March 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: Intel: Update bxt_da7219_max98357a to add a new
by Harsha Priya 21 Mar '17

21 Mar '17

This patch adds a platform clock widget to turn off the clock only when both headset capture and headset playback are not in use. This removes turning off the clock in hw_free so that the clock is on when either capture or playback of headset is in progress. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 97 +++++++++++++++------------ 1 file changed, 53 insertions(+), 44 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 2cda06c..3a8c4d9 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -55,6 +55,54 @@ enum { BXT_DPCM_AUDIO_HDMI3_PB, }; +static inline struct snd_soc_dai *bxt_get_codec_dai(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + + if (!strncmp(rtd->codec_dai->name, BXT_DIALOG_CODEC_DAI, + strlen(BXT_DIALOG_CODEC_DAI))) + return rtd->codec_dai; + } + + return NULL; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + int ret = 0; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + + codec_dai = bxt_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_MCLK, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop PLL: %d\n", ret); + } else if(SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, + DA7219_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN); + if (ret) + dev_err(card->dev, "can't set codec sysclk configuration\n"); + + ret = snd_soc_dai_set_pll(codec_dai, 0, + DA7219_SYSCLK_PLL_SRM, 0, DA7219_PLL_FREQ_OUT_98304); + if (ret) + dev_err(card->dev, "failed to start PLL: %d\n", ret); + } + + return ret; +} + static const struct snd_kcontrol_new broxton_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -69,6 +117,8 @@ enum { SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), SND_SOC_DAPM_SPK("HDMI3", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD|SND_SOC_DAPM_PRE_PMU), }; static const struct snd_soc_dapm_route broxton_map[] = { @@ -109,6 +159,9 @@ enum { /* DMIC */ {"dmic01_hifi", NULL, "DMIC01 Rx"}, {"DMIC01 Rx", NULL, "DMIC AIF"}, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, }; static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, @@ -243,49 +296,6 @@ static int bxt_fe_startup(struct snd_pcm_substream *substream) .startup = bxt_fe_startup, }; -static int broxton_da7219_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; - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - DA7219_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(codec_dai->dev, "can't set codec sysclk configuration\n"); - - ret = snd_soc_dai_set_pll(codec_dai, 0, - DA7219_SYSCLK_PLL_SRM, 0, DA7219_PLL_FREQ_OUT_98304); - if (ret < 0) { - dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret); - return -EIO; - } - - return ret; -} - -static int broxton_da7219_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_pll(codec_dai, 0, - DA7219_SYSCLK_MCLK, 0, 0); - if (ret < 0) { - dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret); - return -EIO; - } - - return ret; -} - -static const struct snd_soc_ops broxton_da7219_ops = { - .hw_params = broxton_da7219_hw_params, - .hw_free = broxton_da7219_hw_free, -}; - static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -467,7 +477,6 @@ static int broxton_refcap_startup(struct snd_pcm_substream *substream) SND_SOC_DAIFMT_CBS_CFS, .ignore_pmdown_time = 1, .be_hw_params_fixup = broxton_ssp_fixup, - .ops = &broxton_da7219_ops, .dpcm_playback = 1, .dpcm_capture = 1, }, -- 1.9.1

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[alsa-devel] [PATCH 1/4] dt-bindings: Document the dmas and dma-names properties for bcm2835-analog-audio
by Michael Zoran 21 Mar '17

21 Mar '17

Add documentation for the DMA properties required when using the PWM controller of the bcm2835 for analog audio output. Signed-off-by: Michael Zoran <mzoran(a)crowfest.net> --- Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt b/Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt index cf573e85b11d..08fc06fc083a 100644 --- a/Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt +++ b/Documentation/devicetree/bindings/pwm/pwm-bcm2835.txt @@ -1,14 +1,21 @@ BCM2835 PWM controller (Raspberry Pi controller) Required properties: -- compatible: should be "brcm,bcm2835-pwm" +- compatible: should be "brcm,bcm2835-pwm" or "brcm,bcm2835-analog-audio" - reg: physical base address and length of the controller's registers - clocks: This clock defines the base clock frequency of the PWM hardware system, the period and the duty_cycle of the PWM signal is a multiple of the base period. + +Required properties(PWM Output): - #pwm-cells: Should be 2. See pwm.txt in this directory for a description of the cells format. +Required properties(Audio Output): +- dmas: Should contain one entry pointing to the DMA channel used to + transfer audio data +- dma-names: Should contain "tx" + Examples: pwm@2020c000 { -- 2.11.0

