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

[alsa-devel] [PATCH v3] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 08 Dec '14

08 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. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- include/sound/wm8960.h | 1 + sound/soc/codecs/wm8960.c | 56 ++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 54 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..ea5f0b7 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 +1070,38 @@ 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); + + 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 +1119,7 @@ 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

3 7

[alsa-devel] [PATCH] ASoC: wm5102: Initialize dac_comp_lock mutex
by Mark Brown 08 Dec '14

08 Dec '14

Commit d74bcaaeb6682 (ASoC: wm5102: Move ultrasonic response settings lock to the driver level) created a driver local mutex for protecting the ultrasonic response settings but neglected to initialize that mutex, causing loud complaints from lockep and potential runtime failures. Fix this by initializing the mutex. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm5102.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 8c99b7cf47e0..f439ae052128 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1900,6 +1900,8 @@ static int wm5102_probe(struct platform_device *pdev) return -ENOMEM; platform_set_drvdata(pdev, wm5102); + mutex_init(&arizona->dac_comp_lock); + wm5102->core.arizona = arizona; wm5102->core.num_inputs = 6; -- 2.1.3

2 1

[alsa-devel] [PATCH 00/15 v4] ALSA: Enhancement for existed FireWire drivers
by Takashi Sakamoto 08 Dec '14

08 Dec '14

This patchset is the rest of patches to enable ALSA Dice driver to handle more models and to improve OXFW driver. This patchset is for linux-next branch. == Updates from my previous post: [alsa-devel] [PATCH 00/30 v3] ALSA: Enhancement for existed FireWire drivers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084791.h… - move mutex operations to a condition in pcm.hw_params (dice/oxfw) - fix naming regression (oxfw) == Supplemental patch for userspace library This patchset adds Hwdep functionality for OXFW driver, thus userspace library needs a patch for entries of snd_hwdep_iface_t. Please apply below patch. [alsa-devel] [PATCH alsa-lib] hwdep: add OXFW driver support http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084824.h… Takashi Sakamoto (15): ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for capturing PCM samples ALSA: dice: Add support for MIDI capture/playback ALSA: oxfw: Change the way to name card ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 10 +- sound/firewire/dice/Makefile | 4 +- sound/firewire/dice/dice-midi.c | 194 ++++++++++ sound/firewire/dice/dice-pcm.c | 252 +++++++++---- sound/firewire/dice/dice-stream.c | 332 ++++++++++++---- sound/firewire/dice/dice-transaction.c | 5 - sound/firewire/dice/dice.c | 37 +- sound/firewire/dice/dice.h | 33 +- sound/firewire/oxfw/Makefile | 3 +- sound/firewire/oxfw/oxfw-command.c | 153 ++++++++ sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++++++++ sound/firewire/oxfw/oxfw-midi.c | 207 ++++++++++ sound/firewire/oxfw/oxfw-pcm.c | 425 +++++++++++++++------ sound/firewire/oxfw/oxfw-proc.c | 113 ++++++ sound/firewire/oxfw/oxfw-stream.c | 667 +++++++++++++++++++++++++++++++-- sound/firewire/oxfw/oxfw.c | 181 +++++++-- sound/firewire/oxfw/oxfw.h | 104 ++++- 19 files changed, 2539 insertions(+), 377 deletions(-) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/oxfw/oxfw-command.c create mode 100644 sound/firewire/oxfw/oxfw-hwdep.c create mode 100644 sound/firewire/oxfw/oxfw-midi.c create mode 100644 sound/firewire/oxfw/oxfw-proc.c -- 2.1.0

2 19

[alsa-devel] [PATCH] sound: pci: asihpi: hpifunc.c: Remove some unused functions
by Rickard Strandqvist 07 Dec '14

07 Dec '14

Removes some functions that are not used anywhere: hpi_subsys_get_version_ex() hpi_instream_wait_start() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/asihpi/hpi.h | 4 ---- sound/pci/asihpi/hpifunc.c | 27 --------------------------- 2 files changed, 31 deletions(-) diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 2088724..8338b6e 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -1164,8 +1164,6 @@ u16 hpi_stream_estimate_buffer_size(struct hpi_format *pF, /* SubSystem */ /*************/ -u16 hpi_subsys_get_version_ex(u32 *pversion_ex); - u16 hpi_subsys_get_num_adapters(int *pn_num_adapters); u16 hpi_subsys_get_adapter(int iterator, u32 *padapter_index, @@ -1289,8 +1287,6 @@ u16 hpi_instream_read_buf(u32 h_instream, u8 *pb_read_buf, u32 bytes_to_read); u16 hpi_instream_start(u32 h_instream); -u16 hpi_instream_wait_start(u32 h_instream); - u16 hpi_instream_stop(u32 h_instream); u16 hpi_instream_reset(u32 h_instream); diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 510e56c..8fd02dd 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -93,18 +93,6 @@ static inline void hpi_send_recvV1(struct hpi_message_header *m, hpi_send_recv((struct hpi_message *)m, (struct hpi_response *)r); } -u16 hpi_subsys_get_version_ex(u32 *pversion_ex) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM, - HPI_SUBSYS_GET_VERSION); - hpi_send_recv(&hm, &hr); - *pversion_ex = hr.u.s.data; - return hr.error; -} - u16 hpi_subsys_get_num_adapters(int *pn_num_adapters) { struct hpi_message hm; @@ -950,21 +938,6 @@ u16 hpi_instream_start(u32 h_instream) return hr.error; } -u16 hpi_instream_wait_start(u32 h_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_WAIT_START); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - u16 hpi_instream_stop(u32 h_instream) { struct hpi_message hm; -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 07 Dec '14

07 Dec '14

sound/soc/xtensa/xtfpga-i2s.c:666:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- xtfpga-i2s.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -663,7 +663,6 @@ static struct platform_driver xtfpga_i2s .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, },

1 0

[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