June 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: wm_adsp: Add basic debugfs entries
by Richard Fitzgerald 09 Jun '15

09 Jun '15

This patch adds some debugfs nodes to get information about the currently running firmware. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 4 + sound/soc/codecs/wm5110.c | 9 ++- sound/soc/codecs/wm_adsp.c | 182 ++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm_adsp.h | 25 ++++++- 4 files changed, 212 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 11eba0e..22fc9e9 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1875,6 +1875,8 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; + wm_adsp_init_debugfs(&priv->core.adsp[0], codec); + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 2); if (ret != 0) return ret; @@ -1893,6 +1895,8 @@ static int wm5102_codec_remove(struct snd_soc_codec *codec) { struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); + wm_adsp_cleanup_debugfs(&priv->core.adsp[0]); + priv->core.arizona->dapm = NULL; return 0; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index d65364e..f9ca1a9 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1599,7 +1599,10 @@ static struct snd_soc_dai_driver wm5110_dai[] = { static int wm5110_codec_probe(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); - int ret; + int i, ret; + + for (i = 0; i < WM5110_NUM_ADSP; i++) + wm_adsp_init_debugfs(&priv->core.adsp[i], codec); priv->core.arizona->dapm = &codec->dapm; @@ -1621,6 +1624,10 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) static int wm5110_codec_remove(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < WM5110_NUM_ADSP; i++) + wm_adsp_cleanup_debugfs(&priv->core.adsp[i]); priv->core.arizona->dapm = NULL; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6e358b5..c8def3ea 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -23,6 +23,7 @@ #include <linux/slab.h> #include <linux/vmalloc.h> #include <linux/workqueue.h> +#include <linux/debugfs.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -243,6 +244,26 @@ struct wm_coeff_ctl { unsigned int flags; }; +#ifdef CONFIG_DEBUG_FS +static void wm_adsp_debugfs_save_wmfwname(struct wm_adsp *dsp, const char *s); +static void wm_adsp_debugfs_save_binname(struct wm_adsp *dsp, const char *s); +static void wm_adsp_debugfs_clear(struct wm_adsp *dsp); +#else +static inline void wm_adsp_debugfs_save_wmfwname(struct wm_adsp *dsp, + const char *s) +{ +} + +static inline void wm_adsp_debugfs_save_binname(struct wm_adsp *dsp, + const char *s) +{ +} + +static inline void wm_adsp_debugfs_clear(struct wm_adsp *dsp) +{ +} +#endif + static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1104,6 +1125,8 @@ static int wm_adsp_load(struct wm_adsp *dsp) adsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, regions, pos - firmware->size); + wm_adsp_debugfs_save_wmfwname(dsp, file); + out_fw: regmap_async_complete(regmap); wm_adsp_buf_free(&buf_list); @@ -1218,11 +1241,12 @@ static int wm_adsp1_setup_algs(struct wm_adsp *dsp) n_algs = be32_to_cpu(adsp1_id.n_algs); dsp->fw_id = be32_to_cpu(adsp1_id.fw.id); + dsp->fw_id_version = be32_to_cpu(adsp1_id.fw.ver); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", dsp->fw_id, - (be32_to_cpu(adsp1_id.fw.ver) & 0xff0000) >> 16, - (be32_to_cpu(adsp1_id.fw.ver) & 0xff00) >> 8, - be32_to_cpu(adsp1_id.fw.ver) & 0xff, + (dsp->fw_id_version & 0xff0000) >> 16, + (dsp->fw_id_version & 0xff00) >> 8, + dsp->fw_id_version & 0xff, n_algs); alg_region = wm_adsp_create_region(dsp, WMFW_ADSP1_ZM, @@ -1321,11 +1345,12 @@ static int wm_adsp2_setup_algs(struct wm_adsp *dsp) n_algs = be32_to_cpu(adsp2_id.n_algs); dsp->fw_id = be32_to_cpu(adsp2_id.fw.id); + dsp->fw_id_version = be32_to_cpu(adsp2_id.fw.ver); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", dsp->fw_id, - (be32_to_cpu(adsp2_id.fw.ver) & 0xff0000) >> 16, - (be32_to_cpu(adsp2_id.fw.ver) & 0xff00) >> 8, - be32_to_cpu(adsp2_id.fw.ver) & 0xff, + (dsp->fw_id_version & 0xff0000) >> 16, + (dsp->fw_id_version & 0xff00) >> 8, + dsp->fw_id_version & 0xff, n_algs); alg_region = wm_adsp_create_region(dsp, WMFW_ADSP2_XM, @@ -1601,6 +1626,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) adsp_warn(dsp, "%s.