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

[alsa-devel] [RESEND PATCH] conf/cards: add VC4-HDMI card
by Boris Brezillon 10 Apr '17

10 Apr '17

Add a conf file for the VC4-HDMI sound card. Signed-off-by: Boris Brezillon <boris.brezillon(a)free-electrons.com> --- Hi, Sorry for the noise, but I didn't send this patch to the alsa-lib maintainer on my first attempt. This patch is adding a card config file for the audio sound card whose driver has been submitted here [1] (not accepted yet). Since I am a total newbie to the alsa world, I'd like to get some feedback on this patch. Also, the card only supports 2 to 8 channels, and I wonder if we should add a plug element to support mono streams, and where this element should be added (after or before the iec958 element). Regards, Boris [1]https://www.spinics.net/lists/arm-kernel/msg565182.html src/conf/cards/Makefile.am | 1 + src/conf/cards/aliases.conf | 1 + src/conf/cards/vc4-hdmi.conf | 64 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 66 insertions(+) create mode 100644 src/conf/cards/vc4-hdmi.conf diff --git a/src/conf/cards/Makefile.am b/src/conf/cards/Makefile.am index e8b530e80a8f..00999f0186d6 100644 --- a/src/conf/cards/Makefile.am +++ b/src/conf/cards/Makefile.am @@ -51,6 +51,7 @@ cfg_files = aliases.conf \ TRID4DWAVENX.conf \ USB-Audio.conf \ YMF744.conf \ + vc4-hdmi.conf \ VIA686A.conf \ VIA8233.conf \ VIA8233A.conf \ diff --git a/src/conf/cards/aliases.conf b/src/conf/cards/aliases.conf index 60f9d26974fd..18a920f41e46 100644 --- a/src/conf/cards/aliases.conf +++ b/src/conf/cards/aliases.conf @@ -56,6 +56,7 @@ AV200 cards.CMI8788 CMI8786 cards.CMI8788 CMI8787 cards.CMI8788 pistachio cards.pistachio-card +VC4-HDMI cards.vc4-hdmi <confdir:pcm/default.conf> <confdir:pcm/dmix.conf> diff --git a/src/conf/cards/vc4-hdmi.conf b/src/conf/cards/vc4-hdmi.conf new file mode 100644 index 000000000000..027804a145d1 --- /dev/null +++ b/src/conf/cards/vc4-hdmi.conf @@ -0,0 +1,64 @@ +# +# Configuration for the VC4-HDMI sound card using software IEC958 +# subframe conversion +# + +<confdir:pcm/front.conf> + +vc4-hdmi.pcm.front.0 { + @args [ CARD ] + @args.CARD { + type string + } + type hw + card $CARD +} + +# default with dmix +vc4-hdmi.pcm.default { + @args [ CARD ] + @args.CARD { + type string + } + type asym + playback.pcm { + type plug + slave.pcm { + @func concat + strings [ "dmix:" $CARD ] + } + } +} + +<confdir:pcm/iec958.conf> + +vc4-hdmi.pcm.iec958.0 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { + type string + } + @args.AES0 { + type integer + } + @args.AES1 { + type integer + } + @args.AES2 { + type integer + } + @args.AES3 { + type integer + } + type iec958 + slave { + format IEC958_SUBFRAME_LE + pcm { + type plug + slave.pcm { + type hw + card $CARD + } + } + } + status [ $AES0 $AES1 $AES2 $AES3 ] +} -- 2.7.4

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[alsa-devel] [PATCH] ALSA: pcm: dmix: Disable var_periodsize as default
by Takashi Iwai 10 Apr '17

10 Apr '17

The recently added variable period-size feature for dmix & co seems causing a regression on some old applications; e.g. Audacious aborts with an error: ALSA error: snd_pcm_hw_params_set_buffer_time_near failed: Invalid argument. As a quick fix, disable the feature as default. One can still enable it via asoundrc. Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=1033179 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 8a75c421dd7b..89468d3537d0 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -1875,7 +1875,7 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, rec->ipc_gid = -1; rec->slowptr = 1; rec->max_periods = 0; - rec->var_periodsize = 1; + rec->var_periodsize = 0; rec->direct_memory_access = 1; /* read defaults */ -- 2.11.1

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[alsa-devel] [PATCH] [alsa-lib] Removed compilation warning from test/namehint.c
by soodvarun78＠gmail.com 10 Apr '17

