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

[alsa-devel] [PATCH 1/2] ASoC:rt286: set combo jack by dmi
by bardliao＠realtek.com 06 Nov '14

06 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch enables combo jack configuration according to dmi. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * rename intel_platform to force_combo_jack_table --- --- sound/soc/codecs/rt286.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 97daa80..d4acb64 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -17,6 +17,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> @@ -1205,6 +1206,16 @@ static const struct acpi_device_id rt286_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); +static struct dmi_system_id force_combo_jack_table[] __initdata = { + { + .ident = "Intel Wilson Beach", + .matches = { + DMI_MATCH(DMI_BOARD_NAME, "Wilson Beach SDS") + } + }, + { } +}; + static int rt286_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1240,6 +1251,9 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; + if (dmi_check_system(force_combo_jack_table)) + rt286->pdata.cbj_en = true; + regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); for (i = 0; i < RT286_POWER_REG_LEN; i++) -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Remove '.owner' field of driver
by Manish Badarkhe 06 Nov '14

06 Nov '14

Remove '.owner' field of driver structure as it is going to be filled via "module_i2c_driver" call. Signed-off-by: Manish Badarkhe <manishvb(a)ti.com> --- :100644 100644 0307909... 191b423... M sound/soc/codecs/tlv320aic31xx.c sound/soc/codecs/tlv320aic31xx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 0307909..191b423 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1303,7 +1303,6 @@ MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); static struct i2c_driver aic31xx_i2c_driver = { .driver = { .name = "tlv320aic31xx-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tlv320aic31xx_of_match), }, .probe = aic31xx_i2c_probe, -- 1.7.9.5

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Re: [alsa-devel] Restore hidden register for default value
by Kailang 06 Nov '14

06 Nov '14

Hi Takashi, I mail it again. BR, Kailang > -----Original Message----- > From: Kailang > Sent: Friday, October 31, 2014 3:12 PM > To: 'Takashi Iwai' > Cc: (alsa-devel(a)alsa-project.org) > Subject: RE: Restore hidden register for default value > > Hi Takashi, > > It's just the default value refill in initial state. > It's only for ALC668(ALC3661). > The patch was tested pass with issue machine in ODM site. > > Many Thanks. > > BR, > Kailang > > > -----Original Message----- > > From: Takashi Iwai [mailto:tiwai@suse.de] > > Sent: Wednesday, October 22, 2014 4:00 PM > > To: Kailang > > Subject: Re: Restore hidden register for default value > > > > At Wed, 22 Oct 2014 07:56:43 +0000, > > Kailang wrote: > > > > > > Hi Takashi, > > > > > > We want to post ALC668 for hidden register issue. > > > > > > Could I post like below array for codec initial state? > > > > > > WORD coefIndexDefault[][2] = {{0x1, > > 0xbebe},{0x2,0xaaaa},{0x3,0x0},{0x4,0x180},{0x6,0x0}, > > > > > {0x7,0xf80},{0x8,0x31},{0xa,0x60},{0xb,0x0},{0xc,0x7cf7},{0xd, > > 0x1080},{0xe,0x7f7f}, > > > > > {0xf,0xcccc},{0x10,0xddcc},{0x11,0x1},{0x13,0x0},{0x14,0x2aa0} > > ,{0x17,0xA940}, > > > > > {0x19,0},{0x1A,0},{0x1B,0},{0x1c,0x0},{0x1d,0x0},{0x1e,0x7418} > > ,{0x1f,0x804}, > > > > > {0x20,0x4200},{0x21,0x468},{0x22,0x8ccc},{0x23,0x250},{0x24,0x7418}, > > > > > > > > > {0x27,0x0000},{0x28,0x8ccc},{0x2a,0xff00},{0x2b,0x8000},{0xa7,0xff00}} > > > ; Because codec sometimes has wrong value for hidden > register after > > > resume back or reboot. > > > It needed to refill the default value with upper array in > > initial state. > > > But the hidden registers was more for 59 records. > > > Could I post it with no remark? Because it just the default > > value. Thanks. > > > > Some comments would be nice, of course, but not mandatory. > > > > One concern is exactly which codecs should this be applied. > > There are a bunch of codecs compatible with alc668. It's only for > > ALC668, and not for others (ALC668, 670, whatever). > > > > > > Takashi > > > > ------Please consider the environment before printing this e-mail. > >

