June 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] gitcompile fails
by Mark Brand 03 Jun '14

03 Jun '14

Fresh git clones of alsa-driver and alsa-kmirror produce a ./gitcompile failure for me: ln -s ../../alsa-kernel/include/pcm-indirect.h cp ../../alsa-kernel/include/pcm.h . patch -p0 -i pcm.patch pcm.h patching file pcm.h Hunk #1 FAILED at 280. Hunk #2 succeeded at 291 (offset -3 lines). [snip] Hunk #11 succeeded at 566 (offset -4 lines). 1 out of 11 hunks FAILED -- saving rejects to file pcm.h.rej make[3]: *** [pcm.h] Error 1 make[3]: Leaving directory `/home/mark/workspace/alsa/alsa-driver/alsa/include/sound' Since I've seen this before (http://mailman.alsa-project.org/pipermail/alsa-devel/2012-July/053419.html), would I be forgiven for assuming that this is once again as a result of a mismatch between alsa-driver and alsa-kmirror? Any suggestion will be much appreciated, thank you. Mark Brand.

1 0

[alsa-devel] [PATCH] ASoC: free jack GPIOs before the sound card is freed
by Stephen Warren 03 Jun '14

03 Jun '14

From: Stephen Warren <swarren(a)nvidia.com> This is the same change as commit fb6b8e71448a "ASoC: tegra: free jack GPIOs before the sound card is freed", but applied to all other ASoC machine drivers where code inspection indicates the same problem exists. That commit's description is: ========== snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to generate an initial jack status report. If sound card initialization fails, that work item needs to be cancelled, so it doesn't run after the card has been freed. Specifically, freeing the card calls snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which is called from the work queue item. snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine drivers do call this function in the platform driver remove() callback. However, this happens after the sound card is freed, at least when the card is freed due to errors late during snd_soc_instantiate_card(). This leaves a window where the work item can execute after the card is freed. In next-20140522, sound card initialization does fail for unrelated reasons, and hits the problem described above. To solve this, fix the Tegra ASoC machine drivers to clean up the Jack GPIOs during the snd_soc_card's .remove() callback, which is executed before the overall card object is freed. also, guard the cleanup call based on whether we actually setup up the GPIOs in the first place. Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove function to match where the GPIOs get set up. However, there is no such callback. ========== Note that I have not even compile-tested this in most cases, since most of the drivers rely on specific mach-* support I don't have enabled, and don't support COMPILE_TEST. Testing by the relevant board maintainers would be useful. Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> (maintainer:OMAP AUDIO SUPPORT) Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> (maintainer:OMAP AUDIO SUPPORT) Cc: Philipp Zabel <philipp.zabel(a)gmail.com> (maintainer:ARM/H4700 (HP IPA...) Cc: Paul Parsons <lost.distance(a)yahoo.com> (maintainer:ARM/H4700 (HP IPA...) Cc: Eric Miao <eric.y.miao(a)gmail.com> (maintainer:PXA2xx/PXA3xx SUP...) Cc: Russell King <linux(a)arm.linux.org.uk> (maintainer:PXA2xx/PXA3xx SUP...) Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> (maintainer:PXA2xx/PXA3xx SUP...) Cc: Ben Dooks <ben-linux(a)fluff.org> (maintainer:ARM/SAMSUNG ARM A...) Cc: Kukjin Kim <kgene.kim(a)samsung.com> (maintainer:ARM/SAMSUNG ARM A...) Cc: Sangbeom Kim <sbkim73(a)samsung.com> (supporter:SAMSUNG AUDIO (AS...) Cc: linux-omap(a)vger.kernel.org (open list:OMAP AUDIO SUPPORT) Cc: linux-samsung-soc(a)vger.kernel.org (moderated list:ARM/SAMSUNG ARM A...) Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Mark, do you need this patch split up by architecture for topic branches? I assume not if you've already sent your pull requests. This applies fine on top of tag asoc-v3.16. --- sound/soc/omap/ams-delta.c | 14 ++++++++++---- sound/soc/omap/omap-twl4030.c | 28 ++++++++++++++-------------- sound/soc/omap/rx51.c | 18 +++++++++--------- sound/soc/pxa/hx4700.c | 9 ++++++++- sound/soc/samsung/h1940_uda1380.c | 11 +++++++++-- sound/soc/samsung/rx1950_uda1380.c | 12 ++++++++++-- sound/soc/samsung/smartq_wm8987.c | 11 +++++++++-- 7 files changed, 69 insertions(+), 34 deletions(-) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index bb243c663e6b..1f41951d8b7f 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -527,6 +527,15 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int ams_delta_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_gpios), + ams_delta_hook_switch_gpios); + + return 0; +} + /* DAI glue - connects codec <--> CPU */ static struct snd_soc_dai_link ams_delta_dai_link = { .name = "CX20442", @@ -543,6 +552,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = { static struct snd_soc_card ams_delta_audio_card = { .name = "AMS_DELTA", .owner = THIS_MODULE, + .remove = ams_delta_card_remove, .dai_link = &ams_delta_dai_link, .num_links = 1, @@ -579,10 +589,6 @@ static int ams_delta_remove(struct platform_device *pdev) dev_warn(&pdev->dev, "failed to unregister V253 line discipline\n"); - snd_soc_jack_free_gpios(&ams_delta_hook_switch, - ARRAY_SIZE(ams_delta_hook_switch_gpios), - ams_delta_hook_switch_gpios); - snd_soc_unregister_card(card); card->dev = NULL; return 0; diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 64141db311b2..b4e282871658 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -231,6 +231,19 @@ static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd) return ret; } +static int omap_twl4030_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card); + + if (priv->jack_detect > 0) + snd_soc_jack_free_gpios(&priv->hs_jack, + ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + return 0; +} + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { { @@ -258,6 +271,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { /* Audio machine driver */ static struct snd_soc_card omap_twl4030_card = { .owner = THIS_MODULE, + .remove = omap_twl4030_card_remove, .dai_link = omap_twl4030_dai_links, .num_links = ARRAY_SIZE(omap_twl4030_dai_links), @@ -353,19 +367,6 @@ static int omap_twl4030_probe(struct platform_device *pdev) return 0; } -static int omap_twl4030_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card); - - if (priv->jack_detect > 0) - snd_soc_jack_free_gpios(&priv->hs_jack, - ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - - return 0; -} - static const struct of_device_id omap_twl4030_of_match[] = { {.compatible = "ti,omap-twl4030", }, { }, @@ -380,7 +381,6 @@ static struct platform_driver omap_twl4030_driver = { .of_match_table = omap_twl4030_of_match, }, .probe = omap_twl4030_probe, - .remove = omap_twl4030_remove, }; module_platform_driver(omap_twl4030_driver); diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 6951dc812055..47a10290535b 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -334,6 +334,14 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; } +static int rx51_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios), + rx51_av_jack_gpios); + + return 0; +} + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link rx51_dai[] = { { @@ -368,6 +376,7 @@ static struct snd_soc_codec_conf rx51_codec_conf[] = { static struct snd_soc_card rx51_sound_card = { .name = "RX-51", .owner = THIS_MODULE, + .remove = rx51_card_remove, .dai_link = rx51_dai, .num_links = ARRAY_SIZE(rx51_dai), .aux_dev = rx51_aux_dev, @@ -499,14 +508,6 @@ static int rx51_soc_probe(struct platform_device *pdev) return 0; } -static int rx51_soc_remove(struct platform_device *pdev) -{ - snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios), - rx51_av_jack_gpios); - - return 0; -} - #if defined(CONFIG_OF) static const struct of_device_id rx51_audio_of_match[] = { { .compatible = "nokia,n900-audio", }, @@ -522,7 +523,6 @@ static struct platform_driver rx51_soc_driver = { .of_match_table = of_match_ptr(rx51_audio_of_match), }, .probe = rx51_soc_probe, - .remove = rx51_soc_remove, }; module_platform_driver(rx51_soc_driver); diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index dcc9b04bd92c..6b81acaffddd 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -152,6 +152,13 @@ static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd) return err; } +static int hx4700_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio); + + return 0; +} + /* hx4700 digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link hx4700_dai = { .name = "ak4641", @@ -170,6 +177,7 @@ static struct snd_soc_dai_link hx4700_dai = { static struct snd_soc_card snd_soc_card_hx4700 = { .name = "iPAQ hx4700", .owner = THIS_MODULE, + .remove = hx4700_card_remove, .dai_link = &hx4700_dai, .num_links = 1, .dapm_widgets = hx4700_dapm_widgets, @@ -206,7 +214,6 @@ static int hx4700_audio_probe(struct platform_device *pdev) static int hx4700_audio_remove(struct platform_device *pdev) { - snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio); snd_soc_unregister_card(&snd_soc_card_hx4700); gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0); diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 9f2fb69dbaae..720357f11a7f 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -189,6 +189,14 @@ static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int h1940_uda1380_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + + return 0; +} + /* s3c24xx digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link h1940_uda1380_dai[] = { { @@ -206,6 +214,7 @@ static struct snd_soc_dai_link h1940_uda1380_dai[] = { static struct snd_soc_card h1940_asoc = { .name = "h1940", .owner = THIS_MODULE, + .remove = h1940_uda1380_card_remove, .dai_link = h1940_uda1380_dai, .num_links = ARRAY_SIZE(h1940_uda1380_dai), @@ -257,8 +266,6 @@ err_out: static void __exit h1940_exit(void) { platform_device_unregister(s3c24xx_snd_device); - snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), - hp_jack_gpios); gpio_free(S3C_GPIO_END + 9); } diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 5b3e504d3a32..192aa9fc102f 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -31,6 +31,7 @@ #include "s3c24xx-i2s.h" static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd); +static int rx1950_uda1380_card_remove(struct snd_soc_pcm_runtime *rtd); static int rx1950_startup(struct snd_pcm_substream *substream); static int rx1950_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); @@ -116,6 +117,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static struct snd_soc_card rx1950_asoc = { .name = "rx1950", .owner = THIS_MODULE, + .remove = rx1950_uda1380_card_remove, .dai_link = rx1950_uda1380_dai, .num_links = ARRAY_SIZE(rx1950_uda1380_dai), @@ -234,6 +236,14 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int rx1950_uda1380_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), + hp_jack_gpios); + + return 0; +} + static int __init rx1950_init(void) { int ret; @@ -278,8 +288,6 @@ err_gpio: static void __exit rx1950_exit(void) { platform_device_unregister(s3c24xx_snd_device); - snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), - hp_jack_gpios); gpio_free(S3C2410_GPA(1)); } diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index df55db5d3554..271a904277a1 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -182,6 +182,14 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) return err; } +static int smartq_wm8987_card_remove(struct snd_soc_pcm_runtime *rtd) +{ + snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + return 0; +} + static struct snd_soc_dai_link smartq_dai[] = { { .name = "wm8987", @@ -198,6 +206,7 @@ static struct snd_soc_dai_link smartq_dai[] = { static struct snd_soc_card snd_soc_smartq = { .name = "SmartQ", .owner = THIS_MODULE, + .remove = smartq_wm8987_card_remove, .dai_link = smartq_dai, .num_links = ARRAY_SIZE(smartq_dai), @@ -259,8 +268,6 @@ err_unregister_device: static void __exit smartq_exit(void) { gpio_free(S3C64XX_GPK(12)); - snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios), - smartq_jack_gpios); platform_device_unregister(smartq_snd_device); } -- 1.8.1.5