10 Apr '17

From: varun <soodvarun78(a)gmail.com> Changed hint variable from char ** to void ** to match function snd_device_name_hint Signed-off-by: varun <soodvarun78(a)gmail.com> --- test/namehint.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/test/namehint.c b/test/namehint.c index e978d5c..18bad1d 100644 --- a/test/namehint.c +++ b/test/namehint.c @@ -4,7 +4,8 @@ int main(int argc, char *argv[]) { const char *iface = "pcm"; - char **hints, **n; + void **hints; + char **n; int err; if (argc > 1) @@ -12,7 +13,7 @@ int main(int argc, char *argv[]) err = snd_device_name_hint(-1, iface, &hints); if (err < 0) errx(1, "snd_device_name_hint error: %s", snd_strerror(err)); - n = hints; + n = (char **)hints; while (*n != NULL) { printf("%s\n", *n); n++; -- 1.9.1

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[alsa-devel] [PATCH v2] aplay: Introduce and use xwrite helper
by Daniel Baluta 10 Apr '17

10 Apr '17

Write can return less then requested bytes, but we treat this as an error thus ending up with confusing error messages. Fix this by introducing xwrite helper, which makes sure all bytes are written or an error is returned. With this patch an usecase where disk is filled by recording will print: $ /mnt/msc/audio.wav: No space left on device instead of random messages like: $/mnt/msc/audio.wav: No such file or directory Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- Changes since v1: * return the correct number of bytes aplay/aplay.c | 55 +++++++++++++++++++++++++++++++++++++------------------ 1 file changed, 37 insertions(+), 18 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index ee480f2..f793c82 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -429,6 +429,25 @@ enum { OPT_FATAL_ERRORS, }; +/* + * make sure we write all bytes or return an error + */ +static ssize_t xwrite(int fd, const void *buf, size_t count) +{ + ssize_t written; + size_t offset = 0; + + while (offset < count) { + written = write(fd, buf + offset, count - offset); + if (written <= 0) + return written; + + offset += written; + }; + + return offset; +} + static long parse_long(const char *str, int *err) { long val; @@ -2428,7 +2447,7 @@ static void voc_play(int fd, int ofs, char *name) l = nextblock; if (l) { if (output && !quiet_mode) { - if (write(2, data, l) != l) { /* to stderr */ + if (xwrite(2, data, l) != l) { /* to stderr */ error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2480,7 +2499,7 @@ static void begin_voc(int fd, size_t cnt) vh.version = LE_SHORT(VOC_ACTUAL_VERSION); vh.coded_ver = LE_SHORT(0x1233 - VOC_ACTUAL_VERSION); - if (write(fd, &vh, sizeof(VocHeader)) != sizeof(VocHeader)) { + if (xwrite(fd, &vh, sizeof(VocHeader)) != sizeof(VocHeader)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2489,14 +2508,14 @@ static void begin_voc(int fd, size_t cnt) bt.type = 8; bt.datalen = 4; bt.datalen_m = bt.datalen_h = 0; - if (write(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { + if (xwrite(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { error(_("write error")); prg_exit(EXIT_FAILURE); } eb.tc = LE_SHORT(65536 - 256000000L / (hwparams.rate << 1)); eb.pack = 0; eb.mode = 1; - if (write(fd, &eb, sizeof(VocExtBlock)) != sizeof(VocExtBlock)) { + if (xwrite(fd, &eb, sizeof(VocExtBlock)) != sizeof(VocExtBlock)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2506,13 +2525,13 @@ static void begin_voc(int fd, size_t cnt) bt.datalen = (u_char) (cnt & 0xFF); bt.datalen_m = (u_char) ((cnt & 0xFF00) >> 8); bt.datalen_h = (u_char) ((cnt & 0xFF0000) >> 16); - if (write(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { + if (xwrite(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { error(_("write error")); prg_exit(EXIT_FAILURE); } vd.tc = (u_char) (256 - (1000000 / hwparams.rate)); vd.pack = 0; - if (write(fd, &vd, sizeof(VocVoiceData)) != sizeof(VocVoiceData)) { + if (xwrite(fd, &vd, sizeof(VocVoiceData)) != sizeof(VocVoiceData)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2581,10 +2600,10 @@ static void begin_wave(int fd, size_t cnt) cd.type = WAV_DATA; cd.length = LE_INT(cnt); - if (write(fd, &h, sizeof(WaveHeader)) != sizeof(WaveHeader) || - write(fd, &cf, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader) || - write(fd, &f, sizeof(WaveFmtBody)) != sizeof(WaveFmtBody) || - write(fd, &cd, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader)) { + if (xwrite(fd, &h, sizeof(WaveHeader)) != sizeof(WaveHeader) || + xwrite(fd, &cf, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader) || + xwrite(fd, &f, sizeof(WaveFmtBody)) != sizeof(WaveFmtBody) || + xwrite(fd, &cd, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2614,7 +2633,7 @@ static void begin_au(int fd, size_t cnt) } ah.sample_rate = BE_INT(hwparams.rate); ah.channels = BE_INT(hwparams.channels); - if (write(fd, &ah, sizeof(AuHeader)) != sizeof(AuHeader)) { + if (xwrite(fd, &ah, sizeof(AuHeader)) != sizeof(AuHeader)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2628,7 +2647,7 @@ static void end_voc(int fd) size_t cnt; char dummy = 0; /* Write a Terminator */ - if (write(fd, &dummy, 1) != 1) { + if (xwrite(fd, &dummy, 1) != 1) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2644,7 +2663,7 @@ static void end_voc(int fd) bt.datalen_m = (u_char) ((cnt & 0xFF00) >> 8); bt.datalen_h = (u_char) ((cnt & 0xFF0000) >> 16); if (lseek64(fd, length_seek, SEEK_SET) == length_seek) - write(fd, &bt, sizeof(VocBlockType)); + xwrite(fd, &bt, sizeof(VocBlockType)); if (fd != 1) close(fd); } @@ -2664,9 +2683,9 @@ static void end_wave(int fd) filelen = fdcount + 2*sizeof(WaveChunkHeader) + sizeof(WaveFmtBody) + 4; rifflen = filelen > 0x7fffffff ? LE_INT(0x7fffffff) : LE_INT(filelen); if (lseek64(fd, 4, SEEK_SET) == 4) - write(fd, &rifflen, 4); + xwrite(fd, &rifflen, 4); if (lseek64(fd, length_seek, SEEK_SET) == length_seek) - write(fd, &cd, sizeof(WaveChunkHeader)); + xwrite(fd, &cd, sizeof(WaveChunkHeader)); if (fd != 1) close(fd); } @@ -2679,7 +2698,7 @@ static void end_au(int fd) length_seek = (char *)&ah.data_size - (char *)&ah; ah.data_size = fdcount > 0xffffffff ? 0xffffffff : BE_INT(fdcount); if (lseek64(fd, length_seek, SEEK_SET) == length_seek) - write(fd, &ah.data_size, sizeof(ah.data_size)); + xwrite(fd, &ah.data_size, sizeof(ah.data_size)); if (fd != 1) close(fd); } @@ -3078,7 +3097,7 @@ static void capture(char *orig_name) in_aborting = 1; break; } - if (write(fd, audiobuf, c) != c) { + if (xwrite(fd, audiobuf, c) != c) { perror(name); in_aborting = 1; break; @@ -3187,7 +3206,7 @@ static void capturev_go(int* fds, unsigned int channels, off64_t count, int rtyp break; rv = r * bits_per_sample / 8; for (channel = 0; channel < channels; ++channel) { - if ((size_t)write(fds[channel], bufs[channel], rv) != rv) { + if ((size_t)xwrite(fds[channel], bufs[channel], rv) != rv) { perror(names[channel]); prg_exit(EXIT_FAILURE); } -- 2.7.4

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Re: [alsa-devel] DSD native format on SoC
by Michael Trimarchi 09 Apr '17

