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

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Ensure DSP controls are always persistent
by Charles Keepax 21 Feb '15

21 Feb '15

From: Nikesh Oswal <Nikesh.Oswal(a)wolfsonmicro.com> Currently DSP controls are persistent (across DSP On/Off) only if they were set whilst the DSP is off. This change makes the controls persistent irrespective of when they are set. Signed-off-by: Nikesh Oswal <Nikesh.Oswal(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 5 ++--- 1 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ff67b33..185ee85 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -420,10 +420,9 @@ static int wm_coeff_put(struct snd_kcontrol *kcontrol, memcpy(ctl->cache, p, ctl->len); - if (!ctl->enabled) { - ctl->set = 1; + ctl->set = 1; + if (!ctl->enabled) return 0; - } return wm_coeff_write_control(kcontrol, p, ctl->len); } -- 1.7.2.5

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[alsa-devel] [PATCH] ALSA: firewire-lib: fix an unexpected byte sequence for micro sign
by Takashi Sakamoto 21 Feb '15

21 Feb '15

The sign for microsecond (U+0085, MICRO SIGN) was encoded to '0x c2 b5' by UTF-8 character encoding scheme. But the byte sequence was converted to '0x c3 82 c2 b5' in a previous commit. As a result, the byte sequence cannot represent microsecond sign in UTF-8 or ASCII. This may confuse developers. This commit replaces the sign to string expression with 'microseconds' to purge superfluous troubles. Fixes: 5c697e5b46ef("ALSA: firewire-lib: remove rx_blocks_for_midi quirk") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 0d58018..7b7b183 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -33,7 +33,7 @@ */ #define MAX_MIDI_RX_BLOCKS 8 -#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 Âµs */ +#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */ /* isochronous header parameters */ #define ISO_DATA_LENGTH_SHIFT 16 -- 2.1.0

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[alsa-devel] [PATCH v3 0/5] alsa-lib: audio timestamping evolutions
by Pierre-Louis Bossart 21 Feb '15

21 Feb '15

alsa-lib side of kernel patches already merged on topic/timestamp. Pierre-Louis Bossart (5): sync asound.h with kernel version add support for get/set_audio_htstamp_config pcm: add helper functions to query timestamping capabilities pcm: add support for new STATUS_EXT ioctl test: fix audio_time with new get/set audio_tstamp_config include/pcm.h | 47 ++++- include/sound/asound.h | 37 +++- src/pcm/pcm.c | 73 +++++++- src/pcm/pcm_hw.c | 16 +- test/audio_time.c | 491 +++++++++++++++++++++++++++++++------------------ 5 files changed, 475 insertions(+), 189 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v6 0/7] audio timestamping evolutions
by Pierre-Louis Bossart 21 Feb '15

21 Feb '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields V3/4: Addressed feedback from Jaroslav: no change to STATUS ioctl, new functionality introduced in STATUS_EXT ioctl with r/w params. bumped PCM protocol for detection on STATUS_EXT in userspace used 32-bit word for accuracy report, no mantissa/exponent packing rolled back info_ext changes, all INFO fields remain in same word Merged comments from Liam (code simplifications) fixed packing Added driver_tstamp field in case there is a delay in passing the tstamp and audio_tstamp over IPC. v5: no functional change, only addressed Takashi comments on patch 2,3,4 fixed used of bool with true/false and ! in comparisons move trigger_tstamp_pending_update flag to snd_usb_substream, will be used by ALSA core when other devices need it v6: rebased on v3.19-rc7 after merge of initial 4 patches address Takashi's comments: simplified parameter handling, const and zero-clear cleanups in pcm.h, fix git-bisect issue Pierre-Louis Bossart (7): ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: add .get_time_info ALSA: hda: replace .wallclock by .get_time_info ALSA: core: remove .wall_clock ALSA: bump PCM protocol to 2.0.13 Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 66 +++++++++- include/uapi/sound/asound.h | 36 +++++- sound/core/pcm_compat.c | 28 ++++- sound/core/pcm_lib.c | 88 ++++++++----- sound/core/pcm_native.c | 41 +++++- sound/pci/hda/hda_controller.c | 40 ++++-- 7 files changed, 444 insertions(+), 55 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

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[alsa-devel] [PATCH 0/4] Internal PCM fixes / cleanups
by Takashi Iwai 20 Feb '15

20 Feb '15

Hi, while reading the internal PCM (BE) handling code again, I noticed a few potential bugs. Here are fix and cleanup patches. Takashi === Takashi Iwai (4): ALSA: pcm: Minor refactoring in snd_pcm_attach_substream() ALSA: pcm: Don't add internal PCMs to PCM device list ALSA: pcm: Don't notify internal PCMs ALSA: pcm: Don't ignore internal PCMs in snd_pcm_dev_disconnect() sound/core/pcm.c | 92 +++++++++++++++++++++++++------------------------------- 1 file changed, 41 insertions(+), 51 deletions(-) -- 2.3.0

