June 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 2/6] ASoC: fsl_ssi: AC'97 DAI driver needs probe method too
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

AC'97 DAI driver struct need the same probe method as I2S one to setup DMA params in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9c46c7d..2ce9e1d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1101,6 +1101,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2,

2 1

[alsa-devel] [PATCH 3/6] ASoC: fsl_ssi: enable AC'97 asymmetric rates
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

AC'97 bus can support asymmetric playback/capture rates so enable them in this case in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2ce9e1d..f3034b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1377,7 +1377,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; }

3 3

[alsa-devel] [PATCH 4/6] ASoC: fsl_ssi: add AC'97 ops setting check and cleanup
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Check whether setting AC'97 ops succeeded and clean them on removal so the fsl_ssi driver can be reloaded. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f3034b9..0b4fcd9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1340,7 +1340,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1480,6 +1484,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; }

2 1

[alsa-devel] [PATCH 5/6] ASoC: fsl_ssi: instantiate AC'97 CODEC
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Instantiate AC'97 CODEC in fsl_ssi driver AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0b4fcd9..e79dc16 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1460,6 +1460,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card:

2 1

[alsa-devel] RME MADIFX mainline inclusion
by Adrian Knoth 10 Aug '15

10 Aug '15

Hi Takashi! Back in the days (late 2012 I guess), I wrote the MADIFX driver based on the existing RME hdspm.c. I never had physical access to this 1399 Euro audio interface, only the OSX driver and a shell login to one of IOSONO's machines which temporarily featured such a card. I know there are other users like University of Washington (DXARTS) and probably some more (the usual WFS suspects). Long story short: I don't have such a card and since I no longer work in Linux audio, I guess I won't have access in the foreseeable future. That said, I think the driver should be part of the mainline kernel, so it gets all the love the other drivers get (static code analysis, API changes etc.) I think I've ported everything you guys have done to hdspm: I've read the commitlog for hdspm and applied those changes to the madifx driver. Code is here: https://github.com/adiknoth/madifx/commits/tokernel Status: * PCM playback/capture working (SS and DS tested, QS untested) * MIDI working * All card settings working (e.g. TX64, SMUX, AESpro, WC-Term, WC-singlespeed...) * Slave Mode/External clock selection working * Mirror-MADI1-to-Out2+3 maybe working (untested) * Redundancy mode maybe working (untested) * ioctls implemented (see ioctl.c) * Static mixer working (fixed 1:1 mapping) * DSP NOT working. RME doesn't intend to release any information regarding the DSP. * Adjustable mixer NOT working (needs new userspace tools) * Levelmetering NOT working (maybe wrong, needs new userspace tools) I have no idea who could write the new mixer code. Apparently, demand is low, maybe zero. checkpatch reports four warnings of which one is a false-positive, the other one doesn't make sense and only two are real but somewhat ugly to fix (line over 80 with really long array indices). In its current state, can I send you three patches (Kconfig, madifx.h, madifx.c) that add said driver to sound/pci/rme9652? Cheers

6 16

[alsa-devel] DP MST audio support
by Dave Airlie 10 Aug '15

10 Aug '15

Hi, So I have a branch that makes no sound, http://cgit.freedesktop.org/~airlied/linux/log/?h=dp-mst-audio and I'm not sure where I need to turn to next, The Intel docs I've read are kinda vague, assuming you know lots of things I clearly don't. so in theory my branch, sets up the SDP stream to the monitor in the payload creation, enables the codec in the intel GPU driver, and passes the ELD to the audio driver. The audio driver uses the device list to get the presence/valid bits per device, and manages to retrieve the ELD. I even create ELD files in /proc/asound/HDMI/ that have sensible values in them So it looks like I'm just missing some routing somewhere, most likely in the audio driver, then again I could be missing a lot more than that. Just looking for any ideas or knowledge people may have locked in their brains or inside their firewalls. Dave.

