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

[alsa-devel] RawMIDI behaviour with MidiFace 4x4
by Kalvas, Taneli 25 Mar '15

25 Mar '15

Dear All, I am developing a patch librarian for a synthesizer. Because the communication happens with Midi sysex and timing is not relevant I chose RawMIDI interface and not the alsa sequencer. The problem I have encountered is that writing to the midi port with snd_rawmidi_write() seems to broken. The function accepts any size of data block and claims to have sent everything. In reality only a part of the message is sent. For a large block size I can see the activity light blinking for fraction of a second. This happens both in blocking and non-blocking mode. I would expect the snd_rawmidi_write() to accept only a part of the message. My hardware is Miditech Midiface 4x4, a USB midi interface: $ cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xe0610000 irq 45 1 [MIDI4x4 ]: USB-Audio - MIDI4x4 MIDIPLUS MIDI4x4 at usb-0000:00:14.0-1, full speed usbid: 1acc:1a0b My alsa is quite recent (aplay shows version 1.0.27.2), I am running Ubuntu with 3.13.0-39-generic kernel. I have tested the system by connecting the output port to an input port on the hardware and having another program output the received bytes on stdout. My observations: - As long as the buffer size is 276 bytes or less it gets sent ok - Larger buffer sizes cause corruption and roughly 300 bytes of received. - Alsa reports midi ring buffer size of 4096. - snd_rawmidi_status_get_xruns() on both ends reports 0 errors. - If I write N=128 byte blocks and usleep for 352*N after each write, everything works ok. Is this due to hardware incompatibility with alsa driver or what is going on? Is there a way to connect two programs via a virtual RawMIDI port so that I could test without using the hardware? Any other diagnostics that I could do? My test programs: write.cpp: int main( void ) { int intstatus; snd_rawmidi_t *midiout; if( (intstatus = snd_rawmidi_open(NULL, &midiout, "hw:1,0,1", 0)) ) { std::cerr << "unable to open MIDI out port: " << snd_strerror(intstatus) << "\n"; exit( 1 ); } int N = 3000; // 276 bytes gets through ok. unsigned char msg[N]; for( int a = 0; a < N; a++ ) { msg[a] = a % 0x80; } msg[0] = 0xF0; msg[N-1] = 0xF7; ssize_t wrstatus; if( (wrstatus = snd_rawmidi_write(midiout, msg, N)) < 0 ) { std::cerr << "unable to write MIDI port: " << snd_strerror(wrstatus) << "\n"; return( -1 ); } std::cout << "write status = " << wrstatus << "\n"; if( (intstatus = snd_rawmidi_drain( midiout )) ) { std::cerr << "unable to drain: " << snd_strerror(intstatus) << "\n"; exit( 1 ); } // Sleep for the amount of time needed for transfer + 10 % extra + 100 ms usleep( 352*N + 100000 ); snd_rawmidi_status_t *stat; snd_rawmidi_status_malloc( &stat ); snd_rawmidi_status( midiout, stat ); int err = snd_rawmidi_status_get_xruns( stat ); std::cout << "xruns = " << err << "\n"; snd_rawmidi_close( midiout ); return( 0 ); } -------------------- read.cpp: bool quit = false; void signal_int( int ) { quit = true; } int main( void ) { int mode = SND_RAWMIDI_NONBLOCK; int intstat; std::string portname = "hw:1,0,0"; //std::string portname = "virtual"; snd_rawmidi_t* midiin; if( (intstat = snd_rawmidi_open(&midiin, NULL, portname.c_str(), mode)) < 0 ) { std::cerr << "unable to open MIDI port " << portname << ": " << snd_strerror(intstat) << "\n"; exit( 1 ); } int count = 0; unsigned char buf[1]; std::cout << std::dec << std::setfill(' ') << std::setw(6) << count << " "; signal( SIGINT, signal_int ); while( !quit ) { int read = snd_rawmidi_read( midiin, buf, 1 ); if( read == 0 || read == -EBUSY || read == -EAGAIN ) { usleep( 320 ); continue; } else if( read < 0 ) { std::cerr << "unable to read MIDI port" << portname << ": " << snd_strerror(read) << "\n"; exit( 1 ); } std::cout << std::hex << std::setfill('0') << "0x" << std::setw(2) << (int)buf[0] << " " << std::flush; count++; if( count % 8 == 0 ) { std::cout << "\n"; std::cout << std::dec << std::setfill(' ') << std::setw(6) << count << " " << std::flush; } } snd_rawmidi_status_t *stat; snd_rawmidi_status_malloc( &stat ); snd_rawmidi_status( midiin, stat ); int err = snd_rawmidi_status_get_xruns( stat ); std::cout << "\nxruns = " << err << "\n"; snd_rawmidi_close( midiin ); return( 0 ); } Thanks. Taneli -- Taneli Kalvas Ph.D., Researcher Department of Physics, room FL114 P.O. Box 35 (YFL) 40014 University of Jyväskylä, Finland Mobile: +358-44-314-1602 Fax: +358-14-617-411 Email: taneli.kalvas(a)jyu.fi

