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

Re: [alsa-devel] ALSA: pcm_direct.c SND_PCM_DIRECT_MAGIC
by Takashi Iwai 24 Sep '15

24 Sep '15

On Wed, 23 Sep 2015 18:07:46 +0200, Mike Bjorge wrote: > > I see, interesting. > > I am running into an issue where pcm_dmix.c is occasionally failing with an > "unable to create IPC shm instance" (pcm_dmix.c@1004) because in > pcm_direct.c@134, (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) when > buf.shm_nattch == 2. > > Do you know if there are any likely causes that would result in > dmix->shmptr->magic not being set correctly? What value is read there? If the value is completely different, it means that some other programs use the same shmem for other purpose. It'd be just a conflict of shmem key. Takashi > > Thanks for your help, > Mike > > > > On Wed, Sep 23, 2015 at 6:11 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > On Sat, 19 Sep 2015 01:13:42 +0200, > > Mike Bjorge wrote: > > > > > > Hi Takashi, > > > > > > I know this is a bit out of the blue, but I was wondering if you could > > > explain the point of SND_PCM_DIRECT_MAGIC and dmix->shmptr->magic? > > > > > > Specifically, what it's doing in pcm_direct.c from this commit? > > > > > http://git.alsa-project.org/?p=alsa-lib.git;a=blobdiff;f=src/pcm/pcm_direct… > > > > It's to check the consistency of shared data via shmem. > > If we change the data structure in an incompatible way, we'd change > > the magic key so that each instance can detect it. > > > > > > Takashi > > > [2 <text/html; UTF-8 (quoted-printable)>] >

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[alsa-devel] [GIT PULL] ASoC fixes for v4.3
by Mark Brown 24 Sep '15

24 Sep '15

The following changes since commit 1f93e4a96c9109378204c147b3eec0d0e8100fde: Linux 4.3-rc2 (2015-09-20 14:32:34 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.3-rc2 for you to fetch changes up to ed14ee0eea8b6808025356cecc87a8007885263f: Merge remote-tracking branches 'asoc/fix/wm8960' and 'asoc/fix/wm8962' into asoc-linus (2015-09-23 11:01:12 -0700) ---------------------------------------------------------------- ASoC: Fixes for v4.3 A disappointingly large set of fixes, though none of them very big and very widely spread over many different drivers. Nothing especially stands out, it's mostly all device specific and relatively minor. ---------------------------------------------------------------- Adam Thomson (1): ASoC: fsl_ssi: Fix checking of dai format for AC97 mode Anatol Pomozov (1): ASoC: Document snd-soc-dummy-dai purpose Arnaud Pouliquen (1): ASoC: sti: check return of of_property_read Axel Lin (2): ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM ASoC: au1x: psc-i2s: Fix unused variable 'ret' warning John Lin (1): ASoC: rt5645: Add struct dmi_system_id "Google Ultima" for chrome platform Jyri Sarha (1): ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver Koro Chen (1): ASoC: mediatek: Increase periods_min in capture Liam Girdwood (1): ASoC: intel: Fix SSP port configuration after RTD3 resume. Maciej S. Szmigiero (1): ASoC: fsl-asoc-card: put ASRC OF node in case of unknown device Mark Brown (6): Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-fix-rt5645 Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/doc', 'asoc/fix/fsl-card', 'asoc/fix/fsl-ssi', 'asoc/fix/intel' and 'asoc/fix/maintainers' into asoc-linus Merge remote-tracking branches 'asoc/fix/mtk', 'asoc/fix/psc', 'asoc/fix/pxa', 'asoc/fix/spear', 'asoc/fix/sti' and 'asoc/fix/wm0010' into asoc-linus Merge remote-tracking branches 'asoc/fix/wm8960' and 'asoc/fix/wm8962' into asoc-linus Oder Chiou (4): ASoC: rt5645: Remove incorrect settings ASoC: rt5645: Use the type SOC_DAPM_SINGLE_AUTODISABLE to prevent the weird sound in runtime of power up ASoC: rt5645: Increase the delay time to remove the pop sound ASoC: rt5645: Prevent the pop sound in case of playback and the jack is plugging Peter Ujfalusi (2): ASoC: davinci-mcasp: Revise the FIFO threshold calculation ASoC: davinci-mcasp: Fix devm_kasprintf format string Richard Fitzgerald (1): MAINTAINERS: Update website and git repo for Wolfson Microelectronics Robert Jarzmik (2): ASoC: fix broken pxa SoC support ASoC: pxa: pxa2xx-ac97: fix dma requestor lines Sudip Mukherjee (4): ASoC: wm0010: fix memory leak ASoC: dapm: fix memory leak ASoC: wm0010: fix memory leak ASoC: wm0010: fix error path Zidan Wang (4): ASoC: wm8960: correct the min gain value of some PGA ASoC: wm8960: correct gain value for input PGA and add microphone PGA ASoC: wm8962: remove 64k sample rate support ASoC: wm8960: correct the max register value of mic boost pga MAINTAINERS | 9 +++------ sound/arm/Kconfig | 15 ++++++++------- sound/soc/au1x/psc-i2s.c | 1 - sound/soc/codecs/rt5645.c | 22 +++++++++++++--------- sound/soc/codecs/wm0010.c | 23 +++++++++-------------- sound/soc/codecs/wm8960.c | 26 ++++++++++++++++++-------- sound/soc/codecs/wm8962.c | 3 ++- sound/soc/davinci/davinci-mcasp.c | 14 +++++++------- sound/soc/fsl/fsl-asoc-card.c | 3 ++- sound/soc/fsl/fsl_ssi.c | 5 +++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 20 ++++++++++++-------- sound/soc/mediatek/mtk-afe-pcm.c | 17 +++++++++++++++++ sound/soc/pxa/Kconfig | 2 -- sound/soc/pxa/pxa2xx-ac97.c | 4 ++-- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-utils.c | 9 +++++++++ sound/soc/spear/Kconfig | 2 +- sound/soc/sti/uniperif_player.c | 14 ++++++++++---- sound/soc/sti/uniperif_reader.c | 6 +++++- 19 files changed, 122 insertions(+), 75 deletions(-)

