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

[alsa-devel] [PATCH 0/2] Update codec driver for dual mics configuration
by Carlo Caione 29 Nov '17

29 Nov '17

From: Carlo Caione <carlo(a)endlessm.com> The RT5651 codec has 3 analog input ports. Some laptops (like the KIANO laptop) have two different analog mics on two ports, leaving the headset on the remaining one. We need to add a new quirk to support also this configuration. Carlo Caione (2): ASoC: intel: byt: Introduce new map for dual mics ASoC: intel: byt: Fix quirk for KIANO laptop sound/soc/intel/boards/bytcr_rt5651.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) -- 2.14.1

2 6

[alsa-devel] [PATCH] ALSA: usb-audio: Fix the enable parameter behavior
by Tamaki Nishino 29 Nov '17

29 Nov '17

The enable parameter doesn't seem to work as expected. Even when the enable parameter is set to false in /etc/modprobe.d/alsa-base.conf like "options snd-usb-audio index=-2 vid=0x1415 pid=0x1d27 enable=0", the specified device is not disabled. This patch fixes the enable parameter behavior to disable the device correctly. Signed-off-by: Tamaki Nishino <otamachan(a)gmail.com> --- sound/usb/card.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 23d1d23..8018d56 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -585,15 +585,24 @@ static int usb_audio_probe(struct usb_interface *intf, * now look for an empty slot and create a new card instance */ for (i = 0; i < SNDRV_CARDS; i++) - if (enable[i] && ! usb_chip[i] && + if (!usb_chip[i] && (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) && (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) { - err = snd_usb_audio_create(intf, dev, i, quirk, - id, &chip); - if (err < 0) + if (enable[i]) { + err = snd_usb_audio_create(intf, dev, i, quirk, + id, &chip); + if (err < 0) + goto __error; + chip->pm_intf = intf; + break; + } else if (vid[i] != -1 || pid[i] != -1) { + dev_info(&dev->dev, + "device (%04x:%04x) is disabled\n", + USB_ID_VENDOR(id), + USB_ID_PRODUCT(id)); + err = -ENOENT; goto __error; - chip->pm_intf = intf; - break; + } } if (!chip) { dev_err(&dev->dev, "no available usb audio device\n"); -- 2.7.4

1 0

[alsa-devel] Applied "ASoC: soc-core: add missing EXPORT_SYMBOL_GPL() for snd_soc_disconnect_sync" to the asoc tree
by Mark Brown 29 Nov '17

29 Nov '17

The patch ASoC: soc-core: add missing EXPORT_SYMBOL_GPL() for snd_soc_disconnect_sync has been applied to the asoc tree at https://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 df532185e8720baff120f55eb46058d270445d56 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 29 Nov 2017 02:38:27 +0000 Subject: [PATCH] ASoC: soc-core: add missing EXPORT_SYMBOL_GPL() for snd_soc_disconnect_sync Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- 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 9047046c749c..345baa4f10c0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1401,6 +1401,7 @@ void snd_soc_disconnect_sync(struct device *dev) snd_card_disconnect_sync(component->card->snd_card); } +EXPORT_SYMBOL_GPL(snd_soc_disconnect_sync); /** * snd_soc_add_dai_link - Add a DAI link dynamically -- 2.15.0

1 0

[alsa-devel] [PATCH] ASoC: rsnd: TDM 6ch needs 8ch clock for hw refine
by Kuninori Morimoto 29 Nov '17