09 Apr '17

Hi all On Mon, Mar 13, 2017 at 05:42:43PM +0900, Naoki Matsumoto wrote: > Retry send because alsa-dev ML doesn't delivery yet. > > -------- Original Message -------- > From: Naoki Matsumoto > Sent: Monday, Mar 13, 2017 10:39 AM GMT+0900 > To: michael(a)amarulasolutions.com > Cc: alsa-devel(a)alsa-project.org > Subject: DSD native format on SoC > > Hello > > I know only xmos-native-dsd(DSD_U32_BE). > I'll share what I know. > I know information about DSD only fragmentally... > Problem here is SoC subsytem for me https://github.com/zonque/alsa-dsd-player.git Cannot set sample format tyring U16_LE DSD. I have changed a bit the core and some components but seems that does not go in. Any idea? diff --git a/sound/soc/codecs/pcm179x.c b/sound/soc/codecs/pcm179x.c index cc34b16..0dea529 100644 --- a/sound/soc/codecs/pcm179x.c +++ b/sound/soc/codecs/pcm179x.c @@ -89,18 +89,19 @@ static int pcm179x_startup(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct pcm179x_private *priv = snd_soc_codec_get_drvdata(codec); - u64 formats = PCM1792A_FORMATS; + u64 formats = PCM1795_FORMATS; switch (priv->codec_model) { - case PCM1795: - formats = PCM1795_FORMATS; + case PCM1792A: + formats = PCM1792A_FORMATS; break; default: break; } - snd_pcm_hw_constraint_mask64(substream->runtime, - SNDRV_PCM_HW_PARAM_FORMAT, formats); + if (formats != PCM1795_FORMATS) + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, formats); msleep(50); return 0; @@ -227,7 +228,7 @@ static struct snd_soc_dai_driver pcm179x_dai = { .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 10000, .rate_max = 200000, - .formats = PCM179X_FORMATS, }, + .formats = PCM1795_FORMATS, }, .ops = &pcm179x_dai_ops, }; @@ -252,9 +253,9 @@ static struct snd_soc_codec_driver soc_codec_dev_pcm179x = { }; const struct of_device_id pcm179x_of_match[] = { - { .compatible = "ti,pcm1792a", }, - { .compatible = "ti,pcm1795", .data = (void *)PCM1795, }, - { .compatible = "ti,pcm1796", }, + { .compatible = "ti,pcm1792a", .data = (void *)PCM1792A }, + { .compatible = "ti,pcm1795", }, + { .compatible = "ti,pcm1796", .data = (void *)PCM1792A }, { } }; MODULE_DEVICE_TABLE(of, pcm179x_of_match); diff --git a/sound/soc/codecs/pcm179x.h b/sound/soc/codecs/pcm179x.h index 4c00047..0665ec8 100644 --- a/sound/soc/codecs/pcm179x.h +++ b/sound/soc/codecs/pcm179x.h @@ -22,7 +22,8 @@ #define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) -#define PCM1795_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define PCM1795_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_DSD_U16_LE) extern const struct regmap_config pcm179x_regmap_config; extern const struct of_device_id pcm179x_of_match[]; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index c55cbfa..d3afa89 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -55,7 +55,7 @@ config SND_SOC_ROCHCHIP_DACMAX tristate "SoC Audio support for DACMAX boards using a pcm1792 codec" depends on SND_SOC_ROCKCHIP && SPI && GPIOLIB select SND_SOC_ROCKCHIP_I2S - select SND_SOC_PCM179X + select SND_SOC_PCM179X_SPI help Say Y or M here if you want to add support for SoC audio on Dacmax boards using the pcm1792a codec. diff --git a/sound/soc/rockchip/dacmax.c b/sound/soc/rockchip/dacmax.c index fb302a8..fa79549 100644 --- a/sound/soc/rockchip/dacmax.c +++ b/sound/soc/rockchip/dacmax.c @@ -22,6 +22,7 @@ * */ +#define DEBUG #include <linux/module.h> #include <linux/of.h> #include <linux/of_gpio.h> @@ -39,6 +40,7 @@ #define CLK1 (1 << 1) #define CLK0 (1 << 2) #define W32 (1 << 3) +#define DSD_EN (1 << 4) #define DAI_NAME_SIZE 32 @@ -52,6 +54,7 @@ struct dacmax_data { int clk_1; int clk_2; int w32; + int dsd_enable; }; static const struct snd_soc_dapm_widget dacmax_dapm_widgets[] = { @@ -78,6 +81,10 @@ static void dacmax_change_freq(struct dacmax_data *data, u8 mask) gpio_set_value(data->w32, value); pr_debug("%s: BITSXWORD(%d)\n", __func__, value); + value = (mask & DSD_EN) ? 