August 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: seq: Mark expected switch fall-through
by Gustavo A. R. Silva 04 Aug '18

04 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Notice that in this particular case, I replaced the code comment with a proper "fall through" annotation, which is what GCC is expecting to find. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/core/seq/seq_midi_emul.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index f9f2133..c1975dd 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -318,7 +318,7 @@ do_control(struct snd_midi_op *ops, void *drv, struct snd_midi_channel_set *chse break; case MIDI_CTL_MSB_DATA_ENTRY: chan->control[MIDI_CTL_LSB_DATA_ENTRY] = 0; - /* go through here */ + /* fall through */ case MIDI_CTL_LSB_DATA_ENTRY: if (chan->param_type == SNDRV_MIDI_PARAM_TYPE_REGISTERED) rpn(ops, drv, chan, chset); -- 2.7.4

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[alsa-devel] [PATCH] ASoC: smd845: fix memory leak
by Gustavo A. R. Silva 03 Aug '18

03 Aug '18

In case memory resources for *card* were allocated, release them before return. Addresses-Coverity-ID: 1472244 ("Resource leak") Fixes: 6b1687bf76ef ("ASoC: qcom: add sdm845 sound card support") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- NOTE: Notice that there are many places in which the return values of functions snd_soc_dai_set_sysclk() and snd_soc_dai_set_fmt() are not being checked. I wonder if this is intentional. Otherwise, I can send a patch for that. Please, confirm. Thanks! sound/soc/qcom/sdm845.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index be0cb11..c1adb77 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -222,8 +222,10 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) /* Allocate the private data */ data = kzalloc(sizeof(*data), GFP_KERNEL); - if (!data) - return -ENOMEM; + if (!data) { + ret = -ENOMEM; + goto data_alloc_fail; + } card->dev = dev; dev_set_drvdata(dev, card); @@ -248,6 +250,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) kfree(card->dai_link); parse_dt_fail: kfree(data); +data_alloc_fail: kfree(card); return ret; } -- 2.7.4

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[alsa-devel] [PATCH 0/7] ASoC: Mark expected switch fall-throughs
by Gustavo A. R. Silva 03 Aug '18

03 Aug '18

Hi all, In preparation to enabling -Wimplicit-fallthrough, this patchset aims to add some annotations in order to mark switch cases where we are expecting to fall through. Thanks Gustavo A. R. Silva (7): ASoC: davinci-i2s: mark expected switch fall-through ASoC: fsl_esai: Mark expected switch fall-through ASoC: Intel: skl-pcm: Mark expected switch fall-through ASoC: omap-dmic: Mark expected switch fall-throughs ASoC: omap-mcpdm: MArk expected switch fall-throughs ASoC: samsung: i2s: Mark expected switch fall-through ASoC: core: mark expected switch fall-through sound/soc/davinci/davinci-i2s.c | 1 + sound/soc/fsl/fsl_esai.c | 1 + sound/soc/intel/skylake/skl-pcm.c | 1 + sound/soc/omap/omap-dmic.c | 2 ++ sound/soc/omap/omap-mcpdm.c | 4 ++++ sound/soc/samsung/i2s.c | 1 + sound/soc/soc-core.c | 1 + 7 files changed, 11 insertions(+) -- 2.7.4

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[alsa-devel] [PATCH 0/6] ALSA: Remove empty init and exit
by Takashi Iwai 03 Aug '18