4 6

[alsa-devel] Possible GPL-licensed header included in alsa-lib
by Clemens Lang 27 Jul '15

27 Jul '15

Hi, I recently reviewed the alsa-lib 1.0.27.2 source code from a licensing perspective to ensure compliant usage and distribution. It seems the intention is for the libraries to be covered by LGPL. I had a look at include/list.h, which seems to be licensed under GPL-2.0. It does not have a license header or any other copyright information, but it states it was taken from Linux 2.4.0: include/list.h, line 5: | This code was taken from the Linux 2.4.0 kernel. [jaroslav] I'm not sure one could argue that the header was trivial, especially since it does not only contain constants, structures and interface definitions, but also implementations of inline functions. I found one previous inquiry about a similar issue[1], but it does not cover the include/list.h header. Two questions: 1. Have the copyright holders of the code in question agreed to re-license under a different, LGPL-preserving license and, if yes, is there some documentation about that? 2. If the header is indeed covered by GPL-2.0, would you accept patches to replace it with a different implementation, e.g. Rusty Russell's MIT-licensed list from ccan[2]? [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-March/038267.html [2] http://ccodearchive.net/info/list.html Thank you for your time, Clemens Lang -- BMW Car IT GmbH Clemens Lang Spezialist Entwicklung Lise-Meitner-Straße 14 89081 Ulm Mail: clemens.lang(a)bmw-carit.de Web: http://www.bmw-carit.de -------------------------------------------------------------------- BMW Car IT GmbH Geschäftsführer: Michael Würtenberger und Reinhard Stolle Sitz und Registergericht: München HRB 134810 --------------------------------------------------------------------

3 12

Re: [alsa-devel] ASoC: Add max98925 codec driver
by Dan Carpenter 20 Jul '15

20 Jul '15

Hello Anish Kumar, The patch 1ff2765182d1: "ASoC: Add max98925 codec driver" from Mar 9, 2015, leads to the following static checker warning: sound/soc/codecs/max98925.c:274 max98925_rate_value() error: buffer overflow 'rate_table' 9 <= 9 sound/soc/codecs/max98925.c 259 static inline int max98925_rate_value(struct snd_soc_codec *codec, 260 int rate, int clock, int *value, int *n, int *m) 261 { 262 int ret = -EINVAL; 263 int i; 264 265 for (i = 0; i < ARRAY_SIZE(rate_table); i++) { 266 if (rate_table[i].rate >= rate) { 267 *value = rate_table[i].sr; 268 *n = rate_table[i].divisors[clock][0]; 269 *m = rate_table[i].divisors[clock][1]; 270 ret = 0; 271 break; 272 } 273 } 274 dev_dbg(codec->dev, "%s: sample rate is %d, returning %d\n", 275 __func__, rate_table[i].rate, *value); ^^^^^^^^^^^^^^^^^^ If we don't break then we're off by one. 276 return ret; 277 } regards, dan carpenter

2 2

[alsa-devel] [PATCH] speaker-test: Add option to specify signal scale
by Julian Scheel 17 Jul '15