%d: %zu bytes at end of file\n", file, blocks, pos - firmware->size); + wm_adsp_debugfs_save_binname(dsp, file); + out_fw: regmap_async_complete(regmap); release_firmware(firmware); @@ -1869,6 +1896,10 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + wm_adsp_debugfs_clear(dsp); + + dsp->fw_id = 0; + dsp->fw_id_version = 0; dsp->running = false; regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, @@ -1929,4 +1960,143 @@ int wm_adsp2_init(struct wm_adsp *dsp) } EXPORT_SYMBOL_GPL(wm_adsp2_init); +#ifdef CONFIG_DEBUG_FS +static void wm_adsp_debugfs_save_wmfwname(struct wm_adsp *dsp, const char *s) +{ + kfree(dsp->wmfw_file_name); + dsp->wmfw_file_name = kstrdup(s, GFP_KERNEL); +} + +static void wm_adsp_debugfs_save_binname(struct wm_adsp *dsp, const char *s) +{ + kfree(dsp->bin_file_name); + dsp->bin_file_name = kstrdup(s, GFP_KERNEL); +} + +static void wm_adsp_debugfs_clear(struct wm_adsp *dsp) +{ + kfree(dsp->wmfw_file_name); + kfree(dsp->bin_file_name); + dsp->wmfw_file_name = NULL; + dsp->bin_file_name = NULL; +} + +static ssize_t wm_adsp_debugfs_string_read(struct wm_adsp *dsp, + char __user *user_buf, + size_t count, loff_t *ppos, + const char *string) +{ + char *temp; + int len; + ssize_t ret; + + if (!string || !dsp->running) + return 0; + + temp = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!temp) + return -ENOMEM; + + len = snprintf(temp, PAGE_SIZE, "%s\n", string); + ret = simple_read_from_buffer(user_buf, count, ppos, temp, len); + kfree(temp); + return ret; +} + +static ssize_t wm_adsp_debugfs_wmfw_read(struct file *file, + char __user *user_buf, + size_t count, loff_t *ppos) +{ + struct wm_adsp *dsp = file->private_data; + + return wm_adsp_debugfs_string_read(dsp, user_buf, count, ppos, + dsp->wmfw_file_name); +} + +static ssize_t wm_adsp_debugfs_bin_read(struct file *file, + char __user *user_buf, + size_t count, loff_t *ppos) +{ + struct wm_adsp *dsp = file->private_data; + + return wm_adsp_debugfs_string_read(dsp, user_buf, count, ppos, + dsp->bin_file_name); +} + +static const struct { + const char *name; + const struct file_operations fops; +} wm_adsp_debugfs_fops[] = { + { + .name = "wmfw_file", + .fops = { + .open = simple_open, + .read = wm_adsp_debugfs_wmfw_read, + }, + }, + { + .name = "bin_file", + .fops = { + .open = simple_open, + .read = wm_adsp_debugfs_bin_read, + }, + }, +}; + +void wm_adsp_init_debugfs(struct wm_adsp *dsp, struct snd_soc_codec *codec) +{ + struct dentry *root = NULL; + char *root_name; + int i; + + if (!codec->component.debugfs_root) { + adsp_err(dsp, "No codec debugfs root\n"); + goto err; + } + + root_name = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!root_name) + goto err; + + snprintf(root_name, PAGE_SIZE, "dsp%d", dsp->num); + root = debugfs_create_dir(root_name, codec->component.debugfs_root); + kfree(root_name); + + if (!root) + goto err; + + if (!debugfs_create_bool("running", S_IRUGO, root, &dsp->running)) + goto err; + + if (!debugfs_create_x32("fw_id", S_IRUGO, root, &dsp->fw_id)) + goto err; + + if (!debugfs_create_x32("fw_version", S_IRUGO, root, + &dsp->fw_id_version)) + goto err; + + for (i = 0; i < ARRAY_SIZE(wm_adsp_debugfs_fops); ++i) { + if (!debugfs_create_file(wm_adsp_debugfs_fops[i].name, + S_IRUGO, root, dsp, + &wm_adsp_debugfs_fops[i].fops)) + goto err; + } + + dsp->debugfs_root = root; + return; + +err: + debugfs_remove_recursive(root); + adsp_err(dsp, "Failed to create debugfs\n"); +} +EXPORT_SYMBOL_GPL(wm_adsp_init_debugfs); + + +void wm_adsp_cleanup_debugfs(struct wm_adsp *dsp) +{ + debugfs_remove_recursive(dsp->debugfs_root); +} +EXPORT_SYMBOL_GPL(wm_adsp_cleanup_debugfs); +#endif + MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 0e5f07c..1d563fe 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -46,17 +46,25 @@ struct wm_adsp { struct list_head alg_regions; int fw_id; + int fw_id_version; const struct wm_adsp_region *mem; int num_mems; int fw; int fw_ver; - bool running; + u32 running; struct list_head ctl_list; struct work_struct boot_work; + +#ifdef CONFIG_DEBUG_FS + struct dentry *debugfs_root; + char *wmfw_file_name; + char *bin_file_name; +#endif + }; #define WM_ADSP1(wname, num) \ @@ -79,6 +87,21 @@ extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); + +#ifdef CONFIG_DEBUG_FS +void wm_adsp_init_debugfs(struct wm_adsp *dsp, struct snd_soc_codec *codec); +void wm_adsp_cleanup_debugfs(struct wm_adsp *dsp); +#else +static inline void wm_adsp_init_debugfs(struct wm_adsp *dsp, + struct snd_soc_codec *codec) +{ +} + +void wm_adsp_cleanup_debugfs(struct wm_adsp *dsp) +{ +} +#endif + int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, -- 1.7.2.5