03 Aug '18

Hi, here is a patchset of trivial cleanups I noticed while working on the recent sequencer bug. There are a bunch of driver codes that have empty module init and exit entries, and we can remove them gracefully. Takashi === Takashi Iwai (6): ALSA: compress: Remove empty init and exit ALSA: drivers: Remove empty init and exit ALSA: isa: Remove empty init and exit ALSA: i2c: Remove empty init and exit ALSA: pci: Remove empty init and exit ALSA: synth: Remove empty init and exit sound/core/compress_offload.c | 12 ------------ sound/drivers/mpu401/mpu401_uart.c | 16 ---------------- sound/drivers/opl3/opl3_lib.c | 16 ---------------- sound/drivers/opl4/opl4_lib.c | 12 ------------ sound/drivers/vx/vx_core.c | 15 --------------- sound/i2c/cs8427.c | 12 ------------ sound/i2c/i2c.c | 13 ------------- sound/i2c/other/ak4xxx-adda.c | 12 ------------ sound/i2c/tea6330t.c | 16 ---------------- sound/isa/es1688/es1688_lib.c | 16 ---------------- sound/isa/gus/gus_main.c | 16 ---------------- sound/isa/sb/sb16_csp.c | 16 ---------------- sound/isa/sb/sb16_main.c | 16 ---------------- sound/isa/sb/sb8_main.c | 16 ---------------- sound/isa/sb/sb_common.c | 16 ---------------- sound/isa/wss/wss_lib.c | 16 ---------------- sound/pci/ac97/ac97_codec.c | 16 ---------------- sound/pci/ice1712/ak4xxx.c | 12 ------------ sound/synth/emux/emux.c | 17 ----------------- sound/synth/util_mem.c | 16 ---------------- 20 files changed, 297 deletions(-) -- 2.18.0

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[alsa-devel] [PATCH] [ALSA] sound/isa/gus: fix spelling mistake "acumulator" -> "accumulator"
by Colin King 03 Aug '18

03 Aug '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial spelling mistake fix in debug message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/gus/gus_io.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/gus/gus_io.c b/sound/isa/gus/gus_io.c index ca79878d8d8c..2fd32ef22c30 100644 --- a/sound/isa/gus/gus_io.c +++ b/sound/isa/gus/gus_io.c @@ -461,7 +461,7 @@ void snd_gf1_print_voice_registers(struct snd_gus_card * gus) printk(KERN_INFO " -%i- GFA1 effect address = 0x%x\n", voice, snd_gf1_i_read_addr(gus, 0x11, ctrl & 4)); printk(KERN_INFO " -%i- GFA1 effect volume = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x16)); printk(KERN_INFO " -%i- GFA1 effect volume final = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x1d)); - printk(KERN_INFO " -%i- GFA1 effect acumulator = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x14)); + printk(KERN_INFO " -%i- GFA1 effect accumulator = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x14)); } if (mode & 0x20) { printk(KERN_INFO " -%i- GFA1 left offset = 0x%x (%i)\n", voice, snd_gf1_i_read16(gus, 0x13), snd_gf1_i_read16(gus, 0x13) >> 4); -- 2.17.1

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[alsa-devel] [PATCH 0/4] ALSA: Mark expected switch fall-throughs
by Gustavo A. R. Silva 03 Aug '18

03 Aug '18

Hi all, In preparation to enabling -Wimplicit-fallthrough, this patchset aims to add some annotations in order to mark switch cases where we are expecting to fall through. Thanks Gustavo A. R. Silva (4): ALSA: galaxy: Mark expected switch fall-through ALSA: opti92x: mark expected switch fall-through ALSA: opti9xx: mark expected switch fall-through ALSA: es18xx: mark expected switch fall-through sound/isa/es18xx.c | 1 + sound/isa/galaxy/galaxy.c | 3 +++ sound/isa/opti9xx/miro.c | 5 +++++ sound/isa/opti9xx/opti92x-ad1848.c | 3 +++ 4 files changed, 12 insertions(+) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: Intel: bxt: Use refcap device for mono recording
by Yong Zhi 03 Aug '18

03 Aug '18

The refcap capture device supports mono recording only, this patch adds the channel constraints, also change stream_name to "Wake on Voice". Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index be6e4b4..097a089 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -319,6 +319,15 @@ static const unsigned int rates_16000[] = { 16000, }; +static const unsigned int ch_mono[] = { + 1, +}; + +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + static const struct snd_pcm_hw_constraint_list constraints_16000 = { .count = ARRAY_SIZE(rates_16000), .list = rates_16000, @@ -326,6 +335,11 @@ static const struct snd_pcm_hw_constraint_list constraints_16000 = { static int broxton_refcap_startup(struct snd_pcm_substream *substream) { + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_16000); @@ -372,7 +386,7 @@ static struct snd_soc_dai_link broxton_dais[] = { [BXT_DPCM_AUDIO_REF_CP] = { .name = "Bxt Audio Reference cap", - .stream_name = "Refcap", + .stream_name = "Wake on Voice", .cpu_dai_name = "Reference Pin", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", -- 2.7.4