29 Nov '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas sound needs 8ch clock if TDM 6ch mode, and needs 2ch clock for 6ch or 8ch sound if Multi SSI mode. And these are related to before/after CTU (= Channel Transfer Unit). To calculate these we already has rsnd_runtime_channel_for_ssi() which returns runtime necessary channels. But, it based on runtime->channels which is not yet set when hw refine. We need to use hw_params instead of runtime->xxx when hw refine, and it is not needed after runtime was set. This patch adds new hw_params on rsnd_dai_stream, and it will be removed on rsnd_hw_params(). This is very temporary durty code, but it seems no choice at this point. Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 92 +++++++++++++++++++++++++++++++++++------------- sound/soc/sh/rcar/rsnd.h | 15 ++++++-- 2 files changed, 80 insertions(+), 27 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index a40f6a0..fa3d911 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -197,16 +197,27 @@ int rsnd_io_is_working(struct rsnd_dai_stream *io) return 0; } -int rsnd_runtime_channel_original(struct rsnd_dai_stream *io) +int rsnd_runtime_channel_original_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params) { struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - return runtime->channels; + /* + * params will be added when refine + * see + * __rsnd_soc_hw_rule_rate() + * __rsnd_soc_hw_rule_channels() + */ + if (params) + return params_channels(params); + else + return runtime->channels; } -int rsnd_runtime_channel_after_ctu(struct rsnd_dai_stream *io) +int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params) { - int chan = rsnd_runtime_channel_original(io); + int chan = rsnd_runtime_channel_original_with_params(io, params); struct rsnd_mod *ctu_mod = rsnd_io_to_mod_ctu(io); if (ctu_mod) { @@ -219,12 +230,13 @@ int rsnd_runtime_channel_after_ctu(struct rsnd_dai_stream *io) return chan; } -int rsnd_runtime_channel_for_ssi(struct rsnd_dai_stream *io) +int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params) { struct rsnd_dai *rdai = rsnd_io_to_rdai(io); int chan = rsnd_io_is_play(io) ? - rsnd_runtime_channel_after_ctu(io) : - rsnd_runtime_channel_original(io); + rsnd_runtime_channel_after_ctu_with_params(io, params) : + rsnd_runtime_channel_original_with_params(io, params); /* Use Multi SSI */ if (rsnd_runtime_is_ssi_multi(io)) @@ -616,8 +628,6 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - rsnd_dai_stream_init(io, substream); - ret = rsnd_dai_call(init, io, priv); if (ret < 0) goto dai_trigger_end; @@ -639,7 +649,6 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, ret |= rsnd_dai_call(quit, io, priv); - rsnd_dai_stream_quit(io); break; default: ret = -EINVAL; @@ -784,8 +793,9 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv, return snd_interval_refine(iv, &p); } -static int rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule, + int is_play) { struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); @@ -793,25 +803,37 @@ static int rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_soc_dai *dai = rule->private; struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; /* * possible sampling rate limitation is same as * 2ch if it supports multi ssi + * and same as 8ch if TDM 6ch (see rsnd_ssi_config_init()) */ ic = *ic_; - if (1 < rsnd_rdai_ssi_lane_get(rdai)) { - ic.min = 2; - ic.max = 2; - } + ic.min = + ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params); return rsnd_soc_hw_rule(priv, rsnd_soc_hw_rate_list, ARRAY_SIZE(rsnd_soc_hw_rate_list), &ic, ir); } +static int