Alsa-devel September 2018

alsa-devel@alsa-project.org

98 participants
231 discussions

[alsa-devel] [PATCH alsa-lib] ioplug: Fix comment in struct snd_pcm_ioplug
by Andre Guedes 02 Oct '18

02 Oct '18

This patch fixes the comment about 'pcm' field in 'struct snd_pcm_ioplug' which wrongly refers to snd_pcm_extplug_create(). Signed-off-by: Andre Guedes <andre.guedes(a)intel.com> --- include/pcm_ioplug.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/pcm_ioplug.h b/include/pcm_ioplug.h index b16fc8b8..81ac8613 100644 --- a/include/pcm_ioplug.h +++ b/include/pcm_ioplug.h @@ -105,7 +105,7 @@ struct snd_pcm_ioplug { */ void *private_data; /** - * PCM handle filled by #snd_pcm_extplug_create() + * PCM handle filled by #snd_pcm_ioplug_create() */ snd_pcm_t *pcm; -- 2.14.4

3 2

[alsa-devel] [PATCH - Multiple plugins 0/4] Fix leaks on multiple plugins
by Andre Guedes 02 Oct '18

02 Oct '18

Hi all, While I was working on AAF plugin RFC v2 series [1], I spotted several leaks in some plugins when handling *_hw_constraint() error cases. This series fixes these issues. Regards, Andre [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-September/140290.… Andre Guedes (4): a52: Fix leaks when a52_set_hw_constraint() fails jack: Fix leaks when jack_set_hw_constraint() fails oss: Fix leaks when oss_hw_constraint() fails usb_stream: Fix leaks when us_set_hw_constraint() fails a52/pcm_a52.c | 2 +- jack/pcm_jack.c | 1 + oss/pcm_oss.c | 2 +- usb_stream/pcm_usb_stream.c | 1 + 4 files changed, 4 insertions(+), 2 deletions(-) -- 2.14.4

3 6

[alsa-devel] [PATCH] ALSA: firewire-lib: add PCM rules to obsolete PCM constraints based on LCM of SYT_INTERVAL
by Takashi Sakamoto 02 Oct '18

02 Oct '18

In blocking mode of IEC 61883-1/6, when one isochronous packet includes data for events, the data is for the same number of events as SYT_INTERVAL decided according to sampling transmission frequency (SFC). IEC 61883-1/6 engine of ALSA firewire stack applies constraints of period and buffer size of PCM intermediate buffer of PCM substream. At present, this constraint is designed to round the size up/down to 32 frames. This value comes from the least common multiple (LCM) of SYT_INTERVAL. Although this looks to work well, in lower sampling rate, applications are not allowed to set size of period quite near period time constraint (at present 5 msec per period). This commit adds PCM rules for period/buffer size and rate to obsoletes the constraints based on LCM. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 78 +++++++++++++++++++++++++++++++---- 1 file changed, 71 insertions(+), 7 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index cb9acfe60f6a..fcd965f1d69e 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -140,6 +140,59 @@ const unsigned int amdtp_rate_table[CIP_SFC_COUNT] = { }; EXPORT_SYMBOL(amdtp_rate_table); +static int apply_constraint_to_size(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *s = hw_param_interval(params, rule->var); + const struct snd_interval *r = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval t = { + .min = s->min, .max = s->max, .integer = 1, + }; + int i; + + for (i = 0; i < CIP_SFC_COUNT; ++i) { + unsigned int rate = amdtp_rate_table[i]; + unsigned int step = amdtp_syt_intervals[i]; + + if (!snd_interval_test(r, rate)) + continue; + + t.min = roundup(t.min, step); + t.max = rounddown(t.max, step); + } + + if (snd_interval_checkempty(&t)) + return -EINVAL; + + return snd_interval_refine(s, &t); +} + +static int apply_constraint_to_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + const struct snd_interval *s = hw_param_interval_c(params, rule->deps[0]); + struct snd_interval t = { + .min = UINT_MAX, .max = 0, .integer = 1, + }; + int i; + + for (i = 0; i < CIP_SFC_COUNT; ++i) { + unsigned int step = amdtp_syt_intervals[i]; + unsigned int rate = amdtp_rate_table[i]; + + if (s->min % step || s->max % step) + continue; + + t.min = min(t.min, rate); + t.max = max(t.max, rate); + } + + return snd_interval_refine(r, &t); +} + /** * amdtp_stream_add_pcm_hw_constraints - add hw constraints for PCM substream * @s: the AMDTP stream, which must be initialized. @@ -194,16 +247,27 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, * number equals to SYT_INTERVAL. So the number is 8, 16 or 32, * depending on its sampling rate. For accurate period interrupt, it's * preferrable to align period/buffer sizes to current SYT_INTERVAL. - * - * TODO: These constraints can be improved with proper rules. - * Currently apply LCM of SYT_INTERVALs. */ - err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto end; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + apply_constraint_to_rate, NULL, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + if (err < 0) + goto end; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + apply_constraint_to_size, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto end; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + apply_constraint_to_rate, NULL, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); if (err < 0) goto end; - err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32); end: return err; } -- 2.17.1

