May 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/3] ASoC: trivial compile warning fixes
by Takashi Iwai 07 May '19

07 May '19

Hi, here are the three fixes for trivial compile warnings I caught during build test. Takashi === Takashi Iwai (3): ASoC: da7219: Fix a compile warning at CONFIG_COMMON_CLK=n ASoC: SOF: Fix a compile warning with CONFIG_PCI=n ASoC: rockchip: Fix an uninitialized variable compile warning sound/soc/codecs/da7219.c | 2 ++ sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/sof/ops.c | 2 +- 3 files changed, 4 insertions(+), 2 deletions(-) -- 2.16.4

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[alsa-devel] [PATCH] ASoC: sprd: Add reserved DMA memory support
by Baolin Wang 07 May '19

07 May '19

For Spreadtrum audio platform driver, it need allocate a larger DMA buffer dynamically to copy audio data between userspace and kernel space, but that will increase the risk of memory allocation failure especially the system is under heavy load situation. To make sure the audio can work in this scenario, we usually reserve one region of memory to be used as a shared pool of DMA buffers for the platform component. So add of_reserved_mem_device_init_by_idx() function to initialize the shared pool of DMA buffers to be used by the platform component. Signed-off-by: Baolin Wang <baolin.wang(a)linaro.org> --- sound/soc/sprd/sprd-pcm-dma.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sprd/sprd-pcm-dma.c b/sound/soc/sprd/sprd-pcm-dma.c index 9be6d4b..d38ebbb 100644 --- a/sound/soc/sprd/sprd-pcm-dma.c +++ b/sound/soc/sprd/sprd-pcm-dma.c @@ -6,6 +6,7 @@ #include <linux/dma/sprd-dma.h> #include <linux/kernel.h> #include <linux/module.h> +#include <linux/of_reserved_mem.h> #include <linux/platform_device.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -530,8 +531,14 @@ static void sprd_pcm_free(struct snd_pcm *pcm) static int sprd_soc_platform_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; int ret; + ret = of_reserved_mem_device_init_by_idx(&pdev->dev, np, 0); + if (ret) + dev_warn(&pdev->dev, + "no reserved DMA memory for audio platform device\n"); + ret = devm_snd_soc_register_component(&pdev->dev, &sprd_soc_component, NULL, 0); if (ret) -- 1.7.9.5

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[alsa-devel] [RFC PATCH] ASoC: codec: hdac_hdmi: no checking monitor in hw_params
by libin.yang＠intel.com 07 May '19

07 May '19

From: Libin Yang <libin.yang(a)intel.com> This patch move the check of monitor from hw_params to trigger callback. The original code will check the monitor presence in hw_params. If the monitor doesn't exist, hw_params will return -ENODEV. Mostly this is OK. However, pulseaudio will check the pcm devices when kernel is booting up. It will try to open, set hw_params, prepare such pcm devices. We can't guarantee that the monitor will be connected when kernel is booting up. Especially, hdac_hdmi will export 3 pcms at most. It's hard to say users will connect 3 monitors to the HDMI/DP ports. This will cause pulseaudio fail in parsing the pcm devices because the driver will return -ENODEV in hw_params. This patch tries to move the check of monitor presence into trigger callback. This can "trick" the pulseaudio the pcm is ready. This bug is found when we try to enable HDMI detection in gnome-sound-setting for ASoC hdac_hdmi. After we enable the hdmi in UCM, pulseaudio will try to parse the hdmi pcm devices. It will cause failure if there are no monitors connected. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 44 +++++++++++++++++++++++++++++++------------- 1 file changed, 31 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 4de1fbf..f482e09 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -455,24 +455,11 @@ static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hparams, struct snd_soc_dai *dai) { struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); - struct hdac_device *hdev = hdmi->hdev; struct hdac_hdmi_dai_port_map *dai_map; - struct hdac_hdmi_port *port; struct hdac_hdmi_pcm *pcm; int format; dai_map = &hdmi->dai_map[dai->id]; - port = dai_map->port; - - if (!port) - return -ENODEV; - - if ((!port->eld.monitor_present) || (!port->eld.eld_valid)) { - dev_err(&hdev->dev, - "device is not configured for this pin:port%d:%d\n", - port->pin->nid, port->id); - return -ENODEV; - } format = snd_hdac_calc_stream_format(params_rate(hparams), params_channels(hparams), params_format(hparams), @@ -630,6 +617,36 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream, dai_map->port = NULL; } +static int hdac_hdmi_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct hdac_hdmi_port *port; + struct hdac_hdmi_dai_port_map *dai_map; + struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); + struct hdac_device *hdev = hdmi->hdev; + + /* + * When start, if there is no monitor, + * It should not start audio. + */ + if (cmd == SNDRV_PCM_TRIGGER_START) { + dai_map = &hdmi->dai_map[dai->id]; + port = dai_map->port; + + if (!port) + return -ENODEV; + + if ((!port->eld.monitor_present) || (!port->eld.eld_valid)) { + dev_err(&hdev->dev, + "device is not configured for this pin:port%d:%d\n", + port->pin->nid, port->id); + return -ENODEV; + } + } + + return 0; +} + static int hdac_hdmi_query_cvt_params(struct hdac_device *hdev, struct hdac_hdmi_cvt *cvt) { @@ -1389,6 +1406,7 @@ static const struct snd_soc_dai_ops hdmi_dai_ops = { .startup = hdac_hdmi_pcm_open, .shutdown = hdac_hdmi_pcm_close, .hw_params = hdac_hdmi_set_hw_params, + .trigger = hdac_hdmi_pcm_trigger, .set_tdm_slot = hdac_hdmi_set_tdm_slot, }; -- 2.7.4

