014 15:49:25 +0200 Am Dienstag, den 16. September 2014 um 15:22:27 Uhr (+0200) schrieb Johannes Kroll: > > I guess I'll need to find the time to port the patch to a recent kernel > when I upgrade, unless someone else does it first. The patch still works on our recent kernels (we are on debian testing with a 3.14-2 kernel), atm there is no porting necessary. I can also confirm that the driver has no known issues on our systems (we use a couple of Scarlett 18i20 in different configurations). It would be great, though if the patch could get integrated into alsa. At the moment it is quite painful for our students with linux laptops as they have to reapply the patch on every upgrade. -- Orm Finnendahl ------------------------------------------------------------------ Prof. Orm Finnendahl Komposition Hochschule für Musik und Darstellende Kunst Eschersheimer Landstr. 29-39 60322 Frankfurt am Main ----- End forwarded message -----

2 1

Re: [alsa-devel] New codec support for ALC256
by Takashi Iwai 17 Dec '14

17 Dec '14

At Wed, 17 Dec 2014 07:56:56 +0000, Kailang wrote: > > Attach the patch. Don't we need the addition to alc_fill_eapd_coef()? thanks, Takashi

2 2

Re: [alsa-devel] New Dell desktop support
by Takashi Iwai 17 Dec '14

17 Dec '14

At Wed, 17 Dec 2014 09:18:56 +0000, Kailang wrote: > > Hi Takashi, > > Attach is the new SSID for Dell desktop for ALC255. > I add it into fixup table first. > If it has more machine that verb table was the same, than let's move it to pin fixup table. OK, applied now with Cc to stable. thanks, Takashi

1 0

[alsa-devel] New codec support for ALC256
by Kailang 17 Dec '14

17 Dec '14

Hi Takashi, Attach is the patch for new codec support. Thanks. BR, Kailang

1 0

[alsa-devel] [GIT PULL] ASoC fixes for v3.19
by Mark Brown 16 Dec '14

16 Dec '14

The following changes since commit 1810afd3e1ded09c53d4e966dddce3c7d484521f: Merge remote-tracking branches 'asoc/topic/wm9090', 'asoc/topic/wm9712' and 'asoc/topic/wm9713' into asoc-next (2014-12-08 13:12:36 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.19-2 for you to fetch changes up to c52e5698efef1d9999d744be001be62d9fcabf55: Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/intel', 'asoc/fix/rt5645', 'asoc/fix/rt5677' and 'asoc/fix/samsung' into asoc-linus (2014-12-16 11:52:34 +0000) ---------------------------------------------------------------- ASoC: Merge window fixes for v3.19 A smattering of driver specific fixes here, some more cleanup of the Exynos7 and Intel ACPI support plus a fix for register cache sync on the pcm512x driver which has been lurking for a while. ---------------------------------------------------------------- Bard Liao (1): ASoC: rt5645: Fix potential crash in jd function Ben Zhang (1): ASoC: rt5677: add REGMAP_I2C and REGMAP_IRQ dependency Kevin Strasser (1): ASoC: Intel: fix possible acpi enumeration panic Mark Brown (1): Merge remote-tracking branches 'asoc/fix/atmel', 'asoc/fix/intel', 'asoc/fix/rt5645', 'asoc/fix/rt5677' and 'asoc/fix/samsung' into asoc-linus Matthieu Crapet (1): ASoC: atmel_ssc_dai/trivial: typo fix Peter Rosin (1): ASoC: pcm512x: Trigger auto-increment of register addresses on i2c Sylwester Nawrocki (1): ASoC: samsung: i2s: Add missing assignment of variant_regs Wei Yongjun (1): ASoC: Intel: fix return value check in sst_acpi_probe() sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/Kconfig | 2 ++ sound/soc/codecs/pcm512x-i2c.c | 7 ++++++- sound/soc/codecs/rt5645.c | 4 ++++ sound/soc/intel/sst/sst_acpi.c | 10 +++++----- sound/soc/samsung/i2s.c | 2 ++ 6 files changed, 20 insertions(+), 7 deletions(-)

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[alsa-devel] [PATCH] ASoC: dapm: Don't use async I/O
by Mark Brown 16 Dec '14

16 Dec '14

The only user of the async I/O support in ASoC is SPI which was using it to avoid needless context thrashing and minimise controller runtime PM bounces. The SPI framework has now been enhanced so that even normal spi_sync() calls won't suffer these effects so we don't need to handle this in ASoC and in fact it can be more efficient not to since we don't need to set up and tear down the buffers needed to manage asynchronous I/O. The async completions that DAPM does are left in place so drivers can use them, they are very cheap if there is no asynchronous work queued. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5136bb1f982..fe06b028130f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -517,8 +517,8 @@ static int soc_dapm_update_bits(struct snd_soc_dapm_context *dapm, { if (!dapm->component) return -EIO; - return snd_soc_component_update_bits_async(dapm->component, reg, - mask, value); + return snd_soc_component_update_bits(dapm->component, reg, + mask, value); } static int soc_dapm_test_bits(struct snd_soc_dapm_context *dapm, -- 2.1.3

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[alsa-devel] [PATCH RFC 0/9] audio timestamping evolutions
by Pierre-Louis Bossart 16 Dec '14

16 Dec '14

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new audio_timestamp definitions provided here. 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 5 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. TODO: for start_at, need to take audio tstamp snapshots even if stream is not running. Pierre-Louis Bossart (9): ALSA: core: don't override timestamp unconditionally ALSA: core: allow for trigger_tstamp snapshot in .trigger ALSA: hda: read trigger_timestamp immediately after starting DMA ALSA: usb: update trigger timestamp on first non-zero URB submitted ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: replace .wall_clock by .get_time_info ALSA: hda: replace .wallclock by .get_time_info Documentation/sound/alsa/timestamping.txt | 171 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 52 ++++++++- include/uapi/sound/asound.h | 20 +++- sound/core/pcm_compat.c | 7 +- sound/core/pcm_lib.c | 81 +++++++++----- sound/core/pcm_native.c | 22 +++- sound/pci/hda/hda_controller.c | 41 +++++-- sound/usb/pcm.c | 9 ++ 8 files changed, 355 insertions(+), 48 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 2.1.0

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[alsa-devel] [PATCH] ASoC: Intel: fix possible acpi enumeration panic
by Kevin Strasser 16 Dec '14

16 Dec '14

A crash can occur on some platforms where adsp is enumerated but codec is not matched. Check that the codec_id string is valid before attempting to match. Signed-off-by: Kevin Strasser <kevin.strasser(a)intel.com> --- sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 31124aa..dd72e58 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -236,7 +236,7 @@ static struct sst_machines *sst_acpi_find_machine( struct sst_machines *mach; bool found = false; - for (mach = machines; mach->codec_id; mach++) + for (mach = machines; mach->codec_id[0]; mach++) if (ACPI_SUCCESS(acpi_get_devices(mach->codec_id, sst_acpi_mach_match, &found, NULL)) && found) -- 1.9.1

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