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

[alsa-devel] [PATCH] ALSA: usb-audio: Don't resubmit pending URBs at MIDI error recovery
by Takashi Iwai 06 Dec '14

06 Dec '14

In snd_usbmidi_error_timer(), the driver tries to resubmit MIDI input URBs to reactivate the MIDI stream, but this causes the error when some of URBs are still pending like: WARNING: CPU: 0 PID: 0 at ../drivers/usb/core/urb.c:339 usb_submit_urb+0x5f/0x70() URB ef705c40 submitted while active CPU: 0 PID: 0 Comm: swapper/0 Not tainted 3.16.6-2-desktop #1 Hardware name: FOXCONN TPS01/TPS01, BIOS 080015 03/23/2010 c0984bfa f4009ed4 c078deaf f4009ee4 c024c884 c09a135c f4009f00 00000000 c0984bfa 00000153 c061ac4f c061ac4f 00000009 00000001 ef705c40 e854d1c0 f4009eec c024c8d3 00000009 f4009ee4 c09a135c f4009f00 f4009f04 c061ac4f Call Trace: [<c0205df6>] try_stack_unwind+0x156/0x170 [<c020482a>] dump_trace+0x5a/0x1b0 [<c0205e56>] show_trace_log_lvl+0x46/0x50 [<c02049d1>] show_stack_log_lvl+0x51/0xe0 [<c0205eb7>] show_stack+0x27/0x50 [<c078deaf>] dump_stack+0x45/0x65 [<c024c884>] warn_slowpath_common+0x84/0xa0 [<c024c8d3>] warn_slowpath_fmt+0x33/0x40 [<c061ac4f>] usb_submit_urb+0x5f/0x70 [<f7974104>] snd_usbmidi_submit_urb+0x14/0x60 [snd_usbmidi_lib] [<f797483a>] snd_usbmidi_error_timer+0x6a/0xa0 [snd_usbmidi_lib] [<c02570c0>] call_timer_fn+0x30/0x130 [<c0257442>] run_timer_softirq+0x1c2/0x260 [<c0251493>] __do_softirq+0xc3/0x270 [<c0204732>] do_softirq_own_stack+0x22/0x30 [<c025186d>] irq_exit+0x8d/0xa0 [<c0795228>] smp_apic_timer_interrupt+0x38/0x50 [<c0794a3c>] apic_timer_interrupt+0x34/0x3c [<c0673d9e>] cpuidle_enter_state+0x3e/0xd0 [<c028bb8d>] cpu_idle_loop+0x29d/0x3e0 [<c028bd23>] cpu_startup_entry+0x53/0x60 [<c0bfac1e>] start_kernel+0x415/0x41a For avoiding these errors, check the pending URBs and skip resubmitting such ones. Reported-and-tested-by: Stefan Seyfried <stefan.seyfried(a)googlemail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index d3d49525a16b..5bfb695547f8 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -365,6 +365,8 @@ static void snd_usbmidi_error_timer(unsigned long data) if (in && in->error_resubmit) { in->error_resubmit = 0; for (j = 0; j < INPUT_URBS; ++j) { + if (atomic_read(&in->urbs[j]->use_count)) + continue; in->urbs[j]->dev = umidi->dev; snd_usbmidi_submit_urb(in->urbs[j], GFP_ATOMIC); } -- 2.2.0

2 2

[alsa-devel] [PATCH] ASoC: samsung: Fix error handling for clock lookup
by Mark Brown 05 Dec '14

05 Dec '14

Return the error code we got from clk_get() and check to make sure that clk_prepare_enable() worked. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 25919e090983..500b9ba5ebaa 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -972,6 +972,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; + int ret; if (other && other->clk) { /* If this is probe on secondary */ samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, @@ -989,9 +990,14 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) if (IS_ERR(i2s->clk)) { dev_err(&i2s->pdev->dev, "failed to get i2s_clock\n"); iounmap(i2s->addr); - return -ENOENT; + return PTR_ERR(i2s->clk); + } + + ret = clk_prepare_enable(i2s->clk); + if (ret != 0) { + dev_err(&i2s->pdev->dev, "failed to enable clock: %d\n", ret); + return ret; } - clk_prepare_enable(i2s->clk); samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); -- 2.1.3

1 0

[alsa-devel] [PATCH v2 1/2] ALSA: sound/atmel/ac97c.c: Add missing clock prepare
by Alexander Stein 05 Dec '14

05 Dec '14

Clocks must be prepared before enabling them. Do this in one step. Replace clk_enable with clk_prepare_enable and clk_disable with clk_disable_unprepare. This fixes the following warning: ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1 at drivers/clk/clk.c:895 __clk_enable+0x24/0x9c() Modules linked in: CPU: 0 PID: 1 Comm: swapper Tainted: G W 3.18.0-rc7+ #245 [<c000dce8>] (unwind_backtrace) from [<c000bcf0>] (show_stack+0x10/0x14) [<c000bcf0>] (show_stack) from [<c001664c>] (warn_slowpath_common+0x60/0x80) [<c001664c>] (warn_slowpath_common) from [<c00166fc>] (warn_slowpath_null+0x18/0x20) [<c00166fc>] (warn_slowpath_null) from [<c02fd7ac>] (__clk_enable+0x24/0x9c) [<c02fd7ac>] (__clk_enable) from [<c02fdbb4>] (clk_enable+0x18/0x2c) [<c02fdbb4>] (clk_enable) from [<c0322688>] (atmel_ac97c_probe+0x154/0x694) [<c0322688>] (atmel_ac97c_probe) from [<c0235e08>] (platform_drv_probe+0x48/0x94) [<c0235e08>] (platform_drv_probe) from [<c02345f8>] (driver_probe_device+0x138/0x350) [<c02345f8>] (driver_probe_device) from [<c02348bc>] (__driver_attach+0x68/0x8c) [<c02348bc>] (__driver_attach) from [<c0232bd0>] (bus_for_each_dev+0x70/0x84) [<c0232bd0>] (bus_for_each_dev) from [<c0233cd8>] (bus_add_driver+0xfc/0x1f8) [<c0233cd8>] (bus_add_driver) from [<c0234f0c>] (driver_register+0x9c/0xe0) [<c0234f0c>] (driver_register) from [<c0008ac4>] (do_one_initcall+0x110/0x1c8) [<c0008ac4>] (do_one_initcall) from [<c053cd58>] (kernel_init_freeable+0xf8/0x1b8) [<c053cd58>] (kernel_init_freeable) from [<c03c0414>] (kernel_init+0x8/0xe4) [<c03c0414>] (kernel_init) from [<c00096d0>] (ret_from_fork+0x14/0x24) ---[ end trace cb88537fdc8fa201 ]--- atmel_ac97c fffa0000.sound: AC'97 0 does not respond - RESET atmel_ac97c fffa0000.sound: AC'97 0 access is not valid [0xffffffff], removing mixer. ------------[ cut here ]------------ Signed-off-by: Alexander Stein <alexanders83(a)web.de> --- sound/atmel/ac97c.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 86f07e9..fa283d3 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -989,7 +989,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); } - clk_enable(pclk); + clk_prepare_enable(pclk); retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, @@ -1167,7 +1167,7 @@ err_ioremap: err_request_irq: snd_card_free(card); err_snd_card_new: - clk_disable(pclk); + clk_disable_unprepare(pclk); clk_put(pclk); return retval; } @@ -1184,7 +1184,7 @@ static int atmel_ac97c_suspend(struct device *pdev) if (test_bit(DMA_TX_READY, &chip->flags)) dw_dma_cyclic_stop(chip->dma.tx_chan); } - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); return 0; } @@ -1194,7 +1194,7 @@ static int atmel_ac97c_resume(struct device *pdev) struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; - clk_enable(chip->pclk); + clk_prepare_enable(chip->pclk); if (cpu_is_at32ap7000()) { if (test_bit(DMA_RX_READY, &chip->flags)) dw_dma_cyclic_start(chip->dma.rx_chan); @@ -1222,7 +1222,7 @@ static int atmel_ac97c_remove(struct platform_device *pdev) ac97c_writel(chip, COMR, 0); ac97c_writel(chip, MR, 0); - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); clk_put(chip->pclk); iounmap(chip->regs); free_irq(chip->irq, chip); -- 2.2.0

2 2

[alsa-devel] [PATCH v7] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 05 Dec '14

05 Dec '14

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 154 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 157 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3a4d7da..9b62457 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -533,6 +534,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130..db60701 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -942,6 +942,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b60ff56..f4956e3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1292,7 +1292,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1304,7 +1305,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9c..7e961ca 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -754,6 +754,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* skip control creation for links with 1 config */ + if (w->num_params == 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* reset 'walked' bit for each dapm path */ static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, struct list_head *sink) @@ -2721,6 +2751,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3206,6 +3239,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + /* Select the configuration set by alsa control */ + config = &config[w->params_select]; + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3274,8 +3310,39 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + /* Can't change the config when widget is already powered */ + if (w->power) + return -EBUSY; + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3283,14 +3350,52 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + 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); + } else { + w_param_text[count] = devm_kmemdup(card->dev, + config->stream_name, + strlen(config->stream_name) + 1, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3298,6 +3403,28 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + /* duplicate w_param_enum on heap so that memory persists */ + private_value = + (unsigned long) devm_kmemdup(card->dev, + (void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + /* duplicate kcontrol_dai_link on heap so that memory persists */ + template.kcontrol_news = devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3305,15 +3432,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + kfree(w); +outfree_kcontrol_news: + kfree(template.kcontrol_news); +outfree_private_value: + kfree((void *)private_value); +outfree_link_name: + kfree(link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + kfree(w_param_text[count]); + kfree(w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