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[alsa-devel] ALSA -> pulseaudio channel mapping, what is it (and why?)
by frederik＠ofb.net 07 May '19

07 May '19

Dear ALSA developers, I ran into a problem which seems to be caused by some lines in "alsa-plugins-1.1.8/pulse/pcm_pulse.c". The problem is that if I create a 6-channel PulseAudio device, and then I configure an ALSA device in ~/.asoundrc to output to the PulseAudio device, the channels of the ALSA device are mixed together in various ways and don't correspond 1-1 with the channels of the PulseAudio device. I find this to be counter-intuitive. If I wanted channels to be mixed and remapped in various ways, I would use the PulseAudio module module-remap-sink on the PulseAudio side, since AFAICT ALSA does not support named channels. I would not expect remixing to happen in the ALSA->Pulse bridge, since the relevant PCM plugin contains no channel map parameter. I don't even know what channel map I should give the PulseAudio device in order to prevent remixing from occurring. The file pcm_pulse.c contains the lines: for (c = pcm->ss.channels; c > 0; c--) if (pa_channel_map_init_auto(&map, c, PA_CHANNEL_MAP_ALSA)) break; This refers to a macro in /usr/include/pulse/channelmap.h: /** \cond fulldocs */ PA_CHANNEL_MAP_ALSA, /**< The default mapping used by ALSA. This mapping is probably * not too useful since ALSA's default channel mapping depends on * the device string used. */ /** \endcond */ The comment says that the ALSA mapping is "not too useful" - so why are we using it? Also, how do I find out what it is, if it "depends on the device string used"? My preference would simply be to have a 1-1 channel map in pcm_pulse, so that a PulseAudio device with 6 channels will turn into an ALSA device with THE SAME 6 CHANNELS. No mixing. Mixing is confusing in this context, and seems undocumented outside of the code. I tried using aux0 thru aux5 as the PulseAudio channel map, to prevent geometry-based remixing. However, this results in silence - apparently none of the 6 ALSA channels end up going anywhere. That was also unexpected. Since I am playing 6-channel audio to a 6-channel device, which emulates another 6-channel device, I would have thought that the channel mapping wouldn't be complicated. Instead, it appears that the PulseAudio sink needs to have channels with specific names in order to receive data from ALSA. Can we fix this, or improve the user experience? Thanks, Frederick

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[alsa-devel] [PATCH v2 1/2] ALSA: hda/hdmi - Read the pin sense from register when repolling
by Hui Wang 07 May '19

07 May '19

The driver will check the monitor presence when resuming from suspend, starting poll or interrupt triggers. In these 3 situations, the jack_dirty will be set to 1 first, then the hda_jack.c reads the pin_sense from register, after reading the register, the jack_dirty will be set to 0. But hdmi_repoll_work() is enabled in these 3 situations, It will read the pin_sense a couple of times subsequently, since the jack_dirty is 0 now, It does not read the register anymore, instead it uses the shadow pin_sense which is read at the first time. It is meaningless to check the shadow pin_sense a couple of times, we need to read the register to check the real plugging state, so we set the jack_dirty to 1 in the hdmi_repoll_work(). Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8b3ac690efa3..410d94301941 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1663,6 +1663,11 @@ static void hdmi_repoll_eld(struct work_struct *work) container_of(to_delayed_work(work), struct hdmi_spec_per_pin, work); struct hda_codec *codec = per_pin->codec; struct hdmi_spec *spec = codec->spec; + struct hda_jack_tbl *jack; + + jack = snd_hda_jack_tbl_get(codec, per_pin->pin_nid); + if (jack) + jack = jack->jack_dirty = 1; if (per_pin->repoll_count++ > 6) per_pin->repoll_count = 0; -- 2.17.1

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[alsa-devel] alsa-utils: axfer/test: fix compiler warnings due to uninitialized return value
by GitHub pull_request - opened 06 May '19

