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[PATCH - maemo fixes 1/1] maemo plugin has two crashes I was able to see in a valgrind log from another user:
by stian.skjelstad＠gmail.com 02 May '19

02 May '19

From: Stian Skjelstad <stian.skjelstad(a)gmail.com> * maximum write size was calculated in words (16bit), but checked against byte-size length. This causes memcpy later to overflow the buffer (normally by up to 12KB). * remove a double free (by marking free'd data with NULL) * mmap returns MMAP_FAILED on error, not NULL I suspect that this plugin/driver might have other issues aswell, since I am unable to find any logic for checking DSP buffer status, and no implementation for odelay reporting. Author: Stian Skjelstad <stian.skjelstad(a)gmail.com> Signed-off-by: Stian Skjelstad <stian.skjelstad(a)gmail.com> diff --git a/maemo/alsa-dsp.c b/maemo/alsa-dsp.c index 7e04f6a..3307384 100644 --- a/maemo/alsa-dsp.c +++ b/maemo/alsa-dsp.c @@ -135,18 +135,18 @@ static snd_pcm_sframes_t alsa_dsp_transfer(snd_pcm_ioplug_t * io, snd_pcm_alsa_dsp_t *alsa_dsp = io->private_data; DENTER(); char *buf; - int words; + int bytes, words; ssize_t result; - words = size * alsa_dsp->bytes_per_frame; - words /= 2; - DPRINT("***** Info: words %d size %lu bpf: %d\n", words, size, - alsa_dsp->bytes_per_frame); - if (words > alsa_dsp->dsp_protocol->mmap_buffer_size) { - DERROR("Requested too much data transfer (playing only %d)\n", - alsa_dsp->dsp_protocol->mmap_buffer_size); - words = alsa_dsp->dsp_protocol->mmap_buffer_size; + bytes = size * alsa_dsp->bytes_per_frame; + DPRINT("***** Info: samples %lu * bpf %d => bytes %d\n", + size, alsa_dsp->bytes_per_frame, bytes); + if (bytes > alsa_dsp->dsp_protocol->mmap_buffer_size) { + DERROR("Requested too much data transfer (requested %d, playing only %d)\n", + bytes, alsa_dsp->dsp_protocol->mmap_buffer_size); + bytes = alsa_dsp->dsp_protocol->mmap_buffer_size; } + words = bytes / 2; if (alsa_dsp->dsp_protocol->state != STATE_PLAYING) { DPRINT("I did nothing - No start sent\n"); alsa_dsp_start(io); diff --git a/maemo/dsp-ctl.c b/maemo/dsp-ctl.c index 5bcda36..ac05942 100644 --- a/maemo/dsp-ctl.c +++ b/maemo/dsp-ctl.c @@ -93,6 +93,7 @@ static void dsp_ctl_close(snd_ctl_ext_t * ext) snd_ctl_dsp_t *dsp_ctl = ext->private_data; DENTER(); free(dsp_ctl->controls); + dsp_ctl->controls = NULL; free_control_list(&(dsp_ctl->playback_devices)); free_control_list(&(dsp_ctl->recording_devices)); // free(dsp_ctl); diff --git a/maemo/dsp-protocol.c b/maemo/dsp-protocol.c index af67251..32193d3 100644 --- a/maemo/dsp-protocol.c +++ b/maemo/dsp-protocol.c @@ -194,7 +194,7 @@ int dsp_protocol_open_node(dsp_protocol_t * dsp_protocol, const char *device) mmap((void *)0, dsp_protocol->mmap_buffer_size, PROT_READ | PROT_WRITE, MAP_SHARED, dsp_protocol->fd, 0); - if (dsp_protocol->mmap_buffer == NULL) { + if (dsp_protocol->mmap_buffer == MAP_FAILED) { report_dsp_protocol("Cannot mmap data buffer", dsp_protocol); ret = -ENOMEM; goto unlock; -- 2.17.1

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[PATCH 1/5] pcm: direct: Add generic hw_ptr_alignment function for dmix, dshare and dsnoop
by vanitha.channaiah＠in.bosch.com 30 Apr '19