2 1

[alsa-devel] usb-audio: I'd like to change two things in usb audio, but I'm not sure if it's right
by Johan Aires Rastén 05 Dec '14

05 Dec '14

So I'll try again, this time with the proper cc:s and a second question. I've come across two issues in the usb audio module that are causing problems with my sound card, but I'm thinking that changing them might break something for some other card. In sound/usb/mixer.c function check_input_term(...), for the case UAC_INPUT_TERMINAL and UAC_VERSION_2: The recursive call to get clock selectors messes up the naming of my terminals. At this stage the terminals are already properly named Mic and Line but the call overwrites them both with the unnamed clock selector "Unit 22". Are other UAC2 cards different, so that the clock selector contains the correct terminal name, or is there some other reasoning here that I'm not seeing? Could the call be removed, or perhaps only done if no terminal name has been found yet? In sound/usb/stream.c function snd_usb_add_audio_stream(...), near the end of the function: A newly created pcm is added with "list_add(&as->list, &chip->pcm_list);" which causes devices to be listed in reverse order, i.e. device 0 will be at the end of the list, and this leads to that capture streams end up in the wrong device under certain circumstances. The problem arises when the code under "look for an empty stream" earlier in the same function executes. It will attach the capture stream to the first compatible existing stream, but since they are reversed it will find the highest numbered device. In my case analog input will be grouped with digital out (device 1) instead of analog out (device 0). This is assuming that playback streams are defined before capture streams but it could happen in the other direction as long as there are several streams of one type. I also suspect that this is what causes the M-Audio Audiophile USB to have swapped inputs, for which there's a fix in USB-Audio.conf ( http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/cards/USB-Aud… ). Thanks to debianuser on IRC for helping me find it. I would like to change this call to use list_add_tail for adding new items instead, but I'm worried that there are quirks or other fixes that expect the current behaviour. There are very few other references to pcm_list and none of them seem to care about ordering. Sorry for being bad at referencing code, I'm new at this. //Johan Rastén

2 2

[alsa-devel] [PATCH 1/2] ALSA: sound/atmel/ac97c.c: Add missing clock prepare
by Alexander Stein 05 Dec '14

05 Dec '14

Replace clk_enable with clk_prepare_enable and clk_disable with clk_disable_unprepare. Signed-off-by: Alexander Stein <alexanders83(a)web.de> --- sound/atmel/ac97c.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b2a3225..ff2f743 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -989,7 +989,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); } - clk_enable(pclk); + clk_prepare_enable(pclk); retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, @@ -1167,7 +1167,7 @@ err_ioremap: err_request_irq: snd_card_free(card); err_snd_card_new: - clk_disable(pclk); + clk_disable_unprepare(pclk); clk_put(pclk); return retval; } @@ -1184,7 +1184,7 @@ static int atmel_ac97c_suspend(struct device *pdev) if (test_bit(DMA_TX_READY, &chip->flags)) dw_dma_cyclic_stop(chip->dma.tx_chan); } - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); return 0; } @@ -1194,7 +1194,7 @@ static int atmel_ac97c_resume(struct device *pdev) struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; - clk_enable(chip->pclk); + clk_prepare_enable(chip->pclk); if (cpu_is_at32ap7000()) { if (test_bit(DMA_RX_READY, &chip->flags)) dw_dma_cyclic_start(chip->dma.rx_chan); @@ -1222,7 +1222,7 @@ static int atmel_ac97c_remove(struct platform_device *pdev) ac97c_writel(chip, COMR, 0); ac97c_writel(chip, MR, 0); - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); clk_put(chip->pclk); iounmap(chip->regs); free_irq(chip->irq, chip); -- 2.2.0

2 3

[alsa-devel] DMA buffer alignment
by Carlo Caione 05 Dec '14

05 Dec '14

Hi, I'm working with a SoC sound hw that requires that all the data written into the DMA memory is aligned a 64 byte boundary. Actually the alignment is required since the format of the data into the DMA memory is (for a 2 channel 16bit audio): 16 samples for the left channel followed by 16 samples for the right channel. If I try to write anything that is not aligned to the 64 bytes I got audio corruption (high pitched noise). Several user-space programs (like aplay) work well with my hw, since the buffer sent from the user-space is always (padded) of period-size bytes, that is aligned to my 64 bytes boundary. Others (like speaker-test) are not working because sometimes they snd_pcm_writei() a buffer that is not multiple of 64 bytes. In general every time I write into the DMA area a buffer that is not aligned, my hw pointer is not aligned anymore with the 64 bytes and the noise starts. IIUC it is not possible to force the user-space programs to send buffers with a fixed size so the problem should be solved in kernel-space (am I wrong on this point?). Any suggestion on how to tackle this problem? Thanks, -- Carlo Caione

3 4

[alsa-devel] [PATCH v3] ASoC: rt5677: make volume TLV closer to reality
by Dylan Reid 05 Dec '14

05 Dec '14

The volume blocks have an step of 0.375dB, but TLV uses 0.01dB for units. Only use the resolution supported, ignoring the LSB of the volume register. This results in half the steps and 0.75dB per step, but reports accurate levels through TLV. Update the masks to reflect that these are registers have the LSB ignored. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/rt5677.c | 22 +++++++++++----------- sound/soc/codecs/rt5677.h | 24 ++++++++++++------------ 2 files changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 133010d..81fe146 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -763,9 +763,9 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 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); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 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); @@ -817,13 +817,13 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5677_DAC1_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC2 Playback Volume", RT5677_DAC2_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC3 Playback Volume", RT5677_DAC3_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC4 Playback Volume", RT5677_DAC4_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), /* IN1/IN2 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5677_IN1, RT5677_BST_SFT1, 8, 0, bst_tlv), @@ -842,19 +842,19 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { RT5677_L_MUTE_SFT, RT5677_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("ADC1 Capture Volume", RT5677_STO1_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC2 Capture Volume", RT5677_STO2_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC3 Capture Volume", RT5677_STO3_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC4 Capture Volume", RT5677_STO4_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5677_MONO_ADC_DIG_VOL, - RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 127, 0, + RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), /* Sidetone Control */ diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index dbd9ffd..c0a625f 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -306,10 +306,10 @@ #define RT5677_R_MUTE_SFT 7 #define RT5677_VOL_R_MUTE (0x1 << 6) #define RT5677_VOL_R_SFT 6 -#define RT5677_L_VOL_MASK (0x3f << 8) -#define RT5677_L_VOL_SFT 8 -#define RT5677_R_VOL_MASK (0x3f) -#define RT5677_R_VOL_SFT 0 +#define RT5677_L_VOL_MASK (0x7f << 9) +#define RT5677_L_VOL_SFT 9 +#define RT5677_R_VOL_MASK (0x7f << 1) +#define RT5677_R_VOL_SFT 1 /* LOUT1 Control (0x01) */ #define RT5677_LOUT1_L_MUTE (0x1 << 15) @@ -447,16 +447,16 @@ #define RT5677_SEL_DAC2_R_SRC_SFT 0 /* Stereo1 ADC Digital Volume Control (0x1c) */ -#define RT5677_STO1_ADC_L_VOL_MASK (0x7f << 8) -#define RT5677_STO1_ADC_L_VOL_SFT 8 -#define RT5677_STO1_ADC_R_VOL_MASK (0x7f) -#define RT5677_STO1_ADC_R_VOL_SFT 0 +#define RT5677_STO1_ADC_L_VOL_MASK (0x3f << 9) +#define RT5677_STO1_ADC_L_VOL_SFT 9 +#define RT5677_STO1_ADC_R_VOL_MASK (0x3f << 1) +#define RT5677_STO1_ADC_R_VOL_SFT 1 /* Mono ADC Digital Volume Control (0x1d) */ -#define RT5677_MONO_ADC_L_VOL_MASK (0x7f << 8) -#define RT5677_MONO_ADC_L_VOL_SFT 8 -#define RT5677_MONO_ADC_R_VOL_MASK (0x7f) -#define RT5677_MONO_ADC_R_VOL_SFT 0 +#define RT5677_MONO_ADC_L_VOL_MASK (0x3f << 9) +#define RT5677_MONO_ADC_L_VOL_SFT 9 +#define RT5677_MONO_ADC_R_VOL_MASK (0x3f << 1) +#define RT5677_MONO_ADC_R_VOL_SFT 1 /* Stereo 1/2 ADC Boost Gain Control (0x1e) */ #define RT5677_STO1_ADC_L_BST_MASK (0x3 << 14) -- 2.2.0.rc0.207.ga3a616c

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[alsa-devel] 'No audio' bug on lenovo ultra dock
by Alexander Lochmann 05 Dec '14

