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

[alsa-devel] [PATCH] ALSA: usb-audio: fix CM6206 register definitions
by Amadeusz Sławiński 11 Jan '19

11 Jan '19

fix typo after a recent commit causing headphones to have no sound Fixes: ad43d528a7ac (ALSA: usb-audio: Define registers for CM6206) Signed-off-by: Amadeusz Sławiński <amade(a)asmblr.net> --- sound/usb/quirks.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 96340f23f86d..ebbadb3a7094 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -768,7 +768,7 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) * REG1: PLL binary search enable, soft mute enable. */ CM6206_REG1_PLLBIN_EN | - CM6206_REG1_SOFT_MUTE_EN | + CM6206_REG1_SOFT_MUTE_EN, /* * REG2: enable output drivers, * select front channels to the headphone output, -- 2.20.1

3 2

[alsa-devel] [PATCH] ASoC: core: Fix deferral of machine drivers
by Jon Hunter 11 Jan '19

11 Jan '19

Commit daecf46ee0e5 ("ASoC: soc-core: use snd_soc_dai_link_component for platform") added a new member to the snd_soc_dai_link structure for storing a pointer for a platform link component. The pointer for this platform link component was allocated, if not already populated by the machine driver, using devm_kzalloc() such that the memory would be automatically freed on error or removal of the soundcard. However, this introduced a new problem, because if the probing of the soundcard is deferred, then although the memory allocated for the platform link component is freed, if the snd_soc_dai_link structure is declared statically by the machine driver, then the pointer in the DAI link structure will never be clearer. This means that when the soundcard is probed again, memory for the platform link component will not be allocated again because the address of the pointer was not cleared and this causes sound core to access memory that is no longer valid. In most cases this causes the following error condition to be triggered and causes probing the soundcard to fail. tegra-snd-sgtl5000 sound: ASoC: Both platform name/of_node are set for sgtl5000 Unfortunately, because this platform link component is allocated before the DAI links are added to the soundcard, there is no easy way to clear this pointer on teardown if an error occurs. The pointer for this platform link component was added for future proofing and communalising the structures for storing various data. Although a machine driver maybe used by more than one platform and so this platform data may vary from platform to platform, there is only ever a single instance for a given platform. Therefore, rather than dynamically allocate the platform link component structure, make it a static member of the snd_soc_dai_link to fix the problem. It should be noted that if the platform_name of platform_of_node members of the snd_soc_dai_link structure are populated, these will always be used regardless of if the new platform.name or platform.of_node members are populated. Fixes: daecf46ee0e5 ("ASoC: soc-core: use snd_soc_dai_link_component for platform") Reported-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- include/sound/simple_card_utils.h | 2 +- include/sound/soc.h | 2 +- sound/soc/generic/audio-graph-card.c | 4 +++- sound/soc/generic/simple-card-utils.c | 4 ++-- sound/soc/generic/simple-card.c | 6 ++++-- sound/soc/soc-core.c | 18 +++++++----------- 6 files changed, 18 insertions(+), 18 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 6d69ed2bd7b1..6d5842c3c09f 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -75,7 +75,7 @@ void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai); &dai_link->codec_dai_name, \ list_name, cells_name, NULL) #define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name) \ - asoc_simple_card_parse_dai(node, dai_link->platform, \ + asoc_simple_card_parse_dai(node, &dai_link->platform, \ &dai_link->platform_of_node, \ NULL, list_name, cells_name, NULL) int asoc_simple_card_parse_dai(struct device_node *node, diff --git a/include/sound/soc.h b/include/sound/soc.h index 8ec1de856ee7..8b7ffc60006a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -925,7 +925,7 @@ struct