July 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: audio-graph-scu-card: fix balance of of_node_get/put()
by Kuninori Morimoto 31 Jul '17

31 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> audio-graph-scu-card with CONFIG_OF_DYNAMIC triggers several errors at kernel boot, like the following one OF: ERROR: Bad of_node_put() on /soc/sound@ec500000/ports/port@1 This patch doesn't call unnecessary of_node_put() which was called in other function (= of_graph_get_port_parent(), of_phandle_iterator_next()) Reported-by: Antonio Borneo <borneo.antonio(a)gmail.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-scu-card.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index dcd2df3..bff1e6a 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -215,7 +215,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) codec_ep = of_graph_get_remote_endpoint(cpu_ep); rcpu_ep = of_graph_get_remote_endpoint(codec_ep); - of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); of_node_put(rcpu_ep); @@ -248,11 +247,9 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); + codec_port = of_graph_get_port_parent(codec_ep); /* call put */ - of_node_put(cpu_port); of_node_put(cpu_ep); - of_node_put(codec_ep); of_node_put(codec_port); if (codec) { @@ -304,11 +301,9 @@ static int asoc_graph_get_dais_count(struct device *dev) cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); + codec_port = of_graph_get_port_parent(codec_ep); /* call put */ - of_node_put(cpu_port); of_node_put(cpu_ep); - of_node_put(codec_ep); of_node_put(codec_port); if (cpu_ep) -- 1.9.1

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[alsa-devel] snd_ctl_add_enumerated_elem_set and TLV
by Fulup Ar Foll 31 Jul '17

31 Jul '17

Hi, I'm using snd_ctl_add_enumerated_elem_set to create user controls, I do not understand why it set "TLV readable" option at true by default. I have a simple enumeration without TLV. I use snd_ctl_elem_write but do not call snd_ctl_elem_tlv_write. I do no understand how to prevent snd_ctl_elem_info_is_tlv_readable to return true for this control, obviously even if TLV is set as readable, it is empty and reading fail. Example fulup@fulup-desktop:~/Workspace/AlsaPulse> amixer -D hw:USB cget numid=237 numid=237,iface=MIXER,name='Hal-VolRamp-Mode' ; type=ENUMERATED,access=rw---RW-,values=1,items=4 ; Item #0 'None' ; Item #1 'Normal' ; Item #2 'Smooth' ; Item #3 'Emergency' : values=1 amixer: Control hw:USB element TLV read error: No such device or address Question: is there something special to set when a control does not have a TLV ? Fulup PS: if I write a fake TLV everything goes fine, but this is not what I expect for ENUM.

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[alsa-devel] [PATCH] ASoC: fix balance of of_node_get()/of_node_put()
by Antonio Borneo 30 Jul '17

30 Jul '17

On Hikey target board, enabling CONFIG_OF_DYNAMIC triggers several errors at kernel boot, like the following one OF: ERROR: Bad of_node_put() on /soc/i2s@f7118000/ports/port@0/endpoint each followed by stack dump. Fixed by: - removing of_node_put() in the body of of_for_each_phandle(){}, since already provided at each iteration. Add it in case the loop is break out; - adding of_node_get() before calling of_graph_get_port_parent() or asoc_graph_card_dai_link_of(). Tested with kernel v4.13-rc1 with hikey_defconfig taken from https://git.linaro.org/people/john.stultz/android-dev.git branch dev/hikey-mainline-WIP Signed-off-by: Antonio Borneo <borneo.antonio(a)gmail.com> --- To: Liam Girdwood <lgirdwood(a)gmail.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> To: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Cc: Wei Xu <xuwei5(a)hisilicon.com> Cc: John Stultz <john.stultz(a)linaro.org> Cc: linux-arm-kernel(a)lists.infradead.org --- sound/soc/generic/audio-graph-card.c | 14 +++++++++----- sound/soc/generic/simple-card-utils.c | 5 +++++ sound/soc/soc-core.c | 5 +++++ 3 files changed, 19 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index c0f08a6..7574c5f 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -228,10 +228,16 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) */ of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + /* + * asoc_graph_card_dai_link_of() will call + * of_node_put(). So, call of_node_get() here + */ + of_node_get(it.node); ret = asoc_graph_card_dai_link_of(it.node, priv, idx++); - of_node_put(it.node); - if (ret < 0) + if (ret < 0) { + of_node_put(it.node); return ret; + } } return asoc_simple_card_parse_card_name(card, NULL); @@ -244,10 +250,8 @@ static int asoc_graph_get_dais_count(struct device *dev) int count = 0; int rc; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + of_for_each_phandle(&it, rc, node, "dais", NULL, 0) count++; - of_node_put(it.node); - } return count; } diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 26d64fa..144954b 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -250,6 +250,11 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) if (ret != -ENOTSUPP) return ret; + /* + * of_graph_get_port_parent() will call + * of_node_put(). So, call of_node_get() here + */ + of_node_get(ep); node = of_graph_get_port_parent(ep); /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 921622a..a0f39de 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4087,6 +4087,11 @@ int snd_soc_get_dai_id(struct device_node *ep) struct device_node *node; int ret; + /* + * of_graph_get_port_parent() will call + * of_node_put(). So, call of_node_get() here + */ + of_node_get(ep); node = of_graph_get_port_parent(ep); /* -- 1.9.1