3 9

[alsa-devel] hw/sw params API in alsa-lib miscompiled, result: unusable alsa-utils
by Deweloper 21 Mar '17

21 Mar '17

Hi, I built alsa-lib and alsa-utils (v.1.1.3) from sources, using GCC 6. When I tried launching arecord, it failed due to lack of memory. During analysis I found out that: - the message is printed because realloc(buf, period_size) returns NULL - realloc() returns NULL because period_size is 0 - period_size is 0 because snd_pcm_hw_params_get_period_size() puts 0 to *frames (instead of 32768) and returns 32768 (instead of 0) - snd_pcm_hw_params_get_period_size() misbehaves in this way because although a NEW API header is used, the OLD API implementation is actually called So it seems that I've run into exactly the same problem which was described by John Spencer in August 2013: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065031.html > my problem is that the old version is getting called instead of the > new one Did anything change in this matter since August, 2013? --

2 2

[alsa-devel] [PATCH v5 6/6] pcm:plug: save converter config
by sutar.mounesh＠gmail.com 21 Mar '17

21 Mar '17

From: Andreas Pape <apape(a)de.adit-jv.com> Passed config is freed after call to open, thus it is invalid when trying to extract the converter name. So config entry is saved for later usage. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index 266707b..c4375ec 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -50,7 +50,7 @@ typedef struct { snd_pcm_format_t sformat; int schannels; int srate; - const snd_config_t *rate_converter; + snd_config_t *rate_converter; enum snd_pcm_plug_route_policy route_policy; snd_pcm_route_ttable_entry_t *ttable; int ttable_ok; @@ -64,6 +64,10 @@ static int snd_pcm_plug_close(snd_pcm_t *pcm) snd_pcm_plug_t *plug = pcm->private_data; int err, result = 0; free(plug->ttable); + if (plug->rate_converter) { + snd_config_delete(plug->rate_converter); + plug->rate_converter = NULL; + } assert(plug->gen.slave == plug->req_slave); if (plug->gen.close_slave) { snd_pcm_unlink_hw_ptr(pcm, plug->req_slave); @@ -1108,7 +1112,6 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, plug->sformat = sformat; plug->schannels = schannels; plug->srate = srate; - plug->rate_converter = rate_converter; plug->gen.slave = plug->req_slave = slave; plug->gen.close_slave = close_slave; plug->route_policy = route_policy; @@ -1125,6 +1128,15 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, pcm->ops = &snd_pcm_plug_ops; pcm->fast_ops = slave->fast_ops; pcm->fast_op_arg = slave->fast_op_arg; + if (rate_converter) { + err = snd_config_copy(&plug->rate_converter, + (snd_config_t *)rate_converter); + if (err < 0) { + snd_pcm_free(pcm); + free(plug); + return err; + } + } pcm->private_data = plug; pcm->poll_fd = slave->poll_fd; pcm->poll_events = slave->poll_events; -- 2.7.4

3 2

[alsa-devel] [PATCH v2] ALSA: x86: Select CONFIG_HDMI_LPE_AUDIO as default
by Ian W MORRISON 21 Mar '17

21 Mar '17

The upstream kernel builds for distributions such as Ubuntu which now includes binary packages for v4.11 mainline kernel release candidates are promoted as a way of testing upstream kernels to to confirm that upstream has fixed a specific issue (see https://wiki.ubuntu.com/ Kernel/MainlineBuilds). Unfortunately the long awaited patch for providing HDMI audio support for Bay Trail and Cherry Trail devices does not include this support through a module built by default. Through including by default of the two associated CONFIG settings (SND_X86 and HDMI_LPE_AUDIO), upstream kernel builds would automatically provide the much desired HDMI audio support by default. This patch uses a Kconfig 'default' statement to include the driver as default. Changes in version 2: CONFIG_SND_X86 now a bool and changed default m to default y Signed-off-by: Ian W Morrison <linuxium(a)linuxium.com.au> --- sound/x86/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 84c8f8fc..cac2270 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -1,6 +1,7 @@ menuconfig SND_X86 - tristate "X86 sound devices" + bool "X86 sound devices" depends on X86 + default y ---help--- X86 sound devices that don't fall under SoC or PCI categories @@ -9,6 +10,7 @@ if SND_X86 config HDMI_LPE_AUDIO tristate "HDMI audio without HDaudio on Intel Atom platforms" depends on DRM_I915 + default y select SND_PCM help Choose this option to support HDMI LPE Audio mode -- 1.9.1

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[alsa-devel] [PATCH] ASoC: STI: Fix null ptr deference in IRQ handler
by Arnaud Pouliquen 21 Mar '17