05 Dec '14

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! The current linux kernel (3.17 as well as 3.18-rc7) does not support the audio output on a Lenovo Ultra Dock in conjunction with the X240. The issue is discussed in [1], which provides a patch as well. The patch is available at [2]. I'm asking you for integrating it into your linux tree. So it might be merged into the mainline kernel during the current or the next development phase. Thanks in advance! Regards, Alex [1] https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1385680 [2] https://launchpadlibrarian.net/188362011/lenovo-ultradock.patch - -- Technische Universität Dortmund Alexander Lochmann PGP key: 0xBC3EF6FD Otto-Hahn-Str. 16 phone: +49.231.7556141 D-44227 Dortmund fax: +49.231.7556116 http://ess.cs.tu-dortmund.de/Staff/al -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) iQIcBAEBAgAGBQJUgXibAAoJEFk+7QW8Pvb9zSEP/0yezuJXBK9P8ejhfrmH7VxK PPNhsALQmgMa5vTPsepzBsbj5060yiyI8BTkt76h5d6QlLy3eKq8+AE4+hkGJNie sZtvbWPr8C0GP+8lnfRz6EFOdnWdXbEc4x1hSd7JVDgpNynzHvYZQgISVnxA1rFe NTqngYzFM8yBhUEKc0DYzvuCTw9J0VODWro9eD6tjuSI7/yyQmrLln7XihcLs3/3 VWcNYVBDzQOqjwOTJM7wPfprGgalr7vOBWo3WyshBnKcXmsnahswB+HSsiVWqtMd SZBoEfm811ocvtBFGfsuMkWJALtUfw3bBEPm2iV5jzKmK1CoCxCAuqTcUkiNEzCz m6f76LEtGzq10vpXsc9FmX1Mlbrt9mavE2a5cxoaxVEPJ2OVpwAU3ufgguwUkIrt L8E4RjBEHwttDbe4CPz1YSNz4j+OD4bxsRVjQGfpk8mVGTh0ljtno1chqxn0HZ5J +9Hto37DqDzywJnc4gYyhxTcAZ669qvdMow0avyPuEwGK///C9qgkGCgrm0W36Oq u/HU2lz9A6Vqxtzr8t4bVVEhiX+J2+HQAyf2OPx7o4iMyLAFoxvvWRZUJ3qvP1ZQ OlhXIHg22xTiy9WsVLiStSbT8xd6lV1UP/HotUwOjEoQLzv8pVh/5DH3fwTX5cIw jvRMnfYW+6Ai6Nn07Frj =Y9Kt -----END PGP SIGNATURE-----

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[alsa-devel] [PATCH v2 0/2] fsl_ssi misc fix
by Jiada Wang 04 Dec '14

04 Dec '14

Hi, Changes in v2: - fix error path in probe - replace irq_of_parse_and_map with platform_get_irq Changes in v1: - free IRQ before irq_dispose_mapping Jiada Wang (2): ASoC: fsl_ssi: fix error path in probe ASoC: fsl_ssi: use platform_get_irq instead of irq_of_parse_and_map sound/soc/fsl/fsl_ssi.c | 13 +++---------- 1 file changed, 3 insertions(+), 10 deletions(-) -- 1.9.3

2 4

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix module unbound
by Fabio Estevam 04 Dec '14

04 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Trying to remove the snd-soc-fsl-ssi module leads to the following warning: [ 31.515336] ------------[ cut here ]------------ [ 31.520091] WARNING: CPU: 2 PID: 434 at fs/proc/generic.c:521 remove_proc_entry+0x14c/0x16c() [ 31.528708] remove_proc_entry: removing non-empty directory 'irq/79', leaking at least '202c000.ss' [ 31.537911] Modules linked in: snd_soc_wm8962 snd_soc_imx_wm8962 snd_soc_fsl_ssi(-) evbug [ 31.546249] CPU: 2 PID: 434 Comm: rmmod Not tainted 3.18.0-rc6-00028-g3314bf6-dirty #1 [ 31.554235] Backtrace: [ 31.556816] [<80011ea8>] (dump_backtrace) from [<80012044>] (show_stack+0x18/0x1c) [ 31.564416] r6:80142c88 r5:00000000 r4:00000000 r3:00000000 [ 31.570267] [<8001202c>] (show_stack) from [<806980ec>] (dump_stack+0x88/0xa4) [ 31.577588] [<80698064>] (dump_stack) from [<80029d78>] (warn_slowpath_common+0x70/0x94) [ 31.585711] r5:00000009 r4:bb61fd90 [ 31.589423] [<80029d08>] (warn_slowpath_common) from [<80029e40>] (warn_slowpath_fmt+0x38/0x40) [ 31.598187] r8:bb61fdfe r7:be05d76d r6:be05d9a8 r5:00000002 r4:be05d700 [ 31.605054] [<80029e0c>] (warn_slowpath_fmt) from [<80142c88>] (remove_proc_entry+0x14c/0x16c) [ 31.613709] r3:806a79c0 r2:808229a0 [ 31.617371] [<80142b3c>] (remove_proc_entry) from [<80070380>] (unregister_irq_proc+0x94/0xb8) [ 31.625989] r10:00000000 r8:8000ede4 r7:80955f2c r6:0000004f r5:8118e738 r4:be00af00 [ 31.633952] [<800702ec>] (unregister_irq_proc) from [<80069dac>] (free_desc+0x2c/0x64) [ 31.641898] r6:0000004f r5:80955f38 r4:be00af00 [ 31.646604] [<80069d80>] (free_desc) from [<80069e68>] (irq_free_descs+0x4c/0x8c) [ 31.654092] r7:00000081 r6:00000001 r5:0000004f r4:00000001 [ 31.659863] [<80069e1c>] (irq_free_descs) from [<8006fc3c>] (irq_dispose_mapping+0x40/0x5c) [ 31.668247] r6:be17b844 r5:be17b800 r4:0000004f r3:802c5ec0 [ 31.673998] [<8006fbfc>] (irq_dispose_mapping) from [<7f004ea4>] (fsl_ssi_remove+0x58/0x70 [snd_so) [ 31.683948] r4:bb5bba10 r3:00000001 [ 31.687618] [<7f004e4c>] (fsl_ssi_remove [snd_soc_fsl_ssi]) from [<803720a0>] (platform_drv_remove) [ 31.697564] r5:7f0064f8 r4:be17b810 [ 31.701195] [<80372080>] (platform_drv_remove) from [<80370494>] (__device_release_driver+0x78/0xc) [ 31.710361] r5:7f0064f8 r4:be17b810 [ 31.713987] [<8037041c>] (__device_release_driver) from [<80370d20>] (driver_detach+0xbc/0xc0) [ 31.722631] r5:7f0064f8 r4:be17b810 [ 31.726259] [<80370c64>] (driver_detach) from [<80370304>] (bus_remove_driver+0x54/0x98) [ 31.734382] r6:00000800 r5:00000000 r4:7f0064f8 r3:bb67f500 [ 31.740149] [<803702b0>] (bus_remove_driver) from [<80371398>] (driver_unregister+0x30/0x50) [ 31.748617] r4:7f0064f8 r3:bd9f7080 [ 31.752245] [<80371368>] (driver_unregister) from [<80371f3c>] (platform_driver_unregister+0x14/0x) [ 31.761498] r4:7f00655c r3:7f005a70 [ 31.765130] [<80371f28>] (platform_driver_unregister) from [<7f005a84>] (fsl_ssi_driver_exit+0x14/) [ 31.776147] [<7f005a70>] (fsl_ssi_driver_exit [snd_soc_fsl_ssi]) from [<8008ed80>] (SyS_delete_mod) [ 31.786553] [<8008ec64>] (SyS_delete_module) from [<8000ec20>] (ret_fast_syscall+0x0/0x48) [ 31.794824] r6:00c46d18 r5:00000800 r4:00c46d18 [ 31.799530] ---[ end trace 954e8a3a15379e52 ]--- The cause of problem and solution are well explained by Lars-Peter: "The driver creates the mapping by calling irq_of_parse_and_map(), so it also has to dispose the mapping. But the easy way out is to simply use platform_get_irq() instead of irq_of_parse_map(). In this case the mapping is not managed by the device but by the of core, so the device has not to dispose the mapping." Tested on a imx6q-sabresd board. Reported-by: Jiada Wang <jiada_wang(a)mentor.com> Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 32a31d9..c528f16 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1361,7 +1361,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) return PTR_ERR(ssi_private->regs); } - ssi_private->irq = irq_of_parse_and_map(np, 0); + ssi_private->irq = platform_get_irq(pdev, 0); if (!ssi_private->irq) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); return -ENXIO; @@ -1387,7 +1387,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ssi_private->soc->imx) { ret = fsl_ssi_imx_probe(pdev, ssi_private, iomem); if (ret) - goto error_irqmap; + return ret; } ret = snd_soc_register_component(&pdev->dev, &fsl_ssi_component, @@ -1458,10 +1458,6 @@ error_asoc_register: if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); -error_irqmap: - if (ssi_private->use_dma) - irq_dispose_mapping(ssi_private->irq); - return ret; } @@ -1478,9 +1474,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); - if (ssi_private->use_dma) - irq_dispose_mapping(ssi_private->irq); - return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: soc-pcm: do not hw_free BE if it's still used
by Qiao Zhou 04 Dec '14

04 Dec '14

Do not free BE hw if it's still used by other FE during dpcm runtime shutdown. Otherwise the BE runtime state will be STATE_HW_FREE and won't be updated to STATE_CLOSE when shutdown ends, because BE dai shutdown function won't close pcm when detecting BE is still under use. With STATE_HW_FREE, BE can't be triggered start again. This corner case can easily appear when one BE is used by two FE, without this patch "ASoC: dpcm: Fix race between FE/BE updates and trigger"(ea9d0d771fcd32cd56070819749477d511ec9117). One FE tries to shutdown but it's raced against xrun on another FE. It improves the be dai hw_free logic. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/soc-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 57277dd..70e8088 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1664,6 +1664,10 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) continue; + /* do not free hw if this BE is used by other FE */ + if (be->dpcm[stream].users > 1) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Augment existing card DAPM routes in snd_soc_of_parse_audio_routing
by Peter Rosin 04 Dec '14

