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

[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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[alsa-devel] [v2 0/3] ASoC: Intel: Fixes in Kabylake machine driver for clear playback and recording
by Harsha Priya 28 Jul '17

28 Jul '17

This patch series includes fixes for clean playback and recording in kabylake machine driver. It fixes the below issues in the corresponding patches and the details of each fix is provided in the patch's commit message. 1. Fix cracking noise in rt5663 headphones 2. Fix pop noise of dmic recording 3. Enable 4 slot IV feedback for maxim amps Harsha Priya (3): ASoC: Intel: Fix cracking noise in rt5663 headphones for kabylake platform ASoC: Intel: Fix pop noise of dmic recording in kabylake platform using rt5514 ASoC: Intel: Enabling 4 slot IV feedback for max98927 on Kabylake platform .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 28 +++++++++++----------- 1 file changed, 14 insertions(+), 14 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: codecs: add wm8524 codec driver
by Mihai Serban 28 Jul '17

28 Jul '17

WM8524 is a 24-bit 192KHz stereo digital/analog converter (DAC) with integral charge pump and a simple hardware control interface. Product information can be found at: https://www.cirrus.com/products/wm8524/ Signed-off-by: Mihai Serban <mihai.serban(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/wm8524.txt | 16 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8524.c | 265 +++++++++++++++++++++ 4 files changed, 288 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8524.txt create mode 100644 sound/soc/codecs/wm8524.c diff --git a/Documentation/devicetree/bindings/sound/wm8524.txt b/Documentation/devicetree/bindings/sound/wm8524.txt new file mode 100644 index 0000000..20c6200 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8524.txt @@ -0,0 +1,16 @@ +WM8524 audio CODEC + +This device does not use I2C or SPI but a simple Hardware Control Interface. + +Required properties: + + - compatible : "wlf,wm8524" + + - wlf,mute-gpios: a GPIO spec for the MUTE pin. + +Example: + +codec: wm8524@0 { + compatible = "wlf,wm8524"; + wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 024ddc9..010811e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -174,6 +174,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8523 if I2C + select SND_SOC_WM8524 if GPIOLIB select SND_SOC_WM8580 if I2C select SND_SOC_WM8711 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8727 @@ -974,6 +975,10 @@ config SND_SOC_WM8523 tristate "Wolfson Microelectronics WM8523 DAC" depends on I2C +config SND_SOC_WM8524 + tristate "Wolfson Microelectronics WM8524 DAC" + depends on GPIOLIB + config SND_SOC_WM8580 tristate "Wolfson Microelectronics WM8580 and WM8581 CODECs" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a8a4b07..12dc2f7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -183,6 +183,7 @@ snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8523-objs := wm8523.o +snd-soc-wm8524-objs := wm8524.o snd-soc-wm8580-objs := wm8580.o snd-soc-wm8711-objs := wm8711.o snd-soc-wm8727-objs := wm8727.o @@ -416,6 +417,7 @@ obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8523) += snd-soc-wm8523.o +obj-$(CONFIG_SND_SOC_WM8524) += snd-soc-wm8524.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o obj-$(CONFIG_SND_SOC_WM8727) += snd-soc-wm8727.o diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c new file mode 100644 index 0000000..f42ec66 --- /dev/null +++ b/sound/soc/codecs/wm8524.c @@ -0,0 +1,265 @@ +/* + * wm8524.c -- WM8524 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2017 NXP + * + * Based on WM8523 ALSA SoC Audio driver written by Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> + +#define WM8524_NUM_RATES 7 + +/* codec private data */ +struct wm8524_priv { + struct gpio_desc *mute; + unsigned int sysclk; + unsigned int rate_constraint_list[WM8524_NUM_RATES]; + struct snd_pcm_hw_constraint_list rate_constraint; +}; + + +static const struct snd_soc_dapm_widget wm8524_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LINEVOUTL"), +SND_SOC_DAPM_OUTPUT("LINEVOUTR"), +}; + +static const struct snd_soc_dapm_route wm8524_dapm_routes[] = { + { "LINEVOUTL", NULL, "DAC" }, + { "LINEVOUTR", NULL, "DAC" }, +}; + +static const struct { + int value; + int ratio; +} lrclk_ratios[WM8524_NUM_RATES] = { + { 