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[alsa-devel] Fwd: Fwd: Re: Basic questions
by tsepp2001＠t-online.de 30 Jul '17

30 Jul '17

Hi Clemens, thank you. In blocking-mode it works. Then i will follow the non-blocking mode. I modified first the callback: void playback_callback (snd_pcm_sframes_t nframes) { snd_pcm_sframes_t avail; avail = snd_pcm_avail_update(handle); while (avail >= period_size) { snd_pcm_writei(handle, buffer_alsa , nframes); avail = snd_pcm_avail_update(handle); std::cout << " Avail: " << avail << std::endl; } } ... called it at the end: playback_callback(period_size); and I get (for "avail"): Frames: 6894888 buffer_size (frames): 2048 buffer_time (frames): 46439 period_size (frames): 256 period_time (usec): 5804 periods (count): 8 Avail: 1282 Avail: 1027 Avail: 772 Avail: 517 Avail: 261 Avail: 6 the same mistake... Regards Thomas tsepp2001(a)t-online.de wrote: > Am I right to use async mode No. It is not supported by all devices, and almost impossible to use correctly. > to get lowest latency times Latency depends on the buffer size, and how fast your code can react to the notification that some space is available. > and using pause/play/stop? You probably want to use non-blocking mode. Regards, Clemens _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] Gina 3g and Echomixer issue
by Britt Yazel 30 Jul '17

30 Jul '17

Hey guys, I work in an auditory neuroscience research lab, and we are using a PCI Gina 3g to communicate with a Fireface 800 over ADAT. My issue is this, 1) each time I reboot the system the digital mode is set to Coaxial digital, rather than ADAT, and I must set it after each reboot. Is there a way to make this setting persist through reboot, or perhaps a parameter that can be set when the kernel module is loaded to force ADAT? 2) The first time I switch from Coaxial digital to ADAT upon a reboot, I consistently get a loud squeeling sound coming from my studio monitors. I don't think this is the intended behaviuor, and it makes me worried when switching into ADAT mode. Granted, after switching the first time to ADAT, I can switch between coaxial digital and ADAT without the squeeling happeneing, it is just that first time after the reboot. Thanks for the help guys. The Gina 3g, even though it is an old audio device, is still our most reliable sound card. We have tried many others with horrible success, and most other ADAT capable PCI/PCIe sound cards are far too expensive, and don't have robust Linux support. However, if you guys have a good reccomendation of an ADAT capable PCIe sound card with full Linux support, please let me know. -Britt

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[alsa-devel] Basic questions
by tsepp2001＠t-online.de 30 Jul '17