2 1

[alsa-devel] [RFC] set up an sync channel between audio and display driver (i.e. ALSA and DRM)
by Lin, Mengdong 03 Jun '14

03 Jun '14

This RFC is based on previous discussion to set up a generic communication channel between display and audio driver and an internal design of Intel MCG/VPG HDMI audio driver. It's still an initial draft and your advice would be appreciated to improve the design. The basic idea is to create a new avsink module and let both drm and alsa depend on it. This new module provides a framework and APIs for synchronization between the display and audio driver. 1. Display/Audio Client The avsink core provides APIs to create, register and lookup a display/audio client. A specific display driver (eg. i915) or audio driver (eg. HD-Audio driver) can create a client, add some resources objects (shared power wells, display outputs, and audio inputs, register ops) to the client, and then register this client to avisink core. The peer driver can look up a registered client by a name or type, or both. If a client gives a valid peer client name on registration, avsink core will bind the two clients as peer for each other. And we expect a display client and an audio client to be peers for each other in a system. int avsink_new_client ( const char *name, int type, /* client type, display or audio */ struct module *module, void *context, const char *peer_name, struct avsink_client **client_ret); int avsink_free_client (struct avsink_client *client); int avsink_register_client(struct avsink_client *client); int avisink_unregister_client(int client_handle); struct avsink_client *avsink_lookup_client(const char *name, int type); struct avsink_client { const char *name; /* client name */ int type; /* client type*/ void *context; struct module *module; /* top-level module for locking */ struct avsink_client *peer; /* peer client */ /* shared power wells */ struct avsink_power_well *power_well; int num_power_wells; /* endpoints, display outputs or audio inputs */ struct avsink_endpoint * endpoint; int num_endpints; struct avsink_registers_ops *reg_ops; /* ops to access registers of a client */ void *private_data; ... }; On system boots, the avsink module is loaded before the display and audio driver module. And the display and audio driver may be loaded on parallel. * If a specific display driver (eg. i915) supports avsink, it can create a display client, add power wells and display outputs to the client, and then register the display client to the avsink core. Then it may look up if there is any audio client registered, by name or type, and may find an audio client registered by some audio driver. * If an audio driver supports avsink, it usually should look up a registered display client by name or type at first, because it may need the shared power well in GPU and check the display outputs' name to bind the audio inputs. If the display client is not registered yet, the audio driver can choose to wait (maybe in a work queue) or return -EAGAIN for a deferred probe. After the display client is found, the audio driver can register an audio client with the display client's name as the peer name, the avsink core will bind the display and audio clients to each other. Open question: If the display or audio driver is disabled by the black list, shall we introduce a time out to avoid waiting for the other client registered endlessly? 2. Shared power wells (optional) The audio and display devices, maybe only part of them, may share a common power well (e.g. for Intel Haswell and Broadwell). If so, the driver that controls the power well should define a power well object, implement the get/put ops, and add it to its avsink client before registering the client to avsink core. Then the peer client can look up this power well by its name, and get/put this power well as a user. A client can have multiple power well objects. struct avsink_power_well { const char *name; /* name of the power well */ void *context; /* parameter of get/put ops, maybe device pointer for this power well */ struct avsink_power_well_ops *ops }; struct avsink_power_well_ops { int (*get)(void *context); int (*put)(void *context); }; API: int avsink_new_power(struct avsink_client *client, const char *power_name, void * power_context, struct avsink_power_well_ops *ops, struct avsink_power_well **power_ret); struct avsink_power_well *avisnk_lookup_power(const char *name); int avsink_get_power(struct avsink_power_well *power); /* Reqesut the power */ int avsink_put_power(struct avsink_power_well *power); /* Release the power */ For example, the i915 display driver can create a device for the shared power well in Haswell GPU, implement its PM functions, and use the device pointer as the context when creating the power well object, like below struct avsink_power_well_ops i915_power_well_ops = { .get = pm_runtime_get_sync; .put = pm_runtime_put_sync; }; ... avsink_new_power ( display_client, "i915_display_power_well", pdev, /* pointer of the power well device */ &i915_power_well_ops, ...) Power domain is not used here since a single device seems enough to represent a power well. 3. Display output and audio input endpoints A display client should register the display output endpoints and its audio peer client should register the audio input endpoints. A client can have multiple endpoints. The avsink core will bind an audio input and a display output as peer to each other. This is to allow the audio and display driver to synchronize with each other for each display pipeline. All endpoints should be added to a client before the client is registered to avsink core. Dynamic endpoints are not supported now. A display out here represents a physical HDMI/DP output port. And as long as it's usable in the system (i.e. physically connected to the HDMP/DP port on the machine board), the display output should be registered not matter the port is connected to an external display device or not. And if HW and display driver can support DP1.2 daisy chain (multiple DP display devices can be connected to a single port), multiple static displays outputs should be defined for the DP port according to the HW capability. The port & display device number can be indicated by the name (e.g. "i915_DDI_B", "i915_DDI_B_DEV0", "i915_DDI_B_DEV1", or "i915_DDI_B_DEV2"), defined by the display driver. The audio driver can check the endpoints of its peer display client and use an display endpoint's name, or a presumed display endpoint name, as peer name when registering an audio endpoint, thus the avsink core will bind the two display and audio endpoints as peers. struct avsink_endpoint { const char *name; /*name of the endpoint */ int type; /* DISPLAY_OUTPUT or AUDIO_INPUT */ void *context; /* private data, used as parameter of the ops */ struct avsink_endpoint_ops *ops; struct avsink_endpoint *peer; /* peer endpoint */ }; struct avsink_endpoint_ops { int (*get_caps) (enum had_caps_list query_element, void *capabilities, void *context); int (*set_caps) (enum had_caps_list set_element, void *capabilities, void *context); int (*event_handler) (enum avsink_event_type event_type, void *context); }; API: int avsink_new_endpoint (struct avsink_client *client, const char *name, int type, /* DISPLAY_OUTPUT or AUDIO_INPUT*/ void *context, const char *peer_name, /* can be NULL if no clue */ avsink_endpoint_ops *ops, struct avsink_endpoint **endpoint_ret); int avsink_endpoint_get_caps(struct avsink_endpoint *endpoint, enum avsink_caps_list t get_element, void *capabilities); int avsink_endpoint_set_caps(struct avsink_endpoint *endpoint, enum had_caps_list set_element, void *capabilities); int avsink_endpoint_post_event(struct avsink_endpoint *endpoint, enum avsink_event_type event_type); 4. Get/Set caps on an endpoint The display or audio driver can get or set capabilities on an endpoint. Depending on the capability ID, the avsink core will call get_caps/set_caps ops of this endpoint, or call get_caps/set_caps ops of its peer endpoint and return the result to the caller. enum avsink_caps_list { /* capabilities for display output endpoints */ AVSINK_GET_DISPLAY_ELD = 1, AVSINK_GET_DISPLAY_TYPE, /* HDMI or DisplayPort */ AVSINK_GET_DISPLAY_NAME, /* Hope to use display device name under /sys/class/drm, like "card0-DP-1", for user * space to figure out which HDMI/DP output on the drm side corresponds to which audio * stream device on the alsa side */ AVSINK_GET_DISPLAY_SAMPLING_FREQ, /* HDMI TMDS clock or DP link symbol clock, for audio driver to * program N value */ AVSINK_GET_DISPLAY_HDCP_STATUS, AVSINK_GET_DISPLAY_AUDIO_STATUS, /* Whether audio is enabled */ AVSINK_SET_DISPLAY_ENABLE_AUDIO, /* Enable audio */ AVSINK_SET_DISPLAY_DISABLE_AUDIO, /* Disable audio */ AVSINK_SET_DISPLAY_ENABLE_AUDIO_INT, /* Enable audio interrupt */ AVSINK_SET_DISPLAY_DISABLE_AUDIO_INT, /* Disable audio interrupt */ /* capabilities for audio input endpoints */ AVSINK_GET_AUDIO_IS_BUSY, /* Whether there is an active audio streaming */ OTHERS_TBD, }; For example, the audio driver can query ELD info on an audio input endpoint by using caps AVSINK_GET_DISPLAY_ELD, and avsink core will call get_caps() on the peer display output endpoint and return the ELD info to the audio driver. Some audio driver may only use part of these caps. E.g. HD-Audio driver can use bus commands instead of the ops to control the audio on gfx side, so it doesn't use caps like ENABLE/DISABLE_AUDIO or ENABLE/DISABLE_AUDIO. When the display driver want to disable a display pipeline for hot-plug, mode change or power saving, it can use caps AVSINK_GET_AUDIO_IS_BUSY to check if the audio input is busy (active streaming) on this display pipeline. And if audio is busy, the display driver can choose to wait or go ahead to disable display pipeline anyway. For the latter case, the audio input endpoint will be notified by an event and should abort audio streaming. 5. Event handling of endpoints A driver can post events on an endpoint. Depending on the event type, the avsink core will call the endpoint's event handler or pass the event to its peer endpoint and trigger the peer's event handler function if defined. int avsink_endpoint_post_event(struct avsink_endpoint *endpoint, enum avsink_event_type event_type); Now we only defined event types which should be handled by the audio input endpoints. The event types can be extended in the future. enum avsink_event_type { AVSINK_EVENT_DISPLAY_DISABLE = 1, /* The display pipeline is disabled for hot-plug, mode change or * suspend. Audio driver should stop any active streaming. */ AVSINK_EVENT_DISPLAY_ENABLE, /* The display pipeline is enabled after hot-plug, mode change or * resume. Audio driver can restore previously interrupted streaming */ AVSINK_EVENT_DISPLAY_MODE_CHANGE, /* Display mode change event. At this time, the new display mode is * configured but the display pipeline is not enabled yet. Audio driver * can do some configurations such as programing N value */ AVSINK_EVENT_DISPLAY_AUDIO_BUFFER_DONE, /* Audio Buffer done interrupts. Only for audio drivers if DMA and * interrupt are handled by GPU */ AVSINK_EVENT_DISPLAY_AUDIO_BUFFER_UNDERRUN, /* Audio Buffer under run interrupts. Only for audio drivers if * DMA and interrupt are handled by GPU */ }; So for a display driver, it can post an event on a display output endpoint and get processed by the peer audio input endpoint. Or it can also directly post an event on a peer audio input endpoint, by using the 'peer' pointer on a display output endpoint. 6. Display register operation (optional) Some audio driver needs to access GPU audio registers. The register ops are provided by the peer display client. struct avsink_registers_ops { int (*read_register) (uint32_t reg_addr, uint32_t *data, void *context); int (*write_register) (uint32_t reg_addr, uint32_t data, void *context); int (*read_modify_register) (uint32_t reg_addr, uint32_t data, uint32_t mask, void *context); int avsink_define_reg_ops (struct avsink_client *client, struct avsink_registers_ops *ops); And avsink core provides API for the audio driver to access the display registers: int avsink_read_display_register(struct avsink_client *client , uint32_t offset, uint32_t *data); int avsink_write_display_register(struct avsink_client *client , uint32_t offset, uint32_t data); int avsink_read_modify_display_register(struct avsink_client *client, uint32_t offset, uint32_t data, uint32_t mask); If the client is an audio client, the avsink core will find it peer display client and call its register ops; and if the client is a display client, the avsink core will just call its own register ops. Thanks Mengdong