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[alsa-devel] [PATCH] ASoC:rt286: set combo jack by dmi
by bardliao＠realtek.com 06 Nov '14

06 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch enables combo jack configuration according to dmi. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 97daa80..aa572b6 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -17,6 +17,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> @@ -1205,6 +1206,16 @@ static const struct acpi_device_id rt286_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); +static struct dmi_system_id intel_platform[] __initdata = { + { + .ident = "Intel Wilson Beach", + .matches = { + DMI_MATCH(DMI_BOARD_NAME, "Wilson Beach SDS") + } + }, + { } +}; + static int rt286_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1240,6 +1251,9 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; + if (dmi_check_system(intel_platform)) + rt286->pdata.cbj_en = true; + regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); for (i = 0; i < RT286_POWER_REG_LEN; i++) -- 1.8.1.1.439.g50a6b54

3 2

[alsa-devel] [PATCH] ASoC: samsung: Add MODULE_DEVICE_TABLE for Snow
by Andreas Färber 05 Nov '14

05 Nov '14

This enables the snd_soc_snow module to be auto-loaded. Fixes: da5993df2374 ("ASoC: samsung: Add sound card driver for Snow board") Signed-off-by: Andreas Färber <afaerber(a)suse.de> --- Tested using patches for Spring, but should apply to Snow/Peach-pi equally. Rebased onto sound.git for-next branch. sound/soc/samsung/snow.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 0acf5d0eed53..72118a77dd5b 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -110,6 +110,7 @@ static const struct of_device_id snow_of_match[] = { { .compatible = "google,snow-audio-max98095", }, {}, }; +MODULE_DEVICE_TABLE(of, snow_of_match); static struct platform_driver snow_driver = { .driver = { -- 2.1.2

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[alsa-devel] [PATCH] ASoC: ad193x: Keep DAC output stage active in idle
by Lars-Peter Clausen 05 Nov '14

05 Nov '14

Setting the DAC power-down bit for the ad193x will also disable the DAC output amplifier. This will cause audible clicks and pops when starting or stopping playback. To prevent this a new widget is introduced that controls the DAC power-down bit. This widget is connected to both the DAC and a newly introduced VMID widget. This makes sure that the DAC power-down bit is not set as long as a audio sink is connected to the DAC output. At the same time the PLL and SYSCLK will still be disabled when no playback or capture stream is active. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad193x.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 6844d0b..387530b 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -72,11 +72,13 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { }; static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { - SND_SOC_DAPM_DAC("DAC", "Playback", AD193X_DAC_CTRL0, 0, 1), + SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), + SND_SOC_DAPM_VMID("VMID"), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), @@ -87,13 +89,15 @@ static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "SYSCLK" }, + { "DAC Output", NULL, "DAC" }, + { "DAC Output", NULL, "VMID" }, { "ADC", NULL, "SYSCLK" }, { "DAC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC_PWR" }, - { "DAC1OUT", NULL, "DAC" }, - { "DAC2OUT", NULL, "DAC" }, - { "DAC3OUT", NULL, "DAC" }, - { "DAC4OUT", NULL, "DAC" }, + { "DAC1OUT", NULL, "DAC Output" }, + { "DAC2OUT", NULL, "DAC Output" }, + { "DAC3OUT", NULL, "DAC Output" }, + { "DAC4OUT", NULL, "DAC Output" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, { "SYSCLK", NULL, "PLL_PWR" }, -- 1.8.0

2 1

[alsa-devel] [PATCH v2] ALSA: echoaudio: cleanup of unnecessary messages
by Sudip Mukherjee 05 Nov '14