snd_soc_dai_link { */ const char *platform_name; struct device_node *platform_of_node; - struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link_component platform; int id; /* optional ID for machine driver link identification */ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 3ec96cdc683b..e961d45ce141 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -687,7 +687,9 @@ static int graph_probe(struct platform_device *pdev) for (i = 0; i < li.link; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; + dai_link[i].platform.name = dai_props[i].platform.name; + dai_link[i].platform.of_node = dai_props[i].platform.of_node; + dai_link[i].platform.dai_name = dai_props[i].platform.dai_name; } priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 336895f7fd1e..74910c7841ec 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -397,8 +397,8 @@ EXPORT_SYMBOL_GPL(asoc_simple_card_init_dai); int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link) { /* Assumes platform == cpu */ - if (!dai_link->platform->of_node) - dai_link->platform->of_node = dai_link->cpu_of_node; + if (!dai_link->platform.of_node) + dai_link->platform.of_node = dai_link->cpu_of_node; return 0; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 479de236e694..b6402e09bba2 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -732,7 +732,9 @@ static int simple_probe(struct platform_device *pdev) for (i = 0; i < li.link; i++) { dai_link[i].codecs = &dai_props[i].codecs; dai_link[i].num_codecs = 1; - dai_link[i].platform = &dai_props[i].platform; + dai_link[i].platform.name = dai_props[i].platform.name; + dai_link[i].platform.of_node = dai_props[i].platform.of_node; + dai_link[i].platform.dai_name = dai_props[i].platform.dai_name; } priv->dai_props = dai_props; @@ -782,7 +784,7 @@ static int simple_probe(struct platform_device *pdev) codecs->name = cinfo->codec; codecs->dai_name = cinfo->codec_dai.name; - platform = dai_link->platform; + platform = &dai_link->platform; platform->name = cinfo->platform; card->name = (cinfo->card) ? cinfo->card : cinfo->name; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..466099995e44 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -915,7 +915,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, /* find one from the set of registered platforms */ for_each_component(component) { - if (!snd_soc_is_matching_component(dai_link->platform, + if (!snd_soc_is_matching_component(&dai_link->platform, component)) continue; @@ -1026,7 +1026,7 @@ static void soc_remove_dai_links(struct snd_soc_card *card) static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { - struct snd_soc_dai_link_component *platform = dai_link->platform; + struct snd_soc_dai_link_component *platform = &dai_link->platform; /* * FIXME @@ -1034,14 +1034,10 @@ static int snd_soc_init_platform(struct snd_soc_card *card, * this function should be removed in the future */ /* convert Legacy platform link */ - if (!platform) { - platform = devm_kzalloc(card->dev, - sizeof(struct snd_soc_dai_link_component), - GFP_KERNEL); - if (!platform) - return -ENOMEM; + if (dai_link->platform_name || dai_link->platform_of_node) { + dev_dbg(card->dev, + "ASoC: Defaulting to legacy platform data!\n"); - dai_link->platform = platform; platform->name = dai_link->platform_name; platform->of_node = dai_link->platform_of_node; platform->dai_name = NULL; @@ -1123,7 +1119,7 @@ static int soc_init_dai_link(struct snd_soc_card *card, * Platform may be specified by either name or OF node, but * can be left unspecified, and a dummy platform will be used. */ - if (link->platform->name && link->platform->of_node) { + if (link->platform.name && link->platform.of_node) { dev_err(card->dev, "ASoC: Both platform name/of_node are set for %s\n", link->name); @@ -1921,7 +1917,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dev_err(card->dev, "init platform error"); continue; } - dai_link->platform->name = component->name; + dai_link->platform.name = component->name; /* convert non BE into BE */ dai_link->no_pcm = 1; -- 2.7.4

3 7

[alsa-devel] [PATCH 1/5] ALSA: hda: add verbs for stripe control
by Sameer Pujar 11 Jan '19