rsnd_soc_hw_rule_rate_playback(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + return __rsnd_soc_hw_rule_rate(params, rule, 1); +} + +static int rsnd_soc_hw_rule_rate_capture(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + return __rsnd_soc_hw_rule_rate(params, rule, 0); +} -static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule, + int is_play) { struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); @@ -819,22 +841,34 @@ static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, struct snd_soc_dai *dai = rule->private; struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; /* * possible sampling rate limitation is same as * 2ch if it supports multi ssi + * and same as 8ch if TDM 6ch (see rsnd_ssi_config_init()) */ ic = *ic_; - if (1 < rsnd_rdai_ssi_lane_get(rdai)) { - ic.min = 2; - ic.max = 2; - } + ic.min = + ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params); return rsnd_soc_hw_rule(priv, rsnd_soc_hw_channels_list, ARRAY_SIZE(rsnd_soc_hw_channels_list), ir, &ic); } +static int rsnd_soc_hw_rule_channels_playback(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + return __rsnd_soc_hw_rule_channels(params, rule, 1); +} + +static int rsnd_soc_hw_rule_channels_capture(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + return __rsnd_soc_hw_rule_channels(params, rule, 0); +} + static const struct snd_pcm_hardware rsnd_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | @@ -859,6 +893,8 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, int ret; int i; + rsnd_dai_stream_init(io, substream); + /* * Channel Limitation * It depends on Platform design @@ -886,11 +922,17 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, * It depends on Clock Master Mode */ if (rsnd_rdai_is_clk_master(rdai)) { + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - rsnd_soc_hw_rule_rate, dai, + is_play ? rsnd_soc_hw_rule_rate_playback : + rsnd_soc_hw_rule_rate_capture, + dai, SNDRV_PCM_HW_PARAM_CHANNELS, -1); snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - rsnd_soc_hw_rule_channels, dai, + is_play ? rsnd_soc_hw_rule_channels_playback : + rsnd_soc_hw_rule_channels_capture, + dai, SNDRV_PCM_HW_PARAM_RATE, -1); } @@ -915,6 +957,8 @@ static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream, * call rsnd_dai_call without spinlock */ rsnd_dai_call(nolock_stop, io, priv); + + rsnd_dai_stream_quit(io); } static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 57cd2bc..ad65235 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -399,9 +399,18 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, struct device_node *playback, struct device_node *capture); -int rsnd_runtime_channel_original(struct rsnd_dai_stream *io); -int rsnd_runtime_channel_after_ctu(struct rsnd_dai_stream *io); -int rsnd_runtime_channel_for_ssi(struct rsnd_dai_stream *io); +#define rsnd_runtime_channel_original(io) \ + rsnd_runtime_channel_original_with_params(io, NULL) +int rsnd_runtime_channel_original_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params); +#define rsnd_runtime_channel_after_ctu(io) \ + rsnd_runtime_channel_after_ctu_with_params(io, NULL) +int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params); +#define rsnd_runtime_channel_for_ssi(io) \ + rsnd_runtime_channel_for_ssi_with_params(io, NULL) +int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params); int rsnd_runtime_is_ssi_multi(struct rsnd_dai_stream *io); int rsnd_runtime_is_ssi_tdm(struct rsnd_dai_stream *io); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: dma.c: spin lock is no longer needed in IRQ handler
by Kuninori Morimoto 29 Nov '17