1 : 0; + gpio_set_value(data->dsd_enable, value); + pr_debug("%s: DSD ENABLE (%d)\n", __func__, value); + mdelay(20); } @@ -91,6 +98,7 @@ static int dacmax_ext_clock_update(struct dacmax_data *data, params_format(params)); switch (params_format(params)) { + case SNDRV_PCM_FORMAT_DSD_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: break; case SNDRV_PCM_FORMAT_S24_LE: @@ -103,6 +111,8 @@ static int dacmax_ext_clock_update(struct dacmax_data *data, } switch (params_rate(params)) { + case 2822400: + mask |= DSD_EN; case 44100: break; case 48000: @@ -251,6 +261,7 @@ static int dacmax_probe(struct platform_device *pdev) data->clk_1 = clk_1; data->clk_2 = clk_2; data->w32 = w32; + data->dsd_enable = dsd_enable; ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 68fea0a..67fd6ec 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -302,6 +302,7 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S8: val |= I2S_TXCR_VDW(8); break; + case SNDRV_PCM_FORMAT_DSD_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: val |= I2S_TXCR_VDW(16); break; @@ -457,7 +458,8 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = { SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE), + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_DSD_U16_LE), }, .capture = { .stream_name = "Capture", diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a1305f8..1ae0aea 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2980,6 +2980,7 @@ static u64 codec_format_map[] = { SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_U18_3BE, SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE, SNDRV_PCM_FMTBIT_FLOAT64_LE | SNDRV_PCM_FMTBIT_FLOAT64_BE, + SNDRV_PCM_FORMAT_DSD_U16_LE | SNDRV_PCM_FORMAT_DSD_U16_BE, SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE | SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, }; > on Wed, 8 Mar 2017 08:21:23 +0100, Michael Nazzareno Trimarchi wrote: > DSD is a continuous stream of bits that flows over two separate > channels, left and right, synchronized by a clock, while I2S is a > single data wire and an additional wire that states if the incoming > sample refers to the left channel or the right channel. There is no > way to get DSD data other than a circuit which decode DSD streams > coming directly from a conventional source, like, for examples, a > SACD. > > The Volta dac has an internal decoding circuitry that still employs > the I2S standard as a usual 44100Hz/16 bits, but sourcing the shift > clock at twice the DSD standard frequency of 2.8224MHz, since it has > to split two the16 bits words received on a single wire into the two > DSD channels, as the standard wants. > > When DSD is enabled, the control signal PCD/DSD must be high, CK3..CK0 > must be all low. The I2S interface works as a standard 44100Hz/16 bit > and the DSD streaming must be packed into 16 bits lenght words > left/right as per LRCK logic. The BCLK frequency supplied from the > interface is 5644800Hz in case of DSD and 11289600 for DSD2. > I don't know your device. > but I've understood that it's device layer implantation topic. > I think that your sound device need to support SND_PCM_FORMAT_DSD_*. > > Refer:enum snd_pcm_format_t > http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html > > If you use xmos-natvie-dsd, the implement is linux.git/sound/usb/quirks.c > > Can you explain how now DSD can be pass to audio card? The idea is to > declare it on the audio card and then configure it as a PCM card but I > don't know how data are suppose to arrive from alsa userspace. > I know DSD playback on sound device that xmos. > Have you checked this page? > https://github.com/lintweaker/xmos-native-dsd > > As alsa-lib > 1. snd_pcm_open > e.g, USB sound device. In your case, soc internal sound device > 2. snd_ocm_hw_params_set_format > e.g, SND_PCM_FORMAT_DSD_U32_BE > 3. snd_pcm_writei > > Other elements > * DSD decoder(e.g,dsf/dsdiff) > * Player > if you use xmos-native-dsd(DSD_U32_BE) device, we can use MPD Ver0.20.2+ > I don't know other DSD pcm_formats. > > Just information. It may be wrong. > Thank you > > > -- > ********************************************** > Naoki MATSUMOTO > Email:n-matsumoto@melcoinc.co.jp > Tel :050-5830-8916 > ********************************************** -- | Michael Nazzareno Trimarchi Amarula Solutions BV | | COO - Founder Cruquiuskade 47 | | +31(0)851119172 Amsterdam 1018 AM NL | | [`as] http://www.amarulasolutions.com |