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[alsa-devel] [PATCH/linux-next] Doc: sound:oss: Fix typo in sound/oss
by Masanari Iida 09 Jun '15

09 Jun '15

This patch fix some spelling typo found in Documentations/sound/oss. Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> --- Documentation/sound/oss/PSS-updates | 2 +- Documentation/sound/oss/README.OSS | 2 +- Documentation/sound/oss/btaudio | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/oss/PSS-updates b/Documentation/sound/oss/PSS-updates index c84dd75..11914a1 100644 --- a/Documentation/sound/oss/PSS-updates +++ b/Documentation/sound/oss/PSS-updates @@ -41,7 +41,7 @@ pss_no_sound This module parameter is a flag that can be used to tell the driver to just configure non-sound components. 0 configures all components, a non-0 -value will only attept to configure the CDROM and joystick ports. This +value will only attempt to configure the CDROM and joystick ports. This parameter can be used by a user who only wished to use the builtin joystick and/or CDROM port(s) of his PSS sound card. If this driver is loaded with this parameter and with the parameter below set to true then a user can safely unload diff --git a/Documentation/sound/oss/README.OSS b/Documentation/sound/oss/README.OSS index 4be2594..a085ea3 100644 --- a/Documentation/sound/oss/README.OSS +++ b/Documentation/sound/oss/README.OSS @@ -1346,7 +1346,7 @@ implement nice real-time signal processing audio effect software and network telephones. The ACI mixer has to be switched into the "solo" mode for duplex operation in order to avoid feedback caused by the mixer (input hears output signal). You can de-/activate this mode -through toggleing the record button for the wave controller with an +through toggling the record button for the wave controller with an OSS-mixer. The PCM20 contains a radio tuner, which is also controlled by diff --git a/Documentation/sound/oss/btaudio b/Documentation/sound/oss/btaudio index 1a693e6..effdb9a 100644 --- a/Documentation/sound/oss/btaudio +++ b/Documentation/sound/oss/btaudio @@ -29,7 +29,7 @@ Driver Status Still somewhat experimental. The driver should work stable, i.e. it should'nt crash your box. It might not work as expected, have bugs, -not being fully OSS API compilant, ... +not being fully OSS API compliant, ... Latest versions are available from http://bytesex.org/bttv/, the driver is in the bttv tarball. Kernel patches might be available too, -- 2.4.2.387.gf86f31a