30 Apr '19

From: Vanitha Channaiah <vanitha.channaiah(a)in.bosch.com> This patch is same as the fix in below commit commit 6b058fda9dce ("pcm: dmix: Add option to allow alignment of slave pointers") Signed-off-by: Vanitha Channaiah <vanitha.channaiah(a)in.bosch.com> --- src/pcm/pcm_direct.c | 19 +++++++++++++++++++ src/pcm/pcm_direct.h | 8 ++++++++ src/pcm/pcm_dmix.c | 25 ++----------------------- 3 files changed, 29 insertions(+), 23 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 666a8ce..411a035 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -2040,3 +2040,22 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, return 0; } + +void snd_pcm_direct_reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) +{ + dmix->slave_appl_ptr = dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; + + if (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_ROUNDUP || + (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_AUTO && + pcm->buffer_size <= pcm->period_size * 2)) + dmix->slave_appl_ptr = + ((dmix->slave_appl_ptr + dmix->slave_period_size - 1) / + dmix->slave_period_size) * dmix->slave_period_size; + else if (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_ROUNDDOWN || + (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_AUTO && + (dmix->slave_period_size * SEC_TO_MS) / + pcm->rate < LOW_LATENCY_PERIOD_TIME)) + dmix->slave_appl_ptr = dmix->slave_hw_ptr = + ((dmix->slave_hw_ptr / dmix->slave_period_size) * + dmix->slave_period_size); +} diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index da5e280..a71aab1 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -24,6 +24,11 @@ #define DIRECT_IPC_SEMS 1 #define DIRECT_IPC_SEM_CLIENT 0 +/* Seconds representing in Milli seconds */ +#define SEC_TO_MS 1000 +/* slave_period time for low latency requirements in ms */ +#define LOW_LATENCY_PERIOD_TIME 10 + typedef void (mix_areas_t)(unsigned int size, volatile void *dst, void *src, @@ -257,6 +262,8 @@ struct snd_pcm_direct { snd1_pcm_direct_get_chmap #define snd_pcm_direct_set_chmap \ snd1_pcm_direct_set_chmap +#define snd_pcm_direct_reset_slave_ptr \ + snd1_pcm_direct_reset_slave_ptr int snd_pcm_direct_semaphore_create_or_connect(snd_pcm_direct_t *dmix); @@ -341,6 +348,7 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct); int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm); int snd_timer_async(snd_timer_t *timer, int sig, pid_t pid); struct timespec snd_pcm_hw_fast_tstamp(snd_pcm_t *pcm); +void snd_pcm_direct_reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix); struct snd_pcm_direct_open_conf { key_t ipc_key; diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index c5592cd..dcde40d 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -55,9 +55,6 @@ const char *_snd_module_pcm_dmix = ""; #define STATE_RUN_PENDING 1024 #endif -#define SEC_TO_MS 1000 /* Seconds representing in Milli seconds */ -#define LOW_LATENCY_PERIOD_TIME 10 /* slave_period time for low latency requirements in ms */ - /* * */ @@ -560,30 +557,12 @@ static int snd_pcm_dmix_hwsync(snd_pcm_t *pcm) } } -static void reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) -{ - dmix->slave_appl_ptr = dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; - - if (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_ROUNDUP || - (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_AUTO && - pcm->buffer_size <= pcm->period_size * 2)) - dmix->slave_appl_ptr = - ((dmix->slave_appl_ptr + dmix->slave_period_size - 1) - / dmix->slave_period_size) * dmix->slave_period_size; - else if (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_ROUNDDOWN || - (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_AUTO && - (dmix->slave_period_size * SEC_TO_MS) / pcm->rate < LOW_LATENCY_PERIOD_TIME)) - dmix->slave_appl_ptr = dmix->slave_hw_ptr = - ((dmix->slave_hw_ptr / dmix->slave_period_size) * - dmix->slave_period_size); -} - static int snd_pcm_dmix_reset(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; dmix->hw_ptr %= pcm->period_size; dmix->appl_ptr = dmix->last_appl_ptr = dmix->hw_ptr; - reset_slave_ptr(pcm, dmix); + snd_pcm_direct_reset_slave_ptr(pcm, dmix); return 0; } @@ -592,7 +571,7 @@ static int snd_pcm_dmix_start_timer(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) int err; snd_pcm_hwsync(dmix->spcm); - reset_slave_ptr(pcm, dmix); + snd_pcm_direct_reset_slave_ptr(pcm, dmix); err = snd_timer_start(dmix->timer); if (err < 0) return err; -- 2.7.4