06 May '19

alsa-project/alsa-utils pull request #18 was opened from takaswie: Hi, One of test programs for axfer internals has bugs to let compiler generate warnings due to uninitialized return value. This patchset fixes them. ``` Takashi Sakamoto (2): axfer/test: fix uninitialized warning axfer/test: fix uninitialized warning axfer/test/mapper-test.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/18 Patch URL : https://github.com/alsa-project/alsa-utils/pull/18.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] [PATCH] ASoC: Intel: bytcr_5640.c:Refactored if statement and removed buffer
by Nariman 06 May '19

06 May '19

From: Nariman Etemadi <narimantos(a)gmail.com> in function snd_byt_rt5640_mc_probe and removed buffer yt_rt5640_codec_aif_name & byt_rt5640_cpu_dai_name Signed-off-by: Nariman Etemadi <narimantos(a)gmail.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 26 ++++---------------------- 1 file changed, 4 insertions(+), 22 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 940eb27158da..0d91642ca287 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1075,8 +1075,6 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { /* SoC card */ static char byt_rt5640_codec_name[SND_ACPI_I2C_ID_LEN]; -static char byt_rt5640_codec_aif_name[12]; /* = "rt5640-aif[1|2]" */ -static char byt_rt5640_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ static char byt_rt5640_long_name[40]; /* = "bytcr-rt5640-*-spk-*-mic" */ static int byt_rt5640_suspend(struct snd_soc_card *card) @@ -1268,28 +1266,12 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) log_quirks(&pdev->dev); if ((byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) || - (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { - - /* fixup codec aif name */ - snprintf(byt_rt5640_codec_aif_name, - sizeof(byt_rt5640_codec_aif_name), - "%s", "rt5640-aif2"); - - byt_rt5640_dais[dai_index].codec_dai_name = - byt_rt5640_codec_aif_name; - } + (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) + byt_rt5640_dais[dai_index].codec_dai_name = "rt5640-aif2"; if ((byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) || - (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { - - /* fixup cpu dai name name */ - snprintf(byt_rt5640_cpu_dai_name, - sizeof(byt_rt5640_cpu_dai_name), - "%s", "ssp0-port"); - - byt_rt5640_dais[dai_index].cpu_dai_name = - byt_rt5640_cpu_dai_name; - } + (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) + byt_rt5640_dais[dai_index].cpu_dai_name = "ssp0-port"; if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) { priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); -- 2.20.1

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Re: [alsa-devel] [PATCH v2 2/2] regmap: soundwire: fix Kconfig select/depend issue
by Pierre-Louis Bossart 06 May '19

06 May '19

On 5/5/19 11:40 PM, Mark Brown wrote: > On Fri, May 03, 2019 at 09:32:53AM -0500, Pierre-Louis Bossart wrote: > >> As I mentioned it'll compile the bus even if there is no user for it, but >> it's your call: alignment or optimization. > > You can have both. Alignment is a requirement. If you want to optimize > this then it'd be better to optimize all the bus types rather than just > having the one weird bus type that does something different for no > documented reason. Fine, I'll align if this is the requirement. Thanks for the feedback.

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[alsa-devel] [PATCH v2] ASoC: RT5677-SPI: Disable 16Bit SPI Transfers
by Curtis Malainey 06 May '19