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[alsa-devel] [alsa-lib] test/namehint.c: Removed compilation warning
by varun sood 09 Apr '17

09 Apr '17

Removed compilation warning Changed hint variable from char ** to void ** to match function snd_device_name_hint Signed-off-by: varun <soodvarun78(a)gmail.com>--- test/namehint.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/test/namehint.c b/test/namehint.cindex e978d5c..18bad1d 100644--- a/test/namehint.c+++ b/test/namehint.c@@ -4,7 +4,8 @@ int main(int argc, char *argv[]) { const char *iface = "pcm";- char **hints, **n;+ void **hints;+ char **n; int err; if (argc > 1)@@ -12,7 +13,7 @@ int main(int argc, char *argv[]) err = snd_device_name_hint(-1, iface, &hints); if (err < 0) errx(1, "snd_device_name_hint error: %s", snd_strerror(err));- n = hints;+ n = (char **)hints; while (*n != NULL) { printf("%s\n", *n); n++;-- 1.9.1

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[alsa-devel] [PATCH] aplay: Introduce and use xwrite helper
by Daniel Baluta 07 Apr '17

07 Apr '17

Write can return less then requested bytes, but we treat this as an error thus ending up with confusing error messages. Fix this by introducing xwrite helper, which makes sure all bytes are written or an error is returned. With this patch an usecase where disk is filled by recording will print: $ /mnt/msc/audio.wav: No space left on device instead of random messages like: $/mnt/msc/audio.wav: No such file or directory Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- This is a follow up of: http://mailman.alsa-project.org/pipermail/alsa-devel/2017-April/119632.html where I tried to fix the error message instead of write :). aplay/aplay.c | 55 +++++++++++++++++++++++++++++++++++++------------------ 1 file changed, 37 insertions(+), 18 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index ee480f2..74d3d37 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -429,6 +429,25 @@ enum { OPT_FATAL_ERRORS, }; +/* + * make sure we write all bytes or return an error + */ +static ssize_t xwrite(int fd, const void *buf, size_t count) +{ + ssize_t written; + size_t offset = 0; + + while (offset < count) { + written = write(fd, buf + offset, count - offset); + if (written <= 0) + return written; + + offset += written; + }; + + return written; +} + static long parse_long(const char *str, int *err) { long val; @@ -2428,7 +2447,7 @@ static void voc_play(int fd, int ofs, char *name) l = nextblock; if (l) { if (output && !quiet_mode) { - if (write(2, data, l) != l) { /* to stderr */ + if (xwrite(2, data, l) != l) { /* to stderr */ error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2480,7 +2499,7 @@ static void begin_voc(int fd, size_t cnt) vh.version = LE_SHORT(VOC_ACTUAL_VERSION); vh.coded_ver = LE_SHORT(0x1233 - VOC_ACTUAL_VERSION); - if (write(fd, &vh, sizeof(VocHeader)) != sizeof(VocHeader)) { + if (xwrite(fd, &vh, sizeof(VocHeader)) != sizeof(VocHeader)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2489,14 +2508,14 @@ static void begin_voc(int fd, size_t cnt) bt.type = 8; bt.datalen = 4; bt.datalen_m = bt.datalen_h = 0; - if (write(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { + if (xwrite(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { error(_("write error")); prg_exit(EXIT_FAILURE); } eb.tc = LE_SHORT(65536 - 256000000L / (hwparams.rate << 1)); eb.pack = 0; eb.mode = 1; - if (write(fd, &eb, sizeof(VocExtBlock)) != sizeof(VocExtBlock)) { + if (xwrite(fd, &eb, sizeof(VocExtBlock)) != sizeof(VocExtBlock)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2506,13 +2525,13 @@ static void begin_voc(int fd, size_t cnt) bt.datalen = (u_char) (cnt & 0xFF); bt.datalen_m = (u_char) ((cnt & 0xFF00) >> 8); bt.datalen_h = (u_char) ((cnt & 0xFF0000) >> 16); - if (write(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { + if (xwrite(fd, &bt, sizeof(VocBlockType)) != sizeof(VocBlockType)) { error(_("write error")); prg_exit(EXIT_FAILURE); } vd.tc = (u_char) (256 - (1000000 / hwparams.rate)); vd.pack = 0; - if (write(fd, &vd, sizeof(VocVoiceData)) != sizeof(VocVoiceData)) { + if (xwrite(fd, &vd, sizeof(VocVoiceData)) != sizeof(VocVoiceData)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2581,10 +2600,10 @@ static void begin_wave(int fd, size_t cnt) cd.type = WAV_DATA; cd.length = LE_INT(cnt); - if (write(fd, &h, sizeof(WaveHeader)) != sizeof(WaveHeader) || - write(fd, &cf, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader) || - write(fd, &f, sizeof(WaveFmtBody)) != sizeof(WaveFmtBody) || - write(fd, &cd, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader)) { + if (xwrite(fd, &h, sizeof(WaveHeader)) != sizeof(WaveHeader) || + xwrite(fd, &cf, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader) || + xwrite(fd, &f, sizeof(WaveFmtBody)) != sizeof(WaveFmtBody) || + xwrite(fd, &cd, sizeof(WaveChunkHeader)) != sizeof(WaveChunkHeader)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2614,7 +2633,7 @@ static void begin_au(int fd, size_t cnt) } ah.sample_rate = BE_INT(hwparams.rate); ah.channels = BE_INT(hwparams.channels); - if (write(fd, &ah, sizeof(AuHeader)) != sizeof(AuHeader)) { + if (xwrite(fd, &ah, sizeof(AuHeader)) != sizeof(AuHeader)) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2628,7 +2647,7 @@ static void end_voc(int fd) size_t cnt; char dummy = 0; /* Write a Terminator */ - if (write(fd, &dummy, 1) != 1) { + if (xwrite(fd, &dummy, 1) != 1) { error(_("write error")); prg_exit(EXIT_FAILURE); } @@ -2644,7 +2663,7 @@ static void end_voc(int fd) bt.datalen_m = (u_char) ((cnt & 0xFF00) >> 8); bt.datalen_h = (u_char) ((cnt & 0xFF0000) >> 16); if (lseek64(fd, length_seek, SEEK_SET) == length_seek) - write(fd, &bt, sizeof(VocBlockType)); + xwrite(fd, &bt, sizeof(VocBlockType)); if (fd != 1) close(fd); } @@ -2664,9 +2683,9 @@ static void end_wave(int fd) filelen = fdcount + 2*sizeof(WaveChunkHeader) + sizeof(WaveFmtBody) + 4; rifflen = filelen > 0x7fffffff ? LE_INT(0x7fffffff) : LE_INT(filelen); if (lseek64(fd, 4, SEEK_SET) == 4) - write(fd, &rifflen, 4); + xwrite(fd, &rifflen, 4); if (lseek64(fd, length_seek, SEEK_SET) == length_seek) - write(fd, &cd, sizeof(WaveChunkHeader)); + xwrite(fd, &cd, sizeof(WaveChunkHeader)); if (fd != 1) close(fd); } @@ -2679,7 +2698,7 @@ static void end_au(int fd) length_seek = (char *)&ah.data_size - (char *)&ah; ah.data_size = fdcount > 0xffffffff ? 0xffffffff : BE_INT(fdcount); if (lseek64(fd, length_seek, SEEK_SET) == length_seek) - write(fd, &ah.data_size, sizeof(ah.data_size)); + xwrite(fd, &ah.data_size, sizeof(ah.data_size)); if (fd != 1) close(fd); } @@ -3078,7 +3097,7 @@ static void capture(char *orig_name) in_aborting = 1; break; } - if (write(fd, audiobuf, c) != c) { + if (xwrite(fd, audiobuf, c) != c) { perror(name); in_aborting = 1; break; @@ -3187,7 +3206,7 @@ static void capturev_go(int* fds, unsigned int channels, off64_t count, int rtyp break; rv = r * bits_per_sample / 8; for (channel = 0; channel < channels; ++channel) { - if ((size_t)write(fds[channel], bufs[channel], rv) != rv) { + if ((size_t)xwrite(fds[channel], bufs[channel], rv) != rv) { perror(names[channel]); prg_exit(EXIT_FAILURE); } -- 2.7.4

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[alsa-devel] [PATCH v2] aplay: Fix error message when writing captured data
by Daniel Baluta 07 Apr '17

07 Apr '17

Read can return less then requested bytes, but we treat this as an error. Anyhow, errno is not updated in this case and we can end up with a confusing error message. For example, when there is no room to write data into the output file we receive: $ arecord -d 2000 -c 2 -r 192000 -f S16_LE -Dplughw:0,0 audio.wav Recording WAVE '/mnt/msc/audio.wav' : Signed 16 bit Little Endian, Rate 192000 Hz, Stereo audio.wav: No such file or directory Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- aplay/aplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index ee480f2..96f5e1b 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -3079,7 +3079,7 @@ static void capture(char *orig_name) break; } if (write(fd, audiobuf, c) != c) { - perror(name); + fprintf(stderr, _("Couldn't write all data to %s\n"), name); in_aborting = 1; break; } -- 2.7.4