29 Nov '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> DMA handler had needed to calculate pointer before, but it doesn't need now. Thus, we can remove unnecessary spin lock from DMAC handler. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 18 ------------------ 1 file changed, 18 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 4d750bdf..41de234 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -71,25 +71,7 @@ struct rsnd_dma_ctrl { static void __rsnd_dmaen_complete(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - bool elapsed = false; - unsigned long flags; - - /* - * Renesas sound Gen1 needs 1 DMAC, - * Gen2 needs 2 DMAC. - * In Gen2 case, it are Audio-DMAC, and Audio-DMAC-peri-peri. - * But, Audio-DMAC-peri-peri doesn't have interrupt, - * and this driver is assuming that here. - */ - spin_lock_irqsave(&priv->lock, flags); - if (rsnd_io_is_working(io)) - elapsed = true; - - spin_unlock_irqrestore(&priv->lock, flags); - - if (elapsed) rsnd_dai_period_elapsed(io); } -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: nau8825: disable crosstalk by default
by John Hsu 29 Nov '17

29 Nov '17

The driver makes the crosstalk funciton disabled by default which can simplify the codec function. The platform may not need this funciton and reduce the potential risk. Therefore, We change the property "nuvoton,crosstalk-bypass" to "nuvoton,crosstalk-enable". The crosstalk measurement is enabled if the property is set. Otherwise, it is disabled. Besides, add more condition in the entry point of the crosstalk sequence to disable the function completely. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8825.txt | 4 ++-- sound/soc/codecs/nau8825.c | 23 ++++++++++++---------- sound/soc/codecs/nau8825.h | 2 +- 3 files changed, 16 insertions(+), 13 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt index 2f5e973..d16d968 100644 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -69,7 +69,7 @@ Optional properties: - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,crosstalk-bypass: make crosstalk function bypass if set. + - nuvoton,crosstalk-enable: make crosstalk function enable if set. - clocks: list of phandle and clock specifier pairs according to common clock bindings for the clocks described in clock-names @@ -98,7 +98,7 @@ Example: nuvoton,short-key-debounce = <2>; nuvoton,jack-insert-debounce = <7>; nuvoton,jack-eject-debounce = <7>; - nuvoton,crosstalk-bypass; + nuvoton,crosstalk-enable; clock-names = "mclk"; clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 714ce17..d3c1a02 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -815,11 +815,12 @@ static void nau8825_xtalk_work(struct work_struct *work) static void nau8825_xtalk_cancel(struct nau8825 *nau8825) { - /* If the xtalk_protect is true, that means the process is still - * on going. The driver forces to cancel the cross talk task and + /* If the crosstalk is eanbled and the process is on going, + * the driver forces to cancel the crosstalk task and * restores the configuration to original status. */ - if (nau8825->xtalk_protect) { + if (nau8825->xtalk_enable && nau8825->xtalk_state != + NAU8825_XTALK_DONE) { cancel_work_sync(&nau8825->xtalk_work); nau8825_xtalk_clean(nau8825); } @@ -1686,7 +1687,7 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { if (nau8825_is_jack_inserted(regmap)) { event |= nau8825_jack_insert(nau8825); - if (!nau8825->xtalk_bypass && !nau8825->high_imped) { + if (nau8825->xtalk_enable && !nau8825->high_imped) { /* Apply the cross talk suppression in the * headset without high impedance. */ @@ -1732,8 +1733,10 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) nau8825->xtalk_event_mask = event_mask; } } else if (active_irq & NAU8825_IMPEDANCE_MEAS_IRQ) { - schedule_work(&nau8825->xtalk_work); - clear_irq = NAU8825_IMPEDANCE_MEAS_IRQ; + if (nau8825->xtalk_enable) { + schedule_work(&nau8825->xtalk_work); + clear_irq = NAU8825_IMPEDANCE_MEAS_IRQ; + } } else if ((active_irq & NAU8825_JACK_INSERTION_IRQ_MASK) == NAU8825_JACK_INSERTION_DETECTED) { /* One more step to check GPIO status directly. Thus, the @@ -2440,8 +2443,8 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) nau8825->jack_insert_debounce); dev_dbg(dev, "jack-eject-debounce: %d\n", nau8825->jack_eject_debounce); - dev_dbg(dev, "crosstalk-bypass: %d\n", - nau8825->xtalk_bypass); + dev_dbg(dev, "crosstalk-enable: %d\n", + nau8825->xtalk_enable); } static int nau8825_read_device_properties(struct device *dev, @@ -2506,8 +2509,8 @@ static int nau8825_read_device_properties(struct device *dev, &nau8825->jack_eject_debounce); if (ret) nau8825->jack_eject_debounce = 0; - nau8825->xtalk_bypass = device_property_read_bool(dev, - "nuvoton,crosstalk-bypass"); + nau8825->xtalk_enable = device_property_read_bool(dev, + "nuvoton,crosstalk-enable"); nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 8aee5c86..199d6ea 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -476,7 +476,7 @@ struct nau8825 { int xtalk_event_mask; bool xtalk_protect; int imp_rms[NAU8825_XTALK_IMM]; - int xtalk_bypass; + int xtalk_enable; }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, -- 2.6.4

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[alsa-devel] [PATCH v2 2/2] ASoC: fsl_ssi: call _fsl_ssi_set_dai_fmt() just once in AC'97 mode
by Maciej S. Szmigiero 29 Nov '17

29 Nov '17

In AC'97 mode we configure and start SSI RX / TX on probe path via a call to _fsl_ssi_set_dai_fmt() function. We don't need to call this function again later and in fact don't want to do it since this function temporarily sets STCR, SRCR and SCR to some intermediate values. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: The SACCST setup code was split out into a separate commit. sound/soc/fsl/fsl_ssi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 375aaaf6080d..70df6bf832df 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1116,6 +1116,9 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + if (fsl_ssi_is_ac97(ssi_private)) + return 0; + return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt); }

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[alsa-devel] [PATCH v2 1/2] ASoC: fsl_ssi: only enable proper channel slots in AC'97 mode
by Maciej S. Szmigiero 29 Nov '17