30 Jul '17

Hi everybody, I am new to ALSA-API and would be glad, if someone give me an advice to the following questions. Up to now I used Rtaudio for cross compiling the player for windows and Linux. But know I need to go more into ALSA... - Am I right to use async mode to get lowest latency times and using pause/play/stop? - I used the following piece of code. But don't get any output. The callback output shows "avail" with very short count of frames... I write a whole file (raw) to buffer, called buffer_alsa with 6894888 frames. What is going wrong? I get this output: Frames: 6894888 buffer_size (frames): 2048 buffer_time (frames): 46439 period_size (frames): 256 period_time (usec): 5804 periods (count): 8 Avail: 1540 Avail: 1285 Avail: 1030 Avail: 775 Avail: 520 Avail: 265 Avail: 10 Avail: 2 Avail: 2 Avail: 3 Avail: 2 ... The code follows ... Best regards Thomas ----------------------------- void MyCallback(snd_async_handler_t *pcm_callback) { snd_pcm_t *handle = snd_async_handler_get_pcm(pcm_callback); snd_pcm_sframes_t avail; //std::cout << "Period_size: " << period_size << std::endl; avail = snd_pcm_avail_update(handle); while (avail >= period_size) { snd_pcm_writei(handle, buffer_alsa , period_size); avail = snd_pcm_avail_update(handle); std::cout << " Avail: " << avail << std::endl; } } //---------------------------------------------------- snd_pcm_hw_params_alloca(&params); if ((err = snd_pcm_hw_params_any(handle, params)) < 0) std::cout << " Error (reading configuration): " << snd_strerror(err) << std::endl; /* Set parameters */ if ((err = snd_pcm_hw_params_set_access(handle, params,SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) //printf("ERROR: Can't set interleaved mode. %s\n", snd_strerror(err)); std::cout << " Error (can't set interleaved mode): " << snd_strerror(err) << std::endl; if ((err = snd_pcm_hw_params_set_format(handle, params, FORMAT)) < 0) //printf("ERROR: Can't set format. %s\n", snd_strerror(err)); std::cout << " Error (can't set format): " << snd_strerror(err) << std::endl; if ((err = snd_pcm_hw_params_set_channels(handle, params, data.channels)) < 0) //printf("ERROR: Can't set channels number. %s\n", snd_strerror(err)); std::cout << " Error (can't set channels number): " << snd_strerror(err) << std::endl; if ((err = snd_pcm_hw_params_set_rate_near(handle, params, &fs, 0)) < 0) //printf("ERROR: Can't set rate. %s\n", snd_strerror(err)); std::cout << " Error (can't set rate): " << snd_strerror(err) << std::endl; if ((err = snd_pcm_hw_params_set_buffer_size_near(handle, params, &buffer_size)) < 0) std::cout << " Error (can't set buffer size): " << snd_strerror(err) << std::endl; if ((err = snd_pcm_hw_params_set_period_size_near (handle, params, &period_size, NULL)) < 0) std::cout << " Error (can't set period size): " << snd_strerror(err) << std::endl; /* Write parameters */ if ((err = snd_pcm_hw_params(handle, params)) < 0) //printf("ERROR: Can't set hardware parameters. %s\n", snd_strerror(err)); std::cout << " Error (can't set harware params): " << snd_strerror(err) << std::endl; //---------------------- ------------------------------ // Get Parameters snd_pcm_hw_params_get_buffer_size(params, &buffer_sz); std::cout << " buffer_size (frames): " << buffer_sz << std::endl; snd_pcm_hw_params_get_buffer_time(params, &buffer_tm, NULL); std::cout << " buffer_time (frames): " << buffer_tm << std::endl; snd_pcm_hw_params_get_period_size(params, &aframes, 0); std::cout << " period_size (frames): " << aframes << std::endl; snd_pcm_hw_params_get_period_time(params, &data.channels, NULL); // tmp Zeit in us std::cout << " period_time (usec): " << data.channels << std::endl; snd_pcm_hw_params_get_periods(params, &piods, 0); std::cout << " periods (count): " << piods << std::endl; // ---------------------------------------------------------------------------- //snd_pcm_hw_params_free (params); // --------------------- Hardware-Params End ------------------------------ snd_pcm_sw_params_t *sw_params; if ((err = snd_pcm_sw_params_malloc (&sw_params)) < 0) { std::cout << " Error (can't allocate sw params): " << snd_strerror(err) << std::endl; return; } if ((err = snd_pcm_sw_params_current(handle, sw_params)) < 0) { std::cout << "Error (can't return curr sw config): " << snd_strerror(err) << std::endl; return; } if ((err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, buffer_size)) < 0) // 0U oder 4096 ? { // This parameter controls the wake up point. If the count of available samples is equal or greater than this value, then the application will be activated. std::cout << "Error (can't set start thresh mode): " << snd_strerror(err) << std::endl; return; } /* if ((err = snd_pcm_sw_params_set_silence_threshold( handle, sw_params, 0 )) < 0) { std::cout << " Error (can't set silence thres): " << snd_strerror(err) << std::endl; return; } */ if ((err = snd_pcm_sw_params_set_avail_min (handle, sw_params, period_size)) < 0) { std::cout << " Error (can't set min avail count): " << snd_strerror(err) << std::endl; return; } // here are two options for a fix //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); /* snd_pcm_uframes_t val; if ((err = snd_pcm_sw_params_get_boundary(sw_params, &val)) < 0) { std::cout << " Error (can't get boundary): " << snd_strerror(err) << std::endl; return; } if ((err = snd_pcm_sw_params_set_silence_size(handle, sw_params, val)) < 0) { std::cout << " Error (can't set silence): " << snd_strerror(err) << std::endl; return; } */ if ((err = snd_pcm_sw_params(handle, sw_params)) < 0) { std::cout << " Error (can't set sw params): " << snd_strerror(err) << std::endl; return; } snd_pcm_sw_params_free(sw_params); // ------------------------- SW-Params End --------------------------------- if ((err = snd_pcm_prepare(handle)) < 0) { std::cout << " Error (can't prepare audio): " << snd_strerror(err) << std::endl; return; } snd_async_handler_t *pcm_callback; if ((err = snd_async_add_pcm_handler(&pcm_callback, handle, MyCallback, NULL)) < 0) { std::cout << " Error (can't add pcm handler): " << snd_strerror(err) << std::endl; return; } if ((err = snd_pcm_writei(handle, buffer_alsa, 2 * period_size) == -EPIPE)) { std::cout << " Error: XRUN" << std::endl; err = snd_pcm_recover(handle, err, 0); if ((err = snd_pcm_prepare (handle)) < 0) { std::cout << " Error (can't prepare audio): " << snd_strerror(err) << std::endl; return; } } if ((err = snd_pcm_start(handle)) < 0) { std::cout << " Error (can't start audio): " << snd_strerror(err) << std::endl; return; } void MyCallback(snd_async_handler_t *pcm_callback);