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[PATCH 1/1] ASoC: Intel: Skylake: change list empty check to warning
by Patrick Waterlander 29 Apr '19

29 Apr '19

NULL is a valid state for the module list when querying unique ID. Downgrade the empty condition from error to warning. Signed-off-by: Patrick Waterlander <patrick.waterlander(a)intel.com> diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 5951bbdf1f1a..95e6a23538cd 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -324,7 +324,7 @@ static struct skl_module_table *skl_module_get_from_id( struct skl_module_table *module; if (list_empty(&ctx->module_list)) { - dev_err(ctx->dev, "Module list is empty\n"); + dev_warn(ctx->dev, "Module list is empty\n"); return NULL; } -- 2.17.1

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[PATCH] ALSA: pcm: Enable MMAP status and control for ARMv7 and ARMv8
by twischer＠de.adit-jv.com 17 Apr '19

17 Apr '19

From: Timo Wischer <twischer(a)de.adit-jv.com> Since ARMv7 hardware cache coherence is supported. "The SCU maintains coherency between the individual data caches in the Cortex-A5 MPCore processor using a variation of the MOESI protocol" [1]. Therefore this patch enables the MMAP access to the status and control structures. This avoids HWSYYNC ioctl calls and therefore lowers the CPU usage. [1] http://infocenter.arm.com/help/index.jsp?topic=/com.arm.doc.ddi0434c/ BABJECBF.html Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 1d84529..b8019ef 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3225,7 +3225,8 @@ static __poll_t snd_pcm_poll(struct file *file, poll_table *wait) * Only on coherent architectures, we can mmap the status and the control records * for effcient data transfer. On others, we have to use HWSYNC ioctl... */ -#if defined(CONFIG_X86) || defined(CONFIG_PPC) || defined(CONFIG_ALPHA) +#if defined(CONFIG_X86) || defined(CONFIG_PPC) || defined(CONFIG_ALPHA) || \ + (defined(CONFIG_ARM) && defined(CONFIG_CPU_V7)) || defined(CONFIG_ARM64) /* * mmap status record */ -- 2.7.4

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[PATCH] ALSA: usb-audio: Add quirk for Focusrite Scarlett Solo
by Roope Salmi 14 Apr '19

14 Apr '19

The device reports Synch: Synchronous on the playback interface. This causes regular audible napping on sample rates that are not multiples of 1 kHz. Fix to Synch: Asynchronous. Specifically observed on Focusrite Scarlett Solo 2nd generation. I assume the first generation model has a different device ID. A first generation Scarlett 2i2 I was able to test advertised Synch: Asynchronous by default. For example, with a sample rate of 44100 Hz, a silent sample is played every 40.96 seconds (likely 44.0 samples instead of 44.1 transmitted per USB frame on average, 4096 being the size of some internal buffer). There may be some other bug at play here since this doesn't happen on other platforms. However, a feedback endpoint is listed and using it fixes the issue. That is the only change in the quirk, but I didn't find a way to declare only it. Tested on two units and on two different computers. Signed-off-by: Roope Salmi <rpsalmi(a)gmail.com> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 86e80916a..5525f9043 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2770,6 +2770,90 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, +{ + /* + * Focusrite Scarlett Solo 2nd generation + * Reports that playback should use Synch: Synchronous + * while still providing a feedback endpoint. Synchronous causes + * snapping on some sample rates. + * Force it to use Synch: Asynchronous. + */ + USB_DEVICE(0x1235, 0x8205), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .protocol = UAC_VERSION_2, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 44100, + .rate_max = 192000, + .nr_rates = 6, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, + 96000, 176400, 192000 + }, + .clock = 41 + } + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC | + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .protocol = UAC_VERSION_2, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 44100, + .rate_max = 192000, + .nr_rates = 6, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, + 96000, 176400, 192000 + }, + .clock = 41 + } + }, + { + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, /* Access Music devices */ { -- 2.17.1

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[PATCH v2] pcm: null: Do not allow a period size of 0
by twischer＠de.adit-jv.com 08 Apr '19