11 Jan '19

Controllers can support multiple Serial Data Out(SDO) lines, for extended outbound bandwidth, to pump data to all codecs on the link. Codecs can sample data present on SDO. Add verbs AC_VERB_GET_STRIPE_CONTROL and AC_VERB_SET_STRIPE_CONTROL These can be used to program usage of SDO lines for codec. Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Reviewed-by: Mohan Kumar D <mkumard(a)nvidia.com> Reviewed-by: Ravindra Lokhande <rlokhande(a)nvidia.com> --- include/sound/hda_verbs.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h index 2a8573a..e36b775 100644 --- a/include/sound/hda_verbs.h +++ b/include/sound/hda_verbs.h @@ -66,6 +66,7 @@ enum { #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 +#define AC_VERB_GET_STRIPE_CONTROL 0x0f24 #define AC_VERB_GET_CVT_CHAN_COUNT 0x0f2d #define AC_VERB_GET_HDMI_DIP_SIZE 0x0f2e #define AC_VERB_GET_HDMI_ELDD 0x0f2f @@ -110,6 +111,7 @@ enum { #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3 0x71f #define AC_VERB_SET_EAPD 0x788 #define AC_VERB_SET_CODEC_RESET 0x7ff +#define AC_VERB_SET_STRIPE_CONTROL 0x724 #define AC_VERB_SET_CVT_CHAN_COUNT 0x72d #define AC_VERB_SET_HDMI_DIP_INDEX 0x730 #define AC_VERB_SET_HDMI_DIP_DATA 0x731 -- 2.7.4 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

2 9

[alsa-devel] [PATCH] ASoC: msm8916-wcd-digital: Select REGMAP_MMIO to fix build error
by Axel Lin 11 Jan '19

11 Jan '19

Fix below build error: ERROR: "__devm_regmap_init_mmio_clk" [sound/soc/codecs/snd-soc-msm8916-digital.ko] undefined! Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 87cb9c51e6f5..45884859229f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -703,6 +703,7 @@ config SND_SOC_MAX9860 config SND_SOC_MSM8916_WCD_ANALOG tristate "Qualcomm MSM8916 WCD Analog Codec" depends on SPMI || COMPILE_TEST + select REGMAP_MMIO config SND_SOC_MSM8916_WCD_DIGITAL tristate "Qualcomm MSM8916 WCD DIGITAL Codec" -- 2.17.1

1 0

[alsa-devel] [PATCH v2] ASoC: soc-core: defer card probe until all component is added to list
by Rohit kumar 11 Jan '19

11 Jan '19

From: Ajit Pandey <ajitp(a)codeaurora.org> DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- Changes since v1: Moved component find logic to soc-core instead of machine driver as suggested by Mark. sound/soc/soc-core.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3e..eec92f1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 6

[alsa-devel] [PATCH - ALSA JACK plugin 1/2] jack: Replacing jack->channels with jack->num_ports
by laxmi.devi＠in.bosch.com 11 Jan '19

11 Jan '19

From: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> As jack->num_ports and jack->channels hold the same values, jack->channels is redundant and hence removed. Sanity check is added in prepare, to check if io->Channels is same as jack->num_ports. Signed-off-by: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index af2136e..d723b17 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -48,7 +48,6 @@ typedef struct { unsigned int sample_bits; snd_pcm_uframes_t min_avail; - unsigned int channels; snd_pcm_channel_area_t *areas; jack_port_t **ports; @@ -263,6 +262,12 @@ static int snd_pcm_jack_prepare(snd_pcm_ioplug_t *io) snd_pcm_sw_params_t *swparams; int err; + if (io->channels != jack->num_ports) { + SNDERR("Channel count %d not equal to no. of ports %d in JACK", + io->channels, jack->num_ports); + return -EINVAL; + } + jack->hw_ptr = 0; jack->xrun_detected = false; @@ -381,7 +386,7 @@ static int jack_set_hw_constraint(snd_pcm_jack_t *jack) unsigned int psize_list[MAX_PERIODS_MULTIPLE]; unsigned int nframes = jack_get_buffer_size(jack->client); unsigned int jack_buffer_bytes = (snd_pcm_format_size(format, nframes) * - jack->channels); + jack->num_ports); unsigned int i; int err; @@ -398,7 +403,7 @@ static int jack_set_hw_constraint(snd_pcm_jack_t *jack) (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_FORMAT, 1, &format)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_CHANNELS, - jack->channels, jack->channels)) < 0 || + jack->num_ports, jack->num_ports)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_RATE, rate, rate)) < 0 || (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, @@ -485,8 +490,7 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, return err; } - jack->channels = jack->num_ports; - if (jack->channels == 0) { + if (jack->num_ports == 0) { SNDERR("define the %s_ports section", stream == SND_PCM_STREAM_PLAYBACK ? "playback" : "capture"); snd_pcm_jack_free(jack); @@ -514,7 +518,7 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, return -ENOENT; } - jack->areas = calloc(jack->channels, sizeof(snd_pcm_channel_area_t)); + jack->areas = calloc(jack->num_ports, sizeof(snd_pcm_channel_area_t)); if (! jack->areas) { snd_pcm_jack_free(jack); return -ENOMEM; -- 2.7.4