1, 128 }, + { 2, 192 }, + { 3, 256 }, + { 4, 384 }, + { 5, 512 }, + { 6, 768 }, + { 7, 1152 }, +}; + +static int wm8524_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(codec); + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!wm8524->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &wm8524->rate_constraint); + + gpiod_set_value_cansleep(wm8524->mute, 1); + + return 0; +} + +static void wm8524_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(codec); + + gpiod_set_value_cansleep(wm8524->mute, 0); +} + +static int wm8524_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int i, j = 0; + + wm8524->sysclk = freq; + + wm8524->rate_constraint.count = 0; + for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { + val = freq / lrclk_ratios[i].ratio; + /* Check that it's a standard rate since core can't + * cope with others and having the odd rates confuses + * constraint matching. + */ + switch (val) { + case 8000: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + case 176400: + case 192000: + dev_err(codec->dev, "Supported sample rate: %dHz\n", + val); + wm8524->rate_constraint_list[j++] = val; + wm8524->rate_constraint.count++; + break; + default: + dev_dbg(codec->dev, "Skipping sample rate: %dHz\n", + val); + } + } + + /* Need at least one supported rate... */ + if (wm8524->rate_constraint.count == 0) + return -EINVAL; + + return 0; +} + +static int wm8524_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + fmt &= (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK | + SND_SOC_DAIFMT_MASTER_MASK); + + if (fmt != (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS)) { + dev_err(codec_dai->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + return 0; +} + +static int wm8524_mute_stream(struct snd_soc_dai *dai, int mute, int stream) +{ + struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(dai->codec); + + if (wm8524->mute) + gpiod_set_value_cansleep(wm8524->mute, mute); + + return 0; +} + +#define WM8524_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops wm8524_dai_ops = { + .startup = wm8524_startup, + .shutdown = wm8524_shutdown, + .set_sysclk = wm8524_set_dai_sysclk, + .set_fmt = wm8524_set_fmt, + .mute_stream = wm8524_mute_stream, +}; + +static struct snd_soc_dai_driver wm8524_dai = { + .name = "wm8524-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8524_RATES, + .formats = WM8524_FORMATS, + }, + .ops = &wm8524_dai_ops, +}; + +static int wm8524_probe(struct snd_soc_codec *codec) +{ + struct wm8524_priv *wm8524 = snd_soc_codec_get_drvdata(codec); + + wm8524->rate_constraint.list = &wm8524->rate_constraint_list[0]; + wm8524->rate_constraint.count = + ARRAY_SIZE(wm8524->rate_constraint_list); + + return 0; +} + +static const struct snd_soc_codec_driver soc_codec_dev_wm8524 = { + .probe = wm8524_probe, + + .component_driver = { + .dapm_widgets = wm8524_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8524_dapm_widgets), + .dapm_routes = wm8524_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8524_dapm_routes), + }, +}; + +#ifdef CONFIG_OF +static const struct of_device_id wm8524_of_match[] = { + { .compatible = "wlf,wm8524" }, + { /* sentinel*/ } +}; +MODULE_DEVICE_TABLE(of, wm8524_of_match); +#endif + +static int wm8524_codec_probe(struct platform_device *pdev) +{ + struct wm8524_priv *wm8524; + int ret; + + wm8524 = devm_kzalloc(&pdev->dev, sizeof(struct wm8524_priv), + GFP_KERNEL); + if (wm8524 == NULL) + return -ENOMEM; + + platform_set_drvdata(pdev, wm8524); + + wm8524->mute = devm_gpiod_get(&pdev->dev, "wlf,mute", GPIOD_OUT_LOW); + if (IS_ERR(wm8524->mute)) { + ret = PTR_ERR(wm8524->mute); + dev_err(&pdev->dev, "Failed to get mute line: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_wm8524, &wm8524_dai, 1); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); + snd_soc_unregister_platform(&pdev->dev); + } + + return ret; +} + +static int wm8524_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver wm8524_codec_driver = { + .probe = wm8524_codec_probe, + .remove = wm8524_codec_remove, + .driver = { + .name = "wm8524-codec", + .of_match_table = wm8524_of_match, + }, +}; +module_platform_driver(wm8524_codec_driver); + +MODULE_DESCRIPTION("ASoC WM8524 driver"); +MODULE_AUTHOR("Mihai Serban <mihai.serban(a)nxp.com>"); +MODULE_ALIAS("platform:wm8524-codec"); +MODULE_LICENSE("GPL"); -- 2.7.4

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