2 1

[alsa-devel] [PATCH][alsa-lib] timer: fix wrong comment to refer to 'SNDRV_TIMER_PSFLG_*'
by Takashi Sakamoto 02 Oct '18

02 Oct '18

ALSA timer core has a comment referring to 'SNDRV_MIXER_PSFLG_*' in a definition of 'struct snd_timer_params' of UAPI header. I can see this in initial state of ALSA timer core, at least in 'alsa-driver-0.4.0.tar.gz'. This commit fixes the comment. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 6041cab2..4d35c84d 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -750,7 +750,7 @@ struct snd_timer_info { #define SNDRV_TIMER_PSFLG_EARLY_EVENT (1<<2) /* write early event to the poll queue */ struct snd_timer_params { - unsigned int flags; /* flags - SNDRV_MIXER_PSFLG_* */ + unsigned int flags; /* flags - SNDRV_TIMER_PSFLG_* */ unsigned int ticks; /* requested resolution in ticks */ unsigned int queue_size; /* total size of queue (32-1024) */ unsigned int reserved0; /* reserved, was: failure locations */ -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: timer: fix wrong comment to refer to 'SNDRV_TIMER_PSFLG_*'
by Takashi Sakamoto 02 Oct '18

02 Oct '18

ALSA timer core has a comment referring to 'SNDRV_MIXER_PSFLG_*' in a definition of 'struct snd_timer_params' of UAPI header. I can see this in initial state of ALSA timer core, at least in 'alsa-driver-0.4.0.tar.gz'. This commit fixes the comment. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index ed0a120d4f08..404d4b9ffe76 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -752,7 +752,7 @@ struct snd_timer_info { #define SNDRV_TIMER_PSFLG_EARLY_EVENT (1<<2) /* write early event to the poll queue */ struct snd_timer_params { - unsigned int flags; /* flags - SNDRV_MIXER_PSFLG_* */ + unsigned int flags; /* flags - SNDRV_TIMER_PSFLG_* */ unsigned int ticks; /* requested resolution in ticks */ unsigned int queue_size; /* total size of queue (32-1024) */ unsigned int reserved0; /* reserved, was: failure locations */ -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: hda: Fix mismatch for register mask and value in ext controller.
by Keyon Jie 02 Oct '18

