From f297107b5a92fd57e47fda1d8109213dc3117376 Mon Sep 17 00:00:00 2001 From: CentOS Sources Date: Feb 02 2021 06:11:19 +0000 Subject: import alsa-lib-1.2.4-5.el8 --- diff --git a/SOURCES/alsa-git.patch b/SOURCES/alsa-git.patch index c0d6687..5132808 100644 --- a/SOURCES/alsa-git.patch +++ b/SOURCES/alsa-git.patch @@ -1,46 +1,129 @@ -diff --git a/Makefile.am b/Makefile.am -index c484d4da..ff4c963a 100644 ---- a/Makefile.am -+++ b/Makefile.am -@@ -16,7 +16,7 @@ SUBDIRS += alsalisp - endif - endif - SUBDIRS += test utils --EXTRA_DIST=ChangeLog INSTALL TODO NOTES configure gitcompile libtool \ -+EXTRA_DIST=README.md ChangeLog INSTALL TODO NOTES configure gitcompile libtool \ - depcomp version MEMORY-LEAK m4/attributes.m4 - AUTOMAKE_OPTIONS=foreign +From ad8c8e5503980295dd8e5e54a6285d2d7e32eb1e Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 22 Oct 2020 20:57:32 +0200 +Subject: [PATCH 01/49] dlmisc: the snd_plugin_dir_set / snd_plugin_dir must be + declared even for \!DL_ORIGIN_AVAILABLE + +Fixes: 8580c081c2 ("dlsym: add support for ALSA_PLUGIN_DIR environment variable") +BugLink: https://github.com/alsa-project/alsa-lib/issues/91 +Signed-off-by: Jaroslav Kysela +--- + src/dlmisc.c | 2 -- + 1 file changed, 2 deletions(-) + +diff --git a/src/dlmisc.c b/src/dlmisc.c +index c9517c55..f20eb593 100644 +--- a/src/dlmisc.c ++++ b/src/dlmisc.c +@@ -42,11 +42,9 @@ + #ifndef PIC + struct snd_dlsym_link *snd_dlsym_start = NULL; + #endif +-#ifdef DL_ORIGIN_AVAILABLE + static int snd_plugin_dir_set = 0; + static char *snd_plugin_dir = NULL; + #endif +-#endif -diff --git a/include/pcm.h b/include/pcm.h -index 5b078231..e300b951 100644 ---- a/include/pcm.h -+++ b/include/pcm.h -@@ -350,6 +350,20 @@ typedef enum _snd_pcm_tstamp_type { - SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, - } snd_pcm_tstamp_type_t; + #if defined(DL_ORIGIN_AVAILABLE) && defined(HAVE_LIBPTHREAD) + static pthread_mutex_t snd_dlpath_mutex = PTHREAD_MUTEX_INITIALIZER; +-- +2.29.2 + + +From 1d993b37ab4e0b9b6e6b795d7be84ce97b29e20b Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 29 Oct 2020 20:31:19 +0100 +Subject: [PATCH 02/49] dlmisc: fix snd_plugin_dir locking for not + DL_ORIGIN_AVAILABLE + +Fixes: 8580c081c2 ("dlsym: add support for ALSA_PLUGIN_DIR environment variable") +Signed-off-by: Jaroslav Kysela +--- + src/dlmisc.c | 4 +--- + 1 file changed, 1 insertion(+), 3 deletions(-) + +diff --git a/src/dlmisc.c b/src/dlmisc.c +index f20eb593..1dd91356 100644 +--- a/src/dlmisc.c ++++ b/src/dlmisc.c +@@ -46,7 +46,7 @@ static int snd_plugin_dir_set = 0; + static char *snd_plugin_dir = NULL; + #endif -+typedef enum _snd_pcm_audio_tstamp_type { -+ /** -+ * first definition for backwards compatibility only, -+ * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else -+ */ -+ SND_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0, -+ SND_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /**< DMA time, reported as per hw_ptr */ -+ SND_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /**< link time reported by sample or wallclock counter, reset on startup */ -+ SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /**< link time reported by sample or wallclock counter, not reset on startup */ -+ SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /**< link time estimated indirectly */ -+ SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /**< link time synchronized with system time */ -+ SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED -+} snd_pcm_audio_tstamp_type_t; -+ - typedef struct _snd_pcm_audio_tstamp_config { - /* 5 of max 16 bits used */ - unsigned int type_requested:4; +-#if defined(DL_ORIGIN_AVAILABLE) && defined(HAVE_LIBPTHREAD) ++#ifdef HAVE_LIBPTHREAD + static pthread_mutex_t snd_dlpath_mutex = PTHREAD_MUTEX_INITIALIZER; + + static inline void snd_dlpath_lock(void) +@@ -440,12 +440,10 @@ void snd_dlobj_cache_cleanup(void) + free(c); + } + snd_dlobj_unlock(); +-#ifdef DL_ORIGIN_AVAILABLE + snd_dlpath_lock(); + snd_plugin_dir_set = 0; + free(snd_plugin_dir); + snd_plugin_dir = NULL; + snd_dlpath_unlock(); +-#endif + } + #endif +-- +2.29.2 + + +From 2757191e3b85d491705b31c6ecca2dbd96d26ade Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 29 Oct 2020 20:34:17 +0100 +Subject: [PATCH 03/49] pcm: snd_pcm_mmap_readi - fix typo in comment + +\param size frames to be *read* + +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_mmap.c | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +diff --git a/src/pcm/pcm_mmap.c b/src/pcm/pcm_mmap.c +index 9600c383..9cbaae05 100644 +--- a/src/pcm/pcm_mmap.c ++++ b/src/pcm/pcm_mmap.c +@@ -183,7 +183,7 @@ snd_pcm_sframes_t snd_pcm_mmap_writen(snd_pcm_t *pcm, void **bufs, snd_pcm_ufram + * \brief Read interleaved frames from a PCM using direct buffer (mmap) + * \param pcm PCM handle + * \param buffer frames containing buffer +- * \param size frames to be written ++ * \param size frames to be read + * \return a positive number of frames actually read otherwise a + * negative error code + * \retval -EBADFD PCM is not in the right state (#SND_PCM_STATE_PREPARED or #SND_PCM_STATE_RUNNING) +-- +2.29.2 + + +From 39bd0e1a5be3620f9123f7fe72ffa6cb7d463b21 Mon Sep 17 00:00:00 2001 +From: Pierre-Louis Bossart +Date: Thu, 12 Nov 2020 10:29:38 -0600 +Subject: [PATCH 04/49] topology: use inclusive language for bclk + +use bclk_provider for structure fields, 'codec_provider' and +'codec_consumer' for options and modify #defines to use CP and CC +suffixes. + +Signed-off-by: Pierre-Louis Bossart +Signed-off-by: Takashi Iwai +--- + include/sound/uapi/asoc.h | 11 +++++++---- + include/topology.h | 2 +- + src/topology/pcm.c | 36 ++++++++++++++++++++++++++---------- + 3 files changed, 34 insertions(+), 15 deletions(-) + diff --git a/include/sound/uapi/asoc.h b/include/sound/uapi/asoc.h -index 4efb4ec4..f32c5697 100644 +index 4efb4ec4..ceafb1a9 100644 --- a/include/sound/uapi/asoc.h +++ b/include/sound/uapi/asoc.h -@@ -169,16 +169,22 @@ +@@ -169,10 +169,13 @@ #define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP (1 << 3) /* DAI topology BCLK parameter @@ -56,6 +139,143 @@ index 4efb4ec4..f32c5697 100644 +#define SND_SOC_TPLG_BCLK_CS SND_SOC_TPLG_BCLK_CC /* DAI topology FSYNC parameter + * For the backwards capability, by default codec is fsync master +@@ -335,7 +338,7 @@ struct snd_soc_tplg_hw_config { + __u8 clock_gated; /* SND_SOC_TPLG_DAI_CLK_GATE_ value */ + __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ + __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ +- __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ ++ __u8 bclk_provider; /* SND_SOC_TPLG_BCLK_ value */ + __u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ + __u8 mclk_direction; /* SND_SOC_TPLG_MCLK_ value */ + __le16 reserved; /* for 32bit alignment */ +diff --git a/include/topology.h b/include/topology.h +index 1f52e66e..6c970649 100644 +--- a/include/topology.h ++++ b/include/topology.h +@@ -1028,7 +1028,7 @@ struct snd_tplg_hw_config_template { + unsigned char clock_gated; /* SND_SOC_TPLG_DAI_CLK_GATE_ value */ + unsigned char invert_bclk; /* 1 for inverted BCLK, 0 for normal */ + unsigned char invert_fsync; /* 1 for inverted frame clock, 0 for normal */ +- unsigned char bclk_master; /* SND_SOC_TPLG_BCLK_ value */ ++ unsigned char bclk_provider; /* SND_SOC_TPLG_BCLK_ value */ + unsigned char fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ + unsigned char mclk_direction; /* SND_SOC_TPLG_MCLK_ value */ + unsigned short reserved; /* for 32bit alignment */ +diff --git a/src/topology/pcm.c b/src/topology/pcm.c +index 191b7a0a..f05df348 100644 +--- a/src/topology/pcm.c ++++ b/src/topology/pcm.c +@@ -1411,6 +1411,7 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, + snd_config_t *n; + const char *id, *val = NULL; + int ret, ival; ++ bool provider_legacy; + + elem = tplg_elem_new_common(tplg, cfg, NULL, SND_TPLG_TYPE_HW_CONFIG); + if (!elem) +@@ -1451,8 +1452,15 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, + continue; + } + +- if (strcmp(id, "bclk") == 0 || +- strcmp(id, "bclk_master") == 0) { ++ provider_legacy = false; ++ if (strcmp(id, "bclk_master") == 0) { ++ SNDERR("deprecated option %s, please use 'bclk'\n", id); ++ provider_legacy = true; ++ } ++ ++ if (provider_legacy || ++ strcmp(id, "bclk") == 0) { ++ + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + +@@ -1462,11 +1470,19 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, + */ + SNDERR("deprecated bclk value '%s'", val); + +- hw_cfg->bclk_master = SND_SOC_TPLG_BCLK_CS; ++ hw_cfg->bclk_provider = SND_SOC_TPLG_BCLK_CC; + } else if (!strcmp(val, "codec_slave")) { +- hw_cfg->bclk_master = SND_SOC_TPLG_BCLK_CS; ++ SNDERR("deprecated bclk value '%s', use 'codec_consumer'", val); ++ ++ hw_cfg->bclk_provider = SND_SOC_TPLG_BCLK_CC; ++ } else if (!strcmp(val, "codec_consumer")) { ++ hw_cfg->bclk_provider = SND_SOC_TPLG_BCLK_CC; + } else if (!strcmp(val, "codec_master")) { +- hw_cfg->bclk_master = SND_SOC_TPLG_BCLK_CM; ++ SNDERR("deprecated bclk value '%s', use 'codec_provider", val); ++ ++ hw_cfg->bclk_provider = SND_SOC_TPLG_BCLK_CP; ++ } else if (!strcmp(val, "codec_provider")) { ++ hw_cfg->bclk_provider = SND_SOC_TPLG_BCLK_CP; + } + continue; + } +@@ -1623,10 +1639,10 @@ int tplg_save_hw_config(snd_tplg_t *tplg ATTRIBUTE_UNUSED, + if (err >= 0 && hc->fmt) + err = tplg_save_printf(dst, pfx, "\tformat '%s'\n", + get_audio_hw_format_name(hc->fmt)); +- if (err >= 0 && hc->bclk_master) ++ if (err >= 0 && hc->bclk_provider) + err = tplg_save_printf(dst, pfx, "\tbclk '%s'\n", +- hc->bclk_master == SND_SOC_TPLG_BCLK_CS ? +- "codec_slave" : "codec_master"); ++ hc->bclk_provider == SND_SOC_TPLG_BCLK_CC ? ++ "codec_consumer" : "codec_provider"); + if (err >= 0 && hc->bclk_rate) + err = tplg_save_printf(dst, pfx, "\tbclk_freq %u\n", + hc->bclk_rate); +@@ -1791,7 +1807,7 @@ static int set_link_hw_config(struct snd_soc_tplg_hw_config *cfg, + cfg->clock_gated = tpl->clock_gated; + cfg->invert_bclk = tpl->invert_bclk; + cfg->invert_fsync = tpl->invert_fsync; +- cfg->bclk_master = tpl->bclk_master; ++ cfg->bclk_provider = tpl->bclk_provider; + cfg->fsync_master = tpl->fsync_master; + cfg->mclk_direction = tpl->mclk_direction; + cfg->reserved = tpl->reserved; +@@ -2174,7 +2190,7 @@ next: + hw->clock_gated = link->hw_config[i].clock_gated; + hw->invert_bclk = link->hw_config[i].invert_bclk; + hw->invert_fsync = link->hw_config[i].invert_fsync; +- hw->bclk_master = link->hw_config[i].bclk_master; ++ hw->bclk_provider = link->hw_config[i].bclk_provider; + hw->fsync_master = link->hw_config[i].fsync_master; + hw->mclk_direction = link->hw_config[i].mclk_direction; + hw->mclk_rate = link->hw_config[i].mclk_rate; +-- +2.29.2 + + +From 706192341d1d0bbb906d690b227b9dee5c1fc4b5 Mon Sep 17 00:00:00 2001 +From: Pierre-Louis Bossart +Date: Thu, 12 Nov 2020 10:29:39 -0600 +Subject: [PATCH 05/49] topology: use inclusive language for fsync + +use fsync_provider for structure fields, 'codec_provider' and +'codec_consumer' for options and modify #defines to use CP and CC +suffixes. + +Signed-off-by: Pierre-Louis Bossart +Signed-off-by: Takashi Iwai +--- + include/sound/uapi/asoc.h | 11 +++++++---- + include/topology.h | 2 +- + src/topology/pcm.c | 37 ++++++++++++++++++++++++++----------- + 3 files changed, 34 insertions(+), 16 deletions(-) + +diff --git a/include/sound/uapi/asoc.h b/include/sound/uapi/asoc.h +index ceafb1a9..f32c5697 100644 +--- a/include/sound/uapi/asoc.h ++++ b/include/sound/uapi/asoc.h +@@ -178,10 +178,13 @@ + #define SND_SOC_TPLG_BCLK_CS SND_SOC_TPLG_BCLK_CC + + /* DAI topology FSYNC parameter - * For the backwards capability, by default codec is fsync master + * For the backwards capability, by default codec is fsync provider */ @@ -69,225 +289,535 @@ index 4efb4ec4..f32c5697 100644 /* * Block Header. -@@ -335,8 +341,8 @@ struct snd_soc_tplg_hw_config { - __u8 clock_gated; /* SND_SOC_TPLG_DAI_CLK_GATE_ value */ +@@ -339,7 +342,7 @@ struct snd_soc_tplg_hw_config { __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ -- __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ + __u8 bclk_provider; /* SND_SOC_TPLG_BCLK_ value */ - __u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ -+ __u8 bclk_provider; /* SND_SOC_TPLG_BCLK_ value */ + __u8 fsync_provider; /* SND_SOC_TPLG_FSYNC_ value */ __u8 mclk_direction; /* SND_SOC_TPLG_MCLK_ value */ __le16 reserved; /* for 32bit alignment */ __le32 mclk_rate; /* MCLK or SYSCLK freqency in Hz */ diff --git a/include/topology.h b/include/topology.h -index 1f52e66e..d1feee4d 100644 +index 6c970649..4ade20df 100644 --- a/include/topology.h +++ b/include/topology.h -@@ -658,8 +658,8 @@ extern "C" { - * - * id "1" # used for binding to the config - * format "I2S" # physical audio format. -- * bclk "master" # Platform is master of bit clock -- * fsync "slave" # Platform is slave of fsync -+ * bclk "codec_provider" # Codec provides the bit clock -+ * fsync "codec_consumer" # Codec follows the fsync - * } - * - * -@@ -1028,8 +1028,8 @@ struct snd_tplg_hw_config_template { - unsigned char clock_gated; /* SND_SOC_TPLG_DAI_CLK_GATE_ value */ +@@ -1029,7 +1029,7 @@ struct snd_tplg_hw_config_template { unsigned char invert_bclk; /* 1 for inverted BCLK, 0 for normal */ unsigned char invert_fsync; /* 1 for inverted frame clock, 0 for normal */ -- unsigned char bclk_master; /* SND_SOC_TPLG_BCLK_ value */ + unsigned char bclk_provider; /* SND_SOC_TPLG_BCLK_ value */ - unsigned char fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ -+ unsigned char bclk_provider; /* SND_SOC_TPLG_BCLK_ value */ + unsigned char fsync_provider; /* SND_SOC_TPLG_FSYNC_ value */ unsigned char mclk_direction; /* SND_SOC_TPLG_MCLK_ value */ unsigned short reserved; /* for 32bit alignment */ unsigned int mclk_rate; /* MCLK or SYSCLK freqency in Hz */ -diff --git a/src/conf.c b/src/conf.c -index 7df2b4e7..3c943db2 100644 ---- a/src/conf.c -+++ b/src/conf.c -@@ -887,7 +887,7 @@ static inline int get_hexachar(input_t *input) - if (c >= '0' && c <= '9') num |= (c - '0') << 0; - else if (c >= 'a' && c <= 'f') num |= (c - 'a') << 0; - else if (c >= 'A' && c <= 'F') num |= (c - 'A') << 0; -- return c; -+ return num; - } +diff --git a/src/topology/pcm.c b/src/topology/pcm.c +index f05df348..c8f41862 100644 +--- a/src/topology/pcm.c ++++ b/src/topology/pcm.c +@@ -1504,8 +1504,15 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, + continue; + } - static int get_quotedchar(input_t *input) -@@ -1970,11 +1970,14 @@ int _snd_config_load_with_include(snd_config_t *config, snd_input_t *in, - SNDERR("%s:%d:%d:%s", fd->name ? fd->name : "_toplevel_", fd->line, fd->column, str); - goto _end; - } -- if (get_char(&input) != LOCAL_UNEXPECTED_EOF) { -+ err = get_char(&input); -+ fd = input.current; -+ if (err != LOCAL_UNEXPECTED_EOF) { - SNDERR("%s:%d:%d:Unexpected }", fd->name ? fd->name : "", fd->line, fd->column); - err = -EINVAL; - goto _end; - } -+ err = 0; - _end: - while (fd->next) { - fd_next = fd->next; -diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf -index 9b64af3c..b1b74b02 100644 ---- a/src/conf/cards/USB-Audio.conf -+++ b/src/conf/cards/USB-Audio.conf -@@ -39,6 +39,7 @@ USB-Audio.pcm.iec958_device { - # "NoiseBlaster 3000" 42 - "USB Sound Blaster HD" 1 - "Xonar U7" 1 -+ "Xonar U7 MKII" 1 - "ASUS XONAR U5" 1 - "XONAR U5" 1 - "XONAR SOUND CARD" 1 -diff --git a/src/confmisc.c b/src/confmisc.c -index eb8218c1..3ce95c7a 100644 ---- a/src/confmisc.c -+++ b/src/confmisc.c -@@ -419,7 +419,6 @@ int snd_func_concat(snd_config_t **dst, snd_config_t *root, snd_config_t *src, - tmp = realloc(res, len + len1 + 1); - if (tmp == NULL) { - free(ptr); -- free(res); - err = -ENOMEM; - goto __error; - } -@@ -440,8 +439,8 @@ int snd_func_concat(snd_config_t **dst, snd_config_t *root, snd_config_t *src, - err = snd_config_get_id(src, &id); - if (err >= 0) - err = snd_config_imake_string(dst, id, res); -- free(res); - __error: -+ free(res); - return err; - } - #ifndef DOC_HIDDEN -diff --git a/src/dlmisc.c b/src/dlmisc.c -index c9517c55..1dd91356 100644 ---- a/src/dlmisc.c -+++ b/src/dlmisc.c -@@ -42,13 +42,11 @@ - #ifndef PIC - struct snd_dlsym_link *snd_dlsym_start = NULL; - #endif --#ifdef DL_ORIGIN_AVAILABLE - static