05 Nov '14

commit "b5b4a41b392960010fccf1f9ccf8334d612bd450" was dereferencing chip after it has been freed. This patch fixes that and at the same time removes some debugging messages, which are unnecessary, as they are just printing information about entry and exit from a function, and which switch-case it is executing. we can easily get from ftrace the information about the entry and exit from a function. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change from v1: removing some more debug messages sound/pci/echoaudio/darla20_dsp.c | 2 -- sound/pci/echoaudio/darla24_dsp.c | 2 -- sound/pci/echoaudio/echo3g_dsp.c | 2 -- sound/pci/echoaudio/echoaudio.c | 31 ------------------------------- sound/pci/echoaudio/echoaudio_3g.c | 5 ----- sound/pci/echoaudio/echoaudio_dsp.c | 12 ------------ sound/pci/echoaudio/gina20_dsp.c | 6 ------ sound/pci/echoaudio/gina24_dsp.c | 9 --------- sound/pci/echoaudio/indigo_dsp.c | 2 -- sound/pci/echoaudio/indigodj_dsp.c | 2 -- sound/pci/echoaudio/indigodjx_dsp.c | 2 -- sound/pci/echoaudio/indigoio_dsp.c | 2 -- sound/pci/echoaudio/indigoiox_dsp.c | 2 -- sound/pci/echoaudio/layla20_dsp.c | 9 --------- sound/pci/echoaudio/layla24_dsp.c | 8 -------- sound/pci/echoaudio/mia_dsp.c | 2 -- sound/pci/echoaudio/midi.c | 6 ------ sound/pci/echoaudio/mona_dsp.c | 6 ------ 18 files changed, 110 deletions(-) diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index c94e92e..febee5b 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA20)) return -ENODEV; @@ -58,7 +57,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index b1272f88..7b4f6fd 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA24)) return -ENODEV; @@ -57,7 +56,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index bc37168..ae11ce1 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -46,7 +46,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) int err; local_irq_enable(); - dev_dbg(chip->card->dev, "init_hw() - Echo3G\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != ECHO3G)) return -ENODEV; @@ -99,7 +98,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1ef29e5..60e4003 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -79,7 +79,6 @@ static void free_firmware(const struct firmware *fw_entry, dev_dbg(chip->card->dev, "firmware not released (kept in cache)\n"); #else release_firmware(fw_entry); - dev_dbg(chip->card->dev, "firmware released\n"); #endif } @@ -96,7 +95,6 @@ static void free_firmware_cache(struct echoaudio *chip) dev_dbg(chip->card->dev, "release_firmware(%d)\n", i); } - dev_dbg(chip->card->dev, "firmware_cache released\n"); #endif } @@ -354,7 +352,6 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err; - dev_dbg(chip->card->dev, "pcm_analog_in_open\n"); if ((err = pcm_open(substream, num_analog_busses_in(chip) - substream->number)) < 0) return err; @@ -389,7 +386,6 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream) #else max_channels = num_analog_busses_out(chip); #endif - dev_dbg(chip->card->dev, "pcm_analog_out_open\n"); if ((err = pcm_open(substream, max_channels - substream->number)) < 0) return err; if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, @@ -422,7 +418,6 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_in_open\n"); max_channels = num_digital_busses_in(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -464,7 +459,6 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_out_open\n"); max_channels = num_digital_busses_out(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -511,7 +505,6 @@ static int pcm_close(struct snd_pcm_substream *substream) /* Nothing to do here. Audio is already off and pipe will be * freed by its callback */ - dev_dbg(chip->card->dev, "pcm_close\n"); atomic_dec(&chip->opencount); oc = atomic_read(&chip->opencount); @@ -620,7 +613,6 @@ static int init_engine(struct snd_pcm_substream *substream, spin_lock_irq(&chip->lock); set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den); spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_params ok\n"); return 0; } @@ -691,7 +683,6 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) } spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_freed\n"); snd_pcm_lib_free_pages(substream); return 0; } @@ -763,10 +754,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&chip->lock); switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - dev_dbg(chip->card->dev, "pcm_trigger resume\n"); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - dev_dbg(chip->card->dev, "pcm_trigger start\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -788,9 +777,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) chip->pipe_cyclic_mask); break; case SNDRV_PCM_TRIGGER_SUSPEND: - dev_dbg(chip->card->dev, "pcm_trigger suspend\n"); case SNDRV_PCM_TRIGGER_STOP: - dev_dbg(chip->card->dev, "pcm_trigger stop\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -800,7 +787,6 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) err = stop_transport(chip, channelmask); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dev_dbg(chip->card->dev, "pcm_trigger pause\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -937,7 +923,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Analog PCM ok\n"); #ifdef ECHOCARD_HAS_DIGITAL_IO /* PCM#1 Digital inputs, no outputs */ @@ -950,7 +935,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Digital PCM ok\n"); #endif /* ECHOCARD_HAS_DIGITAL_IO */ #else /* ECHOCARD_HAS_VMIXER */ @@ -972,7 +956,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Analog PCM ok\n"); #ifdef ECHOCARD_HAS_DIGITAL_IO /* PCM#1 Digital i/o */ @@ -987,7 +970,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Digital PCM ok\n"); #endif /* ECHOCARD_HAS_DIGITAL_IO */ #endif /* ECHOCARD_HAS_VMIXER */ @@ -1881,10 +1863,8 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id) static int snd_echo_free(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "Stop DSP...