02 Oct '18

E.g. for snd_hdac_ext_bus_link_power_up(), we should set mask to be AZX_MLCTL_SPA(it was 0), and AZX_MLCTL_SPA as value to power up it, here correct it and several similar mismatches. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- sound/hda/ext/hdac_ext_controller.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 5bc4a1d587d4..60cb00fd0c69 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -48,9 +48,11 @@ void snd_hdac_ext_bus_ppcap_enable(struct hdac_bus *bus, bool enable) } if (enable) - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_GPROCEN); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_GPROCEN, AZX_PPCTL_GPROCEN); else - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_GPROCEN, 0); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_GPROCEN, 0); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_enable); @@ -68,9 +70,11 @@ void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_bus *bus, bool enable) } if (enable) - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, 0, AZX_PPCTL_PIE); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_PIE, AZX_PPCTL_PIE); else - snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, AZX_PPCTL_PIE, 0); + snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, + AZX_PPCTL_PIE, 0); } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_ppcap_int_enable); @@ -194,7 +198,8 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) */ int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link) { - snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA); + snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, AZX_MLCTL_SPA); return check_hdac_link_power_active(link, true); } @@ -222,8 +227,8 @@ int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus) int ret; list_for_each_entry(hlink, &bus->hlink_list, list) { - snd_hdac_updatel(hlink->ml_addr, - AZX_REG_ML_LCTL, 0, AZX_MLCTL_SPA); + snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, AZX_MLCTL_SPA); ret = check_hdac_link_power_active(hlink, true); if (ret < 0) return ret; @@ -243,7 +248,8 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus) int ret; list_for_each_entry(hlink, &bus->hlink_list, list) { - snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0); + snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, + AZX_MLCTL_SPA, 0); ret = check_hdac_link_power_active(hlink, false); if (ret < 0) return ret; -- 2.17.1

2 1

[alsa-devel] Re-reading alsa-lib configuration when opening sound device
by Pavel Hofman 02 Oct '18

02 Oct '18

Hi, I am playing with virtual balanced inputs https://www.diyaudio.com/forums/pc-based/326747-virtual-balanced-regular-so… . For proper functionality it needs a regular recalibration of the ttable multiplication coefficients of route plugin in .asoundrc. The audio stack is alsa -> wine -> Arta measurement software. When I stop measuring in arta (without exiting the software), the soundcard gets closed - confirmed in /proc/asound/.../hw_params. Now I change the ttable coeffs. After re-opening the device (continuing with measurement in the running arta instance) the changes in .asoundrc are ignored. I have to restart the arta application for the changes to take effect. When I run e.g. aplay inbetween, it accepts the changes right away. Upon continuing in arta, the original ttable coeffs are still loaded for this instance. I understand it is the way wine handles the device somewhere in https://github.com/wine-mirror/wine/blob/master/dlls/winealsa.drv/mmdevdrv.c . Please when is the alsa-lib config loaded by the library/application? Is there any way to force a config reload from outside? Would e.g. calling snd_config_update before snd_pcm_open on line https://github.com/wine-mirror/wine/blob/master/dlls/winealsa.drv/mmdevdrv.… help? If so, what overhead does it impose? Perhaps I could ask the wine guys to consider such change if it worked and required minimum overhead. I very much appreciate any help. Best regards, Pavel.

2 4

[alsa-devel] alsactl monitor spins at 100% cpu after unplugging USB headphones
by Thomas Gläßle 02 Oct '18

02 Oct '18

Hi, on both my desktop and laptop (archlinux), `alsactl monitor` beginns spinning at 100% cpu when I unplug my USB headset. This happens every time and can be reproduced as follows: - plug in USB headphones - start `alsactl monitor` - unplug headphones It happens only if the headphones are already plugged in when starting alsactl. The increased activity does not lead to spamming stdout or similar, alsactl continues to perform seemingly correct, i.e. reports when I change volume levels etc, but nothing apart from the ordinary. I'm sorry if this is not the right place to report this issue or if the problem is already, I couldn't find anything about it. Below is the output from alsa-info.sh: with headphones: http://www.alsa-project.org/db/?f=0dfc632e3adafb8bc78a182e0a225bcb13bea9fa after unplugging: http://www.alsa-project.org/db/?f=41dbf0735fdd7cd2416a6b4a602505715d25b44f Please let me know if you need further info. Best regards, Thomas

2 2

[alsa-devel] [PATCH linux-next 10/10] ASoC: rsnd: Add kctrl to configure dai's busif
by jiada_wang＠mentor.com 02 Oct '18