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[alsa-devel] [PATCH resend] ASoC: fsl-ssi: Fix bitclock calculation for master mode
by Marc Kleine-Budde 25 Mar '15

25 Mar '15

From: Sascha Hauer <s.hauer(a)pengutronix.de> in fsl_ssi_set_bclk we are looking for the best divider by iterating over all possible dividers. The divider is in the range 1..256, so make this clear by using 'div' as name for the loop counter and iterate from 1 to 256 instead of 0 to 255. This makes obvious that tmprate is calculated wrong, it has to be freq * factor * div, not freq * factor * (i + 2). Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Signed-off-by: Marc Kleine-Budde <mkl(a)pengutronix.de> --- sound/soc/fsl/fsl_ssi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b9fabbf69db6..b8af08e49085 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -578,7 +578,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); struct regmap *regs = ssi_private->regs; int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; - u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; + u32 pm = 999, div2, psr, stccr, mask, afreq, factor, div; unsigned long clkrate, baudrate, tmprate; u64 sub, savesub = 100000; unsigned int freq; @@ -602,8 +602,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, factor = (div2 + 1) * (7 * psr + 1) * 2; - for (i = 0; i < 255; i++) { - tmprate = freq * factor * (i + 2); + for (div = 1; div < 256; div++) { + tmprate = freq * factor * div; if (baudclk_is_used) clkrate = clk_get_rate(ssi_private->baudclk); @@ -618,7 +618,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, continue; clkrate /= factor; - afreq = clkrate / (i + 1); + afreq = clkrate / div; if (freq == afreq) sub = 0; @@ -636,7 +636,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, if (sub < savesub) { baudrate = tmprate; savesub = sub; - pm = i; + pm = div - 1; } /* We are lucky */ -- 2.1.4

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[alsa-devel] [PATCH resend] ASoC: wm8741: Fix rates constraints values
by Sergej Sawazki 25 Mar '15

25 Mar '15

The WM8741 DAC supports the following typical audio sampling rates: 44.1kHz, 88.2kHz, 176.4kHz (eg: with a master clock of 22.5792MHz) 32kHz, 48kHz, 96kHz, 192kHz (eg: with a master clock of 24.576MHz) For the rates lists, we should use 82000 instead of 88235, 176400 instead of 1764000 and 192000 instead of 19200 (seems to be a typo). Signed-off-by: Sergej Sawazki <ce3a(a)gmx.de> --- sound/soc/codecs/wm8741.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 31bb480..9e71c76 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -123,7 +123,7 @@ static struct { }; static const unsigned int rates_11289[] = { - 44100, 88235, + 44100, 88200, }; static const struct snd_pcm_hw_constraint_list constraints_11289 = { @@ -150,7 +150,7 @@ static const struct snd_pcm_hw_constraint_list constraints_16384 = { }; static const unsigned int rates_16934[] = { - 44100, 88235, + 44100, 88200, }; static const struct snd_pcm_hw_constraint_list constraints_16934 = { @@ -168,7 +168,7 @@ static const struct snd_pcm_hw_constraint_list constraints_18432 = { }; static const unsigned int rates_22579[] = { - 44100, 88235, 1764000 + 44100, 88200, 176400 }; static const struct snd_pcm_hw_constraint_list constraints_22579 = { @@ -186,7 +186,7 @@ static const struct snd_pcm_hw_constraint_list constraints_24576 = { }; static const unsigned int rates_36864[] = { - 48000, 96000, 19200 + 48000, 96000, 192000 }; static const struct snd_pcm_hw_constraint_list constraints_36864 = { -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: dapm: Fix build warning
by Charles Keepax 25 Mar '15