int snd_plugin_dir_set = 0; - static char *snd_plugin_dir = NULL; - #endif --#endif +- if (strcmp(id, "fsync") == 0 || +- strcmp(id, "fsync_master") == 0) { ++ provider_legacy = false; ++ if (strcmp(id, "fsync_master") == 0) { ++ SNDERR("deprecated option %s, please use 'fsync'\n", id); ++ provider_legacy = true; ++ } ++ ++ if (provider_legacy || ++ strcmp(id, "fsync") == 0) { ++ + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; --#if defined(DL_ORIGIN_AVAILABLE) && defined(HAVE_LIBPTHREAD) -+#ifdef HAVE_LIBPTHREAD - static pthread_mutex_t snd_dlpath_mutex = PTHREAD_MUTEX_INITIALIZER; +@@ -1515,11 +1522,19 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, + */ + SNDERR("deprecated fsync value '%s'", val); - static inline void snd_dlpath_lock(void) -@@ -442,12 +440,10 @@ void snd_dlobj_cache_cleanup(void) - free(c); - } - snd_dlobj_unlock(); --#ifdef DL_ORIGIN_AVAILABLE - snd_dlpath_lock(); - snd_plugin_dir_set = 0; - free(snd_plugin_dir); - snd_plugin_dir = NULL; - snd_dlpath_unlock(); --#endif - } - #endif +- hw_cfg->fsync_master = SND_SOC_TPLG_FSYNC_CS; ++ hw_cfg->fsync_provider = SND_SOC_TPLG_FSYNC_CC; + } else if (!strcmp(val, "codec_slave")) { +- hw_cfg->fsync_master = SND_SOC_TPLG_FSYNC_CS; ++ SNDERR("deprecated fsync value '%s', use 'codec_consumer'", val); ++ ++ hw_cfg->fsync_provider = SND_SOC_TPLG_FSYNC_CC; ++ } else if (!strcmp(val, "codec_consumer")) { ++ hw_cfg->fsync_provider = SND_SOC_TPLG_FSYNC_CC; + } else if (!strcmp(val, "codec_master")) { +- hw_cfg->fsync_master = SND_SOC_TPLG_FSYNC_CM; ++ SNDERR("deprecated fsync value '%s', use 'codec_provider'", val); ++ ++ hw_cfg->fsync_provider = SND_SOC_TPLG_FSYNC_CP; ++ } else if (!strcmp(val, "codec_provider")) { ++ hw_cfg->fsync_provider = SND_SOC_TPLG_FSYNC_CP; + } + continue; + } +@@ -1648,10 +1663,10 @@ int tplg_save_hw_config(snd_tplg_t *tplg ATTRIBUTE_UNUSED, + hc->bclk_rate); + if (err >= 0 && hc->invert_bclk) + err = tplg_save_printf(dst, pfx, "\tbclk_invert 1\n"); +- if (err >= 0 && hc->fsync_master) +- err = tplg_save_printf(dst, pfx, "\tfsync_master '%s'\n", +- hc->fsync_master == SND_SOC_TPLG_FSYNC_CS ? +- "codec_slave" : "codec_master"); ++ if (err >= 0 && hc->fsync_provider) ++ err = tplg_save_printf(dst, pfx, "\tfsync_provider '%s'\n", ++ hc->fsync_provider == SND_SOC_TPLG_FSYNC_CC ? ++ "codec_consumer" : "codec_provider"); + if (err >= 0 && hc->fsync_rate) + err = tplg_save_printf(dst, pfx, "\tfsync_freq %u\n", + hc->fsync_rate); +@@ -1808,7 +1823,7 @@ static int set_link_hw_config(struct snd_soc_tplg_hw_config *cfg, + cfg->invert_bclk = tpl->invert_bclk; + cfg->invert_fsync = tpl->invert_fsync; + cfg->bclk_provider = tpl->bclk_provider; +- cfg->fsync_master = tpl->fsync_master; ++ cfg->fsync_provider = tpl->fsync_provider; + cfg->mclk_direction = tpl->mclk_direction; + cfg->reserved = tpl->reserved; + cfg->mclk_rate = tpl->mclk_rate; +@@ -2191,7 +2206,7 @@ next: + hw->invert_bclk = link->hw_config[i].invert_bclk; + hw->invert_fsync = link->hw_config[i].invert_fsync; + hw->bclk_provider = link->hw_config[i].bclk_provider; +- hw->fsync_master = link->hw_config[i].fsync_master; ++ hw->fsync_provider = link->hw_config[i].fsync_provider; + hw->mclk_direction = link->hw_config[i].mclk_direction; + hw->mclk_rate = link->hw_config[i].mclk_rate; + hw->bclk_rate = link->hw_config[i].bclk_rate; +-- +2.29.2 + + +From e5c350d7bc6f3d45702059c0ae8d32b3603273c1 Mon Sep 17 00:00:00 2001 +From: Pierre-Louis Bossart +Date: Thu, 12 Nov 2020 10:29:40 -0600 +Subject: [PATCH 06/49] topology: use inclusive language in documentation + +Use codec_provider and codec_consumer for bclk and fsync + +Signed-off-by: Pierre-Louis Bossart +Signed-off-by: Takashi Iwai +--- + include/topology.h | 4 ++-- + 1 file changed, 2 insertions(+), 2 deletions(-) + +diff --git a/include/topology.h b/include/topology.h +index 4ade20df..d1feee4d 100644 +--- a/include/topology.h ++++ b/include/topology.h +@@ -658,8 +658,8 @@ extern "C" { + * + * id "1" # used for binding to the config + * format "I2S" # physical audio format. +- * bclk "master" # Platform is master of bit clock +- * fsync "slave" # Platform is slave of fsync ++ * bclk "codec_provider" # Codec provides the bit clock ++ * fsync "codec_consumer" # Codec follows the fsync + * } + * + * +-- +2.29.2 + + +From 7d36895225d09ae26868997382445cc7ad2e9513 Mon Sep 17 00:00:00 2001 +From: Jonas Holmberg +Date: Wed, 4 Nov 2020 14:49:56 +0100 +Subject: [PATCH 07/49] pcm: set the snd_pcm_ioplug_status() tstamp field + +Set the status tstamp field so that it can be accessed with +snd_pcm_status_get_htstamp(). + +Signed-off-by: Jonas Holmberg +Signed-off-by: Takashi Iwai +--- + src/pcm/pcm_ioplug.c | 1 + + 1 file changed, 1 insertion(+) + diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c -index a437ca32..e141b1f9 100644 +index a437ca32..9b1b8ac3 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c -@@ -107,16 +107,35 @@ static int snd_pcm_ioplug_channel_info(snd_pcm_t *pcm, snd_pcm_channel_info_t *i - return snd_pcm_channel_info_shm(pcm, info, -1); - } - -+static int snd_pcm_ioplug_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) -+{ -+ ioplug_priv_t *io = pcm->private_data; -+ -+ if (io->data->version >= 0x010001 && -+ io->data->callback->delay) -+ return io->data->callback->delay(io->data, delayp); -+ else { -+ snd_pcm_ioplug_hw_ptr_update(pcm); -+ *delayp = snd_pcm_mmap_delay(pcm); -+ } -+ return 0; -+} -+ - static int snd_pcm_ioplug_status(snd_pcm_t *pcm, snd_pcm_status_t * status) - { - ioplug_priv_t *io = pcm->private_data; -+ snd_pcm_sframes_t sd; - - memset(status, 0, sizeof(*status)); +@@ -115,6 +115,7 @@ static int snd_pcm_ioplug_status(snd_pcm_t *pcm, snd_pcm_status_t * status) snd_pcm_ioplug_hw_ptr_update(pcm); status->state = io->data->state; status->trigger_tstamp = io->trigger_tstamp; + gettimestamp(&status->tstamp, pcm->tstamp_type); status->avail = snd_pcm_mmap_avail(pcm); status->avail_max = io->avail_max; -+ if (snd_pcm_ioplug_delay(pcm, &sd) < 0) -+ sd = snd_pcm_mmap_delay(pcm); -+ status->delay = sd; return 0; +-- +2.29.2 + + +From 6d3311fefb1b41c7484fd728e4ad4a6fb5b55ad4 Mon Sep 17 00:00:00 2001 +From: Takashi Iwai +Date: Thu, 10 Dec 2020 10:36:41 +0100 +Subject: [PATCH 08/49] Makefile: Add README.md to EXTRA_DIST + +Signed-off-by: Takashi Iwai +--- + Makefile.am | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +diff --git a/Makefile.am b/Makefile.am +index c484d4da..ff4c963a 100644 +--- a/Makefile.am ++++ b/Makefile.am +@@ -16,7 +16,7 @@ SUBDIRS += alsalisp + endif + endif + SUBDIRS += test utils +-EXTRA_DIST=ChangeLog INSTALL TODO NOTES configure gitcompile libtool \ ++EXTRA_DIST=README.md ChangeLog INSTALL TODO NOTES configure gitcompile libtool \ + depcomp version MEMORY-LEAK m4/attributes.m4 + AUTOMAKE_OPTIONS=foreign + +-- +2.29.2 + + +From ea02dbd20a535085638c63c6b8bec94e628486c4 Mon Sep 17 00:00:00 2001 +From: David Henningsson +Date: Wed, 16 Dec 2020 19:55:22 +0100 +Subject: [PATCH 09/49] pcm: Add snd_pcm_audio_tstamp_type_t constants + +These are mostly a copy-paste from the kernel headers. But since functions +snd_pcm_audio_tstamp_config make use of these they should be added to the +public API as well. + +Reported-by: Alex Moon +Signed-off-by: David Henningsson +Signed-off-by: Takashi Iwai +--- + include/pcm.h | 14 ++++++++++++++ + 1 file changed, 14 insertions(+) + +diff --git a/include/pcm.h b/include/pcm.h +index 5b078231..cf1eea8b 100644 +--- a/include/pcm.h ++++ b/include/pcm.h +@@ -350,6 +350,20 @@ typedef enum _snd_pcm_tstamp_type { + SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, + } snd_pcm_tstamp_type_t; + ++typedef enum _snd_pcm_audio_tstamp_type { ++ /** ++ * first definition for backwards compatibility only, ++ * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else ++ */ ++ SND_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0, ++ SND_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /**< DMA time, reported as per hw_ptr */ ++ SND_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /**< link time reported by sample or wallclock counter, reset on startup */ ++ SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /**< link time reported by sample or wallclock counter, not reset on startup */ ++ SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /**< link time estimated indirectly */ ++ SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /**< link time synchronized with system time */ ++ SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED ++} snd_pcm_audio_tstamp_type_t; ++ + typedef struct _snd_pcm_audio_tstamp_config { + /* 5 of max 16 bits used */ + unsigned int type_requested:4; +-- +2.29.2 + + +From dcda999d0000cbdabe61d3d175943b5f23c30057 Mon Sep 17 00:00:00 2001 +From: David Henningsson +Date: Wed, 16 Dec 2020 20:02:52 +0100 +Subject: [PATCH 10/49] test/audio_time: Make use of SND_PCM_AUDIO_TSTAMP_TYPE + constants + +Also fixup the usage row which seemed wrong to me. + +Signed-off-by: David Henningsson +Signed-off-by: Takashi Iwai +--- + test/audio_time.c | 26 +++++++++++++------------- + 1 file changed, 13 insertions(+), 13 deletions(-) + +diff --git a/test/audio_time.c b/test/audio_time.c +index 530922d9..e4d4a944 100644 +--- a/test/audio_time.c ++++ b/test/audio_time.c +@@ -32,7 +32,7 @@ static void usage(char *command) + "-d, --delay add delay \n" + "-D, --device=NAME select PCM by name \n" + "-p, --playback playback tstamps \n" +- "-t, --ts_type=TYPE Default(0),link(1),link_estimated(2),synchronized(3) \n" ++ "-t, --ts_type=TYPE Compat(0),default(1),link(2),link_absolute(3),link_estimated(4),link_synchronized(5) \n" + "-r, --report show audio timestamp and accuracy validity\n" + , command); } +@@ -201,17 +201,17 @@ int main(int argc, char *argv[]) + goto _exit; + } -@@ -132,20 +151,6 @@ static int snd_pcm_ioplug_hwsync(snd_pcm_t *pcm) - return 0; +- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, 0)) ++ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, SND_PCM_AUDIO_TSTAMP_TYPE_COMPAT)) + printf("Playback supports audio compat timestamps\n"); +- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, 1)) ++ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, SND_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)) + printf("Playback supports audio default timestamps\n"); +- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, 2)) ++ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, SND_PCM_AUDIO_TSTAMP_TYPE_LINK)) + printf("Playback supports audio link timestamps\n"); +- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, 3)) ++ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE)) + printf("Playback supports audio link absolute timestamps\n"); +- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, 4)) ++ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED)) + printf("Playback supports audio link estimated timestamps\n"); +- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, 5)) ++ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED)) + printf("Playback supports audio link synchronized timestamps\n"); + + snd_pcm_sw_params_alloca(&swparams_p); +@@ -269,17 +269,17 @@ int main(int argc, char *argv[]) + goto _exit; + } + +- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, 0)) ++ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, SND_PCM_AUDIO_TSTAMP_TYPE_COMPAT)) + printf("Capture supports audio compat timestamps\n"); +- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, 1)) ++ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, SND_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)) + printf("Capture supports audio default timestamps\n"); +- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, 2)) ++ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, SND_PCM_AUDIO_TSTAMP_TYPE_LINK)) + printf("Capture supports audio link timestamps\n"); +- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, 3)) ++ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE)) + printf("Capture supports audio link absolute timestamps\n"); +- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, 4)) ++ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED)) + printf("Capture supports audio link estimated timestamps\n"); +- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, 5)) ++ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED)) + printf("Capture supports audio link synchronized timestamps\n"); + + snd_pcm_sw_params_alloca(&swparams_c); +-- +2.29.2 + + +From a9cbfecd87777d76a83e5a912ca9bdcf48e31fa9 Mon Sep 17 00:00:00 2001 +From: Takashi Iwai +Date: Tue, 22 Dec 2020 16:10:16 +0100 +Subject: [PATCH 11/49] pcm: Fix a typo in SND_PCM_AUDIO_TSTAMP_TYPE_LAST + definition + +It was wrongly defined with SNDRV_ prefix. Fix it. + +Fixes: ea02dbd20a53 ("pcm: Add snd_pcm_audio_tstamp_type_t constants") +Signed-off-by: Takashi Iwai +--- + include/pcm.h | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +diff --git a/include/pcm.h b/include/pcm.h +index cf1eea8b..e300b951 100644 +--- a/include/pcm.h ++++ b/include/pcm.h +@@ -361,7 +361,7 @@ typedef enum _snd_pcm_audio_tstamp_type { + SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /**< link time reported by sample or wallclock counter, not reset on startup */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /**< link time estimated indirectly */ + SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /**< link time synchronized with system time */ +- SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED ++ SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED + } snd_pcm_audio_tstamp_type_t; + + typedef struct _snd_pcm_audio_tstamp_config { +-- +2.29.2 + + +From 5275d170e05639473e68e5feb349aec65bf26428 Mon Sep 17 00:00:00 2001 +From: Alex Henrie +Date: Sat, 26 Dec 2020 14:35:39 -0700 +Subject: [PATCH 12/49] conf: fix use after free in + _snd_config_load_with_include + +Signed-off-by: Alex Henrie +Signed-off-by: Takashi Iwai +--- + src/conf.c | 4 +++- + 1 file changed, 3 insertions(+), 1 deletion(-) + +diff --git a/src/conf.c b/src/conf.c +index 7df2b4e7..44d1bfde 100644 +--- a/src/conf.c ++++ b/src/conf.c +@@ -1970,7 +1970,9 @@ int _snd_config_load_with_include(snd_config_t *config, snd_input_t *in, + SNDERR("%s:%d:%d:%s", fd->name ? fd->name : "_toplevel_", fd->line, fd->column, str); + goto _end; + } +- if (get_char(&input) != LOCAL_UNEXPECTED_EOF) { ++ err = get_char(&input); ++ fd = input.current; ++ if (err != LOCAL_UNEXPECTED_EOF) { + SNDERR("%s:%d:%d:Unexpected }", fd->name ? fd->name : "", fd->line, fd->column); + err = -EINVAL; + goto _end; +-- +2.29.2 + + +From 0f4821ec1729cd72a3dde3ea63f7857e92a10919 Mon Sep 17 00:00:00 2001 +From: Alex Henrie +Date: Sat, 26 Dec 2020 14:35:40 -0700 +Subject: [PATCH 13/49] ucm: fix bad frees in get_list0 and get_list20 + +Signed-off-by: Alex Henrie +Signed-off-by: Takashi Iwai +--- + src/ucm/main.c | 4 ++-- + 1 file changed, 2 insertions(+), 2 deletions(-) + +diff --git a/src/ucm/main.c b/src/ucm/main.c +index 3871d5aa..754b967e 100644 +--- a/src/ucm/main.c ++++ b/src/ucm/main.c +@@ -666,7 +666,7 @@ static int get_list0(struct list_head *list, + } + return cnt; + __fail: +- snd_use_case_free_list((const char **)res, cnt); ++ snd_use_case_free_list(*result, cnt); + return -ENOMEM; } --static int snd_pcm_ioplug_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) --{ -- ioplug_priv_t *io = pcm->private_data; -- -- if (io->data->version >= 0x010001 && -- io->data->callback->delay) -- return io->data->callback->delay(io->data, delayp); -- else { -- snd_pcm_ioplug_hw_ptr_update(pcm); -- *delayp = snd_pcm_mmap_hw_avail(pcm); -- } -- return 0; --} -- - static int snd_pcm_ioplug_reset(snd_pcm_t *pcm) - { - ioplug_priv_t *io = pcm->private_data; -diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h -index fe77e50d..bec5a408 100644 ---- a/src/pcm/pcm_local.h -+++ b/src/pcm/pcm_local.h -@@ -444,7 +444,7 @@ static inline int __snd_pcm_start(snd_pcm_t *pcm) - static inline snd_pcm_state_t __snd_pcm_state(snd_pcm_t *pcm) +@@ -724,7 +724,7 @@ static int get_list20(struct list_head *list, + } + return cnt; + __fail: +- snd_use_case_free_list((const char **)res, cnt); ++ snd_use_case_free_list(*result, cnt); + return -ENOMEM; + } + +-- +2.29.2 + + +From