04 Dec '14

From: Peter Rosin <peda(a)axentia.se> If a snd_soc_card has any DAPM routes when it calls snd_soc_of_parse_audio_routing, those are clobbered without this change. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/soc-core.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b60ff56ebc0f..377db486852e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4585,7 +4585,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname) { struct device_node *np = card->dev->of_node; - int num_routes; + int num_routes, old_routes; struct snd_soc_dapm_route *routes; int i, ret; @@ -4603,7 +4603,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes), + old_routes = card->num_dapm_routes; + routes = devm_kzalloc(card->dev, + (old_routes + num_routes) * sizeof(*routes), GFP_KERNEL); if (!routes) { dev_err(card->dev, @@ -4611,9 +4613,11 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } + memcpy(routes, card->dapm_routes, old_routes * sizeof(*routes)); + for (i = 0; i < num_routes; i++) { ret = of_property_read_string_index(np, propname, - 2 * i, &routes[i].sink); + 2 * i, &routes[old_routes + i].sink); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -4621,7 +4625,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } ret = of_property_read_string_index(np, propname, - (2 * i) + 1, &routes[i].source); + (2 * i) + 1, &routes[old_routes + i].source); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -4630,7 +4634,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes = num_routes; + card->num_dapm_routes += num_routes; card->dapm_routes = routes; return 0; -- 1.7.10.4

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[alsa-devel] [PATCH 1/2] ASoC: rt5677: Add ACPI device probing
by Ben Zhang 04 Dec '14

04 Dec '14

The rt5677 codec driver looks for ACPI device ID "RT5677CE", which is specified in coreboot. This patch allows platform data to be obtained via ACPI Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/rt5677.c | 52 +++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 50 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d317c68..384281d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -20,6 +20,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/acpi.h> #include <linux/firmware.h> #include <linux/gpio.h> #include <sound/core.h> @@ -4525,6 +4526,43 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) return 0; } +#ifdef CONFIG_ACPI + +static unsigned long long rt5677_parse_acpi_entry(struct device *dev, + acpi_string name) +{ + acpi_handle handle = ACPI_HANDLE(dev); + unsigned long long val; + acpi_status status; + + status = acpi_evaluate_integer(handle, name, NULL, &val); + if (ACPI_FAILURE(status)) { + dev_err(dev, "Failed to parse ACPI entry %s, default to 0: %d\n", + name, status); + return 0; + } + return val; +} + +static void rt5677_parse_acpi(struct rt5677_priv *rt5677, struct device *dev) +{ + rt5677->pdata.dmic2_clk_pin = (enum rt5677_dmic2_clk) + rt5677_parse_acpi_entry(dev, "DCLK"); + rt5677->pdata.in1_diff = (bool)rt5677_parse_acpi_entry(dev, "IN1"); + rt5677->pdata.in2_diff = (bool)rt5677_parse_acpi_entry(dev, "IN2"); + rt5677->pdata.lout1_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT1"); + rt5677->pdata.lout2_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT2"); + rt5677->pdata.lout3_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT3"); + rt5677->pdata.jd1_gpio = rt5677_parse_acpi_entry(dev, "JD1"); + rt5677->pdata.jd2_gpio = rt5677_parse_acpi_entry(dev, "JD2"); + rt5677->pdata.jd3_gpio = rt5677_parse_acpi_entry(dev, "JD3"); +} +#else +static void rt5677_parse_acpi(struct rt5677_priv *rt5677, struct device *dev) +{ +} +#endif + static struct regmap_irq rt5677_irqs[] = { [RT5677_IRQ_JD1] = { .reg_offset = 0, @@ -4604,6 +4642,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, if (pdata) rt5677->pdata = *pdata; + rt5677->pow_ldo2 = -EINVAL; if (i2c->dev.of_node) { ret = rt5677_parse_dt(rt5677, i2c->dev.of_node); if (ret) { @@ -4611,8 +4650,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, ret); return ret; } - } else { - rt5677->pow_ldo2 = -EINVAL; + } else if (ACPI_HANDLE(&i2c->dev)) { + rt5677_parse_acpi(rt5677, &i2c->dev); } if (gpio_is_valid(rt5677->pow_ldo2)) { @@ -4708,10 +4747,19 @@ static int rt5677_i2c_remove(struct i2c_client *i2c) return 0; } +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5677_acpi_id[] = { + { "RT5677CE", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, rt5677_acpi_id); +#endif + static struct i2c_driver rt5677_i2c_driver = { .driver = { .name = "rt5677", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5677_acpi_id), }, .probe = rt5677_i2c_probe, .remove = rt5677_i2c_remove, -- 2.1.0.rc2.206.gedb03e5

6 35

[alsa-devel] [PATCH v2] ASoC: add xtensa xtfpga I2S interface and platform
by Max Filippov 04 Dec '14