\n"); if (chip->comm_page) rest_in_peace(chip); - dev_dbg(chip->card->dev, "Stopped.\n"); if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1898,14 +1878,12 @@ static int snd_echo_free(struct echoaudio *chip) if (chip->iores) release_and_free_resource(chip->iores); - dev_dbg(chip->card->dev, "MMIO freed.\n"); pci_disable_device(chip->pci); /* release chip data */ free_firmware_cache(chip); kfree(chip); - dev_dbg(chip->card->dev, "Chip freed.\n"); return 0; } @@ -1915,7 +1893,6 @@ static int snd_echo_dev_free(struct snd_device *device) { struct echoaudio *chip = device->device_data; - dev_dbg(chip->card->dev, "snd_echo_dev_free()...\n"); return snd_echo_free(chip); } @@ -2008,7 +1985,6 @@ static int snd_echo_create(struct snd_card *card, snd_echo_free(chip); return err; } - dev_dbg(card->dev, "Card init OK\n"); if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_echo_free(chip); @@ -2038,7 +2014,6 @@ static int snd_echo_probe(struct pci_dev *pci, return -ENOENT; } - dev_dbg(&pci->dev, "Echoaudio driver starting...\n"); i = 0; err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); @@ -2191,7 +2166,6 @@ static int snd_echo_suspend(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct echoaudio *chip = dev_get_drvdata(dev); - dev_dbg(dev, "suspend start\n"); snd_pcm_suspend_all(chip->analog_pcm); snd_pcm_suspend_all(chip->digital_pcm); @@ -2218,7 +2192,6 @@ static int snd_echo_suspend(struct device *dev) pci_save_state(pci); pci_disable_device(pci); - dev_dbg(dev, "suspend done\n"); return 0; } @@ -2232,7 +2205,6 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - dev_dbg(dev, "resume start\n"); pci_restore_state(pci); commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); if (commpage_bak == NULL) @@ -2247,7 +2219,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_free(chip); return err; } - dev_dbg(dev, "resume init OK\n"); /* Temporarily set chip->pipe_alloc_mask=0 otherwise * restore_dsp_settings() fails. @@ -2260,7 +2231,6 @@ static int snd_echo_resume(struct device *dev) kfree(commpage_bak); return err; } - dev_dbg(dev, "resume restore OK\n"); memcpy(&commpage->audio_format, &commpage_bak->audio_format, sizeof(commpage->audio_format)); @@ -2286,7 +2256,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_midi_output_trigger(chip->midi_out, 1); #endif - dev_dbg(dev, "resume done\n"); return 0; } diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index 50a2169..2fa66dc 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -328,7 +328,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & @@ -337,13 +336,11 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Echo3G clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Echo3G clock to SPDIF\n"); control_reg |= E3G_SPDIF_CLOCK; if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_SPDIF96) control_reg |= E3G_DOUBLE_SPEED_MODE; @@ -353,12 +350,10 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Echo3G clock to ADAT\n"); control_reg |= E3G_ADAT_CLOCK; control_reg &= ~E3G_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_WORD: - dev_dbg(chip->card->dev, "Set Echo3G clock to WORD\n"); control_reg |= E3G_WORD_CLOCK; if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_WORD96) control_reg |= E3G_DOUBLE_SPEED_MODE; diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index ba9d4f1..1a9427a 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -206,7 +206,6 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) goto la_error; } - dev_dbg(chip->card->dev, "ASIC loaded\n"); free_firmware(fw, chip); return 0; @@ -473,7 +472,6 @@ static int load_dsp(struct echoaudio *chip, u16 *code) chip->dsp_code = code; /* Show which DSP code loaded */ chip->bad_board = FALSE; /* DSP OK */ - dev_dbg(chip->card->dev, "load_dsp: OK!