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[alsa-devel] [PATCH] ASoC: mediatek: Increase periods_min in capture
by Koro Chen 24 Sep '15

24 Sep '15

In capture, there is chance that hw_ptr reported at IRQ is a little smaller than period_size due to internal AFE buffer. In the case of ping-pong buffer: |xxxxxxxxxxxxxxxxxxxxxxxxxxxx--|-----------------------------| hw_ptr < period_size This available buffer will not be read since its size is smaller than avail_min (which is period_size by default), and read thread continues to sleep. If the next hw_ptr is just a little larger than buffer_size, overrun occurs. One more period can hold the possible unread buffer. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index d190fe0..f5baf3c 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -549,6 +549,23 @@ static int mtk_afe_dais_startup(struct snd_pcm_substream *substream, memif->substream = substream; snd_soc_set_runtime_hwparams(substream, &mtk_afe_hardware); + + /* + * Capture cannot use ping-pong buffer since hw_ptr at IRQ may be + * smaller than period_size due to AFE's internal buffer. + * This easily leads to overrun when avail_min is period_size. + * One more period can hold the possible unread buffer. + */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + ret = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIODS, + 3, + mtk_afe_hardware.periods_max); + if (ret < 0) { + dev_err(afe->dev, "hw_constraint_minmax failed\n"); + return ret; + } + } ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) -- 1.7.9.5

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[alsa-devel] Applied "spi: atmel: Remove needless bits_per_word and speed_hz tests" to the asoc tree
by Mark Brown 24 Sep '15