04 Dec '14

XTFPGA boards provides an audio subsystem that consists of TI CDCE706 clock synthesizer, I2S transmitter and TLV320AIC23 audio codec. I2S transmitter has MMIO-based interface that resembles that of the OpenCores I2S transmitter. I2S transmitter is always a master on I2S bus. There's no specialized audio DMA, sample data are transferred to I2S transmitter FIFO by CPU through memory-mapped queue interface. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- Changes v1->v2: - fix module name in the sound/soc/xtensa/Makefile; - split PCM data transfer code and I2S interface setup code; - move clock enabling/disabling to runtime PM code; - document I2S interface RATIO field; - rewrite I2S interface LEVEL field calculation to avoid overflows; - drop select statements from Kconfig; - document RCU usage for tx_substream; - clean up code running in RCU read-side critical section; - use threaded IRQ handler instead of IRQ + tasklet; - remove empty PCM callback handlers; - add const to static structures where applicable; - use platform_set_drvdata instead of dev_set_drvdata; - drop redundant error reporting code; - use platform_get_irq instead of platform_get_resource; - clean up xtfpga_i2s_remove; - add comments. .../devicetree/bindings/sound/cdns,xtfpga-i2s.txt | 18 + MAINTAINERS | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/xtensa/Kconfig | 7 + sound/soc/xtensa/Makefile | 3 + sound/soc/xtensa/xtfpga-i2s.c | 676 +++++++++++++++++++++ 7 files changed, 707 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt create mode 100644 sound/soc/xtensa/Kconfig create mode 100644 sound/soc/xtensa/Makefile create mode 100644 sound/soc/xtensa/xtfpga-i2s.c diff --git a/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt new file mode 100644 index 0000000..befd125 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt @@ -0,0 +1,18 @@ +Bindings for I2S controller built into xtfpga Xtensa bitstreams. + +Required properties: +- compatible: shall be "cdns,xtfpga-i2s". +- reg: memory region (address and length) with device registers. +- interrupts: interrupt for the device. +- clocks: phandle to the clk used as master clock. I2S bus clock + is derived from it. + +Examples: + + i2s0: xtfpga-i2s@0d080000 { + #sound-dai-cells = <0>; + compatible = "cdns,xtfpga-i2s"; + reg = <0x0d080000 0x40>; + interrupts = <2 1>; + clocks = <&cdce706 4>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index acb4523..d7bfeec 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -10204,6 +10204,7 @@ M: Max Filippov <jcmvbkbc(a)gmail.com> L: linux-xtensa(a)linux-xtensa.org S: Maintained F: drivers/spi/spi-xtensa-xtfpga.c +F: sound/soc/xtensa/xtfpga-i2s.c YAM DRIVER FOR AX.25 M: Jean-Paul Roubelat <jpr(a)f6fbb.org> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 0e96233..9ca12e7 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -56,6 +56,7 @@ source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" +source "sound/soc/xtensa/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 534714a..99d7222 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -33,3 +33,4 @@ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ +obj-$(CONFIG_SND_SOC) += xtensa/ diff --git a/sound/soc/xtensa/Kconfig b/sound/soc/xtensa/Kconfig new file mode 100644 index 0000000..c201beb --- /dev/null +++ b/sound/soc/xtensa/Kconfig @@ -0,0 +1,7 @@ +config SND_SOC_XTFPGA_I2S + tristate "XTFPGA I2S master" + select REGMAP_MMIO + help + Say Y or M if you want to add support for codecs attached to the + I2S interface on XTFPGA daughter board. You will also need to select + the drivers for the rest of XTFPGA audio subsystem. diff --git a/sound/soc/xtensa/Makefile b/sound/soc/xtensa/Makefile new file mode 100644 index 0000000..15efbf9 --- /dev/null +++ b/sound/soc/xtensa/Makefile @@ -0,0 +1,3 @@ +snd-soc-xtfpga-i2s-objs := xtfpga-i2s.o + +obj-$(CONFIG_SND_SOC_XTFPGA_I2S) += snd-soc-xtfpga-i2s.o diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c new file mode 100644 index 0000000..bd66be2 --- /dev/null +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -0,0 +1,676 @@ +/* + * Xtfpga I2S controller driver + * + * Copyright (c) 2014 Cadence Design Systems Inc. + * + * 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/clk.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define DRV_NAME "xtfpga-i2s" + +#define XTFPGA_I2S_VERSION 0x00 +#define XTFPGA_I2S_CONFIG 0x04 +#define XTFPGA_I2S_INT_MASK 0x08 +#define XTFPGA_I2S_INT_STATUS 0x0c +#define XTFPGA_I2S_CHAN0_DATA 0x10 +#define XTFPGA_I2S_CHAN1_DATA 0x14 +#define XTFPGA_I2S_CHAN2_DATA 0x18 +#define XTFPGA_I2S_CHAN3_DATA 0x1c + +#define XTFPGA_I2S_CONFIG_TX_ENABLE 0x1 +#define XTFPGA_I2S_CONFIG_INT_ENABLE 0x2 +#define XTFPGA_I2S_CONFIG_LEFT 0x4 +#define XTFPGA_I2S_CONFIG_RATIO_BASE 8 +#define XTFPGA_I2S_CONFIG_RATIO_MASK 0x0000ff00 +#define XTFPGA_I2S_CONFIG_RES_BASE 16 +#define XTFPGA_I2S_CONFIG_RES_MASK 0x003f0000 +#define XTFPGA_I2S_CONFIG_LEVEL_BASE 24 +#define XTFPGA_I2S_CONFIG_LEVEL_MASK 0x0f000000 +#define XTFPGA_I2S_CONFIG_CHANNEL_BASE 28 + +#define XTFPGA_I2S_INT_UNDERRUN 0x1 +#define XTFPGA_I2S_INT_LEVEL 0x2 +#define XTFPGA_I2S_INT_VALID 0x3 + +#define XTFPGA_I2S_FIFO_SIZE 8192 + +/* + * I2S controller operation: + * + * Enabling TX: output 1 period of zeros (starting with left channel) + * and then queued data. + * + * Level status and interrupt: whenever FIFO level is below FIFO trigger, + * level status is 1 and an IRQ is asserted (if enabled). + * + * Underrun status and interrupt: whenever FIFO is empty, underrun status + * is 1 and an IRQ is asserted (if enabled). + */ +struct xtfpga_i2s { + struct device *dev; + struct clk *clk; + struct regmap *regmap; + void __iomem *regs; + + /* current playback substream. NULL if not playing. + * + * Access to that field is synchronized between the interrupt handler + * and userspace through RCU. + * + * Interrupt handler (threaded part) does PIO on substream data in RCU + * read-side critical section. Trigger callback sets and clears the + * pointer when the playback is started and stopped with + * rcu_assign_pointer. When userspace is about to free the playback + * stream in the pcm_close callback it synchronizes with the interrupt + * handler by means of synchronize_rcu call. + */ + struct snd_pcm_substream *tx_substream; + unsigned (*tx_fn)(struct xtfpga_i2s *i2s, + struct snd_pcm_runtime *runtime, + unsigned tx_ptr); + unsigned tx_ptr; /* next frame index in the sample buffer */ + + /* current fifo level estimate. + * Doesn't have to be perfectly accurate, but must be not less than + * the actual FIFO level in order to avoid stall on push attempt. + */ + unsigned tx_fifo_level; + + /* FIFO level at which level interrupt occurs */ + unsigned tx_fifo_low; + + /* maximal FIFO level */ + unsigned tx_fifo_high; +}; + +static bool xtfpga_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + return reg >= XTFPGA_I2S_CONFIG; +} + +static bool xtfpga_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + return reg < XTFPGA_I2S_CHAN0_DATA; +} + +static bool xtfpga_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == XTFPGA_I2S_INT_STATUS; +} + +static const struct regmap_config xtfpga_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = XTFPGA_I2S_CHAN3_DATA, + .writeable_reg = xtfpga_i2s_wr_reg, + .readable_reg = xtfpga_i2s_rd_reg, + .volatile_reg = xtfpga_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +/* Generate functions that do PIO from TX DMA area to FIFO for all supported + * stream formats. + * Functions will be called xtfpga_pcm_tx_<channels>x<sample bits>, e.g. + * xtfpga_pcm_tx_2x16 for 16-bit stereo. + * + * FIFO consists of 32-bit words, one word per channel, always 2 channels. + * If I2S interface is configured with smaller sample resolution, only + * the LSB of each word is used. + */ +#define xtfpga_pcm_tx_fn(channels, sample_bits) \ +static unsigned xtfpga_pcm_tx_##channels##x##sample_bits( \ + struct xtfpga_i2s *i2s, struct snd_pcm_runtime *runtime, \ + unsigned tx_ptr) \ +{ \ + const u##sample_bits (*p)[channels] = \ + (void *)runtime->dma_area; \ +\ + for (; i2s->tx_fifo_level < i2s->tx_fifo_high; \ + i2s->tx_fifo_level += 2) { \ + iowrite32(p[tx_ptr][0], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + iowrite32(p[tx_ptr][channels - 1], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + if (++tx_ptr >= runtime->buffer_size) \ + tx_ptr = 0; \ + } \ + return tx_ptr; \ +} + +xtfpga_pcm_tx_fn(1, 16) +xtfpga_pcm_tx_fn(2, 16) +xtfpga_pcm_tx_fn(1, 32) +xtfpga_pcm_tx_fn(2, 32) + +#undef xtfpga_pcm_tx_fn + +static bool xtfpga_pcm_push_tx(struct xtfpga_i2s *i2s) +{ + struct snd_pcm_substream *tx_substream; + bool tx_active; + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + tx_active = tx_substream && snd_pcm_running(tx_substream); + if (tx_active) { + unsigned tx_ptr = ACCESS_ONCE(i2s->tx_ptr); + unsigned new_tx_ptr = i2s->tx_fn(i2s, tx_substream->runtime, + tx_ptr); + + cmpxchg(&i2s->tx_ptr, tx_ptr, new_tx_ptr); + } + rcu_read_unlock(); + + return tx_active; +} + +static void xtfpga_pcm_refill_fifo(struct xtfpga_i2s *i2s) +{ + unsigned int_status; + unsigned i; + + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + for (i = 0; i < 2; ++i) { + bool tx_active = xtfpga_pcm_push_tx(i2s); + + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + if (tx_active) + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + if (!tx_active || + !(int_status & XTFPGA_I2S_INT_LEVEL)) + break; + + /* After the push the level IRQ is still asserted, + * means FIFO level is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + if (!(int_status & XTFPGA_I2S_INT_LEVEL)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_VALID); + else if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_UNDERRUN); + + if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE); + else + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); +} + +static irqreturn_t xtfpga_i2s_threaded_irq_handler(int irq, void *dev_id) +{ + struct xtfpga_i2s *i2s = dev_id; + struct snd_pcm_substream *tx_substream; + unsigned config, int_status, int_mask; + + regmap_read(i2s->regmap, XTFPGA_I2S_CONFIG, &config); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_MASK, &int_mask); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, &int_status); + + if (!(config & XTFPGA_I2S_CONFIG_INT_ENABLE) || + !(int_status & int_mask & XTFPGA_I2S_INT_VALID)) + return IRQ_NONE; + + /* Update FIFO level estimate in accordance with interrupt status + * register. + */ + if (int_status & XTFPGA_I2S_INT_UNDERRUN) { + i2s->tx_fifo_level = 0; + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); + } else { + /* The FIFO isn't empty, but is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + + if (tx_substream && snd_pcm_running(tx_substream)) { + snd_pcm_period_elapsed(tx_substream); + if (int_status & XTFPGA_I2S_INT_UNDERRUN) + dev_dbg_ratelimited(i2s->dev, "%s: underrun\n", + __func__); + } + rcu_read_unlock(); + + /* Refill FIFO, update allowed IRQ reasons, enable IRQ if FIFO is + * not empty. + */ + xtfpga_pcm_refill_fifo(i2s); + + return IRQ_HANDLED; +} + +static int xtfpga_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, i2s); + return 0; +} + +static int xtfpga_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned srate = params_rate(params); + unsigned channels = params_channels(params); + unsigned period_size = params_period_size(params); + unsigned sample_size = snd_pcm_format_width(params_format(params)); + unsigned freq, ratio, level; + int err; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RES_MASK, + sample_size << XTFPGA_I2S_CONFIG_RES_BASE); + + freq = 256 * srate; + err = clk_set_rate(i2s->clk, freq); + if (err < 0) + return err; + + /* ratio field of the config register controls MCLK->I2S clock + * derivation: I2S clock = MCLK / (2 * (ratio + 2)). + * + * So with MCLK = 256 * sample rate ratio is 0 for 32 bit stereo + * and 2 for 16 bit stereo. + */ + ratio = (freq - (srate * sample_size * 8)) / + (srate * sample_size * 4); + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RATIO_MASK, + ratio << XTFPGA_I2S_CONFIG_RATIO_BASE); + + i2s->tx_fifo_low = XTFPGA_I2S_FIFO_SIZE / 2; + + /* period_size * 2: FIFO always gets 2 samples per frame */ + for (level = 1; + i2s->tx_fifo_low / 2 >= period_size * 2 && + level < (XTFPGA_I2S_CONFIG_LEVEL_MASK >> + XTFPGA_I2S_CONFIG_LEVEL_BASE); ++level) + i2s->tx_fifo_low /= 2; + + i2s->tx_fifo_high = 2 * i2s->tx_fifo_low; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_LEVEL_MASK, + level << XTFPGA_I2S_CONFIG_LEVEL_BASE); + + dev_dbg(i2s->dev, + "%s srate: %u, channels: %u, sample_size: %u, period_size: %u\n", + __func__, srate, channels, sample_size, period_size); + dev_dbg(i2s->dev, "%s freq: %u, ratio: %u, level: %u\n", + __func__, freq, ratio, level); + + return 0; +} + +static int xtfpga_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) + return -EINVAL; + + return 0; +} + +/* PCM */ + +static const struct snd_pcm_hardware xtfpga_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .period_bytes_min = 2, + .period_bytes_max = XTFPGA_I2S_FIFO_SIZE / 2 * 8, + .periods_min = 2, + .periods_max = XTFPGA_I2S_FIFO_SIZE * 8 / 2, + .buffer_bytes_max = XTFPGA_I2S_FIFO_SIZE * 8, + .fifo_size = 16, +}; + +static int xtfpga_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + void *p; + + snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware); + p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + runtime->private_data = p; + + return 0; +} + +static int xtfpga_pcm_close(struct snd_pcm_substream *substream) +{ + synchronize_rcu(); + return 0; +} + +static int xtfpga_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int ret; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + unsigned channels = params_channels(hw_params); + + switch (channels) { + case 1: + case 2: + break; + + default: + return -EINVAL; + + } + + switch (params_format(hw_params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x16 : + xtfpga_pcm_tx_2x16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x32 : + xtfpga_pcm_tx_2x32; + break; + + default: + return -EINVAL; + } + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + return ret; +} + +static int xtfpga_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ACCESS_ONCE(i2s->tx_ptr) = 0; + rcu_assign_pointer(i2s->tx_substream, substream); + xtfpga_pcm_refill_fifo(i2s); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + rcu_assign_pointer(i2s->tx_substream, NULL); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static snd_pcm_uframes_t xtfpga_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + snd_pcm_uframes_t pos = ACCESS_ONCE(i2s->tx_ptr); + + return pos < runtime->buffer_size ? pos : 0; +} + +static int xtfpga_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = xtfpga_pcm_hardware.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, size, size); +} + +static void xtfpga_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static const struct snd_pcm_ops xtfpga_pcm_ops = { + .open = xtfpga_pcm_open, + .close = xtfpga_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = xtfpga_pcm_hw_params, + .trigger = xtfpga_pcm_trigger, + .pointer = xtfpga_pcm_pointer, +}; + +static const struct snd_soc_platform_driver xtfpga_soc_platform = { + .pcm_new = xtfpga_pcm_new, + .pcm_free = xtfpga_pcm_free, + .ops = &xtfpga_pcm_ops, +}; + +static const struct snd_soc_component_driver xtfpga_i2s_component = { + .name = DRV_NAME, +}; + +static const struct snd_soc_dai_ops xtfpga_i2s_dai_ops = { + .startup = xtfpga_i2s_startup, + .hw_params = xtfpga_i2s_hw_params, + .set_fmt = xtfpga_i2s_set_fmt, +}; + +static struct snd_soc_dai_driver xtfpga_i2s_dai[] = { + { + .name = "xtfpga-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &xtfpga_i2s_dai_ops, + }, +}; + +static int xtfpga_i2s_runtime_suspend(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk); + return 0; +} + +static int xtfpga_i2s_runtime_resume(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(i2s->clk); + if (ret) { + dev_err(dev, "clk_prepare_enable failed: %d\n", ret); + return ret; + } + return 0; +} + +static int xtfpga_i2s_probe(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s; + struct resource *mem; + int err, irq; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + err = -ENOMEM; + goto err; + } + platform_set_drvdata(pdev, i2s); + i2s->dev = &pdev->dev; + dev_dbg(&pdev->dev, "dev: %p, i2s: %p\n", &pdev->dev, i2s); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->regs)) { + err = PTR_ERR(i2s->regs); + goto err; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->regs, + &xtfpga_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + err = PTR_ERR(i2s->regmap); + goto err; + } + + i2s->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk)) { + dev_err(&pdev->dev, "couldn't get clock\n"); + err = PTR_ERR(i2s->clk); + goto err; + } + + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, + (0x1 << XTFPGA_I2S_CONFIG_CHANNEL_BASE)); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, XTFPGA_I2S_INT_VALID); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, XTFPGA_I2S_INT_UNDERRUN); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "No IRQ resource\n"); + err = irq; + goto err; + } + err = devm_request_threaded_irq(&pdev->dev, irq, NULL, + xtfpga_i2s_threaded_irq_handler, + IRQF_SHARED | IRQF_ONESHOT, + pdev->name, i2s); + if (err < 0) { + dev_err(&pdev->dev, "request_irq failed\n"); + goto err; + } + + err = snd_soc_register_platform(&pdev->dev, &xtfpga_soc_platform); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register platform\n"); + goto err; + } + err = devm_snd_soc_register_component(&pdev->dev, + &xtfpga_i2s_component, + xtfpga_i2s_dai, + ARRAY_SIZE(xtfpga_i2s_dai)); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register component\n"); + goto err_unregister_platform; + } + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + err = xtfpga_i2s_runtime_resume(&pdev->dev); + if (err) + goto err_pm_disable; + } + return 0; + +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_unregister_platform: + snd_soc_unregister_platform(&pdev->dev); +err: + dev_err(&pdev->dev, "%s: err = %d\n", __func__, err); + return err; +} + +static int xtfpga_i2s_remove(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + if (i2s->regmap && !IS_ERR(i2s->regmap)) { + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + } + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + xtfpga_i2s_runtime_suspend(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id xtfpga_i2s_of_match[] = { + { .compatible = "cdns,xtfpga-i2s", }, + {}, +}; +MODULE_DEVICE_TABLE(of, xtfpga_i2s_of_match); +#endif + +static const struct dev_pm_ops xtfpga_i2s_pm_ops = { + SET_RUNTIME_PM_OPS(xtfpga_i2s_runtime_suspend, + xtfpga_i2s_runtime_resume, NULL) +}; + +static struct platform_driver xtfpga_i2s_driver = { + .probe = xtfpga_i2s_probe, + .remove = xtfpga_i2s_remove, + .driver = { + .name = "xtfpga-i2s", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(xtfpga_i2s_of_match), + .pm = &xtfpga_i2s_pm_ops, + }, +}; + +module_platform_driver(xtfpga_i2s_driver); + +MODULE_AUTHOR("Max Filippov <jcmvbkbc(a)gmail.com>"); +MODULE_DESCRIPTION("xtfpga I2S controller driver"); +MODULE_LICENSE("GPL v2"); -- 1.8.1.4