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[alsa-devel] [PATCH] ALSA: hda - Avoid unnecessary power-up at mixer amp changes
by Takashi Iwai 20 Feb '15

20 Feb '15

When the mixer amp is touched by control elements, we don't have to power up always; if the codec was suspended at the time, we can just update the amp cache and it's reflected to the hardware upon resume. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 33 ++++++++++++++++----------------- 1 file changed, 16 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0734cb35887d..5e755eb6f19a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2148,11 +2148,10 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_read); static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val, - bool init_only) + bool init_only, bool cache_only) { struct hda_amp_info *info; unsigned int caps; - unsigned int cache_only; if (snd_BUG_ON(mask & ~0xff)) mask &= 0xff; @@ -2170,7 +2169,7 @@ static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, return 0; } info->vol[ch] = val; - cache_only = info->head.dirty = codec->cached_write; + info->head.dirty |= cache_only; caps = info->amp_caps; mutex_unlock(&codec->hash_mutex); if (!cache_only) @@ -2194,7 +2193,8 @@ static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val) { - return codec_amp_update(codec, nid, ch, direction, idx, mask, val, false); + return codec_amp_update(codec, nid, ch, direction, idx, mask, val, + false, codec->cached_write); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update); @@ -2241,7 +2241,8 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_stereo); int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch, int dir, int idx, int mask, int val) { - return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true); + return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true, + codec->cached_write); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init); @@ -2383,8 +2384,8 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, maxval = get_amp_max_value(codec, nid, dir, 0); if (val > maxval) val = maxval; - return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, - HDA_AMP_VOLMASK, val); + return codec_amp_update(codec, nid, ch, dir, idx, HDA_AMP_VOLMASK, val, + false, !hda_codec_is_power_on(codec)); } /** @@ -2434,14 +2435,12 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; - snd_hda_power_up(codec); if (chs & 1) { change = update_amp_value(codec, nid, 0, dir, idx, ofs, *valp); valp++; } if (chs & 2) change |= update_amp_value(codec, nid, 1, dir, idx, ofs, *valp); - snd_hda_power_down(codec); return change; } EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_put); @@ -3109,19 +3108,19 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; - snd_hda_power_up(codec); if (chs & 1) { - change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE); + change = codec_amp_update(codec, nid, 0, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE, false, + !hda_codec_is_power_on(codec)); valp++; } if (chs & 2) - change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE); + change |= codec_amp_update(codec, nid, 1, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE, false, + !hda_codec_is_power_on(codec)); hda_call_check_power_status(codec, nid); - snd_hda_power_down(codec); return change; } EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put); -- 2.3.0

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[alsa-devel] [PATCH 0/6] Minor HD-audio cleanups
by Takashi Iwai 20 Feb '15

20 Feb '15

Hi, this is a series of minor cleanups for HD-audio codes. No functional changes but just code refactoring for the further works. The patches are found in topic/hda-cleanup branch of sound git tree, too. Takashi === Takashi Iwai (6): ALSA: hda - Drop hda_bus_template for snd_hda_bus_new() ALSA: hda - Embed struct hda_bus_unsolicited into struct hda_bus ALSA: hda - Introduce azx_has_pm_runtime() macro ALSA: hda - Fold hda_priv.h into hda_controller.h ALSA: hda - Drop azx_mixer_create() ALSA: hda - Split azx_codec_create() to two phases sound/pci/hda/hda_codec.c | 54 +----- sound/pci/hda/hda_codec.h | 22 +-- sound/pci/hda/hda_controller.c | 96 +++++----- sound/pci/hda/hda_controller.h | 397 +++++++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_i915.c | 2 +- sound/pci/hda/hda_intel.c | 23 ++- sound/pci/hda/hda_intel.h | 2 +- sound/pci/hda/hda_local.h | 17 -- sound/pci/hda/hda_priv.h | 406 ----------------------------------------- sound/pci/hda/hda_tegra.c | 9 +- 10 files changed, 477 insertions(+), 551 deletions(-) delete mode 100644 sound/pci/hda/hda_priv.h -- 2.3.0

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[alsa-devel] Correct 5.1 settings for Asus mobo AT4NM10-I with IDT 92HD73C1X5
by Misan Thropos 19 Feb '15

19 Feb '15

I got it to work with the help of "debianuser" on #alsa@freenode and hdajackretask which can be found in alsa-tools-gui (if you use Debian). Had to change some pin settings from default configuration to make it work. You can get the working configuration from: http://www.alsa-project.org/db/?f=71b06cd49f390e0ff4061d069a2efcb93b3148d2 It enables the S/PDIF digital output too. Maybe you like to take that information for a model for IDT 92HD73C1X5. cheers, Misan

