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

[alsa-devel] snd_pcm_hw_params_get_tick_time
by Shérab 12 Oct '14

12 Oct '14

Hello, I'm maintaing a TTS engine for french that use snd_pcm_hw_params_get_tick_time. Compiling the code produces a deprecation warning for this function. I tried to figure out how the code should be modified to not use this function but can't find a way. Any help would be warmly appreciated. Thanks, Shérab.

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[alsa-devel] seq module loading issue
by Adam Goode 10 Oct '14

10 Oct '14

Hi, I've observed a scenario in ALSA that I would like some feedback on solving. On Linux, Chrome uses seq for WebMIDI. Eventually, there will be hotplug detection code in Chrome that looks like this: 1. Connect to announce port 2. Listen for any announcements for client/port start/end 3. Send MIDIConnectionEvent events to JavaScript clients ( http://webaudio.github.io/web-midi-api/#midiconnectionevent-interface) Here is the scenario where this fails: 1. Cold boot a machine with no MIDI devices plugged in 2. Start client, subscribe on the seq announce port 3. Insert USB MIDI device 4. snd_rawmidi module is loaded, triggered by insertion of MIDI device 5. snd_seq_midi is NOT loaded, but marked for loading later At this point, there is no client start event sent to the announce port. This is because of the deferred module loading, which won't trigger except for a handful of operations (snd_seq_client_info_get_client is an example). This is easily replicated with looking at aseqdump -p 0:1. My question is: how to solve this? There are a few options: 1. Periodically poll for new clients. This is fairly ugly, but will work to trigger module loading and won't result in latency once any device has been plugged in, since announce will again work. 2. Somehow trigger module loading if there are any clients listening on the announce port. This seems pretty tricky and requires keeping track of some new state. 3. Do away with deferred module loading in seq. This is pretty invasive and would increase the number of modules loaded, but would result in a lot of deleted code in the kernel and a clean fix. Number 3 is my preferred solution, but I wanted to see if these sorts of patches would be acceptable, since it is invasive and would result in more modules loaded. Thanks for any comments. Adam

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[alsa-devel] ASoC machine driver: how to configure supported frame rates?
by Sergej Sawazki 10 Oct '14

10 Oct '14

Dear all, I'm writing an ALSA SoC machine driver. The codec on my board, as well as the cpu do support the following rates: SNDRV_PCM_RATE_8000_192000 However, the codec, as well as the cpu are clock slaves and the clocks are fixed. Hence, my board does only support the 96kHz sample rate. Can someone give me some advice on how to tell ALSA that my board does only support the rate SNDRV_PCM_RATE_96000? I don't want to modify the codec and the platform driver. Any help would be highly appreciated. Thanks! - Sergej

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[alsa-devel] proposal: snd_pcm_start_at()
by Tim Cussins 10 Oct '14

10 Oct '14

Hi all, I'm Tim: I work at Linn Products Ltd - we make Network Music Players, amongst other things. As you might imagine, synchronised-start is important when multiple devices on the network are rendering the same audio. We'd be interested in contributing a small expansion of the alsa-lib API to support synchronised start. Assuming we can synchronise the audio clocks (I'm aware this is not trivial - It's not the topic of this post), we'd propose something like: int snd_pcm_start_at(snd_pcm_t* pcm, snd_htimestamp_t* tstamp); and playback would begin as close to tstamp as possible. If tstamp is in the past, it would should return an error. Recent work by Takashi Iwai enables client code to set the clock type of timestamps using snd_pcm_sw_params_set_tstamp_type(). This context could quite naturally extend to tstamp argument of snd_pcm_start_at(). Before I get stuck into working up the details under the hood, it'd be good to get some feedback/objections regarding this approach. Thanks in advance! Tim

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[alsa-devel] No sound captured with SGTL5000 on i.MX6 in I²S master mode
by Jean-Michel Hautbois 10 Oct '14