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[alsa-devel] [PATCH] mfd: wm8994: Fix NULL pointer exception on missing pdata
by Inha Song 09 Jun '15

09 Jun '15

If we use the dev_get_platdata, pdata can be NULL in DT environments. Actually, NULL pointer exception is occurs with error messages like this: [ 1.226583] Unable to handle kernel NULL pointer dereference at virtual address 00000044 [ 1.467910] [<c02f1e38>] (wm8994_irq_init) from [<c02f1bd4>] (wm8994_i2c_probe+0x684/0x858) [ 1.476241] [<c02f1bd4>] (wm8994_i2c_probe) from [<c0371c78>] (i2c_device_probe+0x11c/0x17c) In wm8994-core, we already checked wheter platform data is NULL and set the wm8994->pdata to platform data if platform data is not NULL. So, we can use wm8994->pdata instead of platform data. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/wm8994-irq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/mfd/wm8994-irq.c b/drivers/mfd/wm8994-irq.c index a14407e..ec2c891 100644 --- a/drivers/mfd/wm8994-irq.c +++ b/drivers/mfd/wm8994-irq.c @@ -193,7 +193,7 @@ int wm8994_irq_init(struct wm8994 *wm8994) { int ret; unsigned long irqflags; - struct wm8994_pdata *pdata = dev_get_platdata(wm8994->dev); + struct wm8994_pdata *pdata = &wm8994->pdata; if (!wm8994->irq) { dev_warn(wm8994->dev, -- 2.0.0.390.gcb682f8

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[alsa-devel] [PATCH v6 0/3] ALSA: HDA: add extended HDA
by Vinod Koul 09 Jun '15

09 Jun '15

New HDA controllers from Intel have extended HDA capabilities like multilink, pipe processing, SPIB, GTS etc In order to use them we create an extended HDA bus, controller and stream which uses the extended configuration Vinod Koul (3): ALSA: hdac_ext: add extended HDA bus ALSA: hdac_ext: add hdac extended controller ALSA: hdac_ext: add extended stream capabilities include/sound/hdaudio_ext.h | 385 +++++++++++++++++++++++++++++++++ sound/hda/Kconfig | 4 + sound/hda/Makefile | 3 + sound/hda/ext/Makefile | 3 + sound/hda/ext/hdac_ext_bus.c | 115 ++++++++++ sound/hda/ext/hdac_ext_controller.c | 295 +++++++++++++++++++++++++ sound/hda/ext/hdac_ext_stream.c | 414 ++++++++++++++++++++++++++++++++++++ 7 files changed, 1219 insertions(+) create mode 100644 include/sound/hdaudio_ext.h create mode 100644 sound/hda/ext/Makefile create mode 100644 sound/hda/ext/hdac_ext_bus.c create mode 100644 sound/hda/ext/hdac_ext_controller.c create mode 100644 sound/hda/ext/hdac_ext_stream.c -- 1.9.1

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[alsa-devel] [PATCH-v2 1/1] HDA: Intel: enable automatic runtime pm for HDMI codecs by default
by han.lu＠intel.com 09 Jun '15