24 Sep '15

The patch spi: atmel: Remove needless bits_per_word and speed_hz tests has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3f8958cf9aaaac8a2bfc7417cb87db757eeb4465 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Date: Tue, 15 Sep 2015 16:26:22 +0300 Subject: [PATCH] spi: atmel: Remove needless bits_per_word and speed_hz tests SPI core validates both bits_per_word and speed_hz transfer parameters and defaults to spi->bits_per_word and spi->max_speed_hz in case these per transfer parameters are not set. This makes possible to remove two if statements and remove one code block that is never executed. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Acked-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/spi/spi-atmel.c | 21 ++++++--------------- 1 file changed, 6 insertions(+), 15 deletions(-) diff --git a/drivers/spi/spi-atmel.c b/drivers/spi/spi-atmel.c index bf9ed38..1cc1f1e 100644 --- a/drivers/spi/spi-atmel.c +++ b/drivers/spi/spi-atmel.c @@ -871,14 +871,7 @@ static int atmel_spi_set_xfer_speed(struct atmel_spi *as, * Calculate the lowest divider that satisfies the * constraint, assuming div32/fdiv/mbz == 0. */ - if (xfer->speed_hz) - scbr = DIV_ROUND_UP(bus_hz, xfer->speed_hz); - else - /* - * This can happend if max_speed is null. - * In this case, we set the lowest possible speed - */ - scbr = 0xff; + scbr = DIV_ROUND_UP(bus_hz, xfer->speed_hz); /* * If the resulting divider doesn't fit into the @@ -1300,14 +1293,12 @@ static int atmel_spi_one_transfer(struct spi_master *master, return -EINVAL; } - if (xfer->bits_per_word) { - asd = spi->controller_state; - bits = (asd->csr >> 4) & 0xf; - if (bits != xfer->bits_per_word - 8) { - dev_dbg(&spi->dev, + asd = spi->controller_state; + bits = (asd->csr >> 4) & 0xf; + if (bits != xfer->bits_per_word - 8) { + dev_dbg(&spi->dev, "you can't yet change bits_per_word in transfers\n"); - return -ENOPROTOOPT; - } + return -ENOPROTOOPT; } /* -- 2.5.0

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[alsa-devel] [PATCH RFC v4 0/8] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 24 Sep '15

24 Sep '15

Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 The binding for tda998x is taken from Jean Francois' patch series[1] on the same subject. The implementation of the of-node parsing has some minor changes from my self. Here is what I think at least could or should still be done, but non of that stuff does not sounds critical right now. Missing from tda998x driver side - hdmi_codec_ops audio_startup() implementation for audio abort support - multi channel audio support (I would need specs and preferably some HW to test for this). Missing from ASoC side generic implementation: - channel_allocation handling is completely left for the video side driver, see if ASoC side could help in any way - snd_soc_jack functionality to handle hdmi cable plug/unplug events [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095596.html Best regards, Jyri Jean-Francois Moine (1): drm/i2c: tda998x: Add support of a DT graph of ports Jyri Sarha (7): ASoC: hdmi: Remove obsolete dummy HDMI codec ALSA: pcm: add IEC958 channel status helper for hw_params ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders drm/i2c: tda998x: Remove include/sound/tda998x.h and fix graph parsing drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC HDMI codec for audio functionality ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/drm/i2c/tda998x.txt | 51 +++ arch/arm/boot/dts/am335x-boneblack.dts | 90 ++++- drivers/gpu/drm/armada/armada_drv.c | 25 +- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 319 +++++++++++++--- include/drm/i2c/tda998x.h | 24 +- include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 52 ++- sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/hdmi-codec.c | 404 +++++++++++++++++++++ sound/soc/codecs/hdmi.c | 109 ------ 13 files changed, 985 insertions(+), 204 deletions(-) create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c delete mode 100644 sound/soc/codecs/hdmi.c -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Disable power_save_node for Thinkpads
by Takashi Iwai 24 Sep '15

24 Sep '15

Lenovo Thinkpads with recent Realtek codecs seem suffering from click noises at power transition since the introduction of widget power saving in 4.1 kernel. Although this might be solved by some delays in appropriate points, as a quick workaround, just disable the power_save_node feature for now. The gain it gives is relatively small, and this makes the situation back to pre 4.1 time. This patch ended up with a bit more code changes than usual because the existing fixup for Thinkpads is highly chained. Instead of adding yet another chain, combine a few of them into a single fixup entry, as a gratis cleanup. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=943982 Cc: <stable(a)vger.kernel.org> # v4.1+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 31 +++++++++++++++++++------------ 1 file changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a75b5611d1e4..afec6dc9f91f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4188,6 +4188,24 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec, } } +/* fixup for Thinkpad docks: add dock pins, avoid HP parser fixup */ +static void alc_fixup_tpt440_dock(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const struct hda_pintbl pincfgs[] = { + { 0x16, 0x21211010 }, /* dock headphone */ + { 0x19, 0x21a11010 }, /* dock mic */ + { } + }; + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; + codec->power_save_node = 0; /* avoid click noises */ + snd_hda_apply_pincfgs(codec, pincfgs); + } +} + static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -4562,7 +4580,6 @@ enum { ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, - ALC292_FIXUP_TPT440_DOCK2, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -5029,17 +5046,7 @@ static const struct hda_fixup alc269_fixups[] = { }, [ALC292_FIXUP_TPT440_DOCK] = { .type = HDA_FIXUP_FUNC, - .v.func = alc269_fixup_pincfg_no_hp_to_lineout, - .chained = true, - .chain_id = ALC292_FIXUP_TPT440_DOCK2 - }, - [ALC292_FIXUP_TPT440_DOCK2] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x16, 0x21211010 }, /* dock headphone */ - { 0x19, 0x21a11010 }, /* dock mic */ - { } - }, + .v.func = alc_fixup_tpt440_dock, .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, -- 2.5.3