02 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> ssi0, ssi1, ssi2, ssi3, ssi4 and ssi9 have busif0 to busif7, currently only busif0 is used by all ssi. But tdm split and tdm ex-split mode, which require busif1 to busif7. This patch adds kctrl in ssi for dai-link, to make it possible to configure the busif used by each dai-link. Also adds rsnd_dma_addr_update() interface to make it possible to update dma data address when there is change in condition of dai connection. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/dma.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 2 ++ sound/soc/sh/rcar/ssi.c | 31 ++++++++++++++++++++++++++++++- 3 files changed, 64 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 6d1947515dc8..8c284fd73fc1 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -792,6 +792,38 @@ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, return rsnd_dai_connect(*dma_mod, io, (*dma_mod)->type); } +void rsnd_dma_addr_update(struct rsnd_dai_stream *io, + struct rsnd_mod *mod, + struct rsnd_mod *dma_mod) +{ + struct rsnd_mod *mod_from = NULL; + struct rsnd_mod *mod_to = NULL; + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct rsnd_dma_ctrl *dmac = rsnd_priv_to_dmac(priv); + struct device *dev = rsnd_priv_to_dev(priv); + int is_play = rsnd_io_is_play(io); + struct rsnd_dma *dma = rsnd_mod_to_dma(dma_mod); + + if (!dmac) + return; + + rsnd_dma_of_path(mod, io, is_play, &mod_from, &mod_to); + + dma->src_addr = rsnd_dma_addr(io, mod_from, is_play, 1); + dma->dst_addr = rsnd_dma_addr(io, mod_to, is_play, 0); + + if (mod_from && mod_to) { + struct rsnd_dmapp *dmapp = rsnd_dma_to_dmapp(dma); + + dmapp->chcr = rsnd_dmapp_get_chcr(io, mod_from, mod_to) | + PDMACHCR_DE; + } + + dev_dbg(dev, "%s[%d] %pad -> %pad\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), + &dma->src_addr, &dma->dst_addr); +} + int rsnd_dma_probe(struct rsnd_priv *priv) { struct platform_device *pdev = rsnd_priv_to_pdev(priv); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 59b6d89d8edc..6c16d8403f7a 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -233,6 +233,8 @@ u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod); */ int rsnd_dma_attach(struct rsnd_dai_stream *io, struct rsnd_mod *mod, struct rsnd_mod **dma_mod); +void rsnd_dma_addr_update(struct rsnd_dai_stream *io, + struct rsnd_mod *mod, struct rsnd_mod *dma_mod); int rsnd_dma_probe(struct rsnd_priv *priv); struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, struct rsnd_mod *mod, char *name); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4ac4b5b75ae2..b469db238697 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -160,6 +160,15 @@ int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) return io->busif->val; } +static void rsnd_ssi_set_busif(struct rsnd_dai_stream *io, + struct rsnd_mod *mod) +{ + if (!rsnd_ssi_use_busif(io)) + return; + + rsnd_dma_addr_update(io, mod, io->dma); +} + static void rsnd_ssi_status_clear(struct rsnd_mod *mod) { rsnd_mod_write(mod, SSISR, 0); @@ -718,6 +727,10 @@ static int rsnd_ssi_pcm_new(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct snd_soc_pcm_runtime *rtd) { + int is_play = rsnd_io_is_play(io); + int id = rsnd_mod_id(mod); + int ret = 0; + /* * rsnd_rdai_is_clk_master() will be enabled after set_fmt, * and, pcm_new will be called after it. @@ -725,7 +738,23 @@ static int rsnd_ssi_pcm_new(struct rsnd_mod *mod, */ rsnd_ssi_parent_attach(mod, io); - return 0; + if (!rsnd_ssi_use_busif(io)) + return 0; + + if (rsnd_ssi_is_multi_slave(mod, io)) + return 0; + + /* SSI5 to SSI8 only have one BUSIF */ + if (id <= 4 || id == 9) + ret = rsnd_kctrl_new_s(mod, io, rtd, + is_play ? + "SSI Out BUSIF" : + "SSI In BUSIF", + rsnd_kctrl_accept_anytime, + rsnd_ssi_set_busif, + io->busif, 7); + + return ret; } static int rsnd_ssi_common_probe(struct rsnd_mod *mod, -- 2.17.0