1 0

[alsa-devel] [RFC 0/2] get rid of remaining users of attach_adapter
by Wolfram Sang 04 Dec '14

04 Dec '14

The attach_adapter mechanism of the I2C framework is deprecated for years. There are two users left, drivers for old Macintosh computers. I recently got the idea of replacing this mechanism with a custom one with the help of deferred probing. Because I don't have the hardware, I worked on a seperate branch where I modified the windtunnel driver to be runnable on my Renesas Lager board (ARM). I first verified that the attach_adapter method was used, then the driver rightfully failed on detecting the i2c clients. Using my custom mechanism, the same happens: all busses get scanned, then the clients cannot be detected. Since I didn't change the actual detection code, I assume that it should be working on those Macintosh devices as well. The keywest driver is only compile-tested. That being said, I'd be more than happy, if we could find someone willing to test these patches. If they could be applied, we can _finally_ get rid of this legacy mechanism and clean up the i2c core. The branch I used for ARM compilation is here: git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git i2c/attach_adapter_removal_with_arm_compile-experimental Thanks, Wolfram Wolfram Sang (2): macintosh: therm_windtunnel: drop using attach adapter sound: ppc: keywest: drop using attach adapter drivers/macintosh/therm_windtunnel.c | 25 +++++++++++++++++++++++-- sound/ppc/keywest.c | 21 ++++++++++++++++++--- 2 files changed, 41 insertions(+), 5 deletions(-) -- 2.1.3

2 4

[alsa-devel] [PATCH v2] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 04 Dec '14

04 Dec '14

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK to save power. If platform data is not supplied, we will get MCLK from device tree and check the return code to handle probe deferral. Enable runtime power management, and auto enable/disable MCLK in pm_runtime resume and suspend. When wm8960 codec is being used, it will triger resume() to enable MCLK. When codec is not being used, it will triger suspend() to disable MCLK. If we use PLL to generate sysclk, wm8960_set_dai_pll() will enable PLL. But PLL doesn't power down after using codec. So power down PLL in suspend(). Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- include/sound/wm8960.h | 1 + sound/soc/codecs/wm8960.c | 59 ++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 57 insertions(+), 3 deletions(-) diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index e8ce8ee..d7e84ed 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -16,6 +16,7 @@ #define WM8960_DRES_MAX 3 struct wm8960_data { + struct clk *mclk; bool capless; /* Headphone outputs configured in capless mode */ bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae..2c7a97a 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,8 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -978,7 +980,7 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; -static void wm8960_set_pdata_from_of(struct i2c_client *i2c, +static int wm8960_set_pdata_from_of(struct i2c_client *i2c, struct wm8960_data *pdata) { const struct device_node *np = i2c->dev.of_node; @@ -988,6 +990,15 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c, if (of_property_read_bool(np, "wlf,shared-lrclk")) pdata->shared_lrclk = true; + + pdata->mclk = devm_clk_get(&i2c->dev, NULL); + + if (IS_ERR(pdata->mclk)) { + if (PTR_ERR(pdata->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + + return 0; } static int wm8960_i2c_probe(struct i2c_client *i2c, @@ -1008,8 +1019,11 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (pdata) memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); - else if (i2c->dev.of_node) - wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + else if (i2c->dev.of_node) { + ret = wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + if (ret != 0) + return ret; + } ret = wm8960_reset(wm8960->regmap); if (ret != 0) { @@ -1041,6 +1055,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); @@ -1053,6 +1070,41 @@ static int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm8960_runtime_resume(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(wm8960->pdata.mclk)) { + ret = clk_prepare_enable(wm8960->pdata.mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int wm8960_runtime_suspend(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + + /* Power down PLL to save power */ + regmap_update_bits(wm8960->regmap, WM8960_POWER2, 0x1, 0); + + if (!IS_ERR(wm8960->pdata.mclk)) + clk_disable_unprepare(wm8960->pdata.mclk); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8960_pm = { + SET_RUNTIME_PM_OPS(wm8960_runtime_suspend, wm8960_runtime_resume, NULL) +}; + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -1070,6 +1122,7 @@ static struct i2c_driver wm8960_i2c_driver = { .name = "wm8960", .owner = THIS_MODULE, .of_match_table = wm8960_of_match, + .pm = &wm8960_pm, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