2 2

[alsa-devel] [PATCH - ALSA JACK plugin 1/2] jack: Replacing jack->channels with jack->num_ports
by laxmi.devi＠in.bosch.com 11 Jan '19

11 Jan '19

From: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> As jack->num_ports and jack->channels hold the same values, jack->channels is redundant and hence removed. Sanity check is added in prepare, to check if io->Channels is same as jack->num_ports. Signed-off-by: Laxmi Devi <Laxmi.Devi(a)in.bosch.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index af2136e..d723b17 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -48,7 +48,6 @@ typedef struct { unsigned int sample_bits; snd_pcm_uframes_t min_avail; - unsigned int channels; snd_pcm_channel_area_t *areas; jack_port_t **ports; @@ -263,6 +262,12 @@ static int snd_pcm_jack_prepare(snd_pcm_ioplug_t *io) snd_pcm_sw_params_t *swparams; int err; + if (io->channels != jack->num_ports) { + SNDERR("Channel count %d not equal to no. of ports %d in JACK", + io->channels, jack->num_ports); + return -EINVAL; + } + jack->hw_ptr = 0; jack->xrun_detected = false; @@ -381,7 +386,7 @@ static int jack_set_hw_constraint(snd_pcm_jack_t *jack) unsigned int psize_list[MAX_PERIODS_MULTIPLE]; unsigned int nframes = jack_get_buffer_size(jack->client); unsigned int jack_buffer_bytes = (snd_pcm_format_size(format, nframes) * - jack->channels); + jack->num_ports); unsigned int i; int err; @@ -398,7 +403,7 @@ static int jack_set_hw_constraint(snd_pcm_jack_t *jack) (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_FORMAT, 1, &format)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_CHANNELS, - jack->channels, jack->channels)) < 0 || + jack->num_ports, jack->num_ports)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_RATE, rate, rate)) < 0 || (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, @@ -485,8 +490,7 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, return err; } - jack->channels = jack->num_ports; - if (jack->channels == 0) { + if (jack->num_ports == 0) { SNDERR("define the %s_ports section", stream == SND_PCM_STREAM_PLAYBACK ? "playback" : "capture"); snd_pcm_jack_free(jack); @@ -514,7 +518,7 @@ static int snd_pcm_jack_open(snd_pcm_t **pcmp, const char *name, return -ENOENT; } - jack->areas = calloc(jack->channels, sizeof(snd_pcm_channel_area_t)); + jack->areas = calloc(jack->num_ports, sizeof(snd_pcm_channel_area_t)); if (! jack->areas) { snd_pcm_jack_free(jack); return -ENOMEM; -- 2.7.4

1 1

[alsa-devel] [PATCH] ASoC: soc-core: add soc_cleanup_component()
by Kuninori Morimoto 11 Jan '19