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[alsa-devel] [PATCH] aplay: Fix error message when writing captured data
by Daniel Baluta 07 Apr '17

07 Apr '17

Write can return less then requested bytes, but we treat this as an error. Anyhow, errno is not updated in this case and we can end up with a confusing error message. For example, when there is no room to write data into the output file we receive: $ arecord -d 2000 -c 2 -r 192000 -f S16_LE -Dplughw:0,0 audio.wav Recording WAVE '/mnt/msc/audio.wav' : Signed 16 bit Little Endian, Rate 192000 Hz, Stereo audio.wav: No such file or directory Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- aplay/aplay.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index ee480f2..9547d1b 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -3079,7 +3079,7 @@ static void capture(char *orig_name) break; } if (write(fd, audiobuf, c) != c) { - perror(name); + fprintf(stderr, "Couldn't write all data to %s\n", name); in_aborting = 1; break; } -- 2.7.4

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[alsa-devel] [PATCH 35/38] Annotate hardware config module parameters in sound/drivers/
by David Howells 07 Apr '17

07 Apr '17

When the kernel is running in secure boot mode, we lock down the kernel to prevent userspace from modifying the running kernel image. Whilst this includes prohibiting access to things like /dev/mem, it must also prevent access by means of configuring driver modules in such a way as to cause a device to access or modify the kernel image. To this end, annotate module_param* statements that refer to hardware configuration and indicate for future reference what type of parameter they specify. The parameter parser in the core sees this information and can skip such parameters with an error message if the kernel is locked down. The module initialisation then runs as normal, but just sees whatever the default values for those parameters is. Note that we do still need to do the module initialisation because some drivers have viable defaults set in case parameters aren't specified and some drivers support automatic configuration (e.g. PNP or PCI) in addition to manually coded parameters. This patch annotates drivers in sound/drivers/. Suggested-by: Alan Cox <gnomes(a)lxorguk.ukuu.org.uk> Signed-off-by: David Howells <dhowells(a)redhat.com> cc: Jaroslav Kysela <perex(a)perex.cz> cc: Takashi Iwai <tiwai(a)suse.com> cc: alsa-devel(a)alsa-project.org --- sound/drivers/mpu401/mpu401.c | 4 ++-- sound/drivers/mtpav.c | 4 ++-- sound/drivers/serial-u16550.c | 4 ++-- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index fed7e7e2177b..9b86e00d7d95 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -53,9 +53,9 @@ MODULE_PARM_DESC(enable, "Enable MPU-401 device."); module_param_array(pnp, bool, NULL, 0444); MODULE_PARM_DESC(pnp, "PnP detection for MPU-401 device."); #endif -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for MPU-401 device."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for MPU-401 device."); module_param_array(uart_enter, bool, NULL, 0444); MODULE_PARM_DESC(uart_enter, "Issue UART_ENTER command at open."); diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 00b31f92c504..0f6392001e30 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -86,9 +86,9 @@ module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for MotuMTPAV MIDI."); module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for MotuMTPAV MIDI."); -module_param(port, long, 0444); +module_param_hw(port, long, ioport, 0444); MODULE_PARM_DESC(port, "Parallel port # for MotuMTPAV MIDI."); -module_param(irq, int, 0444); +module_param_hw(irq, int, irq, 0444); MODULE_PARM_DESC(irq, "Parallel IRQ # for MotuMTPAV MIDI."); module_param(hwports, int, 0444); MODULE_PARM_DESC(hwports, "Hardware ports # for MotuMTPAV MIDI."); diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 60d51ac4ccfe..88e66ea0306d 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -84,9 +84,9 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for Serial MIDI."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable UART16550A chip."); -module_param_array(port, long, NULL, 0444); +module_param_hw_array(port, long, ioport, NULL, 0444); MODULE_PARM_DESC(port, "Port # for UART16550A chip."); -module_param_array(irq, int, NULL, 0444); +module_param_hw_array(irq, int, irq, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for UART16550A chip."); module_param_array(speed, int, NULL, 0444); MODULE_PARM_DESC(speed, "Speed in bauds.");

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