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[alsa-devel] [PATCH linux-next 09/10] ASoC: rsnd: add busif property to dai stream
by jiada_wang＠mentor.com 02 Oct '18

02 Oct '18

From: Jiada Wang <jiada_wang(a)mentor.com> SSI may use different busif for data transfer, this patch adds busif property to each dai stream, to indicate the busif used by playback/capture stream. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/sh/rcar/core.c | 16 +++++++++++++++- sound/soc/sh/rcar/rsnd.h | 3 +++ sound/soc/sh/rcar/ssi.c | 5 ++++- 3 files changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 40d7dc4f7839..aee3132c72b9 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -560,6 +560,14 @@ struct rsnd_dai *rsnd_rdai_get(struct rsnd_priv *priv, int id) return priv->rdai + id; } +struct rsnd_kctrl_cfg_s *rsnd_busif_get(struct rsnd_priv *priv, int id) +{ + if ((id < 0) || (id >= 2 * rsnd_rdai_nr(priv))) + return NULL; + + return priv->busif + id; +} + static struct snd_soc_dai_driver *rsnd_daidrv_get(struct rsnd_priv *priv, int id) { @@ -1045,6 +1053,9 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, io_playback = &rdai->playback; io_capture = &rdai->capture; + io_playback->busif = rsnd_busif_get(priv, dai_i); + io_capture->busif = rsnd_busif_get(priv, rsnd_rdai_nr(priv) + dai_i); + snprintf(rdai->name, RSND_DAI_NAME_SIZE, "rsnd-dai.%d", dai_i); rdai->priv = priv; @@ -1108,6 +1119,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) struct snd_soc_dai_driver *rdrv; struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_dai *rdai; + struct rsnd_kctrl_cfg_s *busif; int nr; int is_graph; int dai_i; @@ -1123,12 +1135,14 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) rdrv = devm_kcalloc(dev, nr, sizeof(*rdrv), GFP_KERNEL); rdai = devm_kcalloc(dev, nr, sizeof(*rdai), GFP_KERNEL); - if (!rdrv || !rdai) + busif = devm_kcalloc(dev, 2 * nr, sizeof(*busif), GFP_KERNEL); + if (!rdrv || !rdai || !busif) return -ENOMEM; priv->rdai_nr = nr; priv->daidrv = rdrv; priv->rdai = rdai; + priv->busif = busif; /* * parse all dai diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 4464d1d0a042..59b6d89d8edc 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -456,6 +456,7 @@ struct rsnd_dai_stream { struct rsnd_mod *dma; struct rsnd_dai *rdai; struct device *dmac_dev; /* for IPMMU */ + struct rsnd_kctrl_cfg_s *busif; u32 parent_ssi_status; }; #define rsnd_io_to_mod(io, i) ((i) < RSND_MOD_MAX ? (io)->mod[(i)] : NULL) @@ -629,6 +630,8 @@ struct rsnd_priv { */ struct snd_soc_dai_driver *daidrv; struct rsnd_dai *rdai; + struct rsnd_kctrl_cfg_s *busif; + int rdai_nr; }; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 75a3d15084c2..4ac4b5b75ae2 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -154,7 +154,10 @@ int rsnd_ssi_use_busif(struct rsnd_dai_stream *io) int rsnd_ssi_get_busif(struct rsnd_dai_stream *io) { - return 0; /* BUSIF0 only for now */ + if (!rsnd_ssi_use_busif(io)) + return 0; + + return io->busif->val; } static void rsnd_ssi_status_clear(struct rsnd_mod *mod) -- 2.17.0

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Alsa-devel September 2018