11 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We need to cleanup component when soc_probe_component() was failed, or when soc_remove_component() was called. But they are cleanuping component on each way. Same code in many places makes code un-understandable. This patch adds new soc_cleanup_component() and call it from snd_probe_component() and snd_remove_component(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2aa9f95..e0a326f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -939,19 +939,23 @@ static int soc_bind_dai_link(struct snd_soc_card *card, return -EPROBE_DEFER; } +static void soc_cleanup_component(struct snd_soc_component *component) +{ + list_del(&component->card_list); + soc_cleanup_component_debugfs(component); + component->card = NULL; + module_put(component->dev->driver->owner); +} + static void soc_remove_component(struct snd_soc_component *component) { if (!component->card) return; - list_del(&component->card_list); - if (component->driver->remove) component->driver->remove(component); - soc_cleanup_component_debugfs(component); - component->card = NULL; - module_put(component->dev->driver->owner); + soc_cleanup_component(component); } static void soc_remove_dai(struct snd_soc_dai *dai, int order) @@ -1333,6 +1337,7 @@ static int soc_probe_component(struct snd_soc_card *card, component->card = card; dapm->card = card; INIT_LIST_HEAD(&dapm->list); + INIT_LIST_HEAD(&component->card_list); soc_set_name_prefix(card, component); soc_init_component_debugfs(component); @@ -1395,12 +1400,9 @@ static int soc_probe_component(struct snd_soc_card *card, /* see for_each_card_components */ list_add(&component->card_list, &card->component_dev_list); - return 0; - err_probe: - soc_cleanup_component_debugfs(component); - component->card = NULL; - module_put(component->dev->driver->owner); + if (ret < 0) + soc_cleanup_component(component); return ret; } -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: soc-core: reduce if/else nest on soc_probe_link_dais
by Kuninori Morimoto 11 Jan '19

11 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Deep nested code is not readable. Let's reduce if/else nest. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 37 +++++++++++++++++-------------------- 1 file changed, 17 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f3fe476..d20509e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1584,27 +1584,24 @@ static int soc_probe_link_dais(struct snd_soc_card *card, dai_link->stream_name); return ret; } - } else { - - if (!dai_link->params) { - /* create the pcm */ - ret = soc_new_pcm(rtd, num); - if (ret < 0) { - dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", - dai_link->stream_name, ret); - return ret; - } - ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd); - if (ret < 0) - return ret; - ret = soc_link_dai_pcm_new(rtd->codec_dais, - rtd->num_codecs, rtd); - if (ret < 0) - return ret; - } else { - INIT_DELAYED_WORK(&rtd->delayed_work, - codec2codec_close_delayed_work); + } else if (!dai_link->params) { + /* create the pcm */ + ret = soc_new_pcm(rtd, num); + if (ret < 0) { + dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", + dai_link->stream_name, ret); + return ret; } + ret = soc_link_dai_pcm_new(&cpu_dai, 1, rtd); + if (ret < 0) + return ret; + ret = soc_link_dai_pcm_new(rtd->codec_dais, + rtd->num_codecs, rtd); + if (ret < 0) + return ret; + } else { + INIT_DELAYED_WORK(&rtd->delayed_work, + codec2codec_close_delayed_work); } return 0; -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: soc-core: merge card resources cleanup method
by Kuninori Morimoto 11 Jan '19