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[alsa-devel] [PATCH] ASoC: soc-pcm: Remove unused 'debugfs_dpcm_state' entry
by Fabio Estevam 29 Jul '17

29 Jul '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> 'debugfs_dpcm_state' member from structure snd_soc_pcm_runtime is never used at all, so it is safe to remove it. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- include/sound/soc.h | 1 - sound/soc/soc-pcm.c | 5 ++--- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8f9376f..9d6e1db 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1233,7 +1233,6 @@ struct snd_soc_pcm_runtime { struct delayed_work delayed_work; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dpcm_root; - struct dentry *debugfs_dpcm_state; #endif unsigned int num; /* 0-based and monotonic increasing */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7d3859e..24148de 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -3000,8 +3000,7 @@ void soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd) return; } - rtd->debugfs_dpcm_state = debugfs_create_file("state", 0444, - rtd->debugfs_dpcm_root, - rtd, &dpcm_state_fops); + debugfs_create_file("state", 0444, rtd->debugfs_dpcm_root, + rtd, &dpcm_state_fops); } #endif -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: rt5514: Use the local variable to get the data from SPI device
by Mark Brown 28 Jul '17

28 Jul '17

On Mon, Jul 24, 2017 at 03:06:36AM +0000, Oder Chiou wrote: > > On Fri, Jul 21, 2017 at 04:00:47PM +0800, Oder Chiou wrote: > > > The patch uses the local variable to get the data from SPI device to > > > prevent to get all zero data. > > How do you end up with all zero data from reading directly into rxbuf? > While the "runtime->dma_area" put into the rxbuf directly on the Intel > platform, it seemed that had the limitation to let the memory cannot be > written the data using the driver of the SPI controller. We verified the > original code on the platforms of Nvidia and Rockchip, and these are > working fine. Do we understand why that is? Note that IIRC you're not really supposed to DMA to or from the stack, there are some platforms with restrictions there so it's possible you're just moving the problem around here if rxbuf can be on the stack.

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[alsa-devel] [PATCH 0/5] ALSA: constify snd_timer_hardware structures
by Bhumika Goyal 28 Jul '17

28 Jul '17

Declare snd_timer_hardware structures as const. Bhumika Goyal (5): ALSA: sound/core: constify snd_timer_hardware structures ALSA: isa: constify snd_timer_hardware structures ALSA: pci: constify snd_timer_hardware structures ALSA: sparc: constify snd_timer_hardware structure ALSA: opl3: constify snd_timer_hardware structures sound/core/hrtimer.c | 2 +- sound/core/pcm_timer.c | 2 +- sound/core/timer.c | 2 +- sound/drivers/opl3/opl3_lib.c | 4 ++-- sound/isa/ad1816a/ad1816a_lib.c | 2 +- sound/isa/gus/gus_timer.c | 4 ++-- sound/isa/wss/wss_lib.c | 2 +- sound/pci/emu10k1/timer.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 2 +- sound/sparc/cs4231.c | 2 +- 10 files changed, 12 insertions(+), 12 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: Reset hw_ptr on resume trigger
by Cheng-Yi Chiang 28 Jul '17

28 Jul '17

From: "U. Artie Eoff" <ullysses.a.eoff(a)intel.com> Reset the hw_ptr before queuing the restore_stream_context work to eradicate a nasty white audio noise on resume. Signed-off-by: U. Artie Eoff <ullysses.a.eoff(a)intel.com> --- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index 4765ad474544..e0db7070cd42 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -187,8 +187,10 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) sst_byt_stream_start(byt, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_RESUME: - if (pdata->restore_stream == true) + if (pdata->restore_stream == true) { + pcm_data->hw_ptr = 0; schedule_work(&pcm_data->work); + } else sst_byt_stream_resume(byt, pcm_data->stream); break; -- 2.12.2

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