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[alsa-devel] [PATCH 0/19] ASoC: rsnd: move 2nd DMAC from DMAEngine to rsnd driver
by Kuninori Morimoto 19 Feb '15

19 Feb '15

Hi Mark, Arnd, Vinod Cc: Laurent, SH-ML Renesas R-Car sound (= rsnd) needs 2 DMAC which are called as Audio DMAC (= 1st DMAC) and Audio DMAC peri peri (2nd DMAC). And rsnd had assumed that 1st / 2nd DMACs are implemented as DMAEngine. But, in result of DMA ML long term discussion, 2nd DMAC was concluded that it is not a general purpose DMAC (2nd DMAC is for Device to Device inside sound system). Additionally, current DMAEngine can't support Device to Device, and we don't have correct DT bindings for it at this point. So the easiest solution for it is that move it from DMAEngine to rsnd driver. http://thread.gmane.org/gmane.linux.ports.sh.devel/41063/focus=43829 These patches add Audio DMAC peri peri implementation in rsnd driver. I will remove current DMAEngine side Audio DMAC peri peri implement if these are accepted. Arnd, Vinod I'm happy if you can send Ack for these patches 1) - 5) are cleanup / tidyup rsnd driver 6) - 11) adds Audio DMAC peri peri implementation in rsnd driver 12) change dma-names for 1st DMAC on DT 13) - 14) cleanup. these are no longer needed now 15) - 19) DT bindings text update 12) changes dma-names for 1st DMAC. It was assumed that 1st / 2nd DMAC are used as DMAEngine. dma-names had same naming rule for both DMAC to avoid complex DT mapping issue, But now, 2nd DMAC is no longer DMAEngine, we don't need to use this naming rule anymore. Now, no SoC/platform is using DMA for rsnd driver yet in upstream. I believe 12) is no problem for DT. Kuninori Morimoto (19): 1) ASoC: rsnd: remove SH-DMA-BASE specific implementation 2) ASoC: rsnd: remove un-needed parameter from rsnd_dma_init() 3) ASoC: rsnd: remove unused rsnd_dma_available() 4) ASoC: rsnd: remove un-needed parameter from rsnd_dma_quit() 5) ASoC: rsnd: tidyup rsnd_dma_to_mod() macro declaration position 6) ASoC: rsnd: enable to get resource by name 7) ASoC: rsnd: add rsnd_gen_get_phy_addr() to get physical address 8) ASoC: rsnd: add dma.c for Audio DMAC / Audio DMAC peri peri 9) ASoC: rsnd: move rsnd_gen_dma_addr() from gen.c to dma.c 10) ASoC: rsnd: enable to care 1st / 2nd DMAC on rsnd_dma_xxx() 11) ASoC: rsnd: add Audio DMAC peri peri support rework 12) ASoC: rsnd: dma-names cares 1st DMAC only 13) ASoC: rsnd: remove rsnd_dma::addr 14) ASoC: rsnd: remove rsnd_dma::dir 15) ASoC: rsnd: tidyup rcar_sound,ssi sample code 16) ASoC: rsnd: add sample code of rcar_sound,src irq 17) ASoC: rsnd: add sample code of reg-names 18) ASoC: rsnd: add sample code of missing clocks 19) ASoC: rsnd: add sample code of dma entry .../devicetree/bindings/sound/renesas,rsnd.txt | 131 +++-- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/core.c | 224 +------- sound/soc/sh/rcar/dma.c | 583 ++++++++++++++++++++ sound/soc/sh/rcar/gen.c | 145 +---- sound/soc/sh/rcar/rsnd.h | 50 +- sound/soc/sh/rcar/src.c | 5 +- sound/soc/sh/rcar/ssi.c | 5 +- 8 files changed, 725 insertions(+), 420 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] baytrail-pcm-audio: audio broken on Lenovo Thinkpad Tablet 10
by Sebastien Bourdeauducq 18 Feb '15

18 Feb '15

Hi, the Thinkpad Tablet 10 has an impressive amount of problems under Linux, among which the audio is completely broken. The kernel log is: [ 7.532413] rt5640 i2c-10EC5640:00: Device with ID register 6271 is not rt5640/39 [ 7.908208] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [ 7.908231] platform byt-rt5640: Driver byt-rt5640 requests probe deferral [ 7.931549] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [ 7.931557] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed -22 [ 7.967040] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.13.a0 [ 7.967047] baytrail-pcm-audio baytrail-pcm-audio: Build type: a0 [ 7.967051] baytrail-pcm-audio baytrail-pcm-audio: Build date: Apr 2 2014 14:14:39 [ 7.967212] byt-rt5640 byt-rt5640: ASoC: CODEC DAI rt5640-aif1 not registered [ 7.967248] platform byt-rt5640: Driver byt-rt5640 requests probe deferral I'm running kernel 3.19 with the latest firmware from the linux-firmware repository. Sebastien -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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