\n"); return 0; } udelay(100); @@ -674,7 +672,6 @@ static void get_audio_meters(struct echoaudio *chip, long *meters) static int restore_dsp_rettings(struct echoaudio *chip) { int i, o, err; - dev_dbg(chip->card->dev, "restore_dsp_settings\n"); if ((err = check_asic_status(chip)) < 0) return err; @@ -771,7 +768,6 @@ static int restore_dsp_rettings(struct echoaudio *chip) if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) return -EIO; - dev_dbg(chip->card->dev, "restore_dsp_rettings done\n"); return 0; } @@ -865,7 +861,6 @@ Same thing for pause_ and stop_ -trasport below. */ static int start_transport(struct echoaudio *chip, u32 channel_mask, u32 cyclic_mask) { - dev_dbg(chip->card->dev, "start_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -891,7 +886,6 @@ static int start_transport(struct echoaudio *chip, u32 channel_mask, static int pause_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "pause_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -918,7 +912,6 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask) static int stop_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "stop_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -954,8 +947,6 @@ static inline int is_pipe_allocated(struct echoaudio *chip, u16 pipe_index) stopped and unallocated. */ static int rest_in_peace(struct echoaudio *chip) { - dev_dbg(chip->card->dev, - "rest_in_peace() open=%x\n", chip->pipe_alloc_mask); /* Stops all active pipes (just to be sure) */ stop_transport(chip, chip->active_mask); @@ -1018,7 +1009,6 @@ static int init_dsp_comm_page(struct echoaudio *chip) */ static int init_line_levels(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "init_line_levels\n"); memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); @@ -1099,7 +1089,6 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe, it moves data. The DMA counter is in units of bytes, not samples. */ pipe->dma_counter = &chip->comm_page->position[pipe_index]; *pipe->dma_counter = 0; - dev_dbg(chip->card->dev, "allocate_pipes: ok\n"); return pipe_index; } @@ -1110,7 +1099,6 @@ static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe) u32 channel_mask; int i; - dev_dbg(chip->card->dev, "free_pipes: Pipe %d\n", pipe->index); if (snd_BUG_ON(!is_pipe_allocated(chip, pipe->index))) return -EINVAL; if (snd_BUG_ON(pipe->state != PIPE_STATE_STOPPED)) diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index a959eae..5dafe92 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -37,7 +37,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA20)) return -ENODEV; @@ -63,7 +62,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -150,7 +148,6 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) static int set_input_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_input_clock:\n"); switch (clock) { case ECHO_CLOCK_INTERNAL: @@ -159,7 +156,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) chip->spdif_status = GD_SPDIF_STATUS_UNDEF; set_sample_rate(chip, chip->sample_rate); chip->input_clock = clock; - dev_dbg(chip->card->dev, "Set Gina clock to INTERNAL\n"); break; case ECHO_CLOCK_SPDIF: chip->comm_page->gd_clock_state = GD_CLOCK_SPDIFIN; @@ -167,7 +163,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) clear_handshake(chip); send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE); chip->clock_state = GD_CLOCK_SPDIFIN; - dev_dbg(chip->card->dev, "Set Gina20 clock to SPDIF\n"); chip->input_clock = clock; break; default: @@ -209,7 +204,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index c8ea576..6971766 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA24)) return -ENODEV; @@ -79,7 +78,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -156,7 +154,6 @@ static int load_asic(struct echoaudio *chip) control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; err = write_control_reg(chip, control_reg, TRUE); } - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } @@ -238,7 +235,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & @@ -247,13 +243,11 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Gina24 clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Gina24 clock to SPDIF\n"); control_reg |= GML_SPDIF_CLOCK; if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96) control_reg |= GML_DOUBLE_SPEED_MODE; @@ -263,17 +257,14 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Gina24 clock to ADAT\n"); control_reg |= GML_ADAT_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_ESYNC: - dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC\n"); control_reg |= GML_ESYNC_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_ESYNC96: - dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC96\n"); control_reg |= GML_ESYNC_CLOCK | GML_DOUBLE_SPEED_MODE; break; default: diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index cdeb073..54edd67 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 133915c..2cf5cc0 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJ\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJ)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index 26cdfcf..5252863 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 5e6df7c..4e81787 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IO\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 90cdd27..6de3f9b 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IOx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 7f0f6ea..f2024a3 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA20)) return -ENODEV; @@ -65,7 +64,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -190,24 +188,19 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) u16 clock; u32 rate; - dev_dbg(chip->card->dev, "set_input_clock:\n"); rate = 0; switch (clock_source) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Layla20 clock to INTERNAL\n"); rate = chip->sample_rate; clock = LAYLA20_CLOCK_INTERNAL; break; case ECHO_CLOCK_SPDIF: - dev_dbg(chip->card->dev, "Set Layla20 clock to SPDIF\n"); clock = LAYLA20_CLOCK_SPDIF; break; case ECHO_CLOCK_WORD: - dev_dbg(chip->card->dev, "Set Layla20 clock to WORD\n"); clock = LAYLA20_CLOCK_WORD; break; case ECHO_CLOCK_SUPER: - dev_dbg(chip->card->dev, "Set Layla20 clock to SUPER\n"); clock = LAYLA20_CLOCK_SUPER; break; default: @@ -232,7 +225,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) static int set_output_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_output_clock: %d\n", clock); switch (clock) { case ECHO_CLOCK_SUPER: clock = LAYLA20_OUTPUT_CLOCK_SUPER; @@ -286,7 +278,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index eb8f218..4f11e81 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA24)) return -ENODEV; @@ -70,7 +69,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -118,7 +116,6 @@ static int load_asic(struct echoaudio *chip) if (chip->asic_loaded) return 1; - dev_dbg(chip->card->dev, "load_asic\n"); /* Give the DSP a few milliseconds to settle down */ mdelay(10); @@ -152,7 +149,6 @@ static int load_asic(struct echoaudio *chip) err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ, TRUE); - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } @@ -262,7 +258,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) /* Pick the new clock */ switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Layla24 clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: @@ -271,7 +266,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg |= GML_SPDIF_CLOCK; /* Layla24 doesn't support 96KHz S/PDIF */ control_reg &= ~GML_DOUBLE_SPEED_MODE; - dev_dbg(chip->card->dev, "Set Layla24 clock to SPDIF\n"); break; case ECHO_CLOCK_WORD: control_reg |= GML_WORD_CLOCK; @@ -279,14 +273,12 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg |= GML_DOUBLE_SPEED_MODE; else control_reg &= ~GML_DOUBLE_SPEED_MODE; - dev_dbg(chip->card->dev, "Set Layla24 clock to WORD\n"); break; case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; control_reg |= GML_ADAT_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; - dev_dbg(chip->card->dev, "Set Layla24 clock to ADAT\n"); break; default: dev_err(chip->card->dev, diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index ed2f21d..fdad079 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Mia\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != MIA)) return -ENODEV; @@ -67,7 +66,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c index 8d43c5a..d913749 100644 --- a/sound/pci/echoaudio/midi.c +++ b/sound/pci/echoaudio/midi.c @@ -157,7 +157,6 @@ static int snd_echo_midi_input_open(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_in = substream; - dev_dbg(chip->card->dev, "rawmidi_iopen\n"); return 0; } @@ -183,7 +182,6 @@ static int snd_echo_midi_input_close(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_in = NULL; - dev_dbg(chip->card->dev, "rawmidi_iclose\n"); return 0; } @@ -196,7 +194,6 @@ static int snd_echo_midi_output_open(struct snd_rawmidi_substream *substream) chip->tinuse = 0; chip->midi_full = 0; chip->midi_out = substream; - dev_dbg(chip->card->dev, "rawmidi_open\n"); return 0; } @@ -209,7 +206,6 @@ static void snd_echo_midi_output_write(unsigned long data) int bytes, sent, time; unsigned char buf[MIDI_OUT_BUFFER_SIZE - 1]; - dev_dbg(chip->card->dev, "snd_echo_midi_output_write\n"); /* No interrupts are involved: we have to check at regular intervals if the card's output buffer has room for new data. */ sent = bytes = 0; @@ -288,7 +284,6 @@ static int snd_echo_midi_output_close(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_out = NULL; - dev_dbg(chip->card->dev, "rawmidi_oclose\n"); return 0; } @@ -328,6 +323,5 @@ static int snd_echo_midi_create(struct snd_card *card, chip->rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX; - dev_dbg(chip->card->dev, "MIDI ok\n"); return 0; } diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index cc46a8c..843c7a5 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Mona\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != MONA)) return -ENODEV; @@ -72,7 +71,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -302,7 +300,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) u32 control_reg, clocks_from_dsp; int err; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Prevent two simultaneous calls to switch_asic() */ if (atomic_read(&chip->opencount)) @@ -315,7 +312,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Mona clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: @@ -327,7 +323,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) spin_lock_irq(&chip->lock); if (err < 0) return err; - dev_dbg(chip->card->dev, "Set Mona clock to SPDIF\n"); control_reg |= GML_SPDIF_CLOCK; if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96) control_reg |= GML_DOUBLE_SPEED_MODE; @@ -335,7 +330,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_WORD: - dev_dbg(chip->card->dev, "Set Mona clock to WORD\n"); spin_unlock_irq(&chip->lock); err = switch_asic(chip, clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD96); -- 1.8.1.2