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[alsa-devel] ALSA queries
by Chakravarthi Pradeep 03 Aug '18

03 Aug '18

I'm working on ALSA driver for PCIe card. My ALSA driver and it's initializing struct snd_pcm_hardware with below parameter. /************************ code start ************************************************/ static struct snd_pcm_hardware audio_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_RESUME ), .formats = (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S24_LE), .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), .rate_min = 44100, .rate_max = 192000, .channels_min = 2, .channels_max = 8, .buffer_bytes_max = 76800, /*75 kbytes */ .period_bytes_min = 512,//4410, /* (channel (2) * sample_rate (44100) * bit width (24)) / (60 * 8) */ .period_bytes_max = 16*1024, .periods_min = 10, .periods_max = 255, }; /************************ code end ************************************************/ 1) I did not understand what is significance of periods_min , period_bytes_min , period_bytes_max and periods_max. Can you please tell me what is importance of these parameter and what value should be mentioned according into ALSA. 2) snd_pcm_ops trigger callback is getting called in the driver when application sends "start" command. But ALSA driver is stopping by itself after one frame is copied to ALSA framework, without waiting for "stop" command. For instance: In trigger callback , I'm getting these logs after one frame is copied. Trigger:Start (When Play button is selected/clicked in application, Start command is sent to ALSA driver) Dma transfer is completed. Trigger:Stop. (When Stop button is selected/clicked in application, Stop command is sent to ALSA driver. But stop button is not clicked in this case) Thanks in advance. -- Thanks and Regards Chakravarthi Pradeep.K Ph: 91 9980434900

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[alsa-devel] bug report. Envy24 based card (terratec ews88d) won't slave to external clock via s/pdif.
by Tobias Eklund 03 Aug '18

03 Aug '18

Hi, when trying to clock my terratec ews88d from an apogee rosetta AD it just won't. The s/pdif input works but the card won't lock to the s/pdif clock but rather stay at whatever setting it was at before i change to s/pdif in envy24control or mudita24. Since I have another converter slaved via ADAT out I can tell what sample rate is being output via ADAT sync. Works perfectly for setting the sample rate in envy24control but not when i choose external clock and change the rate on the Apogee AD. Also, audio recorded through the s/pdif input is obviously at the wrong rate when changing the rate on the Apogee AD. I found this old discussion which seems to describe the same problem: http://alsa-devel.alsa-project.narkive.com/30mvygRD/m-audio-delta-1010lt-s-… Also, there is a problem with the alsa-info.sh script, it didn't give me url to share. Just the message Your ALSA information is located at Please inform the person helping you. If the alsa-info.sh output is required i'll post it in an e-mail. kind regards, -- Tobias Eklund Fatburs Brunnsgata 31 118 28 Stockholm 0046 707570196

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix wrong sizeof argument
by Gustavo A. R. Silva 02 Aug '18

02 Aug '18

The correct argument to be passed to kzalloc is sizeof(*pd) Addresses-Coverity-ID: 1472224 ("Wrong sizeof argument") Fixes: 7edf3b5e6a45 ("ALSA: usb-audio: AudioStreaming Power Domain parsing") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/usb/stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 8fe3b0e..67cf849 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1037,7 +1037,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, fp->rate_max = UAC3_BADD_SAMPLING_RATE; fp->rates = SNDRV_PCM_RATE_CONTINUOUS; - pd = kzalloc(sizeof(pd), GFP_KERNEL); + pd = kzalloc(sizeof(*pd), GFP_KERNEL); if (!pd) { kfree(fp->rate_table); kfree(fp); -- 2.7.4

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