21 Mar '17

With RTlinux a race condition has been found that leads to NULL ptr crash: - On CPU 0: uni_player_irq_handler is called to treat XRUN "(player->state == UNIPERIF_STATE_STOPPED)" is FALSE so status is checked, dev_err(player->dev, "FIFO underflow error detected") is printed and then snd_pcm_stream_lock should be called to lock stream for stopping. - On CPU 1: application stop and close the stream. Issue is that the stop and shutdown functions are executed while "FIFO underflow error detected" is printed. So when CPU 0 call snd_pcm_stream_lock is player->substream is already null. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- sound/soc/sti/uniperif_player.c | 9 +++------ sound/soc/sti/uniperif_reader.c | 8 +++++--- 2 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 60ae31a..6fa9eee 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -65,8 +65,10 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) unsigned int status; unsigned int tmp; + snd_pcm_stream_lock(player->substream); if (player->state == UNIPERIF_STATE_STOPPED) { /* Unexpected IRQ: do nothing */ + snd_pcm_stream_unlock(player->substream); return IRQ_NONE; } @@ -88,9 +90,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player); /* Stop the player */ - snd_pcm_stream_lock(player->substream); snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(player->substream); } ret = IRQ_HANDLED; @@ -104,9 +104,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player); /* Stop the player */ - snd_pcm_stream_lock(player->substream); snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(player->substream); ret = IRQ_HANDLED; } @@ -138,13 +136,12 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) dev_err(player->dev, "Underflow recovery failed\n"); /* Stop the player */ - snd_pcm_stream_lock(player->substream); snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(player->substream); ret = IRQ_HANDLED; } + snd_pcm_stream_unlock(player->substream); return ret; } diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 5992c6a..9273f59 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -46,9 +46,11 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) struct uniperif *reader = dev_id; unsigned int status; + snd_pcm_stream_lock(reader->substream); if (reader->state == UNIPERIF_STATE_STOPPED) { /* Unexpected IRQ: do nothing */ dev_warn(reader->dev, "unexpected IRQ\n"); + snd_pcm_stream_unlock(reader->substream); return IRQ_HANDLED; } @@ -60,13 +62,13 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) { dev_err(reader->dev, "FIFO error detected\n"); - snd_pcm_stream_lock(reader->substream); snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(reader->substream); - return IRQ_HANDLED; + ret = IRQ_HANDLED; } + snd_pcm_stream_unlock(reader->substream); + return ret; } -- 1.9.1

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[alsa-devel] sound: another deadlock in snd_seq_pool_done
by Dmitry Vyukov 21 Mar '17

21 Mar '17

Hello, The following program creates processes deadlocked in snd_seq_pool_done: https://gist.githubusercontent.com/dvyukov/97efc9cb6d63b1b2c7b737b82cc8b0d6… After few seconds I get: # ps afxu | grep a.out root 8660 2.0 0.0 0 0 pts/0 Zl 16:27 0:00 [a.out] <defunct> # kill -9 8660 # cat /proc/8660/status Name: a.out State: Z (zombie) Tgid: 8660 Ngid: 0 Pid: 8660 PPid: 1 TracerPid: 0 Uid: 0 0 0 0 Gid: 0 0 0 0 FDSize: 0 Groups: 0 NStgid: 8660 NSpid: 8660 NSpgid: 8660 NSsid: 2971 Threads: 2 SigQ: 1/3304 SigPnd: 0000000000000000 ShdPnd: 0000000000000100 SigBlk: 0000000000000000 SigIgn: 0000000180000000 SigCgt: 0000000000000440 CapInh: 0000000000000000 CapPrm: 0000003fffffffff CapEff: 0000003fffffffff CapBnd: 0000003fffffffff CapAmb: 0000000000000000 NoNewPrivs: 0 Seccomp: 0 Cpus_allowed: f Cpus_allowed_list: 0-3 Mems_allowed: 00000000,00000001 Mems_allowed_list: 0 voluntary_ctxt_switches: 12 nonvoluntary_ctxt_switches: 0 # cat /proc/8660/task/*/stack [<ffffffff835406db>] snd_seq_pool_done+0x31b/0x620 sound/core/seq/seq_memory.c:436 [<ffffffff8353a11e>] snd_seq_ioctl_set_client_pool+0x1ae/0x600 sound/core/seq/seq_clientmgr.c:1836 [<ffffffff835382ba>] snd_seq_ioctl+0x2da/0x4d0 sound/core/seq/seq_clientmgr.c:2130 [<ffffffff81aced2f>] vfs_ioctl fs/ioctl.c:45 [inline] [<ffffffff81aced2f>] do_vfs_ioctl+0x1bf/0x1790 fs/ioctl.c:685 [<ffffffff81ad038f>] SYSC_ioctl fs/ioctl.c:700 [inline] [<ffffffff81ad038f>] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:691 [<ffffffff8457dc41>] entry_SYSCALL_64_fastpath+0x1f/0xc2