10 Oct '14

Hi, Sorry if double posting, I have not received confirmation on subscription yesterday... So I try again :/. I have a custom i.MX6 board, and have a sgtl5000 connected on it. I would like to have it used in master mode (it is used only for audio capture) but I get a "pcm_read:2031: read error: Input/output error" when using arecord. I have checked my signals and I get something on I²S which seems to be good. I launch : $> arecord -f cd -Dhw:0,0 -v -V stereo Receive Frame Sync is 44.1KHz Receive Clock Signal is ~1MHz (I can be more precise if needed) Data Receive is present, only when arecord is launched. It stops when there is no capture asked. Both clocks are still present even when no arecord is running. This suggests, for me, that master mode is working. So, I don't know why I can't get any data to my arecord. Is this a left/right justified master mode ? Anything else ? I am wondering if I forgot something in my DT or if I need to do something else : sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000"; model = "imx6q-vbx3-sgtl5000"; ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "LINE_IN", "Line In Jack"; mux-int-port = <1>; mux-ext-port = <5>; }; &audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux>; status = "okay"; }; &i2c2 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay"; codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 201>; VDDA-supply = <&vgen6_reg>; VDDIO-supply = <&vgen6_reg>; sysclk = <24000000>; }; }; &iomuxc { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_hog>; imx6qdl-vbx3 { pinctrl_audmux: audmuxgrp { fsl,pins = < MX6QDLPAD_DISP0_DAT13__AUD5_RXFS 0x130b0 MX6QDL_PAD_DISP0_DAT14__AUD5_RXC 0x130b0 MX6QDL_PAD_DISP0_DAT19__AUD5_RXD 0x130b0 >; }; }; &ssi1 { status = "okay"; codec-handle = <&codec>; fsl,mode = "i2s-master"; fsl,ssi-asynchronous; /* Not sure about it, but does not change anything */ }; Here is the dmesg part : [ 6.026803] sgtl5000 1-000a: sgtl5000 revision 0x11 [ 6.129380] imx-sgtl5000 sound: snd_soc_register_card failed (-517) [ 6.248849] platform sound: Driver imx-sgtl5000 requests probe deferral [ 6.311643] sgtl5000 1-000a: Using internal LDO instead of VDDD [ 6.393196] imx-sgtl5000 sound: sgtl5000 <-> 2028000.ssi mapping ok And the user space commands : $> arecord -l **** List of CAPTURE Hardware Devices **** card 0: imx6qvbx3sgtl50 [imx6q-vbx3-sgtl5000], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $> arecord -v -V stereo -fdat -D hw:0,0 somefile.wav Recording WAVE 'somefile.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Hardware PCM card 0 'imx6q-vbx3-sgtl5000' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 arecord: pcm_read:2031: read error: Input/output error The kernel is 3.17-rc5. Thanks for reading all this, and for any advice :). JM

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[alsa-devel] Help Adjusting HDA HDMI settings
by bencoxdev＠gmail.com 10 Oct '14

10 Oct '14

Hi All, I need to change the of an ALSA driver on a proprietary arm linux box. Specifically the box in its standard configuration is hard coded to have 2 channel audio output only. The hardware on the box I’m assured by the manufacturer is capable of surround 5.1 output on it’s HDMI port. I therefore need to increase the ALSA driver’s capabilities to handle surround 5.1 output. I have the kernel source code for box. Could someone point me in the direction of what would need changing/adding in order to achieve this please. Thanks Ben

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[alsa-devel] [PATCH 1/2] ALSA: au88x0: added reference of vortex_t
by Sudip Mukherjee 10 Oct '14

10 Oct '14

added a pointer of the vortex in the following functions : vortex_alsafmt_aspfmt vortex_Vort3D_InitializeSource a3dsrc_ZeroStateA3D so that we can have a reference of the vortex in the function. this reference of the vortex will actually be used in a later patch to convert the pr_* macro to dev_*. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/au88x0/au88x0.h | 4 ++-- sound/pci/au88x0/au88x0_a3d.c | 6 +++--- sound/pci/au88x0/au88x0_core.c | 5 +++-- sound/pci/au88x0/au88x0_pcm.c | 2 +- 4 files changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index 466a5c8..3a8fefe 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -243,7 +243,7 @@ static int vortex_core_init(vortex_t * card); static int vortex_core_shutdown(vortex_t * card); static void vortex_enable_int(vortex_t * card); static irqreturn_t vortex_interrupt(int irq, void *dev_id); -static int vortex_alsafmt_aspfmt(int alsafmt); +static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v); /* Connection stuff. */ static void vortex_connect_default(vortex_t * vortex, int en); @@ -278,7 +278,7 @@ static void vortex_mix_setvolumebyte(vortex_t * vortex, unsigned char mix, static void vortex_Vort3D_enable(vortex_t * v); static void vortex_Vort3D_disable(vortex_t * v); static void vortex_Vort3D_connect(vortex_t * vortex, int en); -static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en); +static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v); #endif /* Driver stuff. */ diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index 30f760e..bc9cda3a 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -484,7 +484,7 @@ static void a3dsrc_ZeroState(a3dsrc_t * a) } /* Reset entire A3D engine */ -static void a3dsrc_ZeroStateA3D(a3dsrc_t * a) +static void a3dsrc_ZeroStateA3D(a3dsrc_t *a, vortex_t *v) { int i, var, var2; @@ -601,7 +601,7 @@ static void vortex_Vort3D_enable(vortex_t *v) Vort3DRend_Initialize(v, XT_HEADPHONE); for (i = 0; i < NR_A3D; i++) { vortex_A3dSourceHw_Initialize(v, i % 4, i >> 2); - a3dsrc_ZeroStateA3D(&(v->a3d[0])); + a3dsrc_ZeroStateA3D(&v->a3d[0], v); } /* Register ALSA controls */ vortex_a3d_register_controls(v); @@ -676,7 +676,7 @@ static void vortex_Vort3D_connect(vortex_t * v, int en) } /* Initialize one single A3D source. */ -static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en) +static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v) { if (a->vortex == NULL) { pr_warn diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 72e8128..00e2096 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2177,7 +2177,8 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, return -EBUSY; } /* (De)Initialize A3D hardware source. */ - vortex_Vort3D_InitializeSource(&(vortex->a3d[a3d]), en); + vortex_Vort3D_InitializeSource(&vortex->a3d[a3d], en, + vortex); } /* Make SPDIF out exclusive to "spdif" device when in use. */ if ((stream->type == VORTEX_PCM_SPDIF) && (en)) { @@ -2765,7 +2766,7 @@ static int vortex_core_shutdown(vortex_t * vortex) /* Alsa support. */ -static int vortex_alsafmt_aspfmt(int alsafmt) +static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v) { int fmt; diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 5adc6b9..bdde182 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -332,7 +332,7 @@ static int snd_vortex_pcm_prepare(struct snd_pcm_substream *substream) dir = 1; else dir = 0; - fmt = vortex_alsafmt_aspfmt(runtime->format); + fmt = vortex_alsafmt_aspfmt(runtime->format, chip); spin_lock_irq(&chip->lock); if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { vortex_adbdma_setmode(chip, dma, 1, dir, fmt, -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: rt5677: make volume TLV closer to reality
by Dylan Reid 10 Oct '14