09 Jun '15

From: "Lu, Han" <han.lu(a)intel.com> Enable runtime PM of the HDMI audio codec on the latest Intel platforms. So the HD-A controller or HDMI codec can suspend when idle timeout by default and release the GFX power well. The patch influences HSW/BDW/BYT/BSW/SKL. Eariler platforms and third party analog codecs will not be influenced. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 54380ed..e9f274b6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3495,6 +3495,9 @@ static void codec_set_power_save(struct hda_codec *codec, int delay) { struct device *dev = hda_codec_dev(codec); + if (delay == 0 && codec->auto_runtime_pm) + delay = 3000; + if (delay > 0) { pm_runtime_set_autosuspend_delay(dev, delay); pm_runtime_use_autosuspend(dev); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6c57258..12837ab 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -251,6 +251,7 @@ struct hda_codec { unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ unsigned int power_save_node:1; /* advanced PM for each widget */ + unsigned int auto_runtime_pm:1; /* enable automatic codec runtime pm */ #ifdef CONFIG_PM unsigned long power_on_acct; unsigned long power_off_acct; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 407978b..f852734 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2356,6 +2356,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->dp_mst = true; } + /* Enable runtime pm for HDMI audio codec of HSW/BDW/SKL/BYT/BSW */ + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) + codec->auto_runtime_pm = 1; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.9.1

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[alsa-devel] [PATCH 1/1] HDA: Intel: enable automatic runtime pm for HDMI codecs by default
by han.lu＠intel.com 09 Jun '15

09 Jun '15

From: "Lu, Han" <han.lu(a)intel.com> Enable runtime PM of the HDMI audio codec on the latest Intel platforms. So the HD-A controller or HDMI codec can suspend when idle timeout by default and release the GFX power well. The patch influences HSW/BDW/BYT/BSW/SKL. Eariler platforms and third party analog codecs will not be influenced. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 54380ed..804ec24 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3491,10 +3491,13 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_add_new_ctls); #ifdef CONFIG_PM -static void codec_set_power_save(struct hda_codec *codec, int delay) +static void codec_set_power_save(struct hda_codec *codec, int delay, bool force) { struct device *dev = hda_codec_dev(codec); + if (force && delay == 0 && codec->auto_runtime_pm) + delay = 3000; + if (delay > 0) { pm_runtime_set_autosuspend_delay(dev, delay); pm_runtime_use_autosuspend(dev); @@ -3511,15 +3514,16 @@ static void codec_set_power_save(struct hda_codec *codec, int delay) * snd_hda_set_power_save - reprogram autosuspend for the given delay * @bus: HD-audio bus * @delay: autosuspend delay in msec, 0 = off + * @force: force to enable autosuspend * * Synchronize the runtime PM autosuspend state from the power_save option. */ -void snd_hda_set_power_save(struct hda_bus *bus, int delay) +void snd_hda_set_power_save(struct hda_bus *bus, int delay, bool force) { struct hda_codec *c; list_for_each_codec(c, bus) - codec_set_power_save(c, delay); + codec_set_power_save(c, delay, force); } EXPORT_SYMBOL_GPL(snd_hda_set_power_save); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6c57258..9f525ba 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -251,6 +251,7 @@ struct hda_codec { unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ unsigned int power_save_node:1; /* advanced PM for each widget */ + unsigned int auto_runtime_pm:1; /* enable automatic codec runtime pm */ #ifdef CONFIG_PM unsigned long power_on_acct; unsigned long power_off_acct; @@ -482,10 +483,11 @@ const char *snd_hda_get_jack_location(u32 cfg); #define snd_hda_power_down(codec) snd_hdac_power_down(&(codec)->core) #define snd_hda_power_down_pm(codec) snd_hdac_power_down_pm(&(codec)->core) #ifdef CONFIG_PM -void snd_hda_set_power_save(struct hda_bus *bus, int delay); +void snd_hda_set_power_save(struct hda_bus *bus, int delay, bool force); void snd_hda_update_power_acct(struct hda_codec *codec); #else -static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {} +static inline void +snd_hda_set_power_save(struct hda_bus *bus, int delay, bool force) {} #endif #ifdef CONFIG_SND_HDA_PATCH_LOADER diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7933fb0..11d52e5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -797,7 +797,7 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) chip = &hda->chip; if (!hda->probe_continued || chip->disabled) continue; - snd_hda_set_power_save(&chip->bus, power_save * 1000); + snd_hda_set_power_save(&chip->bus, power_save * 1000, false); } mutex_unlock(&card_list_lock); return 0; @@ -2034,7 +2034,7 @@ static int azx_probe_continue(struct azx *chip) chip->running = 1; azx_add_card_list(chip); - snd_hda_set_power_save(&chip->bus, power_save * 1000); + snd_hda_set_power_save(&chip->bus, power_save * 1000, true); if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) pm_runtime_put_noidle(&pci->dev); diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 477742c..bec3a37 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -518,7 +518,7 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; chip->running = 1; - snd_hda_set_power_save(&chip->bus, power_save * 1000); + snd_hda_set_power_save(&chip->bus, power_save * 1000, false); return 0; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 407978b..7941d81 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2356,6 +2356,9 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->dp_mst = true; } + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) + codec->auto_runtime_pm = 1; + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.9.1