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[alsa-devel] [PATCH] ALSA: x86: Make CONFIG_SND_X86 bool
by Takashi Iwai 21 Mar '17

21 Mar '17

CONFIG_SND_X86 is a menu config to filter only for x86-specific drivers in its sub-menu, and this doesn't have to be tristate but rather it should be a bool. Also, like other sub-menu configs, it's more user-friendly to be default=y; it's merely a menu config and the actual drivers are configured in the sub-menu, after all. Fixes: 287599cf2d77 ("ALSA: add Intel HDMI LPE audio driver for BYT/CHT-T") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/x86/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 84c8f8fc597c..8adf4d1bd46e 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -1,6 +1,7 @@ menuconfig SND_X86 - tristate "X86 sound devices" + bool "X86 sound devices" depends on X86 + default y ---help--- X86 sound devices that don't fall under SoC or PCI categories -- 2.11.1

1 0

[alsa-devel] [PATCH v4 6/6] pcm:plug: save converter config
by sutar.mounesh＠gmail.com 21 Mar '17

21 Mar '17

From: Andreas Pape <apape(a)de.adit-jv.com> Passed config is freed after call to open, thus it is invalid when trying to extract the converter name. So config entry is saved for later usage. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index 266707b..f1ae814 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -64,6 +64,10 @@ static int snd_pcm_plug_close(snd_pcm_t *pcm) snd_pcm_plug_t *plug = pcm->private_data; int err, result = 0; free(plug->ttable); + if (plug->rate_converter) { + snd_config_delete((snd_config_t*)plug->rate_converter); + plug->rate_converter = NULL; + } assert(plug->gen.slave == plug->req_slave); if (plug->gen.close_slave) { snd_pcm_unlink_hw_ptr(pcm, plug->req_slave); @@ -1108,7 +1112,6 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, plug->sformat = sformat; plug->schannels = schannels; plug->srate = srate; - plug->rate_converter = rate_converter; plug->gen.slave = plug->req_slave = slave; plug->gen.close_slave = close_slave; plug->route_policy = route_policy; @@ -1125,6 +1128,14 @@ int snd_pcm_plug_open(snd_pcm_t **pcmp, pcm->ops = &snd_pcm_plug_ops; pcm->fast_ops = slave->fast_ops; pcm->fast_op_arg = slave->fast_op_arg; + if (rate_converter) { + if ((err = snd_config_copy((snd_config_t**)&plug->rate_converter, + (snd_config_t *)rate_converter)) < 0) { + snd_pcm_free(pcm); + free(plug); + return err; + } + } pcm->private_data = plug; pcm->poll_fd = slave->poll_fd; pcm->poll_events = slave->poll_events; -- 2.7.4

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Re: [alsa-devel] [RESEND PATCH v2 1/2] ASoC: stm32: add bindings for SAI
by Olivier MOYSAN 21 Mar '17