17 Jul '15

Allow generated signals (sine and noise) to have a specified signal scale instead of using hardcoded limits of 80%. This can be handy for debugging, ie when analysing clipping issues with audio drivers. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- speaker-test/speaker-test.c | 30 ++++++++++++++++++------------ 1 file changed, 18 insertions(+), 12 deletions(-) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index ccf2671..25970af 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -100,6 +100,7 @@ static unsigned int period_time = 0; /* period time in us */ static unsigned int nperiods = 4; /* number of periods */ static double freq = 440.0; /* sinusoidal wave frequency in Hz */ static int test_type = TEST_PINK_NOISE; /* Test type. 1 = noise, 2 = sine wave */ +static float generator_scale = 0.8; /* Scale to use for sine volume */ static pink_noise_t pink; static snd_pcm_uframes_t buffer_size; static snd_pcm_uframes_t period_size; @@ -306,7 +307,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas switch (format) { case SND_PCM_FORMAT_S8: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp8++ = ires >> 24; } else { @@ -315,7 +316,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_S16_LE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp16++ = LE_SHORT(ires >> 16); } else { @@ -324,7 +325,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_S16_BE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp16++ = BE_SHORT(ires >> 16); } else { @@ -333,7 +334,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_FLOAT_LE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0.75 ; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale; fres = res; *samp_f++ = fres; } else { @@ -342,7 +343,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_S32_LE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp32++ = LE_INT(ires); } else { @@ -351,7 +352,7 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas break; case SND_PCM_FORMAT_S32_BE: if (chn==channel) { - res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */ + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; ires = res; *samp32++ = BE_INT(ires); } else { @@ -389,7 +390,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { switch (format) { case SND_PCM_FORMAT_S8: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp8++ = ires >> 24; } else { @@ -398,7 +399,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { break; case SND_PCM_FORMAT_S16_LE: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp16++ = LE_SHORT(ires >> 16); } else { @@ -407,7 +408,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { break; case SND_PCM_FORMAT_S16_BE: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp16++ = BE_SHORT(ires >> 16); } else { @@ -416,7 +417,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { break; case SND_PCM_FORMAT_S32_LE: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp32++ = LE_INT(ires); } else { @@ -425,7 +426,7 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { break; case SND_PCM_FORMAT_S32_BE: if (chn==channel) { - res = generate_pink_noise_sample(&pink) * 0x03fffffff; /* Don't use MAX volume */ + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; ires = res; *samp32++ = BE_INT(ires); } else { @@ -1017,6 +1018,7 @@ static void help(void) "-W,--wavdir Specify the directory containing WAV files\n" "-m,--chmap Specify the channel map to override\n" "-X,--force-frequency force frequencies outside the 30-8000hz range\n" + "-S,--scale Scale of generated test tones in percent (default=80)\n" "\n")); printf(_("Recognized sample formats are:")); for (fmt = supported_formats; *fmt >= 0; fmt++) { @@ -1060,6 +1062,7 @@ int main(int argc, char *argv[]) { {"wavdir", 1, NULL, 'W'}, {"debug", 0, NULL, 'd'}, {"force-frequency", 0, NULL, 'X'}, + {"scale", 1, NULL, 'S'}, #ifdef CONFIG_SUPPORT_CHMAP {"chmap", 1, NULL, 'm'}, #endif @@ -1081,7 +1084,7 @@ int main(int argc, char *argv[]) { while (1) { int c; - if ((c = getopt_long(argc, argv, "hD:r:c:f:F:b:p:P:t:l:s:w:W:d:X" + if ((c = getopt_long(argc, argv, "hD:r:c:f:F:b:p:P:t:l:s:w:W:d:XS:" #ifdef CONFIG_SUPPORT_CHMAP "m:" #endif @@ -1182,6 +1185,9 @@ int main(int argc, char *argv[]) { chmap = optarg; break; #endif + case 'S': + generator_scale = atoi(optarg) / 100.0; + break; default: fprintf(stderr, _("Unknown option '%c'\n"), c); exit(EXIT_FAILURE); -- 2.4.3

2 2

[alsa-devel] [GIT PULL] Braswell firmware update
by Vinod Koul 14 Jul '15

14 Jul '15

Hello Linux firmware maintainers, Please PULL to receive the updated BSW firmware binary which contains power management fixes. We have also updated the version of binary in WHENCE file The following changes since commit 3161bfa479d5e9ed4f46b57df9bcecbbc4f8eb3c: linux-firmware: Update firmware patch for Intel Bluetooth 7260 (B3/B4) (2015-05-13 11:12:48 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git BSW for you to fetch changes up to e5fa14afd09909124fc7618fc4a0ac00fa4f9347: linux-firmware: update audio firmware for Braswell platform (2015-06-17 11:21:47 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: update audio firmware for Braswell platform WHENCE | 1 + intel/fw_sst_22a8.bin | Bin 647390 -> 701614 bytes 2 files changed, 1 insertion(+) Thanks -- ~Vinod

2 2