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[alsa-devel] [PATCH 1/6] ASoC: sn95031: Use table based control setup
by Lars-Peter Clausen 05 Nov '14

05 Nov '14

Makes the code a bit cleaner. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sn95031.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index cf8fa40..6167c59 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -867,9 +867,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_SSR2, 0x10); snd_soc_write(codec, SN95031_SSR3, 0x40); - snd_soc_add_codec_controls(codec, sn95031_snd_controls, - ARRAY_SIZE(sn95031_snd_controls)); - return 0; } @@ -886,6 +883,9 @@ static struct snd_soc_codec_driver sn95031_codec = { .remove = sn95031_codec_remove, .set_bias_level = sn95031_set_vaud_bias, .idle_bias_off = true, + + .controls = sn95031_snd_controls, + .num_controls = ARRAY_SIZE(sn95031_snd_controls), .dapm_widgets = sn95031_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(sn95031_dapm_widgets), .dapm_routes = sn95031_audio_map, -- 1.8.0

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix device_del() sysfs warnings at disconnect
by Takashi Iwai 05 Nov '14

05 Nov '14

Some USB-audio devices show weird sysfs warnings at disconnecting the devices, e.g. usb 1-3: USB disconnect, device number 3 ------------[ cut here ]------------ WARNING: CPU: 0 PID: 973 at fs/sysfs/group.c:216 device_del+0x39/0x180() sysfs group ffffffff8183df40 not found for kobject 'midiC1D0' Call Trace: [<ffffffff814a3e38>] ? dump_stack+0x49/0x71 [<ffffffff8103cb72>] ? warn_slowpath_common+0x82/0xb0 [<ffffffff8103cc55>] ? warn_slowpath_fmt+0x45/0x50 [<ffffffff813521e9>] ? device_del+0x39/0x180 [<ffffffff81352339>] ? device_unregister+0x9/0x20 [<ffffffff81352384>] ? device_destroy+0x34/0x40 [<ffffffffa00ba29f>] ? snd_unregister_device+0x7f/0xd0 [snd] [<ffffffffa025124e>] ? snd_rawmidi_dev_disconnect+0xce/0x100 [snd_rawmidi] [<ffffffffa00c0192>] ? snd_device_disconnect+0x62/0x90 [snd] [<ffffffffa00c025c>] ? snd_device_disconnect_all+0x3c/0x60 [snd] [<ffffffffa00bb574>] ? snd_card_disconnect+0x124/0x1a0 [snd] [<ffffffffa02e54e8>] ? usb_audio_disconnect+0x88/0x1c0 [snd_usb_audio] [<ffffffffa015260e>] ? usb_unbind_interface+0x5e/0x1b0 [usbcore] [<ffffffff813553e9>] ? __device_release_driver+0x79/0xf0 [<ffffffff81355485>] ? device_release_driver+0x25/0x40 [<ffffffff81354e11>] ? bus_remove_device+0xf1/0x130 [<ffffffff813522b9>] ? device_del+0x109/0x180 [<ffffffffa01501d5>] ? usb_disable_device+0x95/0x1f0 [usbcore] [<ffffffffa014634f>] ? usb_disconnect+0x8f/0x190 [usbcore] [<ffffffffa0149179>] ? hub_thread+0x539/0x13a0 [usbcore] [<ffffffff810669f5>] ? sched_clock_local+0x15/0x80 [<ffffffff81066c98>] ? sched_clock_cpu+0xb8/0xd0 [<ffffffff81070730>] ? bit_waitqueue+0xb0/0xb0 [<ffffffffa0148c40>] ? usb_port_resume+0x430/0x430 [usbcore] [<ffffffffa0148c40>] ? usb_port_resume+0x430/0x430 [usbcore] [<ffffffff8105973e>] ? kthread+0xce/0xf0 [<ffffffff81059670>] ? kthread_create_on_node+0x1c0/0x1c0 [<ffffffff814a8b7c>] ? ret_from_fork+0x7c/0xb0 [<ffffffff81059670>] ? kthread_create_on_node+0x1c0/0x1c0 ---[ end trace 40b1928d1136b91e ]--- This comes from the fact that usb-audio driver may receive the disconnect callback multiple times, per each usb interface. When a device has both audio and midi interfaces, it gets called twice, and currently the driver tries to release resources at the last call. At this point, the first parent interface has been already deleted, thus deleting a child of the first parent hits such a warning. For fixing this problem, we need to call snd_card_disconnect() and cancel pending operations at the very first disconnect while the release of the whole objects waits until the last disconnect call. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=80931 Reported-and-tested-by: Tomas Gayoso <tgayoso(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 7ecd0e8a5c51..f61ebb17cc64 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -591,18 +591,19 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, { struct snd_card *card; struct list_head *p; + bool was_shutdown; if (chip == (void *)-1L) return; card = chip->card; down_write(&chip->shutdown_rwsem); + was_shutdown = chip->shutdown; chip->shutdown = 1; up_write(&chip->shutdown_rwsem); mutex_lock(&register_mutex); - chip->num_interfaces--; - if (chip->num_interfaces <= 0) { + if (!was_shutdown) { struct snd_usb_endpoint *ep; snd_card_disconnect(card); @@ -622,6 +623,10 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, list_for_each(p, &chip->mixer_list) { snd_usb_mixer_disconnect(p); } + } + + chip->num_interfaces--; + if (chip->num_interfaces <= 0) { usb_chip[chip->index] = NULL; mutex_unlock(&register_mutex); snd_card_free_when_closed(card); -- 2.1.3

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[alsa-devel] [RFC] AXD Audio Processing IP ALSA support - Questions
by Qais Yousef 05 Nov '14

05 Nov '14

Hi, I have several questions on the best way to add AXD support in ALSA. The discussion of the previous patch can be found here: https://lkml.org/lkml/2014/10/28/465 Questions: 1- What is the best example to follow to add a simple mp3 support for AXD? The only one I find is in sst-mfld-platform-compress.c in sound/soc/intel directory but it's a bit confusing. I think because it's sharing code with several other sst drivers/platforms. I find the documentation for compress_offload generally lacking. Is there a plan to improve on that? Being a new comer to ALSA framework api, I'm confused what is the correct way to do things :-/ So far I know I need to call snd_soc_register_card(). I thought snd_compress_register() (from compress_driver.h) is how you add compressed nodes to the card but apparently not. It looks like I need to define a compress_dai? Hmmm. 2- Is tinycompress the only userland support for compress_offload? Is there anyone working on gstreamer and omx plugins to support that? Would tinycompress be part of alsa-utils and alsa-libs in the future? I know it needs more work at the moment but it'd be nice if compress_offload support is part of the standard alsa-utils and alsa-libs. 3- Can we get an example of how transcoding (back to disk) is supposed to be working? 4- How can we reconfigure complex audio effect components (like shelving filters) which need filter co-effecient changes to be applied all at once atomically to avoid instability? My plan is to add mp3 driver first then build on that to add the rest of the features. Hopefully I can minimise the amount of work that needs doing (due to me doing it wrongly) and get features upstream sooner. Thanks, Qais

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