21 Mar '17

Hello Rob, On 03/18/2017 09:43 PM, Rob Herring wrote: > On Thu, Mar 16, 2017 at 5:50 AM, Olivier MOYSAN <olivier.moysan(a)st.com> wrote: >> Hello Rob, >> >> Thanks for your feedback. >> >> On 03/15/2017 06:46 PM, Rob Herring wrote: >>> On Tue, Mar 07, 2017 at 10:44:31AM +0100, olivier moysan wrote: >>>> This patch adds documentation of device tree bindings for the >>>> STM32 SAI ASoC driver. >>>> >>>> Signed-off-by: olivier moysan <olivier.moysan(a)st.com> >>>> --- >>>> .../devicetree/bindings/sound/st,stm32-sai.txt | 79 ++++++++++++++++++++++ >>>> 1 file changed, 79 insertions(+) >>>> create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt >>>> >>>> diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt >>>> new file mode 100644 >>>> index 0000000..b3be26d >>>> --- /dev/null >>>> +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt >>>> @@ -0,0 +1,79 @@ >>>> +STMicroelectronics STM32 Serial Audio Interface (SAI). >>>> + >>>> +The SAI interface (Serial Audio Interface) offers a wide set of audio protocols >>>> +as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. >>>> +The SAI contains two independent audio sub-blocks. Each sub-block has >>>> +its own clock generator and I/O lines controller. >>>> + >>>> +Required properties: >>>> + - compatible: Should be "st,stm32f4-sai" >>>> + - reg: Base address and size of SAI common register set. >>>> + - clocks: Must contain phandle and clock specifier pairs for each entry >>>> + in clock-names. >>>> + - clock-names: Must contain "clk_x8k" and "clk_x11k" >>>> + "clk_x8k": SAI parent clock for sampling rates multiple of 8kHz. >>>> + "clk_x11k": SAI parent clock for sampling rates multiple of 11.025kHz. >>> >>> clk_ is redundant. >>> >> >> Do you mean, removing clk_ prefix ? > > Yes. > >> >>>> + - interrupts: cpu DAI interrupt line shared by SAI sub-blocks >>>> + >>>> +Note: Each SAI controller should have an alias correctly numbered >>>> +in "aliases" node, to provide explicit name to CPU DAI. >>> >>> No, it shouldn't. Why do you need that? >>> >> >> Yes, this is not mandatory. This is convenient to identify sai >> controller, as we may have up to 8 instances. >> I can make it optional, or remove it if you think it is not relevant. > > Remove it. > One more thing regarding aliases. In future patches, I will add alsa controls to configure sai. Some stm32 socs exhibit several sai (4 instances for H7) It will be necessary to identify to which sai each control apply, through the control names: SAI1A control X SAI2A control X ... sub-node compatible already allows to identify A or B sub-instance. An alternative to alias, may be to use dev_name information to identify sai instance. But this does not produce user friendly control names. Another way is to use compatible also. The drawback here is to multiply compatible names. (many soc families and many instances) What is your recommendation ? If there is another solution, I did not consider, please let me know. >> >>> >>>> + >>>> +Optional properties: >>>> + - resets: Reference to a reset controller asserting the SAI >>>> + >>>> +SAI subnodes: >>>> +Two subnodes corresponding to SAI sub-block instances A et B can be defined. >>>> +Subnode can be omitted for unsused sub-block. >>>> + >>>> +SAI subnodes required properties: >>>> + - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b" >>>> + for SAI sub-block A or B respectively. >>>> + - reg: Base address and size of SAI sub-block register set. >>>> + - clocks: Must contain one phandle and clock specifier pair >>>> + for sai_ck which feeds the internal clock generator. >>>> + - clock-names: Must contain "sai_ck". >>>> + - dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt >>>> + - dma-names: identifier string for each DMA request line >>>> + "tx": if sai sub-block is configured as playback DAI >>>> + "rx": if sai sub-block is configured as capture DAI >>>> + - pinctrl-names: should contain only value "default" >>>> + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/pinctrl-stm32.txt >>>> + >>>> +Example: >>>> +sai1: sai1@40015800 { >>>> + compatible = "st,stm32f4-sai"; >>>> + reg = <0x40015800 0x400>; >> >> reg = <0x40015800 0x4>; >> >>>> + clocks = <&rcc 1 CLK_SAIQ_PDIV>, <&rcc 1 CLK_I2SQ_PDIV>; >>>> + clock-names = "clk_x8k", "clk_x11k"; >>>> + interrupts = <87>; >>>> + resets = <&rcc 310>; >>>> + >>>> + sai1b: sai@1 { >>> >>> unit address needs a reg property. >>> >>> audio-controller@1 >>> >> >> Right. This needs to be updated as follows: >> sai1b: sai1b@40015824 { > > audio-controller@... > >> ... >> reg = <0x40015824 0x1C>; >> ... >> }; >> >>>> + #sound-dai-cells = <0>; >>>> + compatible = "st,stm32-sai-sub-b"; >>>> + clocks = <&rcc 1 CLK_SAI2>; >>>> + clock-names = "sai_ck"; >>>> + dmas = <&dma2 5 0 0x400 0x1>; >>>> + dma-names = "tx"; >>>> + }; >>>> +}; >>>> + >>>> +sound { >>>> + compatible = "simple-audio-card"; >>> >>> Perhaps you want to use the graph-card here. It's getting close to >>> merging. >>> >> >> mclk-fs property is required when SAI is configured as master. >> It seems that mclk-fs is not supported in current version >> of audio graph card. So I thing I should keep simple card for the time >> being. Anyway, I plan further updates for SAI, so I will >> have opportunity to change to graph card at that time. > > Well, it's not easy to switch later and maintain compatibility. Is it > complicated to add mclk-fs property? The intent is graph-card supports > everything simple-card supports. > > Rob > Right, it should not be too complicated to add mclk-fs support to graph-card. BRs olivier

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