c56278cfda55b9f5d106890c7354176dba8c8f4d Mon Sep 17 00:00:00 2001 +From: Alex Henrie +Date: Sat, 26 Dec 2020 14:35:41 -0700 +Subject: [PATCH 14/49] rawmidi: fix memory leak in snd_rawmidi_virtual_open + +Signed-off-by: Alex Henrie +Signed-off-by: Takashi Iwai +--- + src/rawmidi/rawmidi_virt.c | 3 ++- + 1 file changed, 2 insertions(+), 1 deletion(-) + +diff --git a/src/rawmidi/rawmidi_virt.c b/src/rawmidi/rawmidi_virt.c +index 2c4c27f5..884b8ff8 100644 +--- a/src/rawmidi/rawmidi_virt.c ++++ b/src/rawmidi/rawmidi_virt.c +@@ -315,7 +315,7 @@ int snd_rawmidi_virtual_open(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp, + int merge, int mode) { - if (!pcm->fast_ops->state) -- return -ENOSYS; -+ return SND_PCM_STATE_OPEN; - return pcm->fast_ops->state(pcm->fast_op_arg); + int err; +- snd_rawmidi_t *rmidi; ++ snd_rawmidi_t *rmidi = NULL; + snd_rawmidi_virtual_t *virt = NULL; + struct pollfd pfd; + +@@ -392,6 +392,7 @@ int snd_rawmidi_virtual_open(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp, + free(*inputp); + if (outputp) + free(*outputp); ++ free(rmidi); + return err; } -diff --git a/src/pcm/pcm_mmap.c b/src/pcm/pcm_mmap.c -index 9600c383..9cbaae05 100644 ---- a/src/pcm/pcm_mmap.c -+++ b/src/pcm/pcm_mmap.c -@@ -183,7 +183,7 @@ snd_pcm_sframes_t snd_pcm_mmap_writen(snd_pcm_t *pcm, void **bufs, snd_pcm_ufram - * \brief Read interleaved frames from a PCM using direct buffer (mmap) - * \param pcm PCM handle - * \param buffer frames containing buffer -- * \param size frames to be written -+ * \param size frames to be read - * \return a positive number of frames actually read otherwise a - * negative error code - * \retval -EBADFD PCM is not in the right state (#SND_PCM_STATE_PREPARED or #SND_PCM_STATE_RUNNING) +-- +2.29.2 + + +From aa28847248f2f416d9ecc6cbfa75192455400f6f Mon Sep 17 00:00:00 2001 +From: Alex Henrie +Date: Sat, 26 Dec 2020 14:35:43 -0700 +Subject: [PATCH 15/49] timer: fix sizeof operator mismatch in + snd_timer_query_hw_open + +Signed-off-by: Alex Henrie +Signed-off-by: Takashi Iwai +--- + src/timer/timer_query_hw.c | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +diff --git a/src/timer/timer_query_hw.c b/src/timer/timer_query_hw.c +index dad228c8..d8bac6e7 100644 +--- a/src/timer/timer_query_hw.c ++++ b/src/timer/timer_query_hw.c +@@ -104,7 +104,7 @@ int snd_timer_query_hw_open(snd_timer_query_t **handle, const char *name, int mo + close(fd); + return -SND_ERROR_INCOMPATIBLE_VERSION; + } +- tmr = (snd_timer_query_t *) calloc(1, sizeof(snd_timer_t)); ++ tmr = (snd_timer_query_t *) calloc(1, sizeof(snd_timer_query_t)); + if (tmr == NULL) { + close(fd); + return -ENOMEM; +-- +2.29.2 + + +From 2e470d59d84a563a9efa81572ddc48d9858ceb92 Mon Sep 17 00:00:00 2001 +From: Alex Henrie +Date: Sat, 26 Dec 2020 14:35:44 -0700 +Subject: [PATCH 16/49] pcm: remove dead assignments from + snd_pcm_rate_(commit_area|grab_next_period) + +Signed-off-by: Alex Henrie +Signed-off-by: Takashi Iwai +--- + src/pcm/pcm_rate.c | 2 -- + 1 file changed, 2 deletions(-) + +diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c +index 5bf7dbb9..dc38e95e 100644 +--- a/src/pcm/pcm_rate.c ++++ b/src/pcm/pcm_rate.c +@@ -746,7 +746,6 @@ static int snd_pcm_rate_commit_area(snd_pcm_t *pcm, snd_pcm_rate_t *rate, + if (result < 0) + return result; + __partial: +- xfer = 0; + cont = slave_frames; + if (cont > slave_size) + cont = slave_size; +@@ -846,7 +845,6 @@ static int snd_pcm_rate_grab_next_period(snd_pcm_t *pcm, snd_pcm_uframes_t hw_of + if (result < 0) + return result; + __partial: +- xfer = 0; + cont = slave_frames; + if (cont > rate->gen.slave->period_size) + cont = rate->gen.slave->period_size; +-- +2.29.2 + + +From fc719bfe4e97bbfce899cc7e67df08ecd573c5d3 Mon Sep 17 00:00:00 2001 +From: Alex Henrie +Date: Sat, 26 Dec 2020 14:35:45 -0700 +Subject: [PATCH 17/49] pcm_multi: remove dead assignment from + _snd_pcm_multi_open + +Signed-off-by: Alex Henrie +Signed-off-by: Takashi Iwai +--- + src/pcm/pcm_multi.c | 1 - + 1 file changed, 1 deletion(-) + diff --git a/src/pcm/pcm_multi.c b/src/pcm/pcm_multi.c index 53c414d5..5fa09b9b 100644 --- a/src/pcm/pcm_multi.c @@ -300,24 +830,237 @@ index 53c414d5..5fa09b9b 100644 for (idx = 0; idx < channels_count; ++idx) channels_sidx[idx] = -1; idx = 0; +-- +2.29.2 + + +From 9ebd29d2d31df0f8967f3e60cf24bb63c95d4984 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Sun, 27 Dec 2020 13:17:26 +0100 +Subject: [PATCH 18/49] conf: fix get_hexachar() return value + +Reported-by: Alex Henrie +Signed-off-by: Jaroslav Kysela +--- + src/conf.c | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +diff --git a/src/conf.c b/src/conf.c +index 44d1bfde..6cfe65bf 100644 +--- a/src/conf.c ++++ b/src/conf.c +@@ -887,7 +887,7 @@ static inline int get_hexachar(input_t *input) + if (c >= '0' && c <= '9') num |= (c - '0') << 0; + else if (c >= 'a' && c <= 'f') num |= (c - 'a') << 0; + else if (c >= 'A' && c <= 'F') num |= (c - 'A') << 0; +- return c; ++ return num; + } + + static int get_quotedchar(input_t *input) +-- +2.29.2 + + +From 36aff79747b23b9535e81befe8b7b2972837bce3 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Sun, 27 Dec 2020 13:29:33 +0100 +Subject: [PATCH 19/49] pcm: fix __snd_pcm_state() return value + +The __snd_pcm_state() must return a valid state, not an error value +when the plugin callback is not defined. Use the first state +SND_PCM_STATE_OPEN - the other functions will return the error +code depending on this state. + +Link: https://lore.kernel.org/alsa-devel/20201226213547.175071-10-alexhenrie24@gmail.com/ +Reported-by: Alex Henrie +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_local.h | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h +index fe77e50d..bec5a408 100644 +--- a/src/pcm/pcm_local.h ++++ b/src/pcm/pcm_local.h +@@ -444,7 +444,7 @@ static inline int __snd_pcm_start(snd_pcm_t *pcm) + static inline snd_pcm_state_t __snd_pcm_state(snd_pcm_t *pcm) + { + if (!pcm->fast_ops->state) +- return -ENOSYS; ++ return SND_PCM_STATE_OPEN; + return pcm->fast_ops->state(pcm->fast_op_arg); + } + +-- +2.29.2 + + +From a92ad2fea4b36d7282f4a571275fb492dd1bba75 Mon Sep 17 00:00:00 2001 +From: Alex Henrie +Date: Sun, 27 Dec 2020 18:43:15 -0700 +Subject: [PATCH 20/49] confmisc: fix memory leak in snd_func_concat + +Signed-off-by: Alex Henrie +Signed-off-by: Takashi Iwai +--- + src/confmisc.c | 3 +-- + 1 file changed, 1 insertion(+), 2 deletions(-) + +diff --git a/src/confmisc.c b/src/confmisc.c +index eb8218c1..3ce95c7a 100644 +--- a/src/confmisc.c ++++ b/src/confmisc.c +@@ -419,7 +419,6 @@ int snd_func_concat(snd_config_t **dst, snd_config_t *root, snd_config_t *src, + tmp = realloc(res, len + len1 + 1); + if (tmp == NULL) { + free(ptr); +- free(res); + err = -ENOMEM; + goto __error; + } +@@ -440,8 +439,8 @@ int snd_func_concat(snd_config_t **dst, snd_config_t *root, snd_config_t *src, + err = snd_config_get_id(src, &id); + if (err >= 0) + err = snd_config_imake_string(dst, id, res); +- free(res); + __error: ++ free(res); + return err; + } + #ifndef DOC_HIDDEN +-- +2.29.2 + + +From 6c24cd2e60cc9c1fca809ffeeb7ffe8af94b201e Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Tue, 29 Dec 2020 11:15:46 +0100 +Subject: [PATCH 21/49] conf: fix return code in _snd_config_load_with_include + +Fixes: 5275d170e0 ("conf: fix use after free in _snd_config_load_with_include") +BugLink: https://github.com/alsa-project/alsa-lib/issues/108 +Signed-off-by: Jaroslav Kysela +--- + src/conf.c | 1 + + 1 file changed, 1 insertion(+) + +diff --git a/src/conf.c b/src/conf.c +index 6cfe65bf..3c943db2 100644 +--- a/src/conf.c ++++ b/src/conf.c +@@ -1977,6 +1977,7 @@ int _snd_config_load_with_include(snd_config_t *config, snd_input_t *in, + err = -EINVAL; + goto _end; + } ++ err = 0; + _end: + while (fd->next) { + fd_next = fd->next; +-- +2.29.2 + + +From afe6ff3b33ee6e5ea3511fe458bfd4e516b10bcf Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Wed, 30 Dec 2020 09:31:10 +0100 +Subject: [PATCH 22/49] pcm: plugin status - fix the return value (regression) + +The snd_pcm_plugin_avail_update() error code in snd_pcm_plugin_status() +should not be reported to the caller. The state errors can be determined +using the state member in the status structure. + +Fixes: 4f90392f07e ("pcm: fix the snd_pcm_plugin_status() avail and delay fields") +BugLink: https://github.com/alsa-project/alsa-lib/issues/107 +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_plugin.c | 8 +++----- + 1 file changed, 3 insertions(+), 5 deletions(-) + diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c -index 5739cfc2..7ed6f25a 100644 +index 5739cfc2..76a524fa 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c -@@ -142,12 +142,6 @@ static int snd_pcm_plugin_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) - int err = snd_pcm_delay(plugin->gen.slave, &sd); +@@ -541,19 +541,17 @@ static snd_pcm_sframes_t snd_pcm_plugin_avail_update(snd_pcm_t *pcm) + static int snd_pcm_plugin_status(snd_pcm_t *pcm, snd_pcm_status_t * status) + { + snd_pcm_plugin_t *plugin = pcm->private_data; +- snd_pcm_sframes_t err, avail; ++ snd_pcm_sframes_t err; + + /* sync with the latest hw and appl ptrs */ +- avail = snd_pcm_plugin_avail_update(pcm); +- if (avail < 0) +- return avail; ++ snd_pcm_plugin_avail_update(pcm); + + err = snd_pcm_status(plugin->gen.slave, status); if (err < 0) return err; -- if (pcm->stream == SND_PCM_STREAM_CAPTURE && -- pcm->access != SND_PCM_ACCESS_RW_INTERLEAVED && -- pcm->access != SND_PCM_ACCESS_RW_NONINTERLEAVED) { -- sd += snd_pcm_mmap_capture_avail(pcm); -- } -- - *delayp = sd; + status->appl_ptr = *pcm->appl.ptr; + status->hw_ptr = *pcm->hw.ptr; +- status->avail = avail; ++ status->avail = snd_pcm_mmap_avail(pcm); + status->delay = snd_pcm_mmap_delay(pcm); + return 0; + } +-- +2.29.2 + + +From 49eea5d7bc7d39f9a4cc1f3b3a813e90b90519eb Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Sun, 3 Jan 2021 16:16:10 +0100 +Subject: [PATCH 23/49] pcm: plugin status - revert the recent changes + +It's no reason to sync the avail/delay fields using the mirrored +buffer pointers. The slave information must be valid. + +The original report probably tries to fix something for +the specific plugin. Revert all changes. + +Fixes: afe6ff3b33e ("pcm: plugin status - fix the return value (regression)") +Fixes: 4f90392f07e ("pcm: fix the snd_pcm_plugin_status() avail and delay fields") +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_plugin.c | 2 -- + 1 file changed, 2 deletions(-) + +diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c +index 76a524fa..ea60eb98 100644 +--- a/src/pcm/pcm_plugin.c ++++ b/src/pcm/pcm_plugin.c +@@ -551,8 +551,6 @@ static int snd_pcm_plugin_status(snd_pcm_t *pcm, snd_pcm_status_t * status) + return err; + status->appl_ptr = *pcm->appl.ptr; + status->hw_ptr = *pcm->hw.ptr; +- status->avail = snd_pcm_mmap_avail(pcm); +- status->delay = snd_pcm_mmap_delay(pcm); return 0; } -@@ -460,101 +454,121 @@ snd_pcm_plugin_mmap_commit(snd_pcm_t *pcm, + +-- +2.29.2 + + +From fa1895aa2b4f3f154e537bee92860fe793045643 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Sun, 3 Jan 2021 16:34:04 +0100 +Subject: [PATCH 24/49] pcm: plugin - tidy snd_pcm_plugin_avail_update() + +No functional changes - move the code to snd_pcm_plugin_sync_hw_ptr() +and put the mmap capture updates to separate function for readability. + +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_plugin.c | 142 +++++++++++++++++++++++-------------------- + 1 file changed, 76 insertions(+), 66 deletions(-) + +diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c +index ea60eb98..83793397 100644 +--- a/src/pcm/pcm_plugin.c ++++ b/src/pcm/pcm_plugin.c +@@ -460,82 +460,92 @@ snd_pcm_plugin_mmap_commit(snd_pcm_t *pcm, return xfer > 0 ? xfer : err; } @@ -476,123 +1219,180 @@ index 5739cfc2..7ed6f25a 100644 } static int snd_pcm_plugin_status(snd_pcm_t *pcm, snd_pcm_status_t * status) - { +-- +2.29.2 + + +From 28cc099d9ea3962b033cb1cb9c3e07db828d9ff7 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Sun, 3 Jan 2021 16:41:29 +0100 +Subject: [PATCH 25/49] pcm: plugin - optimize sync in snd_pcm_plugin_status() + +Do hw_ptr sync only once after the status call. This avoids +double update. + +Also, the application pointer must not change when +the status is called. Add assert() call for this condition. + +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_plugin.c | 7 ++----- + 1 file changed, 2 insertions(+), 5 deletions(-) + +diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c +index 83793397..6a815145 100644 +--- a/src/pcm/pcm_plugin.c ++++ b/src/pcm/pcm_plugin.c +@@ -553,14 +553,11 @@ static int snd_pcm_plugin_status(snd_pcm_t *pcm, snd_pcm_status_t * status) snd_pcm_plugin_t *plugin = pcm->private_data; -- snd_pcm_sframes_t err, avail; -- -- /* sync with the latest hw and appl ptrs */ -- avail = snd_pcm_plugin_avail_update(pcm); -- if (avail < 0) -- return avail; -+ snd_pcm_sframes_t err, diff; + snd_pcm_sframes_t err; +- /* sync with the latest hw and appl ptrs */ +- snd_pcm_plugin_avail_update(pcm); +- err = snd_pcm_status(plugin->gen.slave, status); if (err < 0) return err; - status->appl_ptr = *pcm->appl.ptr; - status->hw_ptr = *pcm->hw.ptr; -- status->avail = avail; -- status->delay = snd_pcm_mmap_delay(pcm); ++ assert(status->appl_ptr == *pcm->appl.ptr); + snd_pcm_plugin_sync_hw_ptr(pcm, status->hw_ptr, status->avail); -+ /* -+ * For capture stream, the situation is more complicated, because -+ * snd_pcm_plugin_avail_update() commits the data to the slave pcm. -+ * It means that the slave appl_ptr is updated. Calculate diff and -+ * update the delay and avail. -+ * -+ * This resolves the data inconsistency for immediate calls: -+ * snd_pcm_avail_update() -+ * snd_pcm_status() -+ */ -+ if (pcm->stream == SND_PCM_STREAM_CAPTURE) { -+ status->appl_ptr = *pcm->appl.ptr; -+ diff = pcm_frame_diff(status->appl_ptr, *pcm->appl.ptr, pcm->boundary); -+ status->avail += diff; -+ status->delay += diff; -+ } else { -+ assert(status->appl_ptr == *pcm->appl.ptr); -+ } return 0; } -diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c -index 5bf7dbb9..770aafea 100644 ---- a/src/pcm/pcm_rate.c -+++ b/src/pcm/pcm_rate.c -@@ -561,17 +561,16 @@ snd_pcm_rate_read_areas1(snd_pcm_t *pcm, - - static inline void snd_pcm_rate_sync_hwptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr) - { -- snd_pcm_rate_t *rate = pcm->private_data; +-- +2.29.2 + + +From 21549e6583f1dfe5b2fe1aaa88345e4645dd96d0 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Wed, 30 Dec 2020 17:49:33 +0100 +Subject: [PATCH 26/49] Revert "pcm_plugin: fix delay" + +This reverts commit aba87e509898ec9ddb3e319267d7c267409ff100. + +The commit does not look good. The plugins must handle +the delay value correctly. + +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_plugin.c | 6 ------ + 1 file changed, 6 deletions(-) + +diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c +index 6a815145..5787a43d 100644 +--- a/src/pcm/pcm_plugin.c ++++ b/src/pcm/pcm_plugin.c +@@ -142,12 +142,6 @@ static int snd_pcm_plugin_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) + int err = snd_pcm_delay(plugin->gen.slave, &sd); + if (err < 0) + return err; +- if (pcm->stream == SND_PCM_STREAM_CAPTURE && +- pcm->access != SND_PCM_ACCESS_RW_INTERLEAVED && +- pcm->access != SND_PCM_ACCESS_RW_NONINTERLEAVED) { +- sd += snd_pcm_mmap_capture_avail(pcm); +- } - -- snd_pcm_sframes_t slave_hw_ptr_diff = slave_hw_ptr - rate->last_slave_hw_ptr; -+ snd_pcm_rate_t *rate; -+ snd_pcm_sframes_t slave_hw_ptr_diff; - snd_pcm_sframes_t last_slave_hw_ptr_frac; - - if (pcm->stream != SND_PCM_STREAM_PLAYBACK) - return; + *delayp = sd; + return 0; + } +-- +2.29.2 + + +From 6ca01c07ee13435d6c2db4e9121d9a86cc4b1457 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Wed, 30 Dec 2020 19:14:25 +0100 +Subject: [PATCH 27/49] pcm: ioplug - fix the delay calculation in the status + callback + +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_ioplug.c | 32 ++++++++++++++++++-------------- + 1 file changed, 18 insertions(+), 14 deletions(-) + +diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c +index 9b1b8ac3..f2315a10 100644 +--- a/src/pcm/pcm_ioplug.c ++++ b/src/pcm/pcm_ioplug.c +@@ -107,9 +107,24 @@ static int snd_pcm_ioplug_channel_info(snd_pcm_t *pcm, snd_pcm_channel_info_t *i + return snd_pcm_channel_info_shm(pcm, info, -1); + } -- if (slave_hw_ptr_diff < 0) -- slave_hw_ptr_diff += rate->gen.slave->boundary; /* slave boundary wraparound */ -- else if (slave_hw_ptr_diff == 0) -+ rate = pcm->private_data; -+ slave_hw_ptr_diff = pcm_frame_diff(slave_hw_ptr, rate->last_slave_hw_ptr, rate->gen.slave->boundary); -+ if (slave_hw_ptr_diff == 0) - return; - last_slave_hw_ptr_frac = rate->last_slave_hw_ptr % rate->gen.slave->period_size; - /* While handling fraction part fo slave period, rounded value will be -@@ -612,11 +611,7 @@ static snd_pcm_uframes_t snd_pcm_rate_playback_internal_delay(snd_pcm_t *pcm) ++static int snd_pcm_ioplug_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) ++{ ++ ioplug_priv_t *io = pcm->private_data; ++ ++ if (io->data->version >= 0x010001 && ++ io->data->callback->delay) ++ return io->data->callback->delay(io->data, delayp); ++ else { ++ snd_pcm_ioplug_hw_ptr_update(pcm); ++ *delayp = snd_pcm_mmap_hw_avail(pcm); ++ } ++ return 0; ++} ++ + static int snd_pcm_ioplug_status(snd_pcm_t *pcm, snd_pcm_status_t * status) { - snd_pcm_rate_t *rate = pcm->private_data; + ioplug_priv_t *io = pcm->private_data; ++ snd_pcm_sframes_t sd; -- if (rate->appl_ptr < rate->last_commit_ptr) { -- return rate->appl_ptr - rate->last_commit_ptr + pcm->boundary; -- } else { -- return rate->appl_ptr - rate->last_commit_ptr; -- } -+ return pcm_frame_diff(rate->appl_ptr, rate->last_commit_ptr, pcm->boundary); + memset(status, 0, sizeof(*status)); + snd_pcm_ioplug_hw_ptr_update(pcm); +@@ -118,6 +133,9 @@ static int snd_pcm_ioplug_status(snd_pcm_t *pcm, snd_pcm_status_t * status) + gettimestamp(&status->tstamp, pcm->tstamp_type); + status->avail = snd_pcm_mmap_avail(pcm); + status->avail_max = io->avail_max; ++ if (snd_pcm_ioplug_delay(pcm, &sd) < 0) ++ sd = snd_pcm_mmap_delay(pcm); ++ status->delay = sd; + return 0; } - static int snd_pcm_rate_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) -@@ -637,7 +632,7 @@ static int snd_pcm_rate_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) - + snd_pcm_rate_playback_internal_delay(pcm); - } else { - *delayp = rate->ops.output_frames(rate->obj, slave_delay) -- + snd_pcm_mmap_capture_hw_avail(pcm); -+ + snd_pcm_mmap_capture_delay(pcm); - } +@@ -133,20 +151,6 @@ static int snd_pcm_ioplug_hwsync(snd_pcm_t *pcm) return 0; } -@@ -746,7 +741,6 @@ static int snd_pcm_rate_commit_area(snd_pcm_t *pcm, snd_pcm_rate_t *rate, - if (result < 0) - return result; - __partial: -- xfer = 0; - cont = slave_frames; - if (cont > slave_size) - cont = slave_size; -@@ -846,7 +840,6 @@ static int snd_pcm_rate_grab_next_period(snd_pcm_t *pcm, snd_pcm_uframes_t hw_of - if (result < 0) - return result; - __partial: -- xfer = 0; - cont = slave_frames; - if (cont > rate->gen.slave->period_size) - cont = rate->gen.slave->period_size; -@@ -928,10 +921,7 @@ static int snd_pcm_rate_sync_playback_area(snd_pcm_t *pcm, snd_pcm_uframes_t app - if (slave_size < 0) - return slave_size; -- if (appl_ptr < rate->last_commit_ptr) -- xfer = appl_ptr - rate->last_commit_ptr + pcm->boundary; -- else -- xfer = appl_ptr - rate->last_commit_ptr; -+ xfer = pcm_frame_diff(appl_ptr, rate->last_commit_ptr, pcm->boundary); - while (xfer >= pcm->period_size && - (snd_pcm_uframes_t)slave_size >= rate->gen.slave->period_size) { - err = snd_pcm_rate_commit_next_period(pcm, rate->last_commit_ptr % pcm->buffer_size); -@@ -966,29 +956,18 @@ static snd_pcm_sframes_t snd_pcm_rate_mmap_commit(snd_pcm_t *pcm, +-static int snd_pcm_ioplug_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) +-{ +- ioplug_priv_t *io = pcm->private_data; +- +- if (io->data->version >= 0x010001 && +- io->data->callback->delay) +- return io->data->callback->delay(io->data, delayp); +- else { +- snd_pcm_ioplug_hw_ptr_update(pcm); +- *delayp = snd_pcm_mmap_hw_avail(pcm); +- } +- return 0; +-} +- + static int snd_pcm_ioplug_reset(snd_pcm_t *pcm) + { + ioplug_priv_t *io = pcm->private_data; +-- +2.29.2 + + +From 5988bb3ff42827ab4c13f248291ad1767c8d973d Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Sun, 3 Jan 2021 16:56:38 +0100 +Subject: [PATCH 28/49] pcm: rate - tidy up snd_pcm_rate_avail_update() + +No functional change - just move the capture code to +a separate function for the better readability. + +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_rate.c | 37 +++++++++++++++++++++---------------- + 1 file changed, 21 insertions(+), 16 deletions(-) + +diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c +index dc38e95e..7073f8ab 100644 +--- a/src/pcm/pcm_rate.c ++++ b/src/pcm/pcm_rate.c +@@ -964,29 +964,18 @@ static snd_pcm_sframes_t snd_pcm_rate_mmap_commit(snd_pcm_t *pcm, return size; } @@ -625,7 +1425,7 @@ index 5bf7dbb9..770aafea 100644 int err = snd_pcm_rate_grab_next_period(pcm, hw_offset); if (err < 0) return err; -@@ -996,13 +975,29 @@ static snd_pcm_sframes_t snd_pcm_rate_avail_update(snd_pcm_t *pcm) +@@ -994,13 +983,29 @@ static snd_pcm_sframes_t snd_pcm_rate_avail_update(snd_pcm_t *pcm) return (snd_pcm_sframes_t)xfer; xfer += pcm->period_size; size -= pcm->period_size; @@ -656,8 +1456,175 @@ index 5bf7dbb9..770aafea 100644 + return snd_pcm_mmap_avail(pcm); } - static int snd_pcm_rate_htimestamp(snd_pcm_t *pcm, -@@ -1060,9 +1055,7 @@ static int snd_pcm_rate_drain(snd_pcm_t *pcm) + static int snd_pcm_rate_htimestamp(snd_pcm_t *pcm, +-- +2.29.2 + + +From 6cee452eabc5cfdf0a6955033b8ac8f6e12ea883 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Sun, 3 Jan 2021 17:13:50 +0100 +Subject: [PATCH 29/49] pcm: ioplug - fix the delay calculation for old plugins + +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_ioplug.c | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c +index f2315a10..e141b1f9 100644 +--- a/src/pcm/pcm_ioplug.c ++++ b/src/pcm/pcm_ioplug.c +@@ -116,7 +116,7 @@ static int snd_pcm_ioplug_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) + return io->data->callback->delay(io->data, delayp); + else { + snd_pcm_ioplug_hw_ptr_update(pcm); +- *delayp = snd_pcm_mmap_hw_avail(pcm); ++ *delayp = snd_pcm_mmap_delay(pcm); + } + return 0; + } +-- +2.29.2 + + +From fc0f7af9ee7bd4c31c2bab229e8e79eb96e908fa Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Mon, 4 Jan 2021 12:29:00 +0100 +Subject: [PATCH 30/49] pcm: rate - use pcm_frame_diff() in + snd_pcm_rate_playback_internal_delay() + +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_rate.c | 6 +----- + 1 file changed, 1 insertion(+), 5 deletions(-) + +diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c +index 7073f8ab..abb2753f 100644 +--- a/src/pcm/pcm_rate.c ++++ b/src/pcm/pcm_rate.c +@@ -612,11 +612,7 @@ static snd_pcm_uframes_t snd_pcm_rate_playback_internal_delay(snd_pcm_t *pcm) + { + snd_pcm_rate_t *rate = pcm->private_data; + +- if (rate->appl_ptr < rate->last_commit_ptr) { +- return rate->appl_ptr - rate->last_commit_ptr + pcm->boundary; +- } else { +- return rate->appl_ptr - rate->last_commit_ptr; +- } ++ return pcm_frame_diff(rate->appl_ptr, rate->last_commit_ptr, pcm->boundary); + } + + static int snd_pcm_rate_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) +-- +2.29.2 + + +From da5b70d3fae091a8568b80a476c1cf340965805a Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Mon, 4 Jan 2021 12:32:25 +0100 +Subject: [PATCH 31/49] pcm: plugin - fix status code for capture + +The recent updates do not take in account the possible +calls for the capture stream. Fix the avail and delay +inconsistencies (and assert). + +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_plugin.c | 21 +++++++++++++++++++-- + 1 file changed, 19 insertions(+), 2 deletions(-) + +diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c +index 5787a43d..7ed6f25a 100644 +--- a/src/pcm/pcm_plugin.c ++++ b/src/pcm/pcm_plugin.c +@@ -545,13 +545,30 @@ static snd_pcm_sframes_t snd_pcm_plugin_avail_update(snd_pcm_t *pcm) + static int snd_pcm_plugin_status(snd_pcm_t *pcm, snd_pcm_status_t * status) + { + snd_pcm_plugin_t *plugin = pcm->private_data; +- snd_pcm_sframes_t err; ++ snd_pcm_sframes_t err, diff; + + err = snd_pcm_status(plugin->gen.slave, status); + if (err < 0) + return err; +- assert(status->appl_ptr == *pcm->appl.ptr); + snd_pcm_plugin_sync_hw_ptr(pcm, status->hw_ptr, status->avail); ++ /* ++ * For capture stream, the situation is more complicated, because ++ * snd_pcm_plugin_avail_update() commits the data to the slave pcm. ++ * It means that the slave appl_ptr is updated. Calculate diff and ++ * update the delay and avail. ++ * ++ * This resolves the data inconsistency for immediate calls: ++ * snd_pcm_avail_update() ++ * snd_pcm_status() ++ */ ++ if (pcm->stream == SND_PCM_STREAM_CAPTURE) { ++ status->appl_ptr = *pcm->appl.ptr; ++ diff = pcm_frame_diff(status->appl_ptr, *pcm->appl.ptr, pcm->boundary); ++ status->avail += diff; ++ status->delay += diff; ++ } else { ++ assert(status->appl_ptr == *pcm->appl.ptr); ++ } + return 0; + } + +-- +2.29.2 + + +From ac520b2ed11f31e789a4da1d55e360e3eaeac50d Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Mon, 4 Jan 2021 17:14:42 +0100 +Subject: [PATCH 32/49] pcm: rate - use pcm_frame_diff() on related places + +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_rate.c | 20 +++++++------------- + 1 file changed, 7 insertions(+), 13 deletions(-) + +diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c +index abb2753f..3d301d0e 100644 +--- a/src/pcm/pcm_rate.c ++++ b/src/pcm/pcm_rate.c +@@ -561,17 +561,16 @@ snd_pcm_rate_read_areas1(snd_pcm_t *pcm, + + static inline void snd_pcm_rate_sync_hwptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr) + { +- snd_pcm_rate_t *rate = pcm->private_data; +- +- snd_pcm_sframes_t slave_hw_ptr_diff = slave_hw_ptr - rate->last_slave_hw_ptr; ++ snd_pcm_rate_t *rate; ++ snd_pcm_sframes_t slave_hw_ptr_diff; + snd_pcm_sframes_t last_slave_hw_ptr_frac; + + if (pcm->stream != SND_PCM_STREAM_PLAYBACK) + return; + +- if (slave_hw_ptr_diff < 0) +- slave_hw_ptr_diff += rate->gen.slave->boundary; /* slave boundary wraparound */ +- else if (slave_hw_ptr_diff == 0) ++ rate = pcm->private_data; ++ slave_hw_ptr_diff = pcm_frame_diff(slave_hw_ptr, rate->last_slave_hw_ptr, rate->gen.slave->boundary); ++ if (slave_hw_ptr_diff == 0) + return; + last_slave_hw_ptr_frac = rate->last_slave_hw_ptr % rate->gen.slave->period_size; + /* While handling fraction part fo slave period, rounded value will be +@@ -922,10 +921,7 @@ static int snd_pcm_rate_sync_playback_area(snd_pcm_t *pcm, snd_pcm_uframes_t app + if (slave_size < 0) + return slave_size; + +- if (appl_ptr < rate->last_commit_ptr) +- xfer = appl_ptr - rate->last_commit_ptr + pcm->boundary; +- else +- xfer = appl_ptr - rate->last_commit_ptr; ++ xfer = pcm_frame_diff(appl_ptr, rate->last_commit_ptr, pcm->boundary); + while (xfer >= pcm->period_size && + (snd_pcm_uframes_t)slave_size >= rate->gen.slave->period_size) { + err = snd_pcm_rate_commit_next_period(pcm, rate->last_commit_ptr % pcm->buffer_size); +@@ -1059,9 +1055,7 @@ static int snd_pcm_rate_drain(snd_pcm_t *pcm) sw_params.avail_min = 1; snd_pcm_sw_params(rate->gen.slave, &sw_params); @@ -668,7 +1635,37 @@ index 5bf7dbb9..770aafea 100644 ofs = rate->last_commit_ptr % pcm->buffer_size; while (size > 0) { snd_pcm_uframes_t psize, spsize; -@@ -1164,12 +1157,8 @@ static int snd_pcm_rate_status(snd_pcm_t *pcm, snd_pcm_status_t * status) +-- +2.29.2 + + +From b62f66442bfbcea7c11afe4917e196fa95a65083 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Mon, 4 Jan 2021 17:27:32 +0100 +Subject: [PATCH 33/49] pcm: rate - fix the capture delay values + +Use the correct snd_pcm_mmap_capture_delay() function instead +snd_pcm_mmap_capture_hw_avail(). + +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_rate.c | 8 ++------ + 1 file changed, 2 insertions(+), 6 deletions(-) + +diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c +index 3d301d0e..770aafea 100644 +--- a/src/pcm/pcm_rate.c ++++ b/src/pcm/pcm_rate.c +@@ -632,7 +632,7 @@ static int snd_pcm_rate_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) + + snd_pcm_rate_playback_internal_delay(pcm); + } else { + *delayp = rate->ops.output_frames(rate->obj, slave_delay) +- + snd_pcm_mmap_capture_hw_avail(pcm); ++ + snd_pcm_mmap_capture_delay(pcm); + } + return 0; + } +@@ -1157,12 +1157,8 @@ static int snd_pcm_rate_status(snd_pcm_t *pcm, snd_pcm_status_t * status) status->avail = snd_pcm_mmap_playback_avail(pcm); status->avail_max = rate->ops.input_frames(rate->obj, status->avail_max); } else { @@ -682,219 +1679,50 @@ index 5bf7dbb9..770aafea 100644 status->avail = snd_pcm_mmap_capture_avail(pcm); status->avail_max = rate->ops.output_frames(rate->obj, status->avail_max); } -diff --git a/src/rawmidi/rawmidi_virt.c b/src/rawmidi/rawmidi_virt.c -index 2c4c27f5..884b8ff8 100644 ---- a/src/rawmidi/rawmidi_virt.c -+++ b/src/rawmidi/rawmidi_virt.c -@@ -315,7 +315,7 @@ int snd_rawmidi_virtual_open(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp, - int merge, int mode) - { - int err; -- snd_rawmidi_t *rmidi; -+ snd_rawmidi_t *rmidi = NULL; - snd_rawmidi_virtual_t *virt = NULL; - struct pollfd pfd; - -@@ -392,6 +392,7 @@ int snd_rawmidi_virtual_open(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp, - free(*inputp); - if (outputp) - free(*outputp); -+ free(rmidi); - return err; - } - -diff --git a/src/timer/timer_query_hw.c b/src/timer/timer_query_hw.c -index dad228c8..d8bac6e7 100644 ---- a/src/timer/timer_query_hw.c -+++ b/src/timer/timer_query_hw.c -@@ -104,7 +104,7 @@ int snd_timer_query_hw_open(snd_timer_query_t **handle, const char *name, int mo - close(fd); - return -SND_ERROR_INCOMPATIBLE_VERSION; - } -- tmr = (snd_timer_query_t *) calloc(1, sizeof(snd_timer_t)); -+ tmr = (snd_timer_query_t *) calloc(1, sizeof(snd_timer_query_t)); - if (tmr == NULL) { - close(fd); - return -ENOMEM; -diff --git a/src/topology/dapm.c b/src/topology/dapm.c -index 92dc01aa..f6a84a60 100644 ---- a/src/topology/dapm.c -+++ b/src/topology/dapm.c -@@ -836,6 +836,7 @@ int tplg_add_widget_object(snd_tplg_t *tplg, snd_tplg_obj_template_t *t) - default: - SNDERR("widget %s: invalid type %d for ctl %d", - wt->name, ct->type, i); -+ ret = -EINVAL; - break; - } - -diff --git a/src/topology/data.c b/src/topology/data.c -index c2931bd2..5633cdc3 100644 ---- a/src/topology/data.c -+++ b/src/topology/data.c -@@ -860,7 +860,7 @@ static int parse_tuple_set(snd_config_t *cfg, - } - - if ((type == SND_SOC_TPLG_TUPLE_TYPE_WORD -- && tuple_val > UINT_MAX) -+ /* && tuple_val > UINT_MAX */) - || (type == SND_SOC_TPLG_TUPLE_TYPE_SHORT - && tuple_val > USHRT_MAX) - || (type == SND_SOC_TPLG_TUPLE_TYPE_BYTE -diff --git a/src/topology/pcm.c b/src/topology/pcm.c -index 191b7a0a..a473b59b 100644 ---- a/src/topology/pcm.c -+++ b/src/topology/pcm.c -@@ -1411,6 +1411,7 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, - snd_config_t *n; - const char *id, *val = NULL; - int ret, ival; -+ bool provider_legacy; - - elem = tplg_elem_new_common(tplg, cfg, NULL, SND_TPLG_TYPE_HW_CONFIG); - if (!elem) -@@ -1451,8 +1452,15 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, - continue; - } - -- if (strcmp(id, "bclk") == 0 || -- strcmp(id, "bclk_master") == 0) { -+ provider_legacy = false; -+ if (strcmp(id, "bclk_master") == 0) { -+ SNDERR("deprecated option %s, please use 'bclk'\n", id); -+ provider_legacy = true; -+ } -+ -+ if (provider_legacy || -+ strcmp(id, "bclk") == 0) { -+ - if (snd_config_get_string(n, &val) < 0) - return -EINVAL; - -@@ -1462,11 +1470,19 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, - */ - SNDERR("deprecated bclk value '%s'", val); - -- hw_cfg->bclk_master = SND_SOC_TPLG_BCLK_CS; -+ hw_cfg->bclk_provider = SND_SOC_TPLG_BCLK_CC; - } else if (!strcmp(val, "codec_slave")) { -- hw_cfg->bclk_master = SND_SOC_TPLG_BCLK_CS; -+ SNDERR("deprecated bclk value '%s', use 'codec_consumer'", val); -+ -+ hw_cfg->bclk_provider = SND_SOC_TPLG_BCLK_CC; -+ } else if (!strcmp(val, "codec_consumer")) { -+ hw_cfg->bclk_provider = SND_SOC_TPLG_BCLK_CC; - } else if (!strcmp(val, "codec_master")) { -- hw_cfg->bclk_master = SND_SOC_TPLG_BCLK_CM; -+ SNDERR("deprecated bclk value '%s', use 'codec_provider", val); -+ -+ hw_cfg->bclk_provider = SND_SOC_TPLG_BCLK_CP; -+ } else if (!strcmp(val, "codec_provider")) { -+ hw_cfg->bclk_provider = SND_SOC_TPLG_BCLK_CP; - } - continue; - } -@@ -1488,8 +1504,15 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, - continue; - } - -- if (strcmp(id, "fsync") == 0 || -- strcmp(id, "fsync_master") == 0) { -+ provider_legacy = false; -+ if (strcmp(id, "fsync_master") == 0) { -+ SNDERR("deprecated option %s, please use 'fsync'\n", id); -+ provider_legacy = true; -+ } -+ -+ if (provider_legacy || -+ strcmp(id, "fsync") == 0) { -+ - if (snd_config_get_string(n, &val) < 0) - return -EINVAL; - -@@ -1499,11 +1522,19 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, - */ - SNDERR("deprecated fsync value '%s'", val); - -- hw_cfg->fsync_master = SND_SOC_TPLG_FSYNC_CS; -+ hw_cfg->fsync_provider = SND_SOC_TPLG_FSYNC_CC; - } else if (!strcmp(val, "codec_slave")) { -- hw_cfg->fsync_master = SND_SOC_TPLG_FSYNC_CS; -+ SNDERR("deprecated fsync value '%s', use 'codec_consumer'", val); -+ -+ hw_cfg->fsync_provider = SND_SOC_TPLG_FSYNC_CC; -+ } else if (!strcmp(val, "codec_consumer")) { -+ hw_cfg->fsync_provider = SND_SOC_TPLG_FSYNC_CC; - } else if (!strcmp(val, "codec_master")) { -- hw_cfg->fsync_master = SND_SOC_TPLG_FSYNC_CM; -+ SNDERR("deprecated fsync value '%s', use 'codec_provider'", val); -+ -+ hw_cfg->fsync_provider = SND_SOC_TPLG_FSYNC_CP; -+ } else if (!strcmp(val, "codec_provider")) { -+ hw_cfg->fsync_provider = SND_SOC_TPLG_FSYNC_CP; - } - continue; - } -@@ -1623,19 +1654,19 @@ int tplg_save_hw_config(snd_tplg_t *tplg ATTRIBUTE_UNUSED, - if (err >= 0 && hc->fmt) - err = tplg_save_printf(dst, pfx, "\tformat '%s'\n", - get_audio_hw_format_name(hc->fmt)); -- if (err >= 0 && hc->bclk_master) -+ if (err >= 0 && hc->bclk_provider) - err = tplg_save_printf(dst, pfx, "\tbclk '%s'\n", -- hc->bclk_master == SND_SOC_TPLG_BCLK_CS ? -- "codec_slave" : "codec_master"); -+ hc->bclk_provider == SND_SOC_TPLG_BCLK_CC ? -+ "codec_consumer" : "codec_provider"); - if (err >= 0 && hc->bclk_rate) - err = tplg_save_printf(dst, pfx, "\tbclk_freq %u\n", - hc->bclk_rate); - if (err >= 0 && hc->invert_bclk) - err = tplg_save_printf(dst, pfx, "\tbclk_invert 1\n"); -- if (err >= 0 && hc->fsync_master) -- err = tplg_save_printf(dst, pfx, "\tfsync_master '%s'\n", -- hc->fsync_master == SND_SOC_TPLG_FSYNC_CS ? -- "codec_slave" : "codec_master"); -+ if (err >= 0 && hc->fsync_provider) -+ err = tplg_save_printf(dst, pfx, "\tfsync_provider '%s'\n", -+ hc->fsync_provider == SND_SOC_TPLG_FSYNC_CC ? -+ "codec_consumer" : "codec_provider"); - if (err >= 0 && hc->fsync_rate) - err = tplg_save_printf(dst, pfx, "\tfsync_freq %u\n", - hc->fsync_rate); -@@ -1791,8 +1822,8 @@ static int set_link_hw_config(struct snd_soc_tplg_hw_config *cfg, - cfg->clock_gated = tpl->clock_gated; - cfg->invert_bclk = tpl->invert_bclk; - cfg->invert_fsync = tpl->invert_fsync; -- cfg->bclk_master = tpl->bclk_master; -- cfg->fsync_master = tpl->fsync_master; -+ cfg->bclk_provider = tpl->bclk_provider; -+ cfg->fsync_provider = tpl->fsync_provider; - cfg->mclk_direction = tpl->mclk_direction; - cfg->reserved = tpl->reserved; - cfg->mclk_rate = tpl->mclk_rate; -@@ -1989,7 +2020,7 @@ next: - pt->playback = pcm->playback; - pt->capture = pcm->capture; - pt->compress = pcm->compress; -- tplg_log(tplg, 'D', pos, "pcm: playback %d capture %d compress", -+ tplg_log(tplg, 'D', pos, "pcm: playback %d capture %d compress %d", - pt->playback, pt->capture, pt->compress); - pt->num_streams = pcm->num_streams; - pt->flag_mask = pcm->flag_mask; -@@ -2174,8 +2205,8 @@ next: - hw->clock_gated = link->hw_config[i].clock_gated; - hw->invert_bclk = link->hw_config[i].invert_bclk; - hw->invert_fsync = link->hw_config[i].invert_fsync; -- hw->bclk_master = link->hw_config[i].bclk_master; -- hw->fsync_master = link->hw_config[i].fsync_master; -+ hw->bclk_provider = link->hw_config[i].bclk_provider; -+ hw->fsync_provider = link->hw_config[i].fsync_provider; - hw->mclk_direction = link->hw_config[i].mclk_direction; - hw->mclk_rate = link->hw_config[i].mclk_rate; - hw->bclk_rate = link->hw_config[i].bclk_rate; +-- +2.29.2 + + +From 437b5b3aae8b1d9f65289f563deb0fa6356a0fa6 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 7 Jan 2021 17:32:11 +0100 +Subject: [PATCH 34/49] ucm: fix possible memory leak in parse_verb_file() + +Signed-off-by: Jaroslav Kysela +--- + src/ucm/parser.c | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +diff --git a/src/ucm/parser.c b/src/ucm/parser.c +index 75b78826..c8bee1f2 100644 +--- a/src/ucm/parser.c ++++ b/src/ucm/parser.c +@@ -1575,7 +1575,7 @@ static int parse_verb_file(snd_use_case_mgr_t *uc_mgr, + /* in-place evaluation */ + err = uc_mgr_evaluate_inplace(uc_mgr, cfg); + if (err < 0) +- return err; ++ goto _err; + + /* parse master config sections */ + snd_config_for_each(i, next, cfg) { +-- +2.29.2 + + +From d0bb8f84c92357bfd0e024efe5f6d53a963c4d7a Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 7 Jan 2021 17:34:00 +0100 +Subject: [PATCH 35/49] topology: tplg_pprint_integer() fix coverity + uninitalized variable error + +Signed-off-by: Jaroslav Kysela +--- + src/topology/save.c | 2 ++ + 1 file changed, 2 insertions(+) + diff --git a/src/topology/save.c b/src/topology/save.c -index c7a5a801..fecbc6a5 100644 +index c7a5a801..f7af7af3 100644 --- a/src/topology/save.c +++ b/src/topology/save.c @@ -133,6 +133,8 @@ static int tplg_pprint_integer(snd_config_t *n, char **ret) @@ -906,7 +1734,141 @@ index c7a5a801..fecbc6a5 100644 } err = tplg_nice_value_format(buf, sizeof(buf), (unsigned int)lval); if (err < 0) -@@ -178,29 +180,28 @@ static snd_config_t *sort_config(const char *id, snd_config_t *src) +-- +2.29.2 + + +From b8764a061b83284be60adc01c1f7b035c4484ace Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 7 Jan 2021 17:36:04 +0100 +Subject: [PATCH 36/49] topology: tplg_add_widget_object() - do not use invalid + elem_ctl + +Signed-off-by: Jaroslav Kysela +--- + src/topology/dapm.c | 1 + + 1 file changed, 1 insertion(+) + +diff --git a/src/topology/dapm.c b/src/topology/dapm.c +index 92dc01aa..f6a84a60 100644 +--- a/src/topology/dapm.c ++++ b/src/topology/dapm.c +@@ -836,6 +836,7 @@ int tplg_add_widget_object(snd_tplg_t *tplg, snd_tplg_obj_template_t *t) + default: + SNDERR("widget %s: invalid type %d for ctl %d", + wt->name, ct->type, i); ++ ret = -EINVAL; + break; + } + +-- +2.29.2 + + +From 0dbaba95f4bc59602bf2d3fb12a60db7444e5892 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 7 Jan 2021 17:38:50 +0100 +Subject: [PATCH 37/49] topology: tplg_decode_pcm() - add missing log argument + (compress) + +Signed-off-by: Jaroslav Kysela +--- + src/topology/pcm.c | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +diff --git a/src/topology/pcm.c b/src/topology/pcm.c +index c8f41862..a473b59b 100644 +--- a/src/topology/pcm.c ++++ b/src/topology/pcm.c +@@ -2020,7 +2020,7 @@ next: + pt->playback = pcm->playback; + pt->capture = pcm->capture; + pt->compress = pcm->compress; +- tplg_log(tplg, 'D', pos, "pcm: playback %d capture %d compress", ++ tplg_log(tplg, 'D', pos, "pcm: playback %d capture %d compress %d", + pt->playback, pt->capture, pt->compress); + pt->num_streams = pcm->num_streams; + pt->flag_mask = pcm->flag_mask; +-- +2.29.2 + + +From 51e1d486ce4535923692bd1d7e59d5aa3a29cd50 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 7 Jan 2021 17:40:42 +0100 +Subject: [PATCH 38/49] topology: parse_tuple_set() - remove dead condition + code + +Signed-off-by: Jaroslav Kysela +--- + src/topology/data.c | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +diff --git a/src/topology/data.c b/src/topology/data.c +index c2931bd2..5633cdc3 100644 +--- a/src/topology/data.c ++++ b/src/topology/data.c +@@ -860,7 +860,7 @@ static int parse_tuple_set(snd_config_t *cfg, + } + + if ((type == SND_SOC_TPLG_TUPLE_TYPE_WORD +- && tuple_val > UINT_MAX) ++ /* && tuple_val > UINT_MAX */) + || (type == SND_SOC_TPLG_TUPLE_TYPE_SHORT + && tuple_val > USHRT_MAX) + || (type == SND_SOC_TPLG_TUPLE_TYPE_BYTE +-- +2.29.2 + + +From 3f63dc26445ae7c215e48a57af83b6da325f166d Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 7 Jan 2021 17:41:38 +0100 +Subject: [PATCH 39/49] ucm: uc_mgr_substitute_tree() fix use after free + +Signed-off-by: Jaroslav Kysela +--- + src/ucm/ucm_subs.c | 3 ++- + 1 file changed, 2 insertions(+), 1 deletion(-) + +diff --git a/src/ucm/ucm_subs.c b/src/ucm/ucm_subs.c +index f608bb09..df6d736f 100644 +--- a/src/ucm/ucm_subs.c ++++ b/src/ucm/ucm_subs.c +@@ -417,11 +417,12 @@ int uc_mgr_substitute_tree(snd_use_case_mgr_t *uc_mgr, snd_config_t *node) + if (err < 0) + return err; + err = snd_config_set_id(node, s); +- free(s); + if (err < 0) { + uc_error("unable to set substituted id '%s' (old id '%s')", s, id); ++ free(s); + return err; + } ++ free(s); + } + if (snd_config_get_type(node) != SND_CONFIG_TYPE_COMPOUND) { + if (snd_config_get_type(node) == SND_CONFIG_TYPE_STRING) { +-- +2.29.2 + + +From 45f503632acf24877c466a7c1c74d8a26414bf3e Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 7 Jan 2021 17:45:27 +0100 +Subject: [PATCH 40/49] topology: sort_config() cleanups - use goto for the + error path + +Signed-off-by: Jaroslav Kysela +--- + src/topology/save.c | 15 +++++++-------- + 1 file changed, 7 insertions(+), 8 deletions(-) + +diff --git a/src/topology/save.c b/src/topology/save.c +index f7af7af3..fecbc6a5 100644 +--- a/src/topology/save.c ++++ b/src/topology/save.c +@@ -180,29 +180,28 @@ static snd_config_t *sort_config(const char *id, snd_config_t *src) } if (array <= 0) qsort(a, count, sizeof(a[0]), _compar); @@ -943,117 +1905,555 @@ index c7a5a801..fecbc6a5 100644 } static int tplg_check_quoted(const unsigned char *p) -diff --git a/src/ucm/main.c b/src/ucm/main.c -index 3871d5aa..754b967e 100644 ---- a/src/ucm/main.c -+++ b/src/ucm/main.c -@@ -666,7 +666,7 @@ static int get_list0(struct list_head *list, - } - return cnt; - __fail: -- snd_use_case_free_list((const char **)res, cnt); -+ snd_use_case_free_list(*result, cnt); - return -ENOMEM; +-- +2.29.2 + + +From 93752fb4de397554e92a4eb9079f77dabaec7d7f Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 7 Jan 2021 17:49:25 +0100 +Subject: [PATCH 41/49] conf: USB - add "Xonar U7 MKII" to + USB-Audio.pcm.iec958_device + +BugLink: https://github.com/alsa-project/alsa-lib/issues/100 +Signed-off-by: Jaroslav Kysela +--- + src/conf/cards/USB-Audio.conf | 1 + + 1 file changed, 1 insertion(+) + +diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf +index 9b64af3c..b1b74b02 100644 +--- a/src/conf/cards/USB-Audio.conf ++++ b/src/conf/cards/USB-Audio.conf +@@ -39,6 +39,7 @@ USB-Audio.pcm.iec958_device { + # "NoiseBlaster 3000" 42 + "USB Sound Blaster HD" 1 + "Xonar U7" 1 ++ "Xonar U7 MKII" 1 + "ASUS XONAR U5" 1 + "XONAR U5" 1 + "XONAR SOUND CARD" 1 +-- +2.29.2 + + +From 644514e85da169670e4a490b7b15b5ecfcec92c0 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Mon, 18 Jan 2021 21:09:43 +0100 +Subject: [PATCH 42/49] pcm_plugin: set the initial hw_ptr/appl_ptr from the + child pcm + +The direct plugins (dmix & etc.) sets own initial +hw_ptr and appl_ptr. Use this initial settings +to export correct values in snd_pcm_status(). + +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_plugin.c | 32 +++++++++++++++++--------------- + 1 file changed, 17 insertions(+), 15 deletions(-) + +diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c +index 7ed6f25a..ff254eba 100644 +--- a/src/pcm/pcm_plugin.c ++++ b/src/pcm/pcm_plugin.c +@@ -146,15 +146,14 @@ static int snd_pcm_plugin_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) + return 0; } -@@ -724,7 +724,7 @@ static int get_list20(struct list_head *list, - } - return cnt; - __fail: -- snd_use_case_free_list((const char **)res, cnt); -+ snd_use_case_free_list(*result, cnt); - return -ENOMEM; +-static int snd_pcm_plugin_prepare(snd_pcm_t *pcm) ++static int snd_pcm_plugin_call_init_cb(snd_pcm_t *pcm, snd_pcm_plugin_t *plugin) + { +- snd_pcm_plugin_t *plugin = pcm->private_data; ++ snd_pcm_t *slave = plugin->gen.slave; + int err; +- err = snd_pcm_prepare(plugin->gen.slave); +- if (err < 0) +- return err; +- *pcm->hw.ptr = 0; +- *pcm->appl.ptr = 0; ++ ++ assert(pcm->boundary == slave->boundary); ++ *pcm->hw.ptr = *slave->hw.ptr; ++ *pcm->appl.ptr = *slave->appl.ptr; + if (plugin->init) { + err = plugin->init(pcm); + if (err < 0) +@@ -163,6 +162,16 @@ static int snd_pcm_plugin_prepare(snd_pcm_t *pcm) + return 0; } -diff --git a/src/ucm/parser.c b/src/ucm/parser.c -index 75b78826..c8bee1f2 100644 ---- a/src/ucm/parser.c -+++ b/src/ucm/parser.c -@@ -1575,7 +1575,7 @@ static int parse_verb_file(snd_use_case_mgr_t *uc_mgr, - /* in-place evaluation */ - err = uc_mgr_evaluate_inplace(uc_mgr, cfg); ++static int snd_pcm_plugin_prepare(snd_pcm_t *pcm) ++{ ++ snd_pcm_plugin_t *plugin = pcm->private_data; ++ int err; ++ err = snd_pcm_prepare(plugin->gen.slave); ++ if (err < 0) ++ return err; ++ return snd_pcm_plugin_call_init_cb(pcm, plugin); ++} ++ + static int snd_pcm_plugin_reset(snd_pcm_t *pcm) + { + snd_pcm_plugin_t *plugin = pcm->private_data; +@@ -170,14 +179,7 @@ static int snd_pcm_plugin_reset(snd_pcm_t *pcm) + err = snd_pcm_reset(plugin->gen.slave); if (err < 0) -- return err; -+ goto _err; + return err; +- *pcm->hw.ptr = 0; +- *pcm->appl.ptr = 0; +- if (plugin->init) { +- err = plugin->init(pcm); +- if (err < 0) +- return err; +- } +- return 0; ++ return snd_pcm_plugin_call_init_cb(pcm, plugin); + } - /* parse master config sections */ - snd_config_for_each(i, next, cfg) { -diff --git a/src/ucm/ucm_subs.c b/src/ucm/ucm_subs.c -index f608bb09..df6d736f 100644 ---- a/src/ucm/ucm_subs.c -+++ b/src/ucm/ucm_subs.c -@@ -417,11 +417,12 @@ int uc_mgr_substitute_tree(snd_use_case_mgr_t *uc_mgr, snd_config_t *node) - if (err < 0) - return err; - err = snd_config_set_id(node, s); -- free(s); - if (err < 0) { - uc_error("unable to set substituted id '%s' (old id '%s')", s, id); -+ free(s); - return err; - } -+ free(s); + static snd_pcm_sframes_t snd_pcm_plugin_rewindable(snd_pcm_t *pcm) +-- +2.29.2 + + +From 0de72e63d7bd180566c71f6fb01d7719525ee326 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Tue, 19 Jan 2021 11:17:00 +0100 +Subject: [PATCH 43/49] pcm: dmix/dshare - delay calculation fixes and cleanups + +Unfortunately, we cannot use status->avail from slave, because this value +does not wrap to the buffer size and it may even overflow slave boundary +(endless run). We can use only hw_ptr from slave. + +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_dmix.c | 5 ++--- + src/pcm/pcm_dshare.c | 5 ++--- + 2 files changed, 4 insertions(+), 6 deletions(-) + +diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c +index 5b7472d9..be2675af 100644 +--- a/src/pcm/pcm_dmix.c ++++ b/src/pcm/pcm_dmix.c +@@ -488,8 +488,7 @@ static int snd_pcm_dmix_status(snd_pcm_t *pcm, snd_pcm_status_t * status) + case SNDRV_PCM_STATE_DRAINING: + case SNDRV_PCM_STATE_RUNNING: + snd_pcm_dmix_sync_ptr0(pcm, status->hw_ptr); +- status->delay += snd_pcm_mmap_playback_delay(pcm) +- + status->avail - dmix->spcm->buffer_size; ++ status->delay = snd_pcm_mmap_playback_delay(pcm); + break; + default: + break; +@@ -518,7 +517,7 @@ static int snd_pcm_dmix_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_SUSPENDED: + case STATE_RUN_PENDING: +- *delayp = snd_pcm_mmap_playback_hw_avail(pcm); ++ *delayp = snd_pcm_mmap_playback_delay(pcm); + return 0; + case SNDRV_PCM_STATE_XRUN: + return -EPIPE; +diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c +index 8a672572..10243013 100644 +--- a/src/pcm/pcm_dshare.c ++++ b/src/pcm/pcm_dshare.c +@@ -237,8 +237,7 @@ static int snd_pcm_dshare_status(snd_pcm_t *pcm, snd_pcm_status_t * status) + case SNDRV_PCM_STATE_DRAINING: + case SNDRV_PCM_STATE_RUNNING: + snd_pcm_dshare_sync_ptr0(pcm, status->hw_ptr); +- status->delay += snd_pcm_mmap_playback_delay(pcm) +- + status->avail - dshare->spcm->buffer_size; ++ status->delay += snd_pcm_mmap_playback_delay(pcm); + break; + default: + break; +@@ -290,7 +289,7 @@ static int snd_pcm_dshare_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_SUSPENDED: + case STATE_RUN_PENDING: +- *delayp = snd_pcm_mmap_playback_hw_avail(pcm); ++ *delayp = snd_pcm_mmap_playback_delay(pcm); + return 0; + case SNDRV_PCM_STATE_XRUN: + return -EPIPE; +-- +2.29.2 + + +From ae7362c996e0b991c399cc26c6d0b3c8e295e83c Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Wed, 20 Jan 2021 16:58:33 +0100 +Subject: [PATCH 44/49] topology: fix 'parse_tuple_set() - remove dead + condition code' + +The whole SND_SOC_TPLG_TUPLE_TYPE_WORD condition part must be +commented out, otherwise the condition is always true. + +Fixes: 51e1d486ce ("topology: parse_tuple_set() - remove dead condition code") +BugLink: https://github.com/alsa-project/alsa-lib/issues/114 +Signed-off-by: Jaroslav Kysela +--- + src/topology/data.c | 10 +++++----- + 1 file changed, 5 insertions(+), 5 deletions(-) + +diff --git a/src/topology/data.c b/src/topology/data.c +index 5633cdc3..0546d63e 100644 +--- a/src/topology/data.c ++++ b/src/topology/data.c +@@ -859,11 +859,11 @@ static int parse_tuple_set(snd_config_t *cfg, + goto err; + } + +- if ((type == SND_SOC_TPLG_TUPLE_TYPE_WORD +- /* && tuple_val > UINT_MAX */) +- || (type == SND_SOC_TPLG_TUPLE_TYPE_SHORT +- && tuple_val > USHRT_MAX) +- || (type == SND_SOC_TPLG_TUPLE_TYPE_BYTE ++ if (/* (type == SND_SOC_TPLG_TUPLE_TYPE_WORD ++ && tuple_val > UINT_MAX) || */ ++ (type == SND_SOC_TPLG_TUPLE_TYPE_SHORT ++ && tuple_val > USHRT_MAX) || ++ (type == SND_SOC_TPLG_TUPLE_TYPE_BYTE + && tuple_val > UCHAR_MAX)) { + SNDERR("tuple %s: invalid value", id); + goto err; +-- +2.29.2 + + +From f2c1a9f3279a30cbd8dcba1eaf0d52f50b56b886 Mon Sep 17 00:00:00 2001 +From: Takashi Iwai +Date: Fri, 22 Jan 2021 15:48:39 +0100 +Subject: [PATCH 45/49] pcm: direct: Fix the missing appl_ptr update + +A snd_pcm_status() call for the direct plugins receives the status +from the slave PCM, but this doesn't contain a valid appl_ptr, since +the slave PCM for the direct plugins is in a free-wheel mode, hence +the appl_ptr is always zero. This result in the inconsistent +status->appl_ptr and pcm->appl.ptr, hitting the recently introduced +assert() call. + +Fix it by transferring the plugin's appl_ptr to the upper caller. + +BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1181194 +Signed-off-by: Takashi Iwai +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_dmix.c | 1 + + src/pcm/pcm_dshare.c | 1 + + src/pcm/pcm_dsnoop.c | 1 + + 3 files changed, 3 insertions(+) + +diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c +index be2675af..d8495065 100644 +--- a/src/pcm/pcm_dmix.c ++++ b/src/pcm/pcm_dmix.c +@@ -495,6 +495,7 @@ static int snd_pcm_dmix_status(snd_pcm_t *pcm, snd_pcm_status_t * status) } - if (snd_config_get_type(node) != SND_CONFIG_TYPE_COMPOUND) { - if (snd_config_get_type(node) == SND_CONFIG_TYPE_STRING) { -diff --git a/test/audio_time.c b/test/audio_time.c -index 530922d9..e4d4a944 100644 ---- a/test/audio_time.c -+++ b/test/audio_time.c -@@ -32,7 +32,7 @@ static void usage(char *command) - "-d, --delay add delay \n" - "-D, --device=NAME select PCM by name \n" - "-p, --playback playback tstamps \n" -- "-t, --ts_type=TYPE Default(0),link(1),link_estimated(2),synchronized(3) \n" -+ "-t, --ts_type=TYPE Compat(0),default(1),link(2),link_absolute(3),link_estimated(4),link_synchronized(5) \n" - "-r, --report show audio timestamp and accuracy validity\n" - , command); + + status->state = snd_pcm_dmix_state(pcm); ++ status->appl_ptr = *pcm->appl.ptr; /* slave PCM doesn't set appl_ptr */ + status->trigger_tstamp = dmix->trigger_tstamp; + status->avail = snd_pcm_mmap_playback_avail(pcm); + status->avail_max = status->avail > dmix->avail_max ? status->avail : dmix->avail_max; +diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c +index 10243013..dccb137b 100644 +--- a/src/pcm/pcm_dshare.c ++++ b/src/pcm/pcm_dshare.c +@@ -243,6 +243,7 @@ static int snd_pcm_dshare_status(snd_pcm_t *pcm, snd_pcm_status_t * status) + break; + } + status->state = snd_pcm_dshare_state(pcm); ++ status->appl_ptr = *pcm->appl.ptr; /* slave PCM doesn't set appl_ptr */ + status->trigger_tstamp = dshare->trigger_tstamp; + status->avail = snd_pcm_mmap_playback_avail(pcm); + status->avail_max = status->avail > dshare->avail_max ? status->avail : dshare->avail_max; +diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c +index c6e8cd27..695bf4aa 100644 +--- a/src/pcm/pcm_dsnoop.c ++++ b/src/pcm/pcm_dsnoop.c +@@ -193,6 +193,7 @@ static int snd_pcm_dsnoop_status(snd_pcm_t *pcm, snd_pcm_status_t * status) + snd_pcm_status(dsnoop->spcm, status); + state = snd_pcm_state(dsnoop->spcm); + status->state = state == SND_PCM_STATE_RUNNING ? dsnoop->state : state; ++ status->appl_ptr = *pcm->appl.ptr; /* slave PCM doesn't set appl_ptr */ + status->trigger_tstamp = dsnoop->trigger_tstamp; + status->avail = snd_pcm_mmap_capture_avail(pcm); + status->avail_max = status->avail > dsnoop->avail_max ? status->avail : dsnoop->avail_max; +-- +2.29.2 + + +From 26fdcb98e6be186016d53f87d06ae47aabe5b7fc Mon Sep 17 00:00:00 2001 +From: Takashi Iwai +Date: Fri, 22 Jan 2021 15:48:40 +0100 +Subject: [PATCH 46/49] pcm: ioplug: Pass appl_ptr and hw_ptr in + snd_pcm_status() + +The snd_pcm_status() of the ioplug doesn't return the current +positions of hw_ptr and appl_ptr as advertised. Fix it by copying the +current values stored in the plugin data. + +Signed-off-by: Takashi Iwai +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_ioplug.c | 2 ++ + 1 file changed, 2 insertions(+) + +diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c +index e141b1f9..a1463bf6 100644 +--- a/src/pcm/pcm_ioplug.c ++++ b/src/pcm/pcm_ioplug.c +@@ -133,6 +133,8 @@ static int snd_pcm_ioplug_status(snd_pcm_t *pcm, snd_pcm_status_t * status) + gettimestamp(&status->tstamp, pcm->tstamp_type); + status->avail = snd_pcm_mmap_avail(pcm); + status->avail_max = io->avail_max; ++ status->appl_ptr = *pcm->appl.ptr; ++ status->hw_ptr = *pcm->hw.ptr; + if (snd_pcm_ioplug_delay(pcm, &sd) < 0) + sd = snd_pcm_mmap_delay(pcm); + status->delay = sd; +-- +2.29.2 + + +From bcc762f99ad642cc0dd9eaceb51f228eba9fe68c Mon Sep 17 00:00:00 2001 +From: Takashi Iwai +Date: Fri, 22 Jan 2021 15:48:41 +0100 +Subject: [PATCH 47/49] pcm: null: Pass appl_ptr and hw_ptr in snd_pcm_status() + +Just like the previous fix for ioplug, the null plugin also misses the +appl_ptr and hw_ptr updates for snd_pcm_status(). Fix it as well. + +Signed-off-by: Takashi Iwai +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_null.c | 2 ++ + 1 file changed, 2 insertions(+) + +diff --git a/src/pcm/pcm_null.c b/src/pcm/pcm_null.c +index 1d815485..c8ea9b38 100644 +--- a/src/pcm/pcm_null.c ++++ b/src/pcm/pcm_null.c +@@ -96,6 +96,8 @@ static int snd_pcm_null_status(snd_pcm_t *pcm, snd_pcm_status_t * status) + memset(status, 0, sizeof(*status)); + status->state = null->state; + status->trigger_tstamp = null->trigger_tstamp; ++ status->appl_ptr = *pcm->appl.ptr; ++ status->hw_ptr = *pcm->hw.ptr; + gettimestamp(&status->tstamp, pcm->tstamp_type); + status->avail = snd_pcm_null_avail_update(pcm); + status->avail_max = pcm->buffer_size; +-- +2.29.2 + + +From 55d59821ffb8f44395cf2b7a424d6523da58e66a Mon Sep 17 00:00:00 2001 +From: Takashi Iwai +Date: Fri, 22 Jan 2021 15:48:42 +0100 +Subject: [PATCH 48/49] pcm: share: Pass appl_ptr and hw_ptr in + snd_pcm_status() + +This one also has the same problem as others; the appl_ptr and hw_ptr +of share plugin aren't updated in snd_pcm_status() call. Fix it. + +Signed-off-by: Takashi Iwai +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm_share.c | 2 ++ + 1 file changed, 2 insertions(+) + +diff --git a/src/pcm/pcm_share.c b/src/pcm/pcm_share.c +index bff9507d..72509491 100644 +--- a/src/pcm/pcm_share.c ++++ b/src/pcm/pcm_share.c +@@ -711,6 +711,8 @@ static int snd_pcm_share_status(snd_pcm_t *pcm, snd_pcm_status_t *status) + _notrunning: + status->delay = sd + d; + status->state = share->state; ++ status->appl_ptr = *pcm->appl.ptr; ++ status->hw_ptr = *pcm->hw.ptr; + status->trigger_tstamp = share->trigger_tstamp; + _end: + Pthread_mutex_unlock(&slave->mutex); +-- +2.29.2 + + +From 00eafe98eebff8ecd1ecdc58470068bfd610cc8d Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 21 Jan 2021 15:45:49 +0100 +Subject: [PATCH 49/49] pcm: pcm_ioplug - fix the avail_update mmap capture + copy issue + +It seems that the commit "pcm: ioplug: Transfer all available data" +introduced new regressions (wrong memory access). The second issue +is that the avail_update in ioplug does not move appl_ptr nor hw_ptr, +so it's possible that the transfers may be repetitive. + +This patch moves the transfer calls to mmap_begin callback where it +should be. The pointer wraps are handled by design now. + +Fixes: 1714332719 ("pcm: ioplug: Transfer all available data") +BugLink: https://github.com/alsa-project/alsa-lib/pull/103 +Tested-by: Andreas Pape +Signed-off-by: Jaroslav Kysela +--- + src/pcm/pcm.c | 20 ++++++++++----- + src/pcm/pcm_ioplug.c | 60 +++++++++++++++++++++++++------------------- + src/pcm/pcm_local.h | 2 ++ + 3 files changed, 50 insertions(+), 32 deletions(-) + +diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c +index 24030b31..a57ce5d9 100644 +--- a/src/pcm/pcm.c ++++ b/src/pcm/pcm.c +@@ -7218,9 +7218,8 @@ int snd_pcm_mmap_begin(snd_pcm_t *pcm, } -@@ -201,17 +201,17 @@ int main(int argc, char *argv[]) - goto _exit; - } -- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, 0)) -+ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, SND_PCM_AUDIO_TSTAMP_TYPE_COMPAT)) - printf("Playback supports audio compat timestamps\n"); -- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, 1)) -+ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, SND_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)) - printf("Playback supports audio default timestamps\n"); -- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, 2)) -+ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, SND_PCM_AUDIO_TSTAMP_TYPE_LINK)) - printf("Playback supports audio link timestamps\n"); -- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, 3)) -+ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE)) - printf("Playback supports audio link absolute timestamps\n"); -- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, 4)) -+ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED)) - printf("Playback supports audio link estimated timestamps\n"); -- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, 5)) -+ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_p, SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED)) - printf("Playback supports audio link synchronized timestamps\n"); + #ifndef DOC_HIDDEN +-/* locked version */ +-int __snd_pcm_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, +- snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames) ++int __snd_pcm_mmap_begin_generic(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, ++ snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames) + { + snd_pcm_uframes_t cont; + snd_pcm_uframes_t f; +@@ -7229,9 +7228,6 @@ int __snd_pcm_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, - snd_pcm_sw_params_alloca(&swparams_p); -@@ -269,17 +269,17 @@ int main(int argc, char *argv[]) - goto _exit; - } + assert(pcm && areas && offset && frames); -- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, 0)) -+ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, SND_PCM_AUDIO_TSTAMP_TYPE_COMPAT)) - printf("Capture supports audio compat timestamps\n"); -- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, 1)) -+ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, SND_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)) - printf("Capture supports audio default timestamps\n"); -- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, 2)) -+ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, SND_PCM_AUDIO_TSTAMP_TYPE_LINK)) - printf("Capture supports audio link timestamps\n"); -- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, 3)) -+ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE)) - printf("Capture supports audio link absolute timestamps\n"); -- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, 4)) -+ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED)) - printf("Capture supports audio link estimated timestamps\n"); -- if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, 5)) -+ if (snd_pcm_hw_params_supports_audio_ts_type(hwparams_c, SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED)) - printf("Capture supports audio link synchronized timestamps\n"); +- if (pcm->fast_ops->mmap_begin) +- return pcm->fast_ops->mmap_begin(pcm->fast_op_arg, areas, offset, frames); +- + /* fallback for plugins that do not specify new callback */ + xareas = snd_pcm_mmap_areas(pcm); + if (xareas == NULL) +@@ -7250,6 +7246,18 @@ int __snd_pcm_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, + *frames = f; + return 0; + } ++ ++/* locked version */ ++int __snd_pcm_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, ++ snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames) ++{ ++ assert(pcm && areas && offset && frames); ++ ++ if (pcm->fast_ops->mmap_begin) ++ return pcm->fast_ops->mmap_begin(pcm->fast_op_arg, areas, offset, frames); ++ ++ return __snd_pcm_mmap_begin_generic(pcm, areas, offset, frames); ++} + #endif - snd_pcm_sw_params_alloca(&swparams_c); + /** +diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c +index a1463bf6..98184398 100644 +--- a/src/pcm/pcm_ioplug.c ++++ b/src/pcm/pcm_ioplug.c +@@ -697,6 +697,38 @@ static snd_pcm_sframes_t snd_pcm_ioplug_readn(snd_pcm_t *pcm, void **bufs, snd_p + } + } + ++static int snd_pcm_ioplug_mmap_begin_capture(snd_pcm_t *pcm, ++ const snd_pcm_channel_area_t **areas, ++ snd_pcm_uframes_t *offset, ++ snd_pcm_uframes_t *frames) ++{ ++ ioplug_priv_t *io = pcm->private_data; ++ int err; ++ ++ err = __snd_pcm_mmap_begin_generic(pcm, areas, offset, frames); ++ if (err < 0) ++ return err; ++ ++ if (io->data->callback->transfer && ++ pcm->access != SND_PCM_ACCESS_RW_INTERLEAVED && ++ pcm->access != SND_PCM_ACCESS_RW_NONINTERLEAVED) { ++ snd_pcm_sframes_t result; ++ result = io->data->callback->transfer(io->data, *areas, *offset, *frames); ++ if (result < 0) ++ return result; ++ } ++ ++ return err; ++} ++ ++static int snd_pcm_ioplug_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, ++ snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames) ++{ ++ if (pcm->stream == SND_PCM_STREAM_PLAYBACK) ++ return __snd_pcm_mmap_begin_generic(pcm, areas, offset, frames); ++ return snd_pcm_ioplug_mmap_begin_capture(pcm, areas, offset, frames); ++} ++ + static snd_pcm_sframes_t snd_pcm_ioplug_mmap_commit(snd_pcm_t *pcm, + snd_pcm_uframes_t offset, + snd_pcm_uframes_t size) +@@ -707,7 +739,7 @@ static snd_pcm_sframes_t snd_pcm_ioplug_mmap_commit(snd_pcm_t *pcm, + const snd_pcm_channel_area_t *areas; + snd_pcm_uframes_t ofs, frames = size; + +- __snd_pcm_mmap_begin(pcm, &areas, &ofs, &frames); ++ __snd_pcm_mmap_begin_generic(pcm, &areas, &ofs, &frames); + if (ofs != offset) + return -EIO; + return ioplug_priv_transfer_areas(pcm, areas, offset, frames); +@@ -727,31 +759,6 @@ static snd_pcm_sframes_t snd_pcm_ioplug_avail_update(snd_pcm_t *pcm) + return -EPIPE; + + avail = snd_pcm_mmap_avail(pcm); +- if (pcm->stream == SND_PCM_STREAM_CAPTURE && +- pcm->access != SND_PCM_ACCESS_RW_INTERLEAVED && +- pcm->access != SND_PCM_ACCESS_RW_NONINTERLEAVED) { +- if (io->data->callback->transfer) { +- const snd_pcm_channel_area_t *areas; +- snd_pcm_uframes_t offset, size = UINT_MAX; +- snd_pcm_sframes_t result; +- +- __snd_pcm_mmap_begin(pcm, &areas, &offset, &size); +- result = io->data->callback->transfer(io->data, areas, offset, size); +- if (result < 0) +- return result; +- +- /* If the available data doesn't fit in the +- contiguous area at the end of the mmap we +- must transfer the remaining data to the +- beginning of the mmap. */ +- if (size < avail) { +- result = io->data->callback->transfer(io->data, areas, +- 0, avail - size); +- if (result < 0) +- return result; +- } +- } +- } + if (avail > io->avail_max) + io->avail_max = avail; + return (snd_pcm_sframes_t)avail; +@@ -947,6 +954,7 @@ static const snd_pcm_fast_ops_t snd_pcm_ioplug_fast_ops = { + .poll_descriptors_count = snd_pcm_ioplug_poll_descriptors_count, + .poll_descriptors = snd_pcm_ioplug_poll_descriptors, + .poll_revents = snd_pcm_ioplug_poll_revents, ++ .mmap_begin = snd_pcm_ioplug_mmap_begin, + }; + + #endif /* !DOC_HIDDEN */ +diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h +index bec5a408..a63f4be0 100644 +--- a/src/pcm/pcm_local.h ++++ b/src/pcm/pcm_local.h +@@ -420,6 +420,8 @@ int _snd_pcm_poll_descriptor(snd_pcm_t *pcm); + #define _snd_pcm_async_descriptor _snd_pcm_poll_descriptor /* FIXME */ + + /* locked versions */ ++int __snd_pcm_mmap_begin_generic(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, ++ snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames); + int __snd_pcm_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, + snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames); + snd_pcm_sframes_t __snd_pcm_mmap_commit(snd_pcm_t *pcm, +-- +2.29.2 + diff --git a/SOURCES/alsa-ucm-git.patch b/SOURCES/alsa-ucm-git.patch index 43b9d26..3e330d2 100644 --- a/SOURCES/alsa-ucm-git.patch +++ b/SOURCES/alsa-ucm-git.patch @@ -1,23 +1,20 @@ -diff --git a/ucm2/HDA-Intel/HDAudio-DualCodecs.conf b/ucm2/HDA-Intel/HDAudio-DualCodecs.conf -index 8aaabd0..6fd7bd1 100644 ---- a/ucm2/HDA-Intel/HDAudio-DualCodecs.conf -+++ b/ucm2/HDA-Intel/HDAudio-DualCodecs.conf -@@ -4,3 +4,15 @@ SectionUseCase."HiFi" { - File "HiFi-dual.conf" - Comment "Default" - } -+ -+BootSequence [ -+ cset "name='Headphone Playback Volume' 60%" -+ cset "name='Headphone Playback Switch' off" -+ cset "name='Speaker Playback Volume' 60%" -+ cset "name='Speaker Playback Switch' on" -+ cset "name='Front Playback Volume' 100%" -+ cset "name='Front Playback Switch' on" -+ cset "name='Rear-Panel Capture Volume' 100%" -+ cset "name='Rear-Panel Capture Switch' on" -+ cset "name='Input Source' Rear Mic" -+] +From 33be660e4b1e75c19d5332556c3d2636dd3344bf Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Mon, 2 Nov 2020 16:44:51 +0100 +Subject: [PATCH 01/24] fix the ucm2/codecs/hda/hdmi.conf use + +The mentioned commit (dcef48f13d) removed the index for the IEC958 on/off +control which made the HDMI 2+ outputs silent. + +Fixes: dcef48f13d ("HDA-Intel: add support for AMD acp microphone devices") +BugLink: https://github.com/alsa-project/alsa-lib/issues/95 +Signed-off-by: Jaroslav Kysela +--- + ucm2/HDA-Intel/Hdmi.conf | 8 +++++++- + ucm2/codecs/hda/hdmi.conf | 4 ++-- + ucm2/sof-hda-dsp/Hdmi.conf | 3 +++ + 3 files changed, 12 insertions(+), 3 deletions(-) + diff --git a/ucm2/HDA-Intel/Hdmi.conf b/ucm2/HDA-Intel/Hdmi.conf index c8b6e77..c533e49 100644 --- a/ucm2/HDA-Intel/Hdmi.conf @@ -73,145 +70,12 @@ index c8b6e77..c533e49 100644 + Include.hdmi6.File "/codecs/hda/hdmi.conf" } } -diff --git a/ucm2/HDA-Intel/HiFi-acp.conf b/ucm2/HDA-Intel/HiFi-acp.conf -index 15eeca5..123ae15 100644 ---- a/ucm2/HDA-Intel/HiFi-acp.conf -+++ b/ucm2/HDA-Intel/HiFi-acp.conf -@@ -4,6 +4,5 @@ SectionDevice."Mic1" { - Value { - CapturePriority 100 - CapturePCM "hw:${var:AcpCardId}" -- CaptureMixerElem "Capture" - } - } -diff --git a/ucm2/HDA-Intel/HiFi-dual.conf b/ucm2/HDA-Intel/HiFi-dual.conf -index cff948e..f2c6915 100644 ---- a/ucm2/HDA-Intel/HiFi-dual.conf -+++ b/ucm2/HDA-Intel/HiFi-dual.conf -@@ -8,6 +8,8 @@ SectionVerb { - cset "name='Front Playback Switch' on" - cset "name='Rear-Panel Capture Volume' 100%" - cset "name='Rear-Panel Capture Switch' on" -+ cset "name='Headphone Playback Switch' off" -+ cset "name='Speaker Playback Switch' off" - ] - - DisableSequence [ -@@ -27,6 +29,14 @@ SectionDevice."Speaker" { - PlaybackMixerElem "Speaker" - } - -+ EnableSequence [ -+ cset "name='Speaker Playback Switch' on" -+ ] -+ -+ DisableSequence [ -+ cset "name='Speaker Playback Switch' off" -+ ] -+ - ConflictingDevice [ - "Headphones" - ] -@@ -54,6 +64,14 @@ SectionDevice."Headphones" { - JackHWMute "Speaker" - } - -+ EnableSequence [ -+ cset "name='Headphone Playback Switch' on" -+ ] -+ -+ DisableSequence [ -+ cset "name='Headphone Playback Switch' off" -+ ] -+ - ConflictingDevice [ - "Speaker" - ] -diff --git a/ucm2/HDA-Intel/init.conf b/ucm2/HDA-Intel/init.conf -index 1a351a9..6310eb3 100644 ---- a/ucm2/HDA-Intel/init.conf -+++ b/ucm2/HDA-Intel/init.conf -@@ -15,6 +15,7 @@ If.master { - } - True.BootSequence [ - cset "name='Master Playback Volume' 60%" -+ cset "name='Master Playback Switch' on" - ] - } - -diff --git a/ucm2/chtnau8824/HiFi.conf b/ucm2/chtnau8824/HiFi.conf -index 3ccd6b1..105f360 100644 ---- a/ucm2/chtnau8824/HiFi.conf -+++ b/ucm2/chtnau8824/HiFi.conf -@@ -1,4 +1,5 @@ - Define.Speaker "Speaker" -+Define.Mic "InternalMic" - - If.cfg-mspk { - Condition { -@@ -11,6 +12,19 @@ If.cfg-mspk { - } - } - -+If.cfg-mic { -+ Condition { -+ Type RegexMatch -+ String "${CardLongName}" -+ # Medion E22??T models put DMIC in their product version -+ # when using DMICs -+ Regex "(MEDION-E22..T.*-DMIC-)" -+ } -+ True { -+ Define.Mic "DMIC1_2" -+ } -+} -+ - SectionVerb { - - Value { -@@ -34,8 +48,15 @@ SectionVerb { - } - } - --Include.spk.File "/codecs/nau8824/${var:Speaker}.conf" --Include.hp.File "/codecs/nau8824/HeadPhones.conf" -+# The includes using $vars in there path must be conditional otherwise the -+# $var gets expanded before the other If-s above can change the vars. -+If.cfg-includes { -+ Condition { Type String Empty "" } -+ True { -+ Include.spk.File "/codecs/nau8824/${var:Speaker}.conf" -+ Include.hp.File "/codecs/nau8824/HeadPhones.conf" - --Include.mic.File "/codecs/nau8824/InternalMic.conf" --Include.hsmic.File "/codecs/nau8824/HeadsetMic.conf" -+ Include.mic.File "/codecs/nau8824/${var:Mic}.conf" -+ Include.hsmic.File "/codecs/nau8824/HeadsetMic.conf" -+ } -+} -diff --git a/ucm2/chtrt5645/HiFi.conf b/ucm2/chtrt5645/HiFi.conf -index 10b0a1c..30fad50 100644 ---- a/ucm2/chtrt5645/HiFi.conf -+++ b/ucm2/chtrt5645/HiFi.conf -@@ -18,7 +18,7 @@ If.cfg-dmic2 { - Condition { - Type RegexMatch - String "${CardLongName}" -- Regex "(LENOVO.*LenovoMIIX320|MEDION.*Wingman)" -+ Regex "(LENOVO.*LenovoMIIX320|MEDION.*Wingman|Standard-EF20EA-1.0)" - } - True { - Define.AnalogMic "" diff --git a/ucm2/codecs/hda/hdmi.conf b/ucm2/codecs/hda/hdmi.conf -index ba80fef..c7b7ba4 100644 +index ba80fef..32a77f4 100644 --- a/ucm2/codecs/hda/hdmi.conf +++ b/ucm2/codecs/hda/hdmi.conf -@@ -5,14 +5,14 @@ If.hdmi { - } - True { - SectionDevice."HDMI${var:HdmiNum}" { -- Comment "HDMI${var:HdmiNum} Output" -+ Comment "HDMI / DisplayPort ${var:HdmiNum} Output" +@@ -8,11 +8,11 @@ If.hdmi { + Comment "HDMI${var:HdmiNum} Output" EnableSequence [ - cset "name='IEC958 Playback Switch' on" @@ -224,67 +88,6 @@ index ba80fef..c7b7ba4 100644 ] Value { -diff --git a/ucm2/codecs/nau8824/DMIC1_2.conf b/ucm2/codecs/nau8824/DMIC1_2.conf -new file mode 100644 -index 0000000..f54f671 ---- /dev/null -+++ b/ucm2/codecs/nau8824/DMIC1_2.conf -@@ -0,0 +1,30 @@ -+# Stereo DMICs on the DMIC1 and DMIC2 inputs -+ -+SectionDevice."Mic" { -+ Comment "Internal Digital Microphones" -+ -+ Value { -+ CapturePriority 200 -+ CapturePCM "hw:${CardId}" -+ } -+ -+ ConflictingDevice [ -+ "Headset" -+ ] -+ -+ EnableSequence [ -+ # Note needs to be swapped / swap is deliberate! -+ cset "name='ADC CH0 Select' 1" -+ cset "name='ADC CH1 Select' 0" -+ -+ cset "name='Int Mic Switch' on" -+ cset "name='DMIC1 Enable Switch' on" -+ cset "name='DMIC2 Enable Switch' on" -+ ] -+ -+ DisableSequence [ -+ cset "name='Int Mic Switch' off" -+ cset "name='DMIC1 Enable Switch' off" -+ cset "name='DMIC2 Enable Switch' off" -+ ] -+} -diff --git a/ucm2/codecs/nau8824/EnableSeq.conf b/ucm2/codecs/nau8824/EnableSeq.conf -index 433e8e2..3650d06 100644 ---- a/ucm2/codecs/nau8824/EnableSeq.conf -+++ b/ucm2/codecs/nau8824/EnableSeq.conf -@@ -7,8 +7,8 @@ EnableSequence [ - cset "name='DMIC2 Enable Switch' off" - cset "name='DMIC3 Enable Switch' off" - cset "name='DMIC4 Enable Switch' off" -- cset "name='MIC1 Volume' 10" -- cset "name='MIC2 Volume' 10" -+ cset "name='MIC1 Volume' 14" -+ cset "name='MIC2 Volume' 14" - # Button Configuration - cset "name='THD for key media' 10" - cset "name='THD for key voice command' 16" -diff --git a/ucm2/codecs/rt715/init.conf b/ucm2/codecs/rt715/init.conf -index 6e07b2d..15a0571 100644 ---- a/ucm2/codecs/rt715/init.conf -+++ b/ucm2/codecs/rt715/init.conf -@@ -7,4 +7,5 @@ BootSequence [ - cset "name='rt715 ADC 25 Mux' 4" - cset "name='rt715 ADC 27 Capture Switch' 1" - cset "name='rt715 ADC 07 Capture Switch' 1" -+ cset "name='rt715 ADC 07 Capture Volume' 58" - ] diff --git a/ucm2/sof-hda-dsp/Hdmi.conf b/ucm2/sof-hda-dsp/Hdmi.conf index 7b777d1..1726567 100644 --- a/ucm2/sof-hda-dsp/Hdmi.conf @@ -313,18 +116,48 @@ index 7b777d1..1726567 100644 HdmiPrio 700 } Include.hdmi3.File "/codecs/hda/hdmi.conf" -diff --git a/ucm2/sof-hda-dsp/sof-hda-dsp.conf b/ucm2/sof-hda-dsp/sof-hda-dsp.conf -index c8069a6..79b54a9 100644 ---- a/ucm2/sof-hda-dsp/sof-hda-dsp.conf -+++ b/ucm2/sof-hda-dsp/sof-hda-dsp.conf -@@ -24,6 +24,7 @@ If.master { +-- +2.29.2 + + +From f8c2428ccc42488c5e96801cce9f7e28080ecb71 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Mon, 2 Nov 2020 18:15:14 +0100 +Subject: [PATCH 02/24] codecs/hda/hdmi.conf - add DisplayPort to the device + description (Comment) + +Signed-off-by: Jaroslav Kysela +--- + ucm2/codecs/hda/hdmi.conf | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +diff --git a/ucm2/codecs/hda/hdmi.conf b/ucm2/codecs/hda/hdmi.conf +index 32a77f4..c7b7ba4 100644 +--- a/ucm2/codecs/hda/hdmi.conf ++++ b/ucm2/codecs/hda/hdmi.conf +@@ -5,7 +5,7 @@ If.hdmi { } - True.BootSequence [ - cset "name='Master Playback Volume' 60%" -+ cset "name='Master Playback Switch' on" - ] - } + True { + SectionDevice."HDMI${var:HdmiNum}" { +- Comment "HDMI${var:HdmiNum} Output" ++ Comment "HDMI / DisplayPort ${var:HdmiNum} Output" + EnableSequence [ + cset "name='IEC958 Playback Switch',index=${var:HdmiCtlIndex} on" +-- +2.29.2 + + +From 334b12e9a796cddf70779b68418ee7ad52ba1d14 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Mon, 16 Nov 2020 11:48:05 +0100 +Subject: [PATCH 03/24] sof-soundwire: use the codecs/hda/hdmi.conf macro + +Signed-off-by: Jaroslav Kysela +--- + ucm2/sof-soundwire/Hdmi.conf | 83 ++++++++++-------------------------- + 1 file changed, 22 insertions(+), 61 deletions(-) + diff --git a/ucm2/sof-soundwire/Hdmi.conf b/ucm2/sof-soundwire/Hdmi.conf index d0ba790..d5ee9c6 100644 --- a/ucm2/sof-soundwire/Hdmi.conf @@ -431,3 +264,1308 @@ index d0ba790..d5ee9c6 100644 + Include.hdmi3.File "/codecs/hda/hdmi.conf" } } +-- +2.29.2 + + +From 6889736b917eed89c4d775824efd8655400e27e3 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Mon, 7 Dec 2020 09:33:30 +0100 +Subject: [PATCH 04/24] Revert "ucm2: HDA/acp - add Capture simple mixer + element to the ACP Mic device" + +This reverts commit 3320b1ac804595d6c145c1ad7af85917182a3c98. + +This solution does not work. The mute state is shared between +the HDA and AMD ACP in PA, so it may cause the security issue, if +the users do not set the mute manually. + +BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/64 +Signed-off-by: Jaroslav Kysela +--- + ucm2/HDA-Intel/HiFi-acp.conf | 1 - + 1 file changed, 1 deletion(-) + +diff --git a/ucm2/HDA-Intel/HiFi-acp.conf b/ucm2/HDA-Intel/HiFi-acp.conf +index 15eeca5..123ae15 100644 +--- a/ucm2/HDA-Intel/HiFi-acp.conf ++++ b/ucm2/HDA-Intel/HiFi-acp.conf +@@ -4,6 +4,5 @@ SectionDevice."Mic1" { + Value { + CapturePriority 100 + CapturePCM "hw:${var:AcpCardId}" +- CaptureMixerElem "Capture" + } + } +-- +2.29.2 + + +From 13420f94a6858e9c3e3a056a7987d9c0db823394 Mon Sep 17 00:00:00 2001 +From: Hans de Goede +Date: Sun, 6 Dec 2020 13:47:16 +0100 +Subject: [PATCH 05/24] chtnau8824: Fix mono speaker config not working + +Fix the Speaker define / var being changed from "Speaker" to "MonoSpeaker" +not having any effect because the variable gets expanded before the if +making the change is interpreted. + +Signed-off-by: Hans de Goede +Signed-off-by: Jaroslav Kysela +--- + ucm2/chtnau8824/HiFi.conf | 15 +++++++++++---- + 1 file changed, 11 insertions(+), 4 deletions(-) + +diff --git a/ucm2/chtnau8824/HiFi.conf b/ucm2/chtnau8824/HiFi.conf +index 3ccd6b1..d7a5f63 100644 +--- a/ucm2/chtnau8824/HiFi.conf ++++ b/ucm2/chtnau8824/HiFi.conf +@@ -34,8 +34,15 @@ SectionVerb { + } + } + +-Include.spk.File "/codecs/nau8824/${var:Speaker}.conf" +-Include.hp.File "/codecs/nau8824/HeadPhones.conf" ++# The includes using $vars in there path must be conditional otherwise the ++# $var gets expanded before the other If-s above can change the vars. ++If.cfg-includes { ++ Condition { Type String Empty "" } ++ True { ++ Include.spk.File "/codecs/nau8824/${var:Speaker}.conf" ++ Include.hp.File "/codecs/nau8824/HeadPhones.conf" + +-Include.mic.File "/codecs/nau8824/InternalMic.conf" +-Include.hsmic.File "/codecs/nau8824/HeadsetMic.conf" ++ Include.mic.File "/codecs/nau8824/InternalMic.conf" ++ Include.hsmic.File "/codecs/nau8824/HeadsetMic.conf" ++ } ++} +-- +2.29.2 + + +From 7db718571440b13045ebb3150a71745033ff5c0c Mon Sep 17 00:00:00 2001 +From: Hans de Goede +Date: Sun, 6 Dec 2020 13:47:17 +0100 +Subject: [PATCH 06/24] chtnau8824: Add support for laptops using stereo DMICs + +The Medion E2228T uses a NAU8824 codec combined with stereo +DMICs, add support for this. + +Signed-off-by: Hans de Goede +Signed-off-by: Jaroslav Kysela +--- + ucm2/chtnau8824/HiFi.conf | 16 +++++++++++++++- + ucm2/codecs/nau8824/DMIC1_2.conf | 30 ++++++++++++++++++++++++++++++ + 2 files changed, 45 insertions(+), 1 deletion(-) + create mode 100644 ucm2/codecs/nau8824/DMIC1_2.conf + +diff --git a/ucm2/chtnau8824/HiFi.conf b/ucm2/chtnau8824/HiFi.conf +index d7a5f63..105f360 100644 +--- a/ucm2/chtnau8824/HiFi.conf ++++ b/ucm2/chtnau8824/HiFi.conf +@@ -1,4 +1,5 @@ + Define.Speaker "Speaker" ++Define.Mic "InternalMic" + + If.cfg-mspk { + Condition { +@@ -11,6 +12,19 @@ If.cfg-mspk { + } + } + ++If.cfg-mic { ++ Condition { ++ Type RegexMatch ++ String "${CardLongName}" ++ # Medion E22??T models put DMIC in their product version ++ # when using DMICs ++ Regex "(MEDION-E22..T.