2 2

[alsa-devel] [PATCH 4/5] ASoC: dwc: Add devicetree support for Designware I2S
by Andrew Jackson 04 Dec '14

04 Dec '14

Convert to driver to use either platform_data or device-tree for configuration of the device. When using device-tree, the I2S block's configuration is read from the relevant registers: this reduces the amount of information required in the device tree. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- .../devicetree/bindings/sound/designware-i2s.txt | 32 +++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 238 ++++++++++++++++---- 3 files changed, 222 insertions(+), 49 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/designware-i2s.txt b/Documentation/devicetree/bindings/sound/designware-i2s.txt new file mode 100644 index 0000000..cdee591 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/designware-i2s.txt @@ -0,0 +1,32 @@ +DesignWare I2S controller + +Required properties: + - compatible : Must be "snps,designware-i2s" + - reg : Must contain I2S core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + The controller expects two clocks, the clock used for the APB interface and + the clock used as the sampling rate reference clock sample. + - clock-names : "apb_plck" for the clock to the APB interface, "i2sclk" for the sample + rate reference clock. + - dmas: Pairs of phandle and specifier for the DMA channels that are used by + the core. The core expects one or two dma channels, one for transmit and one for + receive. + - dma-names : "tx" for the transmit channel, "rx" for the receive channel. + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + + soc_i2s: i2s@7ff90000 { + compatible = "snps,designware-i2s"; + reg = <0x0 0x7ff90000 0x0 0x1000>; + clocks = <&scpi_i2sclk 0>, <&soc_refclk100mhz>; + clock-names = "i2sclk", "apb_pclk"; + #sound-dai-cells = <0>; + dmas = <&dma0 5>; + dma-names = "tx"; + }; diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index e334900..d50e085 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -1,6 +1,7 @@ config SND_DESIGNWARE_I2S tristate "Synopsys I2S Device Driver" depends on CLKDEV_LOOKUP + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for I2S driver for Synopsys desigwnware I2S device. The device supports upto diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index c497ada..083779d 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -22,6 +22,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> /* common register for all channel */ #define IER 0x000 @@ -54,19 +55,46 @@ #define I2S_COMP_VERSION 0x01F8 #define I2S_COMP_TYPE 0x01FC +/* + * Component parameter register fields - define the I2S block's + * configuration. + */ +#define COMP1_TX_WORDSIZE_3(r) (((r) & GENMASK(27, 25)) >> 25) +#define COMP1_TX_WORDSIZE_2(r) (((r) & GENMASK(24, 22)) >> 22) +#define COMP1_TX_WORDSIZE_1(r) (((r) & GENMASK(21, 19)) >> 19) +#define COMP1_TX_WORDSIZE_0(r) (((r) & GENMASK(18, 16)) >> 16) +#define COMP1_TX_CHANNELS(r) (((r) & GENMASK(10, 9)) >> 9) +#define COMP1_RX_CHANNELS(r) (((r) & GENMASK(8, 7)) >> 7) +#define COMP1_RX_ENABLED(r) (((r) & BIT(6)) >> 6) +#define COMP1_TX_ENABLED(r) (((r) & BIT(5)) >> 5) +#define COMP1_MODE_EN(r) (((r) & BIT(4)) >> 4) +#define COMP1_FIFO_DEPTH_GLOBAL(r) (((r) & GENMASK(3, 2)) >> 2) +#define COMP1_APB_DATA_WIDTH(r) (((r) & GENMASK(1, 0)) >> 0) + +#define COMP2_RX_WORDSIZE_3(r) (((r) & GENMASK(12, 10)) >> 10) +#define COMP2_RX_WORDSIZE_2(r) (((r) & GENMASK(9, 7)) >> 7) +#define COMP2_RX_WORDSIZE_1(r) (((r) & GENMASK(5, 3)) >> 3) +#define COMP2_RX_WORDSIZE_0(r) (((r) & GENMASK(2, 0)) >> 0) + #define MAX_CHANNEL_NUM 8 #define MIN_CHANNEL_NUM 2 +union snd_dma_data { + struct i2s_dma_data pd; + struct snd_dmaengine_dai_dma_data dt; +}; + struct dw_i2s_dev { void __iomem *i2s_base; struct clk *clk; int active; unsigned int capability; struct device *dev; + bool using_pd; /* data related to DMA transfers b/w i2s and DMAC */ - struct i2s_dma_data play_dma_data; - struct i2s_dma_data capture_dma_data; + union snd_dma_data play_dma_data; + union snd_dma_data capture_dma_data; struct i2s_clk_config_data config; int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); }; @@ -153,7 +181,7 @@ static int dw_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - struct i2s_dma_data *dma_data = NULL; + union snd_dma_data *dma_data = NULL; if (!(dev->capability & DWC_I2S_RECORD) && (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) @@ -227,8 +255,7 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, i2s_disable_channels(dev, substream->stream); - /* Iterate over set of channels - independently controlled. - */ + /* Iterate over set of channels - independently controlled. */ do { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { i2s_write_reg(dev->i2s_base, TCR(ch_reg), @@ -251,13 +278,18 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, config->sample_rate = params_rate(params); - if (!dev->i2s_clk_cfg) - return -EINVAL; + if (dev->using_pd) { + ret = dev->i2s_clk_cfg(config); + if (ret < 0) { + dev_err(dev->dev, "runtime audio clk config fail\n"); + return ret; + } + } else { + u32 bitclk; - ret = dev->i2s_clk_cfg(config); - if (ret < 0) { - dev_err(dev->dev, "runtime audio clk config fail\n"); - return ret; + /* TODO: Validate sample rate against permissible set */ + bitclk = config->sample_rate * config->data_width * 2; + clk_set_rate(dev->clk, bitclk); } return 0; @@ -331,6 +363,31 @@ static int dw_i2s_resume(struct snd_soc_dai *dai) #define dw_i2s_resume NULL #endif +/* Maximum resolution of a channel - not uniformly spaced */ +static const u32 fifo_width[] = { + 12, 16, 20, 24, 32, 0, 0, 0 +}; + +/* Width of (DMA) bus */ +static const u32 bus_widths[] = { + DMA_SLAVE_BUSWIDTH_1_BYTE, + DMA_SLAVE_BUSWIDTH_2_BYTES, + DMA_SLAVE_BUSWIDTH_4_BYTES, + DMA_SLAVE_BUSWIDTH_UNDEFINED +}; + +/* PCM format to support channel resolution */ +static const u32 formats[] = { + SNDRV_PCM_FMTBIT_S16_LE, + SNDRV_PCM_FMTBIT_S16_LE, + SNDRV_PCM_FMTBIT_S24_LE, + SNDRV_PCM_FMTBIT_S24_LE, + SNDRV_PCM_FMTBIT_S32_LE, + 0, + 0, + 0 +}; + static int dw_i2s_probe(struct platform_device *pdev) { const struct i2s_platform_data *pdata = pdev->dev.platform_data; @@ -340,11 +397,6 @@ static int dw_i2s_probe(struct platform_device *pdev) unsigned int cap; struct snd_soc_dai_driver *dw_i2s_dai; - if (!pdata) { - dev_err(&pdev->dev, "Invalid platform data\n"); - return -EINVAL; - } - dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); if (!dev) { dev_warn(&pdev->dev, "kzalloc fail\n"); @@ -373,47 +425,114 @@ static int dw_i2s_probe(struct platform_device *pdev) return PTR_ERR(dev->i2s_base); } - cap = pdata->cap; - dev->capability = cap; - dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + if (pdata) { + dev->using_pd = true; + cap = pdata->cap; + dev->capability = cap; + dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + if (!dev->i2s_clk_cfg) { + dev_err(&pdev->dev, "no clock device\n"); + return -ENODEV; + } - /* Set DMA slaves info */ + /* Set DMA slaves info */ + + dev->play_dma_data.pd.data = pdata->play_dma_data; + dev->capture_dma_data.pd.data = pdata->capture_dma_data; + dev->play_dma_data.pd.addr = res->start + I2S_TXDMA; + dev->capture_dma_data.pd.addr = res->start + I2S_RXDMA; + dev->play_dma_data.pd.max_burst = 16; + dev->capture_dma_data.pd.max_burst = 16; + dev->play_dma_data.pd.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->capture_dma_data.pd.addr_width = + DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->play_dma_data.pd.filter = pdata->filter; + dev->capture_dma_data.pd.filter = pdata->filter; + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) + return PTR_ERR(dev->clk); + + if (cap & DWC_I2S_PLAY) { + dev_dbg(&pdev->dev, " designware: play supported\n"); + dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->playback.channels_max = pdata->channel; + dw_i2s_dai->playback.formats = pdata->snd_fmts; + dw_i2s_dai->playback.rates = pdata->snd_rates; + } - dev->play_dma_data.data = pdata->play_dma_data; - dev->capture_dma_data.data = pdata->capture_dma_data; - dev->play_dma_data.addr = res->start + I2S_TXDMA; - dev->capture_dma_data.addr = res->start + I2S_RXDMA; - dev->play_dma_data.max_burst = 16; - dev->capture_dma_data.max_burst = 16; - dev->play_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; - dev->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; - dev->play_dma_data.filter = pdata->filter; - dev->capture_dma_data.filter = pdata->filter; + if (cap & DWC_I2S_RECORD) { + dev_dbg(&pdev->dev, "designware: record supported\n"); + dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->capture.channels_max = pdata->channel; + dw_i2s_dai->capture.formats = pdata->snd_fmts; + dw_i2s_dai->capture.rates = pdata->snd_rates; + } + } else { + /* + * Read component parameter registers to extract + * the I2S block's configuration. + */ + u32 comp1 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_1); + u32 comp2 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_2); + u32 bus_width = bus_widths[COMP1_APB_DATA_WIDTH(comp1)]; + u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1)); + u32 max_size; + + dev->using_pd = false; + + dev->clk = devm_clk_get(&pdev->dev, "i2sclk"); + if (IS_ERR(dev->clk)) + return PTR_ERR(dev->clk); + + /* + * Code presumes all channels are configured with the same + * word size. + */ + if (COMP1_TX_ENABLED(comp1)) { + dev_dbg(&pdev->dev, "playback capable\n"); + + dev->capability |= DWC_I2S_PLAY; + max_size = COMP1_TX_WORDSIZE_0(comp1); + dev->play_dma_data.dt.addr = res->start + I2S_TXDMA; + dev->play_dma_data.dt.addr_width = bus_width; + dev->play_dma_data.dt.chan_name = "TX"; + dev->play_dma_data.dt.fifo_size = fifo_depth * + (fifo_width[max_size]) >> 8; + dev->play_dma_data.dt.maxburst = 16; + dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->playback.channels_max = + 1 << (COMP1_TX_CHANNELS(comp1) + 1); + dw_i2s_dai->playback.formats = formats[max_size]; + dw_i2s_dai->playback.rates = SNDRV_PCM_RATE_8000_192000; + } + if (COMP1_RX_ENABLED(comp1)) { + dev_dbg(&pdev->dev, "record capable\n"); + + dev->capability |= DWC_I2S_RECORD; + max_size = COMP2_RX_WORDSIZE_0(comp2); + dev->capture_dma_data.dt.addr = res->start + I2S_RXDMA; + dev->capture_dma_data.dt.addr_width = bus_width; + dev->capture_dma_data.dt.chan_name = "RX"; + dev->capture_dma_data.dt.fifo_size = fifo_depth * + (fifo_width[max_size] >> 8); + dev->capture_dma_data.dt.maxburst = 16; + dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->capture.channels_max = + 1 << (COMP1_RX_CHANNELS(comp1) + 1); + dw_i2s_dai->capture.formats = formats[max_size]; + dw_i2s_dai->capture.rates = SNDRV_PCM_RATE_8000_192000; + } + } - dev->clk = clk_get(&pdev->dev, NULL); - if (IS_ERR(dev->clk)) - return PTR_ERR(dev->clk); + ret = clk_prepare(dev->clk); + if (ret < 0) + goto err_clk_put; ret = clk_enable(dev->clk); if (ret < 0) goto err_clk_put; - if (cap & DWC_I2S_PLAY) { - dev_dbg(&pdev->dev, " designware: play supported\n"); - dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; - dw_i2s_dai->playback.channels_max = pdata->channel; - dw_i2s_dai->playback.formats = pdata->snd_fmts; - dw_i2s_dai->playback.rates = pdata->snd_rates; - } - - if (cap & DWC_I2S_RECORD) { - dev_dbg(&pdev->dev, "designware: record supported\n"); - dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; - dw_i2s_dai->capture.channels_max = pdata->channel; - dw_i2s_dai->capture.formats = pdata->snd_fmts; - dw_i2s_dai->capture.rates = pdata->snd_rates; - } - dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component, @@ -423,6 +542,15 @@ static int dw_i2s_probe(struct platform_device *pdev) goto err_clk_disable; } + if (!dev->using_pd) { + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, + "Could not register PCM: %d\n", ret); + return ret; + } + } + return 0; err_clk_disable: @@ -443,12 +571,24 @@ static int dw_i2s_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id dw_i2s_of_match[] = { + { .compatible = "snps,designware-i2s", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, dw_i2s_of_match); +#endif + static struct platform_driver dw_i2s_driver = { .probe = dw_i2s_probe, .remove = dw_i2s_remove, .driver = { .name = "designware-i2s", .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = dw_i2s_of_match, +#endif }, }; -- 1.7.1