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[alsa-devel] [linux-sunxi][PATCH 0/3]Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 24 Sep '15

24 Sep '15

From: Marcus Cooper <codekipper(a)gmail.com> This patch set adds support for the Allwinner SPDIF transceiver as present on the A10, A20 and A31 SoC boards. For now just the SPDIF transmitter has been tested on a Mele A2000. In order for this patch set to be functional we require audio clock patches which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery Thanks in advance, CK Marcus Cooper (3): dt-bindings: add sunxi SPDIF transceiver bindings dt-binding: Add sunxi SPDIF machine driver ASOC: sunxi: Add support for the spdif block .../devicetree/bindings/sound/sunxi,spdif.txt | 49 ++ .../bindings/sound/sunxi-audio-spdif.txt | 36 + sound/soc/sunxi/Kconfig | 10 + sound/soc/sunxi/Makefile | 4 + sound/soc/sunxi/sunxi-machine-spdif.c | 110 +++ sound/soc/sunxi/sunxi-spdif.c | 801 +++++++++++++++++++++ 6 files changed, 1010 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/sunxi-audio-spdif.txt create mode 100644 sound/soc/sunxi/sunxi-machine-spdif.c create mode 100644 sound/soc/sunxi/sunxi-spdif.c -- 2.5.3

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[alsa-devel] [PATCH] drm/i915: set proper N/CTS in modeset
by libin.yang＠intel.com 24 Sep '15