25 Mar '15

commit c66150824b8a ("ASoC: dapm: add code to configure dai link parameters") introduced the following build warning: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': sound/soc/soc-dapm.c:3389:4: warning: passing argument 1 of 'snprintf' discards 'const' qualifier from pointer target type snprintf(w_param_text[count], len, This patch fixes this by switching to using devm_kasprintf. This also saves a couple of lines of code. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v2: - Use devm_kasprintf sound/soc/soc-dapm.c | 13 +++++-------- 1 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 049b300..4bd2a6f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3346,7 +3346,6 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, { struct snd_soc_dapm_widget template; struct snd_soc_dapm_widget *w; - size_t len; char *link_name; int ret, count; unsigned long private_value; @@ -3366,28 +3365,26 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w_param_text) return -ENOMEM; - len = strlen(source->name) + strlen(sink->name) + 2; - link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); if (!link_name) { ret = -ENOMEM; goto outfree_w_param; } - snprintf(link_name, len, "%s-%s", source->name, sink->name); for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { dev_warn(card->dapm.dev, "ASoC: anonymous config %d for dai link %s\n", count, link_name); - len = strlen("Anonymous Configuration ") + 3; w_param_text[count] = - devm_kzalloc(card->dev, len, GFP_KERNEL); + devm_kasprintf(card->dev, GFP_KERNEL, + "Anonymous Configuration %d", + count); if (!w_param_text[count]) { ret = -ENOMEM; goto outfree_link_name; } - snprintf(w_param_text[count], len, - "Anonymous Configuration %d", count); } else { w_param_text[count] = devm_kmemdup(card->dev, config->stream_name, -- 1.7.2.5

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[alsa-devel] [ASoC] Free kcontrol memory while unregistering codec
by Vishal T 25 Mar '15

25 Mar '15

Hi, While going through the codec cleanup code (snd_soc_unregister_codec()), I could not find the code which frees memory allocated for snd_kcontrol (while adding the kcontrols to card). Am I missing anything obvious? Sorry if I have created noise. Thanks, V

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[alsa-devel] [PATCH] ASoC: DAPM: Fix build warning
by Charles Keepax 25 Mar '15

25 Mar '15

commit c66150824b8a ("ASoC: dapm: add code to configure dai link parameters") introduced the following build warning: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': sound/soc/soc-dapm.c:3389:4: warning: passing argument 1 of 'snprintf' discards 'const' qualifier from pointer target type snprintf(w_param_text[count], len, This patch fixes this by adding a non-const temporary variable to hold the value as it is created before assigning to the entry in w_param_text. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 049b300..7640370 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3376,18 +3376,20 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { + char *anon_name; + dev_warn(card->dapm.dev, "ASoC: anonymous config %d for dai link %s\n", count, link_name); len = strlen("Anonymous Configuration ") + 3; - w_param_text[count] = - devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!w_param_text[count]) { + anon_name = devm_kzalloc(card->dev, len, GFP_KERNEL); + if (!anon_name) { ret = -ENOMEM; goto outfree_link_name; } - snprintf(w_param_text[count], len, + snprintf(anon_name, len, "Anonymous Configuration %d", count); + w_param_text[count] = anon_name; } else { w_param_text[count] = devm_kmemdup(card->dev, config->stream_name, -- 1.7.2.5

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[alsa-devel] [RFC v2 0/3] ALSA: hda: add hda controller to hda core
by Vinod Koul 25 Mar '15

25 Mar '15

This series starts adding base hda controller code to hda core. There are few more changes like dsp loader etc planned to be added as well. This is based on Takashi's test/hda-core branch with patches posted last week, we have modified naming convention loosely based on takashi's work, Takshi feel free to recommend more changes if you think they are required Jeeja KP (2): ALSA: hda: move common hda controller register defines up ALSA: hda: add hda controller to hda Vinod Koul (1): ALSA: hda: add HDA_MAX_CODECS include/sound/hda_registers.h | 200 +++++++ include/sound/hda_verbs.h | 1 + include/sound/hdaudio.h | 215 +++++++ sound/hda/Makefile | 2 +- sound/hda/hdac_controller.c | 1284 ++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_controller.c | 4 +- sound/pci/hda/hda_controller.h | 203 +------ 7 files changed, 1707 insertions(+), 202 deletions(-) create mode 100644 include/sound/hda_registers.h create mode 100644 sound/hda/hdac_controller.c -- 1.7.9.5

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[alsa-devel] [PATCH v2] ASoC: dapm: Fix build warning
by Charles Keepax 25 Mar '15