06 May '19

The current algorithm allows 3 types of transfers, 16bit, 32bit and burst. According to Realtek, 16bit transfers have a special restriction in that it is restricted to the memory region of 0x18020000 ~ 0x18021000. This region is the memory location of the I2C registers. The current algorithm does not uphold this restriction and therefore fails to complete writes. Since this has been broken for some time it likely no one is using it. Better to simply disable the 16 bit writes. This will allow users to properly load firmware over SPI without data corruption. CC: Ben Zhang <benzh(a)chromium.org> CC: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677-spi.c | 35 ++++++++++++++++------------------- 1 file changed, 16 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 167a02773a0b..84b6bd8b50e1 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -58,13 +58,15 @@ static DEFINE_MUTEX(spi_mutex); * RT5677_SPI_READ/WRITE_32: Transfer 4 bytes * RT5677_SPI_READ/WRITE_BURST: Transfer any multiples of 8 bytes * - * For example, reading 260 bytes at 0x60030002 uses the following commands: - * 0x60030002 RT5677_SPI_READ_16 2 bytes + * Note: + * 16 Bit writes and reads are restricted to the address range + * 0x18020000 ~ 0x18021000 + * + * For example, reading 256 bytes at 0x60030004 uses the following commands: * 0x60030004 RT5677_SPI_READ_32 4 bytes * 0x60030008 RT5677_SPI_READ_BURST 240 bytes * 0x600300F8 RT5677_SPI_READ_BURST 8 bytes * 0x60030100 RT5677_SPI_READ_32 4 bytes - * 0x60030104 RT5677_SPI_READ_16 2 bytes * * Input: * @read: true for read commands; false for write commands @@ -79,15 +81,13 @@ static u8 rt5677_spi_select_cmd(bool read, u32 align, u32 remain, u32 *len) { u8 cmd; - if (align == 2 || align == 6 || remain == 2) { - cmd = RT5677_SPI_READ_16; - *len = 2; - } else if (align == 4 || remain <= 6) { + if (align == 4 || remain <= 4) { cmd = RT5677_SPI_READ_32; *len = 4; } else { cmd = RT5677_SPI_READ_BURST; - *len = min_t(u32, remain & ~7, RT5677_SPI_BURST_LEN); + *len = (((remain - 1) >> 3) + 1) << 3; + *len = min_t(u32, *len, RT5677_SPI_BURST_LEN); } return read ? cmd : cmd + 1; } @@ -108,7 +108,7 @@ static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen) } } -/* Read DSP address space using SPI. addr and len have to be 2-byte aligned. */ +/* Read DSP address space using SPI. addr and len have to be 4-byte aligned. */ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) { u32 offset; @@ -124,7 +124,7 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) if (!g_spi) return -ENODEV; - if ((addr & 1) || (len & 1)) { + if ((addr & 3) || (len & 3)) { dev_err(&g_spi->dev, "Bad read align 0x%x(%zu)\n", addr, len); return -EACCES; } @@ -159,13 +159,13 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) } EXPORT_SYMBOL_GPL(rt5677_spi_read); -/* Write DSP address space using SPI. addr has to be 2-byte aligned. - * If len is not 2-byte aligned, an extra byte of zero is written at the end +/* Write DSP address space using SPI. addr has to be 4-byte aligned. + * If len is not 4-byte aligned, then extra zeros are written at the end * as padding. */ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) { - u32 offset, len_with_pad = len; + u32 offset; int status = 0; struct spi_transfer t; struct spi_message m; @@ -178,22 +178,19 @@ int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) if (!g_spi) return -ENODEV; - if (addr & 1) { + if (addr & 3) { dev_err(&g_spi->dev, "Bad write align 0x%x(%zu)\n", addr, len); return -EACCES; } - if (len & 1) - len_with_pad = len + 1; - memset(&t, 0, sizeof(t)); t.tx_buf = buf; t.speed_hz = RT5677_SPI_FREQ; spi_message_init_with_transfers(&m, &t, 1); - for (offset = 0; offset < len_with_pad;) { + for (offset = 0; offset < len;) { spi_cmd = rt5677_spi_select_cmd(false, (addr + offset) & 7, - len_with_pad - offset, &t.len); + len - offset, &t.len); /* Construct SPI message header */ buf[0] = spi_cmd; -- 2.21.0.1020.gf2820cf01a-goog

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[alsa-devel] [PATCH] ASoC: max98090: Fix restore of DAPM Muxes
by Jon Hunter 06 May '19

06 May '19

The max98090 driver defines 3 DAPM muxes; one for the right line output (LINMOD Mux), one for the left headphone mixer source (MIXHPLSEL Mux) and one for the right headphone mixer source (MIXHPRSEL Mux). The same bit is used for the mux as well as the DAPM enable, and although the mux can be correctly configured, after playback has completed, the mux will be reset during the disable phase. This is preventing the state of these muxes from being saved and restored correctly on system reboot. Fix this by marking these muxes as SND_SOC_NOPM. Note this has been verified this on the Tegra124 Nyan Big which features the MAX98090 codec. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- If this is the correct fix for this issue, can we tag this for stable? Or I can re-send with the stable tag. However, wanted to send out for review first. sound/soc/codecs/max98090.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 30c242c38d99..7619ea31ab50 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1194,14 +1194,14 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { &max98090_right_rcv_mixer_controls[0], ARRAY_SIZE(max98090_right_rcv_mixer_controls)), - SND_SOC_DAPM_MUX("LINMOD Mux", M98090_REG_LOUTR_MIXER, - M98090_LINMOD_SHIFT, 0, &max98090_linmod_mux), + SND_SOC_DAPM_MUX("LINMOD Mux", SND_SOC_NOPM, 0, 0, + &max98090_linmod_mux), - SND_SOC_DAPM_MUX("MIXHPLSEL Mux", M98090_REG_HP_CONTROL, - M98090_MIXHPLSEL_SHIFT, 0, &max98090_mixhplsel_mux), + SND_SOC_DAPM_MUX("MIXHPLSEL Mux", SND_SOC_NOPM, 0, 0, + &max98090_mixhplsel_mux), - SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL, - M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), + SND_SOC_DAPM_MUX("MIXHPRSEL Mux", SND_SOC_NOPM, 0, 0, + &max98090_mixhprsel_mux), SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE, M98090_HPLEN_SHIFT, 0, NULL, 0), -- 2.7.4

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