24 Sep '15

From: Libin Yang <libin.yang(a)intel.com> When modeset occurs and the TMDS frequency is set to some speical values, the N/CTS need to be set manually if audio is playing. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- drivers/gpu/drm/i915/intel_audio.c | 56 ++++++++++++++++++++++++++++++++------ include/drm/i915_component.h | 10 +++++++ 2 files changed, 57 insertions(+), 9 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index e3c32ce..a12bf57 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -128,6 +128,20 @@ static int audio_config_get_n(const struct drm_display_mode *mode, int rate) return 0; } +static uint32_t audio_config_setup_n_reg(int n, uint32_t val) +{ + int n_low, n_up; + uint32_t tmp = val; + + n_low = n & 0xfff; + n_up = (n >> 12) & 0xff; + tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); + tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | + (n_low << AUD_CONFIG_LOWER_N_SHIFT) | + AUD_CONFIG_N_PROG_ENABLE); + return tmp; +} + /* check whether N/CTS/M need be set manually */ static bool audio_rate_need_prog(struct intel_crtc *crtc, struct drm_display_mode *mode) @@ -262,9 +276,14 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, struct drm_i915_private *dev_priv = connector->dev->dev_private; struct intel_crtc *intel_crtc = to_intel_crtc(encoder->base.crtc); enum pipe pipe = intel_crtc->pipe; + struct i915_audio_component *acomp = dev_priv->audio_component; const uint8_t *eld = connector->eld; + struct intel_digital_port *intel_dig_port = + enc_to_dig_port(&encoder->base); + enum port port = intel_dig_port->port; uint32_t tmp; int len, i; + int n, rate; DRM_DEBUG_KMS("Enable audio codec on pipe %c, %u bytes ELD\n", pipe_name(pipe), drm_eld_size(eld)); @@ -302,12 +321,29 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, /* Enable timestamps */ tmp = I915_READ(HSW_AUD_CFG(pipe)); tmp &= ~AUD_CONFIG_N_VALUE_INDEX; - tmp &= ~AUD_CONFIG_N_PROG_ENABLE; tmp &= ~AUD_CONFIG_PIXEL_CLOCK_HDMI_MASK; if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT)) tmp |= AUD_CONFIG_N_VALUE_INDEX; else tmp |= audio_config_hdmi_pixel_clock(mode); + + tmp &= ~AUD_CONFIG_N_PROG_ENABLE; + if (audio_rate_need_prog(intel_crtc, mode)) { + if (!acomp) + rate = 0; + else if (port >= PORT_A && port <= PORT_E) + rate = acomp->aud_sample_rate[port]; + else { + DRM_ERROR("invalid port: %d\n", port); + rate = 0; + } + n = audio_config_get_n(mode, rate); + if (n != 0) + tmp = audio_config_setup_n_reg(n, tmp); + else + DRM_DEBUG_KMS("no suitable N value is found\n"); + } + I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -594,9 +630,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, struct intel_digital_port *intel_dig_port; struct intel_crtc *crtc; struct drm_display_mode *mode; + struct i915_audio_component *acomp = dev_priv->audio_component; enum pipe pipe = -1; u32 tmp; - int n_low, n_up, n; + int n; /* HSW, BDW SKL need this fix */ if (!IS_SKYLAKE(dev_priv) && @@ -630,6 +667,9 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, pipe_name(pipe), port_name(port)); mode = &crtc->config->base.adjusted_mode; + /* port must be valid now, otherwise the pipe will be invalid */ + acomp->aud_sample_rate[port] = rate; + /* 2. check whether to set the N/CTS/M manually or not */ if (!audio_rate_need_prog(crtc, mode)) { tmp = I915_READ(HSW_AUD_CFG(pipe)); @@ -649,15 +689,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, mutex_unlock(&dev_priv->av_mutex); return 0; } - n_low = n & 0xfff; - n_up = (n >> 12) & 0xff; - /* 4. set the N/CTS/M */ + /* 3. set the N/CTS/M */ tmp = I915_READ(HSW_AUD_CFG(pipe)); - tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); - tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | - (n_low << AUD_CONFIG_LOWER_N_SHIFT) | - AUD_CONFIG_N_PROG_ENABLE); + tmp = audio_config_setup_n_reg(n, tmp); I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -678,6 +713,7 @@ static int i915_audio_component_bind(struct device *i915_dev, { struct i915_audio_component *acomp = data; struct drm_i915_private *dev_priv = dev_to_i915(i915_dev); + int i; if (WARN_ON(acomp->ops || acomp->dev)) return -EEXIST; @@ -685,6 +721,8 @@ static int i915_audio_component_bind(struct device *i915_dev, drm_modeset_lock_all(dev_priv->dev); acomp->ops = &i915_audio_component_ops; acomp->dev = i915_dev; + for (i = 0; i < I915_MAX_PORTS; i++) + acomp->aud_sample_rate[i] = 0; dev_priv->audio_component = acomp; drm_modeset_unlock_all(dev_priv->dev); diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index e6d35d7..89dc7d6 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,8 +24,18 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +/* MAX_PORT is the number of port + * It must be sync with I915_MAX_PORTS defined i915_drv.h + * 5 should be enough as only HSW, BDW, SKL need such fix. + */ +#define MAX_PORTS 5 + struct i915_audio_component { struct device *dev; + /** + * @aud_sample_rate: the array of audio sample rate per port + */ + int aud_sample_rate[MAX_PORTS]; const struct i915_audio_component_ops { struct module *owner; -- 1.9.1

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[alsa-devel] Bug: LADSPA implementation - no up/downmix handling
by Andreas Hartmann 24 Sep '15

24 Sep '15

Hi, Using the surround_encoder of the swh-plugins I noticed, that the alsa implementation only assumes an equal number of in- and output channels for ladspa plugins. This occurs from Line 752 onwards in pcm_ladspa.c and leads to a Segfault shortly after. There needs to be a check and correct handling for up/downmix scenarios. Andreas

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[alsa-devel] No stream timing info with alsa pulse plugin?
by sqweek 24 Sep '15

24 Sep '15

Hi guys, I discovered on my machine that alsa_snd_pcm_status_get_tstamp (and also get_htstamp) stop working when my asoundrc invokes the pulse plugin. That is, normally they work fine but once the sound is going via pulse the returned time stamp is always zero. I'm trying to understand why that is. I couldn't really find any discussion of the pulse plugin on the archives so I'm not entirely sure I've found the right list; I was directed here via the pulseaudio bugzilla[1]. [1] https://bugs.freedesktop.org/show_bug.cgi?id=88503 Thanks for any assistance! -sqweek

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