25 Mar '15

commit c66150824b8a ("ASoC: dapm: add code to configure dai link parameters") introduced the following build warning: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_new_pcm': sound/soc/soc-dapm.c:3389:4: warning: passing argument 1 of 'snprintf' discards 'const' qualifier from pointer target type snprintf(w_param_text[count], len, This patch fixes this by adding a non-const temporary variable to hold the value as it is created before assigning to the entry in w_param_text. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Just small fix to commit message sound/soc/soc-dapm.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 049b300..7640370 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3376,18 +3376,20 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { + char *anon_name; + dev_warn(card->dapm.dev, "ASoC: anonymous config %d for dai link %s\n", count, link_name); len = strlen("Anonymous Configuration ") + 3; - w_param_text[count] = - devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!w_param_text[count]) { + anon_name = devm_kzalloc(card->dev, len, GFP_KERNEL); + if (!anon_name) { ret = -ENOMEM; goto outfree_link_name; } - snprintf(w_param_text[count], len, + snprintf(anon_name, len, "Anonymous Configuration %d", count); + w_param_text[count] = anon_name; } else { w_param_text[count] = devm_kmemdup(card->dev, config->stream_name, -- 1.7.2.5

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[alsa-devel] Missing beep controls for Via HDA codec (snd_hda_codec_via)
by W. Trevor King 25 Mar '15

25 Mar '15

My VT1802 chip has a beep-generating node: $ cat /proc/asound/card1/codec#0 Codec: VIA VT1802 ... Vendor Id: 0x11068446 Subsystem Id: 0x15587410 Revision Id: 0x100000 ... Node 0x22 [Beep Generator Widget] wcaps 0x70040c: Mono Amp-Out Amp-Out caps: ofs=0x0a, nsteps=0x12, stepsize=0x05, mute=1 Amp-Out vals: [0x0a] Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 ... But I'm missing the: Control: name=... entries that I think I need to manage this widget from alsamixer. For example, sambrian posts codec output for a CX20561 chip which has [1]: Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] I can patch with something like this: diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 3de6d3d..85a0c08 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1520,8 +1520,10 @@ static int patch_vt2002P(struct hda_codec *codec) spec->gen.mixer_nid = 0x21; override_mic_boost(codec, 0x2b, 0, 3, 40); override_mic_boost(codec, 0x29, 0, 3, 40); - if (spec->codec_type == VT1802) + if (spec->codec_type == VT1802) { + spec->gen.beep_nid = 0x22; fix_vt1802_connections(codec); + } add_secret_dac_path(codec); snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); to get a dmesg entry like: input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card1/hdaudioC1D0/input15 but that doesn't add the beep controls. Looking at patch_analog.c, patch_conexant.c, and patch_realtek.c for inspiration, I expect we'll need an HDA_CODEC_MUTE_BEEP(_MONO) entry, and possibly an HDA_CODEC_VOLUME(_MONO) entry, but I'm not sure if these should be mono or not, or if they should be inputs (following patch_realtek.c) or outputs (following patch_analog.c and patch_conexant.c). I've filled out Via's datasheet-request form [2], but I don't know how likely that is to be accepted. Does anyone have a VT1802 datasheet that they're allowed to share (or some free time to fill in Via beep support on their own)? Or am I barking up the wrong tree? My only previous experience with this code-base was a beep-tweak for a previous box's Realtek chip [3], but I love my beeps and am happy to poke around with a datasheet until I get this working ;). Maybe folks with more experience can just guess the right code without needing to see specs for the VT1802? Cheers, Trevor [1]: http://forums.fedoraforum.org/archive/index.php/t-265822.html [2]: http://www.via.com.tw/en/support/contact_us/ [3]: http://thread.gmane.org/gmane.linux.alsa.devel/121300 -- This email may be signed or encrypted with GnuPG (http://www.gnupg.org). For more information, see http://en.wikipedia.org/wiki/Pretty_Good_Privacy

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Re: [alsa-devel] ASoC: Max98090 Codec DAPM Widget setup in Open Embedded
by noman pouigt 25 Mar '15

25 Mar '15

On Mon, Mar 23, 2015 at 11:53 PM, noman pouigt <variksla(a)gmail.com> wrote: >> I would like to know which file I can use to connect/enable >> DPAM widgets permanently so a simple aplay/record can work? >> >> My environment is: >> - Yocto Linux Kernel 3.10 - (DTS based) >> - Max98090 Codec >> - imx6SL SoC >> >> I thought about doing this in alsa.conf or asound.conf but I couldn't >> find a single example with DAPM widgets . > > I am also stuck with the same problem. I am sure ALSA experts > can help here. > > What i want to know is how to set the dapm widgets in open > embedded? Is it alsa.conf or asound.conf ? > > I need to set something like this: > amixer cset name="LEFT MIXER DAC" 1 > > Using amixer i am able to playback but i don't know how can i > modify conf file to have this dapm widgets associated with > a device. > > Any example would suffice.

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