08 Apr '19

From: Timo Wischer <twischer(a)de.adit-jv.com> Some applications do not expect that get_period_size_min() could return 0. Therefore these applications cannot use the null plugin without this patch. Due to there is no use case for having a period size of 0 this patch disallows a period size of 0 when using the null plugin. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- Hi Takashi, > Why 32? I thought there is also no use case for a period of 1 frame. Therefore I increased it to a 2^n value to also lower the CPU usage in case of null plugin. > Or does it cause a problem? I do not see any issues here. But for example portaudio will automatically choose a period size of 1 frame which will result in a higher CPU load. But anyway I would also be fine with this higher CPU load if you prefer this patch. Best regards Timo src/pcm/pcm_null.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_null.c b/src/pcm/pcm_null.c index ff61624..1d81548 100644 --- a/src/pcm/pcm_null.c +++ b/src/pcm/pcm_null.c @@ -261,7 +261,17 @@ static snd_pcm_sframes_t snd_pcm_null_mmap_commit(snd_pcm_t *pcm, static int snd_pcm_null_hw_refine(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_hw_params_t *params) { - int err = snd_pcm_hw_refine_soft(pcm, params); + int err; + + /* Do not return a period size of 0 because for example portaudio cannot + * handle it. + */ + err = _snd_pcm_hw_param_set_min(params, SND_PCM_HW_PARAM_PERIOD_SIZE, 1, + 0); + if (err < 0) + return err; + + err = snd_pcm_hw_refine_soft(pcm, params); params->info = SND_PCM_INFO_MMAP | SND_PCM_INFO_MMAP_VALID | SND_PCM_INFO_RESUME | SND_PCM_INFO_PAUSE; params->fifo_size = 0; -- 2.7.4

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[PATCH] pcm: null: Do not allow a period size of 0
by twischer＠de.adit-jv.com 05 Apr '19

05 Apr '19

From: Timo Wischer <twischer(a)de.adit-jv.com> Some applications do not expect that get_period_size_min() could return 0. Therefore these applications cannot use the null plugin without this patch. Due to there is no use case for having a period size of 0 this patch disallows a period size of 0 when using the null plugin. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- src/pcm/pcm_null.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_null.c b/src/pcm/pcm_null.c index ff61624..c95fad8 100644 --- a/src/pcm/pcm_null.c +++ b/src/pcm/pcm_null.c @@ -261,7 +261,18 @@ static snd_pcm_sframes_t snd_pcm_null_mmap_commit(snd_pcm_t *pcm, static int snd_pcm_null_hw_refine(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_hw_params_t *params) { - int err = snd_pcm_hw_refine_soft(pcm, params); + int err; + + /* Do not return a period size of 0 because for example portaudio cannot + * handle it. In addition such small buffers would only increase the CPU + * load but there is no benefit when throwing the samples anyway away. + */ + err = _snd_pcm_hw_param_set_min(params, SND_PCM_HW_PARAM_PERIOD_SIZE, + 32, 0); + if (err < 0) + return err; + + err = snd_pcm_hw_refine_soft(pcm, params); params->info = SND_PCM_INFO_MMAP | SND_PCM_INFO_MMAP_VALID | SND_PCM_INFO_RESUME | SND_PCM_INFO_PAUSE; params->fifo_size = 0; -- 2.7.4

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[PATCH 00/10] ALSA: aloop: Support selection of snd_timer
by twischer＠de.adit-jv.com 26 Mar '19

26 Mar '19

With the following patch set any sound timer can be used as timer source for the aloop devices if selected. It can be select by the additional timer_source kernel module parameter: -2 the jiffies timer will be used as done without this patch (default) -1 the sound timer will be selected depending on the linked sound cards (snd_pcm_link()). The sound timer of the first linked card which is not a aloop device nor a dummy device will be used. 0..N Select the sound timer by the card number. This patch set is required for forwarding audio data between a HW sound card and an aloop device without the usage of an asynchronous sample rate converter. Best regards, Timo

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[PATCH] ALSA: aloop: Support S24 sample formats
by twischer＠de.adit-jv.com 25 Mar '19

25 Mar '19

From: Timo Wischer <twischer(a)de.adit-jv.com> Currently snd_aloop supports only S16 and S32 audio sample formats. With this patch the S24 formats are also supported. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- sound/drivers/aloop.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 8c3fbe1..c14e57b 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -337,7 +337,7 @@ static int loopback_prepare(struct snd_pcm_substream *substream) loopback_timer_stop_sync(dpcm); - salign = (snd_pcm_format_width(runtime->format) * + salign = (snd_pcm_format_physical_width(runtime->format) * runtime->channels) / 8; bps = salign * runtime->rate; if (bps <= 0 || salign <= 0) @@ -562,6 +562,8 @@ static const struct snd_pcm_hardware loopback_pcm_hardware = SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE), .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000, -- 2.7.4