11 Jan '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We need to cleanup card resources when snd_soc_instantiate_card() was failed, or when snd_soc_unbind_card() was called. But they are cleanuping card resources on each way. Same code in many places makes code un-understandable. This patch reuses soc_cleanup_card_resources() for cleanuping code resource. Then, it makes avoiding cleanup order. It will be called from snd_soc_instantiate_card() and snd_soc_unbind_card(). Then, original soc_cleanup_card_resources() included snd_soc_flush_all_delayed_work(), but it is now separated. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 103 +++++++++++++++++++++------------------------------ 1 file changed, 43 insertions(+), 60 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8b661eb..f3fe476 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1980,6 +1980,29 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) } } +static int soc_cleanup_card_resources(struct snd_soc_card *card) +{ + /* free the ALSA card at first; this syncs with pending operations */ + if (card->snd_card) + snd_card_free(card->snd_card); + + /* remove and free each DAI */ + soc_remove_dai_links(card); + soc_remove_pcm_runtimes(card); + + /* remove auxiliary devices */ + soc_remove_aux_devices(card); + + snd_soc_dapm_free(&card->dapm); + soc_cleanup_card_debugfs(card); + + /* remove the card */ + if (card->remove) + card->remove(card); + + return 0; +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -1989,6 +2012,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + card->dapm.bias_level = SND_SOC_BIAS_OFF; + card->dapm.dev = card->dev; + card->dapm.card = card; + list_add(&card->dapm.list, &card->dapm_list); + /* check whether any platform is ignore machine FE and using topology */ soc_check_tplg_fes(card); @@ -1996,14 +2024,14 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) for_each_card_prelinks(card, i, dai_link) { ret = soc_bind_dai_link(card, dai_link); if (ret != 0) - goto base_error; + goto probe_end; } /* bind aux_devs too */ for (i = 0; i < card->num_aux_devs; i++) { ret = soc_bind_aux_dev(card, i); if (ret != 0) - goto base_error; + goto probe_end; } /* add predefined DAI links to the list */ @@ -2017,16 +2045,11 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) dev_err(card->dev, "ASoC: can't create sound card for card %s: %d\n", card->name, ret); - goto base_error; + goto probe_end; } soc_init_card_debugfs(card); - card->dapm.bias_level = SND_SOC_BIAS_OFF; - card->dapm.dev = card->dev; - card->dapm.card = card; - list_add(&card->dapm.list, &card->dapm_list); - #ifdef CONFIG_DEBUG_FS snd_soc_dapm_debugfs_init(&card->dapm, card->debugfs_card_root); #endif @@ -2048,7 +2071,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (card->probe) { ret = card->probe(card); if (ret < 0) - goto card_probe_error; + goto probe_end; } /* probe all components used by DAI links on this card */ @@ -2059,7 +2082,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) dev_err(card->dev, "ASoC: failed to instantiate card %d\n", ret); - goto probe_dai_err; + goto probe_end; } } } @@ -2067,7 +2090,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* probe auxiliary components */ ret = soc_probe_aux_devices(card); if (ret < 0) - goto probe_dai_err; + goto probe_end; /* * Find new DAI links added during probing components and bind them. @@ -2079,10 +2102,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) ret = soc_init_dai_link(card, dai_link); if (ret) - goto probe_dai_err; + goto probe_end; ret = soc_bind_dai_link(card, dai_link); if (ret) - goto probe_dai_err; + goto probe_end; } /* probe all DAI links on this card */ @@ -2093,7 +2116,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) dev_err(card->dev, "ASoC: failed to instantiate card %d\n", ret); - goto probe_dai_err; + goto probe_end; } } } @@ -2140,7 +2163,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) { dev_err(card->dev, "ASoC: %s late_probe() failed: %d\n", card->name, ret); - goto probe_aux_dev_err; + goto probe_end; } } @@ -2150,33 +2173,17 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) { dev_err(card->dev, "ASoC: failed to register soundcard %d\n", ret); - goto probe_aux_dev_err; + goto probe_end; } card->instantiated = 1; dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); - mutex_unlock(&card->mutex); - mutex_unlock(&client_mutex); - - return 0; -probe_aux_dev_err: - soc_remove_aux_devices(card); - -probe_dai_err: - soc_remove_dai_links(card); - -card_probe_error: - if (card->remove) - card->remove(card); - - snd_soc_dapm_free(&card->dapm); - soc_cleanup_card_debugfs(card); - snd_card_free(card->snd_card); +probe_end: + if (ret < 0) + soc_cleanup_card_resources(card); -base_error: - soc_remove_pcm_runtimes(card); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); @@ -2205,31 +2212,6 @@ static int soc_probe(struct platform_device *pdev) return snd_soc_register_card(card); } -static int soc_cleanup_card_resources(struct snd_soc_card *card) -{ - /* make sure any delayed work runs */ - snd_soc_flush_all_delayed_work(card); - - /* free the ALSA card at first; this syncs with pending operations */ - snd_card_free(card->snd_card); - - /* remove and free each DAI */ - soc_remove_dai_links(card); - soc_remove_pcm_runtimes(card); - - /* remove auxiliary devices */ - soc_remove_aux_devices(card); - - snd_soc_dapm_free(&card->dapm); - soc_cleanup_card_debugfs(card); - - /* remove the card */ - if (card->remove) - card->remove(card); - - return 0; -} - /* removes a socdev */ static int soc_remove(struct platform_device *pdev) { @@ -2795,6 +2777,7 @@ static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) if (card->instantiated) { card->instantiated = false; snd_soc_dapm_shutdown(card); + snd_soc_flush_all_delayed_work(card); soc_cleanup_card_resources(card); if (!unregister) list_add(&card->list, &unbind_card_list); -- 2.7.4

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