10 23

Re: [alsa-devel] [PATCH] ARM: i.MX: Compile FIQ code for AVIC & TZIC only
by Lothar Waßmann 03 Jun '14

03 Jun '14

Hi, Alexander Shiyan wrote: > There are no reason to compile FIQ specific code for GIC IRQ > controller (i.MX6). This patch restrict compilation for FIQ > specific code for AVIC and TZIC only. > > Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> > --- > arch/arm/mach-imx/Makefile | 6 +++--- > 1 file changed, 3 insertions(+), 3 deletions(-) > > diff --git a/arch/arm/mach-imx/Makefile b/arch/arm/mach-imx/Makefile > index bbe93bb..620a760 100644 > --- a/arch/arm/mach-imx/Makefile > +++ b/arch/arm/mach-imx/Makefile > @@ -1,4 +1,4 @@ > -obj-y := time.o cpu.o system.o irq-common.o > +obj-y := time.o cpu.o system.o > > obj-$(CONFIG_SOC_IMX1) += clk-imx1.o mm-imx1.o > obj-$(CONFIG_SOC_IMX21) += clk-imx21.o mm-imx21.o > @@ -21,8 +21,8 @@ obj-$(CONFIG_COMMON_CLK) += clk-pllv1.o clk-pllv2.o clk-pllv3.o clk-gate2.o \ > obj-$(CONFIG_IMX_HAVE_IOMUX_V1) += iomux-v1.o > obj-$(CONFIG_ARCH_MXC_IOMUX_V3) += iomux-v3.o > > -obj-$(CONFIG_MXC_TZIC) += tzic.o > -obj-$(CONFIG_MXC_AVIC) += avic.o > +obj-$(CONFIG_MXC_TZIC) += tzic.o irq-common.o > +obj-$(CONFIG_MXC_AVIC) += avic.o irq-common.o > > obj-$(CONFIG_MXC_USE_EPIT) += epit.o > obj-$(CONFIG_MXC_DEBUG_BOARD) += 3ds_debugboard.o > This patch introduces a build error in sound/soc/fsl/imx-pcm-fiq.c: sound/built-in.o: In function `imx_pcm_fiq_free': last.c:(.text+0x2aca8): undefined reference to `mxc_set_irq_fiq' sound/built-in.o: In function `imx_pcm_fiq_init': last.c:(.text+0x2b1a4): undefined reference to `mxc_set_irq_fiq' last.c:(.text+0x2b200): undefined reference to `mxc_set_irq_fiq' make[1]: *** [vmlinux] Error 1 make: *** [sub-make] Error 2 This could be fixed with the following patch: --- mxc-fiq.orig/sound/soc/fsl/Kconfig 2014-06-02 11:25:38.315960451 +0200 +++ mxc-fiq/sound/soc/fsl/Kconfig 2014-06-02 11:25:38.411960595 +0200 @@ -15,7 +15,7 @@ config SND_SOC_FSL_SSI tristate "Synchronous Serial Interface module support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && ARCH_MXC + select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_AVIC || MXC_TZIC) help Say Y if you want to add Synchronous Serial Interface (SSI) support for the Freescale CPUs. Lothar Waßmann -- ___________________________________________________________ Ka-Ro electronics GmbH | Pascalstraße 22 | D - 52076 Aachen Phone: +49 2408 1402-0 | Fax: +49 2408 1402-10 Geschäftsführer: Matthias Kaussen Handelsregistereintrag: Amtsgericht Aachen, HRB 4996 www.karo-electronics.de | info(a)karo-electronics.de ___________________________________________________________

3 3

[alsa-devel] [PATCH] firewire-lib: Remove a comment about restriction of asynchronous operation
by Takashi Sakamoto 03 Jun '14

03 Jun '14

The comment for fcp_avc_transaction() describes it doesn't support this type of operation. But it was already supported by this commit. 00a7bb81c20f3e81711e28e0f6c08cee8fd18514 ALSA: firewire-lib: Add support for deferred transaction Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fcp.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c index d48e326..0619597 100644 --- a/sound/firewire/fcp.c +++ b/sound/firewire/fcp.c @@ -218,8 +218,6 @@ struct fcp_transaction { * * @command and @response can point to the same buffer. * - * Asynchronous operation (INTERIM, NOTIFY) is not supported at the moment. - * * Returns the actual size of the response frame, or a negative error code. */ int fcp_avc_transaction(struct fw_unit *unit, -- 1.8.3.2

2 1

[alsa-devel] [PATCH 0/6] ASoC: rsnd: tidyup for DMA support for DT
by Kuninori Morimoto 03 Jun '14

03 Jun '14

Hi Mark These patches are tidyup code and fixup for DMA support of DT #1 - #4 are cleanup code patches #5 - #6 are fixup for DMA for DT Kuninori Morimoto (6): ASoC: rsnd: DT node clean up by using the of_node_put() ASoC: rsnd: save platform_device instead of device ASoC: rsnd: remove rsnd_src_non_ops ASoC: rsnd: module name is unified ASoC: rsnd: care DMA slave channel name for DT ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + sound/soc/sh/rcar/core.c | 87 ++++++++++++++++-- sound/soc/sh/rcar/dvc.c | 20 ++++- sound/soc/sh/rcar/gen.c | 95 ++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 10 ++- sound/soc/sh/rcar/src.c | 50 +++++++---- sound/soc/sh/rcar/ssi.c | 28 ++++-- 7 files changed, 260 insertions(+), 31 deletions(-)

4 9

[alsa-devel] [PATCH] ASoC: rsnd: fixup SSI interrupts style of DeviceTree
by Kuninori Morimoto 03 Jun '14