29 Nov '17

We need to make sure that only proper channel slots (in SACCST register) are enabled at playback start time since some AC'97 CODECs (like VT1613 on UDOO board) were observed requesting via SLOTREQ spurious ones just after an AC'97 link is started but before the CODEC is configured by its driver. When a bit for some channel slot is set in a SLOTREQ request then SSI sets the relevant bit in SACCST automatically, which then 'sticks' until it is manually unset. The SACCST register is not writable directly, we have to use SACCDIS and SACCEN registers to configure it instead (these aren't normal registers: writing a '1' bit at some position in SACCEN sets the relevant bit in SACCST; SACCDIS operates in a similar way but allows unsetting bits in SACCST). Theoretically, this should be necessary only for the very first playback but since some CODECs are so untrustworthy and extra channel slots enabled mean ruined playback let's play safe here and make sure that no extra slots are enabled in SACCST every time a playback is started. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Split out this part from "fsl_ssi: call _fsl_ssi_set_dai_fmt() just once in AC'97 mode" commit, describe the problem and its solution better both in the commit message and in the code, move the SACCST setup code into a separate function and call it from TX config instead of doing it from trigger handler function. sound/soc/fsl/fsl_ssi.c | 52 +++++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 46 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 48bb850a34d9..375aaaf6080d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -574,8 +574,54 @@ static void fsl_ssi_rx_config(struct fsl_ssi_private *ssi_private, bool enable) fsl_ssi_config(ssi_private, enable, &ssi_private->rxtx_reg_val.rx); } +static void fsl_ssi_tx_ac97_saccst_setup(struct fsl_ssi_private *ssi_private) +{ + struct regmap *regs = ssi_private->regs; + + /* no SACC{ST,EN,DIS} regs on imx21-class SSI */ + if (!ssi_private->soc->imx21regs) { + /* + * Note that these below aren't just normal registers. + * They are a way to disable or enable bits in SACCST + * register: + * - writing a '1' bit at some position in SACCEN sets the + * relevant bit in SACCST, + * - writing a '1' bit at some position in SACCDIS unsets + * the relevant bit in SACCST register. + * + * The two writes below first disable all channels slots, + * then enable just slots 3 & 4 ("PCM Playback Left Channel" + * and "PCM Playback Right Channel"). + */ + regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); + regmap_write(regs, CCSR_SSI_SACCEN, 0x300); + } +} + static void fsl_ssi_tx_config(struct fsl_ssi_private *ssi_private, bool enable) { + /* + * Why are we setting up SACCST everytime we are starting a + * playback? + * Some CODECs (like VT1613 CODEC on UDOO board) like to + * (sometimes) set extra bits in their SLOTREQ requests. + * When a bit is set in a SLOTREQ request then SSI sets the + * relevant bit in SACCST automatically (it is enough if a bit was + * set in a SLOTREQ just once, bits in SACCST are 'sticky'). + * If an extra slot gets enabled that's a disaster for playback + * because some of normal left or right channel samples are + * redirected instead to this extra slot. + * + * A workaround implemented in fsl-asoc-card of setting an + * appropriate CODEC register so that slots 3 & 4 (the normal + * stereo playback slots) are used for S/PDIF seems to mostly fix + * this issue on the UDOO board but since this CODEC is so + * untrustworthy let's play safe here and make sure that no extra + * slots are enabled every time a playback is started. + */ + if (enable && fsl_ssi_is_ac97(ssi_private)) + fsl_ssi_tx_ac97_saccst_setup(ssi_private); + fsl_ssi_config(ssi_private, enable, &ssi_private->rxtx_reg_val.tx); } @@ -630,12 +676,6 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) regmap_write(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV); - /* no SACC{ST,EN,DIS} regs on imx21-class SSI */ - if (!ssi_private->soc->imx21regs) { - regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); - regmap_write(regs, CCSR_SSI_SACCEN, 0x300); - } - /* * Enable SSI, Transmit and Receive. AC97 has to communicate with the * codec before a stream is started.

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[alsa-devel] [PATCH] ASoC: fsl_asrc: Fix typo in a field define
by Nicolin Chen 29 Nov '17

29 Nov '17

ASRFSTi_IAEi has an 11-bit offset as its _SHIFT macro defines. So this patch just fixes that. Reported-by: Laurent Charpentier <laurent.charpentier(a)nxp.com> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_asrc.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 0f163ab..52c27a35 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -260,8 +260,8 @@ #define ASRFSTi_OUTPUT_FIFO_SHIFT 12 #define ASRFSTi_OUTPUT_FIFO_MASK (((1 << ASRFSTi_OUTPUT_FIFO_WIDTH) - 1) << ASRFSTi_OUTPUT_FIFO_SHIFT) #define ASRFSTi_IAEi_SHIFT 11 -#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_OAFi_SHIFT) -#define ASRFSTi_IAEi (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_IAEi_SHIFT) +#define ASRFSTi_IAEi (1 << ASRFSTi_IAEi_SHIFT) #define ASRFSTi_INPUT_FIFO_WIDTH 7 #define ASRFSTi_INPUT_FIFO_SHIFT 0 #define ASRFSTi_INPUT_FIFO_MASK ((1 << ASRFSTi_INPUT_FIFO_WIDTH) - 1) -- 2.1.4

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[alsa-devel] [PATCH] ASoC: soc-utils: add const to dummy_codec
by Kuninori Morimoto 29 Nov '17

29 Nov '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Let's align dummy_platform and dummy_codec Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index e30aacb..bcd3da2 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -288,7 +288,7 @@ static int dummy_dma_open(struct snd_pcm_substream *substream) .ops = &dummy_dma_ops, }; -static struct snd_soc_codec_driver dummy_codec; +static const struct snd_soc_codec_driver dummy_codec; #define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ -- 1.9.1

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