10 Oct '14

The volume blocks have an step of 0.375dB, but TLV uses 0.01dB for units. Approximate 0.375 as 0.37. It leaves the minimum value off by 1.15dB, but that is better than being off by ~500 as it was with 37.5 as the step. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Just a suggestion. It seems to me to be the lesser of two evils. This probably won't be the last codec that has a higher resolution than 0.01dB, should we come up with a way to express that? --- sound/soc/codecs/rt5677.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 16aa4d9..f139d5a 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -538,9 +538,11 @@ static bool rt5677_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +/* Actual min=-65.625dB and step=0.375dB but only have 0.01 resolution. */ +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6475, 37, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +/* Actual min=-17.625dB and step=0.375dB but only have 0.01 resolution. */ +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1739, 37, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0); -- 2.1.2.330.g565301e

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[alsa-devel] [PATCH 1/1] ALSA: ALC283 codec - Avoid pop noise on headphones during suspend/resume
by Harsha Priya 09 Oct '14

09 Oct '14

This patch sets the headphones mode to default before suspending which helps avoid the pop noise on headphones Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 69d1236..e839e7a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2884,6 +2884,9 @@ static void alc283_shutup(struct hda_codec *codec) alc_write_coef_idx(codec, 0x43, 0x9004); + /*depop hp during suspend*/ + alc_write_coef_idx(codec, 0x06, 0x2100); + snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); -- 1.9.1

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[alsa-devel] [GIT PULL] ASoC fixes for v3.18
by Mark Brown 09 Oct '14

09 Oct '14

The following changes since commit a5448c88b812390a3622e76d774e10c0da1fb970: Merge remote-tracking branches 'asoc/topic/txx9', 'asoc/topic/wm8978' and 'asoc/topic/wm8994' into asoc-next (2014-10-06 12:49:09 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.18-2 for you to fetch changes up to bdf20b4291eaa3b327398b8dd330065ad8e6d3ce: Merge remote-tracking branches 'asoc/fix/88pm860x', 'asoc/fix/fsl', 'asoc/fix/imx', 'asoc/fix/mc13783', 'asoc/fix/rockchip' and 'asoc/fix/simple' into asoc-linus (2014-10-08 16:44:50 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.18 A few small fixes, mostly small although a few really nasty ones like the interaction between GPIO 0 and simple-card. ---------------------------------------------------------------- Daniel Mack (4): ASoC: soc-dapm: fix use after free ASoC: core: fix use after free in snd_soc_remove_platform() ASoC: 88pm860x-codec: Fix possibly missing string termination ASoC: soc-pcm: fix sig_bits determination in soc_pcm_apply_msb() Geert Uytterhoeven (1): ASoC: simple-card: Initialize headphone and mic GPIO numbers Mark Brown (6): ASoC: mc13783: Ensure we only try to dereference valid of_nodes Merge tag 'asoc-v3.17-rc4' into asoc-linus Merge tag 'asoc-v3.18' into asoc-linus Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/88pm860x', 'asoc/fix/fsl', 'asoc/fix/imx', 'asoc/fix/mc13783', 'asoc/fix/rockchip' and 'asoc/fix/simple' into asoc-linus Sonny Rao (1): ASoC: rockchip-i2s: fix infinite loop in rockchip_snd_txctrl Takashi Iwai (2): ASoC: imx-es8328: Fix missing return code in imx_es8328_probe() ASoC: imx-es8328: Fix of_node_put() call with uninitialized object sound/soc/codecs/88pm860x-codec.c | 2 +- sound/soc/codecs/mc13783.c | 16 ++++++++++------ sound/soc/fsl/imx-es8328.c | 3 ++- sound/soc/generic/simple-card.c | 3 +++ sound/soc/rockchip/rockchip_i2s.c | 4 +++- sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 25 ++++++++++++++----------- sound/soc/soc-pcm.c | 2 +- 8 files changed, 36 insertions(+), 23 deletions(-)

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