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[alsa-devel] [PATCH 2/5] ASoC: dwc: Don't allow negative use counts
by Andrew Jackson 04 Dec '14

04 Dec '14

Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 08f0229..f8946bd 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -280,7 +280,8 @@ static int dw_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dev->active--; + if (dev->active > 0) + dev->active--; i2s_stop(dev, substream); break; default: -- 1.7.1

3 6

[alsa-devel] [PATCH 1/5] ASoC: dwc: Allocate resources with devm_ioremap_resource
by Andrew Jackson 04 Dec '14

04 Dec '14

Prepare for the introduction of device-tree support by re-ordering some of the allocations and using devm_iomap_resource to simplify IO mapping. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 46 ++++++++++++++++----------------------- 1 files changed, 19 insertions(+), 27 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index e961388..08f0229 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -338,31 +338,34 @@ static int dw_i2s_probe(struct platform_device *pdev) return -EINVAL; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "no i2s resource defined\n"); - return -ENODEV; - } - - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name)) { - dev_err(&pdev->dev, "i2s region already claimed\n"); - return -EBUSY; - } - dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); if (!dev) { dev_warn(&pdev->dev, "kzalloc fail\n"); return -ENOMEM; } - dev->i2s_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!dev->i2s_base) { - dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); + if (!dw_i2s_dai) { + dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); return -ENOMEM; } + dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->suspend = dw_i2s_suspend; + dw_i2s_dai->resume = dw_i2s_resume; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no i2s resource defined\n"); + return -ENODEV; + } + + dev->i2s_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(dev->i2s_base)) { + dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + return PTR_ERR(dev->i2s_base); + } + cap = pdata->cap; dev->capability = cap; dev->i2s_clk_cfg = pdata->i2s_clk_cfg; @@ -388,13 +391,6 @@ static int dw_i2s_probe(struct platform_device *pdev) if (ret < 0) goto err_clk_put; - dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); - if (!dw_i2s_dai) { - dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); - ret = -ENOMEM; - goto err_clk_disable; - } - if (cap & DWC_I2S_PLAY) { dev_dbg(&pdev->dev, " designware: play supported\n"); dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; @@ -411,10 +407,6 @@ static int dw_i2s_probe(struct platform_device *pdev) dw_i2s_dai->capture.rates = pdata->snd_rates; } - dw_i2s_dai->ops = &dw_i2s_dai_ops; - dw_i2s_dai->suspend = dw_i2s_suspend; - dw_i2s_dai->resume = dw_i2s_resume; - dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component, -- 1.7.1

3 5

[alsa-devel] [PATCH 3/5] ASoC: dwc: Iterate over all channels
by Andrew Jackson 04 Dec '14

04 Dec '14

On the Designware core, the channels are independent and not combined in higher registers. So as more channels are added, more registers need to be updated. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 32 +++++++++++++++++++------------- 1 files changed, 19 insertions(+), 13 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index f8946bd..c497ada 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -227,19 +227,25 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, i2s_disable_channels(dev, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution); - i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); - irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); - i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); - i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); - } else { - i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution); - i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); - irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); - i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); - i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); - } + /* Iterate over set of channels - independently controlled. + */ + do { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, TCR(ch_reg), + xfer_resolution); + i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); + i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); + } else { + i2s_write_reg(dev->i2s_base, RCR(ch_reg), + xfer_resolution); + i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); + i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); + } + } while (ch_reg-- > 0); i2s_write_reg(dev->i2s_base, CCR, ccr); -- 1.7.1

3 4

[alsa-devel] [PATCH 0/5] ASoC: dwc: Add device tree support to designware I2S
by Andrew Jackson 04 Dec '14

04 Dec '14

This patch set extends the DesignWare I2S driver to provide device tree support and fixes a couple of small faults. Andrew Jackson (5): ASoC: dwc: Allocate resources with devm_ioremap_resource ASoC: dwc: Don't allow negative use counts ASoC: dwc: Iterate over all channels ASoC: dwc: Add devicetree support for Designware I2S ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap .../devicetree/bindings/sound/designware-i2s.txt | 32 ++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 313 ++++++++++++++------ 3 files changed, 260 insertions(+), 86 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt

2 2

[alsa-devel] [PATCH 0/2] rockchip: i2s: patches to add optional clock output from
by Jianqun Xu 04 Dec '14

04 Dec '14

The clock output from rockchip SoCs' CRU to chip outside is from the same source as hclk which is for i2s controller. We make it optional because some audio codec outside not need the master clock from SoC but from other clock source. Jianqun Xu (2): ASoC: rockchip-i2s: dt: add an optional property "i2s_clk_out" ASoC: rockchip: i2s: add support for grabbing output clock to codec Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 8 ++++++-- sound/soc/rockchip/rockchip_i2s.c | 11 +++++++++++ 2 files changed, 17 insertions(+), 2 deletions(-) -- 1.9.1

6 16

[alsa-devel] [PATCH v2] ASoC: rt5677: make volume TLV closer to reality
by Dylan Reid 03 Dec '14

03 Dec '14

The volume blocks have an step of 0.375dB, but TLV uses 0.01dB for units. Switch to use TLV_DB_MINMAX as this allows the minimum to be off by only 0.005dB. This is an improvement from the ~500dB is was misreported by before. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/rt5677.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 16aa4d9..a4c71f9 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -538,9 +538,9 @@ 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); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 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); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 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

3 6

[alsa-devel] [PATCH 0/2] ALSA: ctxfi: Delete an unnecessary check before kfree()
by SF Markus Elfring 03 Dec '14

03 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 20:35:41 +0100 Another update suggestion was taken into account after a patch was applied from static source code analysis. Markus Elfring (2): Deletion of an unnecessary check before the function call "kfree" One function call less in get_daio_rsc() after error detection sound/pci/ctxfi/ctdaio.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) -- 2.1.3

3 10