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[PATCH] [-, JACK, plugin, 1/1] configure: use $prefix/etc instead of /etc
by Alex Ivanov 20 Mar '19

20 Mar '19

Use $sysconfdir as prefix for ALSA_LCONF_DIR by default. Otherwise install fails on non-FHS distros. Signed-off-by: Alex Ivanov <gnidorah(a)ya.ru> --- configure.ac | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 5b80585..7b8dd66 100644 --- a/configure.ac +++ b/configure.ac @@ -219,6 +219,23 @@ AC_DEFINE_UNQUOTED(ALSA_DATA_DIR, "$alsadatadir", [directory containing ALSA dat ALSA_DATA_DIR="$alsadatadir" AC_SUBST(ALSA_DATA_DIR) +dnl ALSA configuration directory +AC_ARG_WITH(alsaconfdir, + AS_HELP_STRING([--with-alsaconfdir=dir], + [path where ALSA configuration files are stored]), + alsaconfdir="$withval", alsaconfdir="") +if test -z "$alsaconfdir"; then + eval dir="$sysconfdir" + case "$dir" in + /*) ;; + *) dir="$dir" + esac + alsaconfdir="$dir/alsa" +fi +AC_DEFINE_UNQUOTED(ALSA_CONF_DIR, "$alsaconfdir", [directory containing ALSA configuration files]) +ALSA_CONF_DIR="$alsaconfdir" +AC_SUBST(ALSA_CONF_DIR) + dnl ALSA add-on global config directory AC_ARG_WITH(alsagconfdir, AS_HELP_STRING([--with-alsagconfdir=dir], @@ -237,7 +254,7 @@ AC_ARG_WITH(alsalconfdir, [path where ALSA local add-on config files are stored]), alsalconfdir="$withval", alsalconfdir="") if test -z "$alsalconfdir"; then - alsalconfdir="/etc/alsa/conf.d" + alsalconfdir="$ALSA_CONF_DIR/conf.d" fi AC_DEFINE_UNQUOTED(ALSA_LCONF_DIR, "$alsalconfdir", [directory containing local ALSA add-on config files]) ALSA_LCONF_DIR="$alsalconfdir" -- 2.19.2

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[PATCH] [-, JACK, plugin, 1/1] jack: use correct port names in 50-jack.conf
by Alex Ivanov 20 Mar '19

20 Mar '19

Change port names in 50-jack.conf from "alsa_pcm" to "system". JACK1 v0.125.0 still provides alsa_pcm:playback_* and alsa_pcm:capture_* port names, but only as aliases: $ jack_lsp -A system:capture_1 alsa_pcm:capture_1 system:capture_2 alsa_pcm:capture_2 system:playback_1 alsa_pcm:playback_1 system:playback_2 alsa_pcm:playback_2 JACK2 v1.9.12 doesn't support alsa_pcm:playback_* and alsa_pcm:capture_* aliases anymore, so 50-jack.conf will fail for JACK2 users: $ jack_lsp -A system:capture_1 alsa_pcm:hw:0:out1 system:capture_2 alsa_pcm:hw:0:out2 system:playback_1 alsa_pcm:hw:0:in1 system:playback_2 alsa_pcm:hw:0:in2 As you see in both cases system:playback_* and system:capture_* are correct port names. Signed-off-by: Alex Ivanov <gnidorah(a)ya.ru> --- jack/50-jack.conf | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/jack/50-jack.conf b/jack/50-jack.conf index d780dfc..9948f2f 100644 --- a/jack/50-jack.conf +++ b/jack/50-jack.conf @@ -1,12 +1,12 @@ pcm.jack { type jack playback_ports { - 0 alsa_pcm:playback_1 - 1 alsa_pcm:playback_2 + 0 system:playback_1 + 1 system:playback_2 } capture_ports { - 0 alsa_pcm:capture_1 - 1 alsa_pcm:capture_2 + 0 system:capture_1 + 1 system:capture_2 } hint { show { -- 2.19.2

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test2, ignore, please
by Jaroslav Kysela 18 Mar '19

18 Mar '19

Ignore, please. -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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