03 Jun '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current SSI interrupts has been set to each port. But it is too ugly style. SSI interrupts will be listed under rcar_sound,ssi node by this patch Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Mark There is still no DT user for this driver at this point. No conflict happens on platform. .../devicetree/bindings/sound/renesas,rsnd.txt | 50 ++++++++------------ sound/soc/sh/rcar/ssi.c | 2 +- 2 files changed, 21 insertions(+), 31 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 8346cab..1fba56b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -53,36 +53,26 @@ rcar_sound: rcar_sound@0xffd90000 { }; rcar_sound,ssi { - ssi0: ssi@0 { - interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi1: ssi@1 { - interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi2: ssi@2 { - interrupts = <0 372 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi3: ssi@3 { - interrupts = <0 373 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi4: ssi@4 { - interrupts = <0 374 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi5: ssi@5 { - interrupts = <0 375 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi6: ssi@6 { - interrupts = <0 376 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi7: ssi@7 { - interrupts = <0 377 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi8: ssi@8 { - interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - }; - ssi9: ssi@9 { - interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - }; + interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>, + <0 371 IRQ_TYPE_LEVEL_HIGH>, + <0 372 IRQ_TYPE_LEVEL_HIGH>, + <0 373 IRQ_TYPE_LEVEL_HIGH>, + <0 374 IRQ_TYPE_LEVEL_HIGH>, + <0 375 IRQ_TYPE_LEVEL_HIGH>, + <0 376 IRQ_TYPE_LEVEL_HIGH>, + <0 377 IRQ_TYPE_LEVEL_HIGH>, + <0 378 IRQ_TYPE_LEVEL_HIGH>, + <0 379 IRQ_TYPE_LEVEL_HIGH>; + ssi0: ssi@0 { }; + ssi1: ssi@1 { }; + ssi2: ssi@2 { }; + ssi3: ssi@3 { }; + ssi4: ssi@4 { }; + ssi5: ssi@5 { }; + ssi6: ssi@6 { }; + ssi7: ssi@7 { }; + ssi8: ssi@8 { }; + ssi9: ssi@9 { }; }; rcar_sound,dai { diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 2df723d..b311bc6 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -588,7 +588,7 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev, /* * irq */ - ssi_info->pio_irq = irq_of_parse_and_map(np, 0); + ssi_info->pio_irq = irq_of_parse_and_map(node, i); /* * DMA -- 1.7.9.5

4 6

[alsa-devel] [PATCH] ASoC: soc-cache: Fix the return value of snd_soc_cache_init when reg_size is 0
by Fabio Estevam 02 Jun '14

02 Jun '14

Commit b59dce53ef6 (ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error.) caused the following probe error: wm8962 0-001a: customer id 0 revision D wm8962 0-001a: ASoC: Failed to set cache compression type: -22 imx-wm8962 sound: snd_soc_register_card failed (-22) imx-wm8962: probe of sound failed with error -22 If reg_size just return success instead. Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/soc-cache.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 8fff5b6..00e70b6 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -73,7 +73,7 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; if (!reg_size) - return -EINVAL; + return 0; mutex_init(&codec->cache_rw_mutex); -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: cache: Fix error code when not using ASoC level cache
by Mark Brown 02 Jun '14

02 Jun '14

From: Mark Brown <broonie(a)linaro.org> It is not an error to have no cache so we shouldn't return an error code and cause our callers to fail, just silently do nothing instead. Thanks to Jarkko for identify the problematic commit. Reported-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Reported-by: Fabio Estevam <festevam(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-cache.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 8fff5b6..00e70b6 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -73,7 +73,7 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; if (!reg_size) - return -EINVAL; + return 0; mutex_init(&codec->cache_rw_mutex); -- 2.0.0.rc2

1 0

[alsa-devel] Failed to set cache compression type
by Fabio Estevam 02 Jun '14

02 Jun '14

Hi, I am booting linux-next 20140602 on a mx6qsabresd and I get the following probe error: wm8962 0-001a: customer id 0 revision D wm8962 0-001a: ASoC: Failed to set cache compression type: -22 imx-wm8962 sound: snd_soc_register_card failed (-22) imx-wm8962: probe of sound failed with error -22 What is the proper way to fix it? Thanks, Fabio Estevam

4 5