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Re: [alsa-devel] [pulseaudio-discuss] snd_pcm_avail no reliable
by Raymond Yau 09 Jun '15

09 Jun '15

> > I tried using a smaller period_size (about 1/16 of hwbuf_size), snd_pcm_avail returned a better value. Period bytes = 2Kbytes which is still larger than the default rewind_safeguard 512 bytes > But cpu usage moved from 1% to 4%, since we are using it in a automotive, we don`t care about power saving, I think it is acceptable. > > But I`m not sure if there is any other down side about "time based scheduling" ? > >

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[alsa-devel] [PATCH] alsa: hda_codec: fix number of devices query on hotplug
by airlied＠gmail.com 09 Jun '15

09 Jun '15

From: Dave Airlie <airlied(a)redhat.com> The new regmap code seems to cache this, which isn't helpful for the hotplug dock situation where this gets updated. Use the uncached query for this. Signed-off-by: Dave Airlie <airlied(a)redhat.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b49feff..b778221 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -436,7 +436,7 @@ static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid) get_wcaps_type(wcaps) != AC_WID_PIN) return 0; - parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN); + parm = snd_hdac_read_parm_uncached(&codec->core, nid, AC_PAR_DEVLIST_LEN); if (parm == -1 && codec->bus->rirb_error) parm = 0; return parm & AC_DEV_LIST_LEN_MASK; -- 2.4.1

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[alsa-devel] [PATCH] PCM: snd_pcm_dmix_drain() maybe blocked after suspend and resume
by Shengjiu Wang 09 Jun '15

09 Jun '15

After suspend and resume, the alsa driver is stopped. But if alsa-lib run into snd_pcm_dmix_drain(), it need to wait avail >= pcm->stop_threshold, otherwise, it will not exit the loop, so finally it is blocked at poll() of snd_pcm_wait_nocheck(pcm, -1). This patch is to add state check after snd_pcm_wait_nocheck(pcm, -1), if the state is SND_PCM_STATE_SUSPENDED, then return error. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- src/pcm/pcm_dmix.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index babde6a..0cc318e 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -647,6 +647,12 @@ static int snd_pcm_dmix_drain(snd_pcm_t *pcm) if (dmix->state == SND_PCM_STATE_DRAINING) { snd_pcm_dmix_sync_area(pcm); snd_pcm_wait_nocheck(pcm, -1); + switch (snd_pcm_state(pcm)) { + case SND_PCM_STATE_SUSPENDED: + return -ESTRPIPE; + default: + break; + } snd_pcm_direct_clear_timer_queue(dmix); /* force poll to wait */ } } while (dmix->state == SND_PCM_STATE_DRAINING); -- 1.7.9.5

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Re: [alsa-devel] Fwd: Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 09 Jun '15

09 Jun '15

> > > > > > No change. > > > http://www.alsa-project.org/db/?f=30b3f0087374b914a20dbe20a618fb892a5d6fd5 > > > > Are there any offical specificaion ? > > > > So far most review only mention stereo speakers and subwoofer, the service > > guide only show how to replace two internal speakers > > Service manual mentions > "Dolby®-certified surround sound system with two built-in stereo speakers and > one subwoofer supporting low-frequency effects" > > so all additional channels what I'm hearing in Windows is emulated. I hope > that it is clear now that my laptop has two internal speakers and one > subwoofer. > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… You need to define your 2.1 chmap if the subwoofer only use left channel Compiling and running PulseAudio from git http://colin.guthr.ie/2010/09/compiling-and-running-pulseaudio-from-git/

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