*-DMIC-)" ++ } ++ True { ++ Define.Mic "DMIC1_2" ++ } ++} ++ + SectionVerb { + + Value { +@@ -42,7 +56,7 @@ If.cfg-includes { + Include.spk.File "/codecs/nau8824/${var:Speaker}.conf" + Include.hp.File "/codecs/nau8824/HeadPhones.conf" + +- Include.mic.File "/codecs/nau8824/InternalMic.conf" ++ Include.mic.File "/codecs/nau8824/${var:Mic}.conf" + Include.hsmic.File "/codecs/nau8824/HeadsetMic.conf" + } + } +diff --git a/ucm2/codecs/nau8824/DMIC1_2.conf b/ucm2/codecs/nau8824/DMIC1_2.conf +new file mode 100644 +index 0000000..f54f671 +--- /dev/null ++++ b/ucm2/codecs/nau8824/DMIC1_2.conf +@@ -0,0 +1,30 @@ ++# Stereo DMICs on the DMIC1 and DMIC2 inputs ++ ++SectionDevice."Mic" { ++ Comment "Internal Digital Microphones" ++ ++ Value { ++ CapturePriority 200 ++ CapturePCM "hw:${CardId}" ++ } ++ ++ ConflictingDevice [ ++ "Headset" ++ ] ++ ++ EnableSequence [ ++ # Note needs to be swapped / swap is deliberate! ++ cset "name='ADC CH0 Select' 1" ++ cset "name='ADC CH1 Select' 0" ++ ++ cset "name='Int Mic Switch' on" ++ cset "name='DMIC1 Enable Switch' on" ++ cset "name='DMIC2 Enable Switch' on" ++ ] ++ ++ DisableSequence [ ++ cset "name='Int Mic Switch' off" ++ cset "name='DMIC1 Enable Switch' off" ++ cset "name='DMIC2 Enable Switch' off" ++ ] ++} +-- +2.29.2 + + +From dacfe2fc7b41b800f4f379ce4b1722462b97cab2 Mon Sep 17 00:00:00 2001 +From: Hans de Goede +Date: Sun, 6 Dec 2020 13:47:18 +0100 +Subject: [PATCH 07/24] chtnau8824: Boost analog mic volumes a bit + +Both Mic1 (internal analog mic) and Mic2 (headset mic) are a bit soft +with the current settings, boost their volumes a bit. + +Signed-off-by: Hans de Goede +Signed-off-by: Jaroslav Kysela +--- + ucm2/codecs/nau8824/EnableSeq.conf | 4 ++-- + 1 file changed, 2 insertions(+), 2 deletions(-) + +diff --git a/ucm2/codecs/nau8824/EnableSeq.conf b/ucm2/codecs/nau8824/EnableSeq.conf +index 433e8e2..3650d06 100644 +--- a/ucm2/codecs/nau8824/EnableSeq.conf ++++ b/ucm2/codecs/nau8824/EnableSeq.conf +@@ -7,8 +7,8 @@ EnableSequence [ + cset "name='DMIC2 Enable Switch' off" + cset "name='DMIC3 Enable Switch' off" + cset "name='DMIC4 Enable Switch' off" +- cset "name='MIC1 Volume' 10" +- cset "name='MIC2 Volume' 10" ++ cset "name='MIC1 Volume' 14" ++ cset "name='MIC2 Volume' 14" + # Button Configuration + cset "name='THD for key media' 10" + cset "name='THD for key voice command' 16" +-- +2.29.2 + + +From 263bd26b1216c933db3d216197a78678d0f8610e Mon Sep 17 00:00:00 2001 +From: Libin Yang +Date: Thu, 19 Nov 2020 20:01:30 +0800 +Subject: [PATCH 08/24] rt715: init: setup ADC07 to a proper volume + +This patch initiates 'rt715 ADC 07 Capture Volume' to a proper volume. + +Signed-off-by: Libin Yang +Signed-off-by: Jaroslav Kysela +--- + ucm2/codecs/rt715/init.conf | 1 + + 1 file changed, 1 insertion(+) + +diff --git a/ucm2/codecs/rt715/init.conf b/ucm2/codecs/rt715/init.conf +index 6e07b2d..15a0571 100644 +--- a/ucm2/codecs/rt715/init.conf ++++ b/ucm2/codecs/rt715/init.conf +@@ -7,4 +7,5 @@ BootSequence [ + cset "name='rt715 ADC 25 Mux' 4" + cset "name='rt715 ADC 27 Capture Switch' 1" + cset "name='rt715 ADC 07 Capture Switch' 1" ++ cset "name='rt715 ADC 07 Capture Volume' 58" + ] +-- +2.29.2 + + +From cf267cce6d553d4d436c166c23d3befc5476acef Mon Sep 17 00:00:00 2001 +From: Hui Wang +Date: Thu, 29 Oct 2020 13:34:02 +0800 +Subject: [PATCH 09/24] sof-hda-dsp: Set Master Playback Switch on in the + BootSequence + +Otherwise, the audio output will be muted by default if there is no +/var/lib/alsa/asound.state. I have experienced this issue When newly +install an OS. + +Signed-off-by: Hui Wang +Signed-off-by: Jaroslav Kysela +--- + ucm2/HDA-Intel/init.conf | 1 + + ucm2/sof-hda-dsp/sof-hda-dsp.conf | 1 + + 2 files changed, 2 insertions(+) + +diff --git a/ucm2/HDA-Intel/init.conf b/ucm2/HDA-Intel/init.conf +index 1a351a9..6310eb3 100644 +--- a/ucm2/HDA-Intel/init.conf ++++ b/ucm2/HDA-Intel/init.conf +@@ -15,6 +15,7 @@ If.master { + } + True.BootSequence [ + cset "name='Master Playback Volume' 60%" ++ cset "name='Master Playback Switch' on" + ] + } + +diff --git a/ucm2/sof-hda-dsp/sof-hda-dsp.conf b/ucm2/sof-hda-dsp/sof-hda-dsp.conf +index c8069a6..79b54a9 100644 +--- a/ucm2/sof-hda-dsp/sof-hda-dsp.conf ++++ b/ucm2/sof-hda-dsp/sof-hda-dsp.conf +@@ -24,6 +24,7 @@ If.master { + } + True.BootSequence [ + cset "name='Master Playback Volume' 60%" ++ cset "name='Master Playback Switch' on" + ] + } + +-- +2.29.2 + + +From 5405544752c18153acc310ee7e4a4bacda040dca Mon Sep 17 00:00:00 2001 +From: Kai-Heng Feng +Date: Thu, 27 Aug 2020 21:18:13 +0800 +Subject: [PATCH 10/24] HDA-Intel/HiFi-dual: Add EnableSequence and + DisableSequence for Speaker and Headphones + +BugLink: https://github.com/alsa-project/alsa-ucm-conf/pull/48 +Signed-off-by: Kai-Heng Feng +Signed-off-by: Jaroslav Kysela +--- + ucm2/HDA-Intel/HiFi-dual.conf | 16 ++++++++++++++++ + 1 file changed, 16 insertions(+) + +diff --git a/ucm2/HDA-Intel/HiFi-dual.conf b/ucm2/HDA-Intel/HiFi-dual.conf +index cff948e..bacfd80 100644 +--- a/ucm2/HDA-Intel/HiFi-dual.conf ++++ b/ucm2/HDA-Intel/HiFi-dual.conf +@@ -27,6 +27,14 @@ SectionDevice."Speaker" { + PlaybackMixerElem "Speaker" + } + ++ EnableSequence [ ++ cset "name='Speaker Playback Switch' on" ++ ] ++ ++ DisableSequence [ ++ cset "name='Speaker Playback Switch' off" ++ ] ++ + ConflictingDevice [ + "Headphones" + ] +@@ -54,6 +62,14 @@ SectionDevice."Headphones" { + JackHWMute "Speaker" + } + ++ EnableSequence [ ++ cset "name='Headphone Playback Switch' on" ++ ] ++ ++ DisableSequence [ ++ cset "name='Headphone Playback Switch' off" ++ ] ++ + ConflictingDevice [ + "Speaker" + ] +-- +2.29.2 + + +From 5947daef124c84fae511233248196576dfbfab7c Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Wed, 16 Dec 2020 08:55:04 +0100 +Subject: [PATCH 11/24] HDA-Intel/HiFi-dual: Add BootSequence and disable + playback channels in verb init + +BugLink: https://github.com/alsa-project/alsa-ucm-conf/pull/48 +Signed-off-by: Jaroslav Kysela +--- + ucm2/HDA-Intel/HDAudio-DualCodecs.conf | 12 ++++++++++++ + ucm2/HDA-Intel/HiFi-dual.conf | 2 ++ + 2 files changed, 14 insertions(+) + +diff --git a/ucm2/HDA-Intel/HDAudio-DualCodecs.conf b/ucm2/HDA-Intel/HDAudio-DualCodecs.conf +index 8aaabd0..6fd7bd1 100644 +--- a/ucm2/HDA-Intel/HDAudio-DualCodecs.conf ++++ b/ucm2/HDA-Intel/HDAudio-DualCodecs.conf +@@ -4,3 +4,15 @@ SectionUseCase."HiFi" { + File "HiFi-dual.conf" + Comment "Default" + } ++ ++BootSequence [ ++ cset "name='Headphone Playback Volume' 60%" ++ cset "name='Headphone Playback Switch' off" ++ cset "name='Speaker Playback Volume' 60%" ++ cset "name='Speaker Playback Switch' on" ++ cset "name='Front Playback Volume' 100%" ++ cset "name='Front Playback Switch' on" ++ cset "name='Rear-Panel Capture Volume' 100%" ++ cset "name='Rear-Panel Capture Switch' on" ++ cset "name='Input Source' Rear Mic" ++] +diff --git a/ucm2/HDA-Intel/HiFi-dual.conf b/ucm2/HDA-Intel/HiFi-dual.conf +index bacfd80..f2c6915 100644 +--- a/ucm2/HDA-Intel/HiFi-dual.conf ++++ b/ucm2/HDA-Intel/HiFi-dual.conf +@@ -8,6 +8,8 @@ SectionVerb { + cset "name='Front Playback Switch' on" + cset "name='Rear-Panel Capture Volume' 100%" + cset "name='Rear-Panel Capture Switch' on" ++ cset "name='Headphone Playback Switch' off" ++ cset "name='Speaker Playback Switch' off" + ] + + DisableSequence [ +-- +2.29.2 + + +From 5634db4be3ed435f3e59f30a71c3c3711162929d Mon Sep 17 00:00:00 2001 +From: Jian-Hong Pan +Date: Tue, 15 Dec 2020 16:54:03 +0800 +Subject: [PATCH 12/24] chtrt5645: Enable Internal MIC of ECS EF20EA + +The ECS EF20EA uses a digital mic connected to the dmic2 pin, rather +then the default analog mic. + +1 [chtrt5645 ]: chtrt5645 - chtrt5645 + Standard-EF20EA-1.0 + +Add a card long name match for this, so that the right config is used. + +BugLink: https://github.com/alsa-project/alsa-ucm-conf/pull/68 +Signed-off-by: Jian-Hong Pan +Signed-off-by: Jaroslav Kysela +--- + ucm2/chtrt5645/HiFi.conf | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +diff --git a/ucm2/chtrt5645/HiFi.conf b/ucm2/chtrt5645/HiFi.conf +index 10b0a1c..30fad50 100644 +--- a/ucm2/chtrt5645/HiFi.conf ++++ b/ucm2/chtrt5645/HiFi.conf +@@ -18,7 +18,7 @@ If.cfg-dmic2 { + Condition { + Type RegexMatch + String "${CardLongName}" +- Regex "(LENOVO.*LenovoMIIX320|MEDION.*Wingman)" ++ Regex "(LENOVO.*LenovoMIIX320|MEDION.*Wingman|Standard-EF20EA-1.0)" + } + True { + Define.AnalogMic "" +-- +2.29.2 + + +From bdd05ac3396a3a09f3818aaa9157eb7ef07676d6 Mon Sep 17 00:00:00 2001 +From: Hans de Goede +Date: Sat, 9 Jan 2021 22:02:51 +0100 +Subject: [PATCH 13/24] bytcr-rt5640: Add support for devices without speakers + and/or an internal mic + +There are kernel patches pending upstream for the bytcr-rt5640 machine-driver +to support devices without speakers and/or an internal mic. + +Since the UCM profile already conditionally loads the speaker and +internal-mic codec include files, these will simply get skipped in this +case, so this almost works with the current UCM profile without changes. + +The only troublesome part is the ConflictingDevice sections in the codec +HeadPhones.conf and HeadsetMic.conf files, which refer to resp. a "Speaker" +and a "Mic" device. Without any of the speaker or mic codec conf files +being included there will not by any "Speaker" / "Mic" devices leading +to an error while parsing the HeadPhones.conf / HeadsetMic.conf files. + +This commit makes the ConflictingDevice section conditional, fixing this. + +Cc: Rasmus Porsager +Signed-off-by: Hans de Goede +Signed-off-by: Jaroslav Kysela +--- + ucm2/bytcr-rt5640/HiFi-Components.conf | 28 +++++++++++++++++++++----- + ucm2/bytcr-rt5640/HiFi-LongName.conf | 28 +++++++++++++++++++++----- + ucm2/codecs/rt5640/HeadPhones.conf | 14 ++++++++++--- + ucm2/codecs/rt5640/HeadsetMic.conf | 14 ++++++++++--- + 4 files changed, 68 insertions(+), 16 deletions(-) + +diff --git a/ucm2/bytcr-rt5640/HiFi-Components.conf b/ucm2/bytcr-rt5640/HiFi-Components.conf +index 2ce4f59..a76b42d 100644 +--- a/ucm2/bytcr-rt5640/HiFi-Components.conf ++++ b/ucm2/bytcr-rt5640/HiFi-Components.conf +@@ -1,10 +1,16 @@ ++Define.HaveSpeaker "" ++Define.HaveInternalMic "" ++ + If.spk { + Condition { + Type String + Haystack "${CardComponents}" + Needle "cfg-spk:2" + } +- True.Include.spk.File "/codecs/rt5640/Speaker.conf" ++ True { ++ Include.spk.File "/codecs/rt5640/Speaker.conf" ++ Define.HaveSpeaker "yes" ++ } + } + + If.mono { +@@ -13,7 +19,10 @@ If.mono { + Haystack "${CardComponents}" + Needle "cfg-spk:1" + } +- True.Include.mspk.File "/codecs/rt5640/MonoSpeaker.conf" ++ True { ++ Include.mspk.File "/codecs/rt5640/MonoSpeaker.conf" ++ Define.HaveSpeaker "yes" ++ } + } + + Include.hs.File "/codecs/rt5640/HeadPhones.conf" +@@ -24,7 +33,10 @@ If.dmic1 { + Haystack "${CardComponents}" + Needle "cfg-mic:dmic1" + } +- True.Include.dmic.File "/codecs/rt5640/DigitalMics.conf" ++ True { ++ Include.dmic.File "/codecs/rt5640/DigitalMics.conf" ++ Define.HaveInternalMic "yes" ++ } + } + + If.in1 { +@@ -33,7 +45,10 @@ If.in1 { + Haystack "${CardComponents}" + Needle "cfg-mic:in1" + } +- True.Include.mic1.File "/codecs/rt5640/IN1-InternalMic.conf" ++ True { ++ Include.mic1.File "/codecs/rt5640/IN1-InternalMic.conf" ++ Define.HaveInternalMic "yes" ++ } + } + + If.in3 { +@@ -42,7 +57,10 @@ If.in3 { + Haystack "${CardComponents}" + Needle "cfg-mic:in3" + } +- True.Include.mic3.File "/codecs/rt5640/IN3-InternalMic.conf" ++ True { ++ Include.mic3.File "/codecs/rt5640/IN3-InternalMic.conf" ++ Define.HaveInternalMic "yes" ++ } + } + + Include.hsmic.File "/codecs/rt5640/HeadsetMic.conf" +diff --git a/ucm2/bytcr-rt5640/HiFi-LongName.conf b/ucm2/bytcr-rt5640/HiFi-LongName.conf +index c07f153..6cb5556 100644 +--- a/ucm2/bytcr-rt5640/HiFi-LongName.conf ++++ b/ucm2/bytcr-rt5640/HiFi-LongName.conf +@@ -1,10 +1,16 @@ ++Define.HaveSpeaker "" ++Define.HaveInternalMic "" ++ + If.spk { + Condition { + Type String + Haystack "${CardLongName}" + Needle "-stereo-spk" + } +- True.Include.spk.File "/codecs/rt5640/Speaker.conf" ++ True { ++ Include.spk.File "/codecs/rt5640/Speaker.conf" ++ Define.HaveSpeaker "yes" ++ } + } + + If.mono { +@@ -13,7 +19,10 @@ If.mono { + Haystack "${CardLongName}" + Needle "-mono-spk" + } +- True.Include.mspk.File "/codecs/rt5640/MonoSpeaker.conf" ++ True { ++ Include.mspk.File "/codecs/rt5640/MonoSpeaker.conf" ++ Define.HaveSpeaker "yes" ++ } + } + + Include.hs.File "/codecs/rt5640/HeadPhones.conf" +@@ -24,7 +33,10 @@ If.dmic1 { + Haystack "${CardLongName}" + Needle "-dmic1-mic" + } +- True.Include.dmic.File "/codecs/rt5640/DigitalMics.conf" ++ True { ++ Include.dmic.File "/codecs/rt5640/DigitalMics.conf" ++ Define.HaveInternalMic "yes" ++ } + } + + If.in1 { +@@ -33,7 +45,10 @@ If.in1 { + Haystack "${CardLongName}" + Needle "-in1-mic" + } +- True.Include.mic1.File "/codecs/rt5640/IN1-InternalMic.conf" ++ True { ++ Include.mic1.File "/codecs/rt5640/IN1-InternalMic.conf" ++ Define.HaveInternalMic "yes" ++ } + } + + If.in3 { +@@ -42,7 +57,10 @@ If.in3 { + Haystack "${CardLongName}" + Needle "-in3-mic" + } +- True.Include.mic3.File "/codecs/rt5640/IN3-InternalMic.conf" ++ True { ++ Include.mic3.File "/codecs/rt5640/IN3-InternalMic.conf" ++ Define.HaveInternalMic "yes" ++ } + } + + Include.hsmic.File "/codecs/rt5640/HeadsetMic.conf" +diff --git a/ucm2/codecs/rt5640/HeadPhones.conf b/ucm2/codecs/rt5640/HeadPhones.conf +index 42151d6..b7707e5 100644 +--- a/ucm2/codecs/rt5640/HeadPhones.conf ++++ b/ucm2/codecs/rt5640/HeadPhones.conf +@@ -1,9 +1,17 @@ + SectionDevice."Headphones" { + Comment "Headphones" + +- ConflictingDevice [ +- "Speaker" +- ] ++ If.have-spk { ++ Condition { ++ Type String ++ Empty "${var:HaveSpeaker}" ++ } ++ False { ++ ConflictingDevice [ ++ "Speaker" ++ ] ++ } ++ } + + EnableSequence [ + cset "name='DAC MIXL INF1 Switch' on" +diff --git a/ucm2/codecs/rt5640/HeadsetMic.conf b/ucm2/codecs/rt5640/HeadsetMic.conf +index 7a8dfb8..f88b611 100644 +--- a/ucm2/codecs/rt5640/HeadsetMic.conf ++++ b/ucm2/codecs/rt5640/HeadsetMic.conf +@@ -1,9 +1,17 @@ + SectionDevice."Headset" { + Comment "Headset Microphone" + +- ConflictingDevice [ +- "Mic" +- ] ++ If.have-mic { ++ Condition { ++ Type String ++ Empty "${var:HaveInternalMic}" ++ } ++ False { ++ ConflictingDevice [ ++ "Mic" ++ ] ++ } ++ } + + EnableSequence [ + cset "name='Headset Mic Switch' on" +-- +2.29.2 + + +From cd02791e9360bcfa2cd4bed40cf5ddabb378ef93 Mon Sep 17 00:00:00 2001 +From: Hans de Goede +Date: Sat, 9 Jan 2021 22:02:52 +0100 +Subject: [PATCH 14/24] rt5640: Move standard DAC setup to EnableSeq.conf + +No matter which output is used, we always need to setup the standard +DAC config. Move this to the shared EnableSeq.conf to avoid having +to duplicate it in various places. + +Signed-off-by: Hans de Goede +Signed-off-by: Jaroslav Kysela +--- + ucm2/codecs/rt5640/EnableSeq.conf | 6 ++++++ + ucm2/codecs/rt5640/HeadPhones.conf | 6 ------ + ucm2/codecs/rt5640/MonoSpeaker.conf | 6 ------ + ucm2/codecs/rt5640/Speaker.conf | 6 ------ + 4 files changed, 6 insertions(+), 18 deletions(-) + +diff --git a/ucm2/codecs/rt5640/EnableSeq.conf b/ucm2/codecs/rt5640/EnableSeq.conf +index cbf4038..145a23b 100644 +--- a/ucm2/codecs/rt5640/EnableSeq.conf ++++ b/ucm2/codecs/rt5640/EnableSeq.conf +@@ -1,5 +1,11 @@ + EnableSequence [ + # RT5640 default output routing ++ cset "name='DAC MIXL INF1 Switch' on" ++ cset "name='DAC MIXR INF1 Switch' on" ++ cset "name='Stereo DAC MIXL DAC L1 Switch' on" ++ cset "name='Stereo DAC MIXR DAC R1 Switch' on" ++ cset "name='Stereo DAC MIXL DAC L2 Switch' on" ++ cset "name='Stereo DAC MIXR DAC R2 Switch' on" + cset "name='OUT MIXL DAC L1 Switch' on" + cset "name='OUT MIXR DAC R1 Switch' on" + +diff --git a/ucm2/codecs/rt5640/HeadPhones.conf b/ucm2/codecs/rt5640/HeadPhones.conf +index b7707e5..a9e4d20 100644 +--- a/ucm2/codecs/rt5640/HeadPhones.conf ++++ b/ucm2/codecs/rt5640/HeadPhones.conf +@@ -14,12 +14,6 @@ SectionDevice."Headphones" { + } + + EnableSequence [ +- cset "name='DAC MIXL INF1 Switch' on" +- cset "name='DAC MIXR INF1 Switch' on" +- cset "name='Stereo DAC MIXL DAC L1 Switch' on" +- cset "name='Stereo DAC MIXR DAC R1 Switch' on" +- cset "name='Stereo DAC MIXL DAC L2 Switch' on" +- cset "name='Stereo DAC MIXR DAC R2 Switch' on" + cset "name='HPO MIX HPVOL Switch' on" + cset "name='Headphone Switch' on" + cset "name='HP Channel Switch' on" +diff --git a/ucm2/codecs/rt5640/MonoSpeaker.conf b/ucm2/codecs/rt5640/MonoSpeaker.conf +index ddc7ba8..23cf34c 100644 +--- a/ucm2/codecs/rt5640/MonoSpeaker.conf ++++ b/ucm2/codecs/rt5640/MonoSpeaker.conf +@@ -6,12 +6,6 @@ SectionDevice."Speaker" { + ] + + EnableSequence [ +- cset "name='DAC MIXL INF1 Switch' on" +- cset "name='DAC MIXR INF1 Switch' on" +- cset "name='Stereo DAC MIXL DAC L1 Switch' on" +- cset "name='Stereo DAC MIXR DAC R1 Switch' on" +- cset "name='Stereo DAC MIXL DAC L2 Switch' on" +- cset "name='Stereo DAC MIXR DAC R2 Switch' on" + cset "name='SPK MIXL DAC L1 Switch' on" + cset "name='SPK MIXR DAC R1 Switch' on" + cset "name='SPOL MIX SPKVOL L Switch' on" +diff --git a/ucm2/codecs/rt5640/Speaker.conf b/ucm2/codecs/rt5640/Speaker.conf +index 411cd13..422cfdf 100644 +--- a/ucm2/codecs/rt5640/Speaker.conf ++++ b/ucm2/codecs/rt5640/Speaker.conf +@@ -6,12 +6,6 @@ SectionDevice."Speaker" { + ] + + EnableSequence [ +- cset "name='DAC MIXL INF1 Switch' on" +- cset "name='DAC MIXR INF1 Switch' on" +- cset "name='Stereo DAC MIXL DAC L1 Switch' on" +- cset "name='Stereo DAC MIXR DAC R1 Switch' on" +- cset "name='Stereo DAC MIXL DAC L2 Switch' on" +- cset "name='Stereo DAC MIXR DAC R2 Switch' on" + cset "name='SPK MIXL DAC L1 Switch' on" + cset "name='SPK MIXR DAC R1 Switch' on" + cset "name='SPOL MIX SPKVOL L Switch' on" +-- +2.29.2 + + +From 95587ae8b75134c4bbc2cec0181e5d529826c9c4 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 14 Jan 2021 18:31:24 +0100 +Subject: [PATCH 15/24] bytcr-rt5640: fix the execution order + +The includes must be run _after_ defines. The Include directives +have precedence. Put everything to If compound. + +Fixes: bdd05ac339 ("bytcr-rt5640: Add support for devices without speakers and/or an internal mic") +Signed-off-by: Jaroslav Kysela +--- + ucm2/bytcr-rt5640/HiFi-Components.conf | 10 ++++++++-- + ucm2/bytcr-rt5640/HiFi-LongName.conf | 10 ++++++++-- + 2 files changed, 16 insertions(+), 4 deletions(-) + +diff --git a/ucm2/bytcr-rt5640/HiFi-Components.conf b/ucm2/bytcr-rt5640/HiFi-Components.conf +index a76b42d..1999717 100644 +--- a/ucm2/bytcr-rt5640/HiFi-Components.conf ++++ b/ucm2/bytcr-rt5640/HiFi-Components.conf +@@ -25,7 +25,10 @@ If.mono { + } + } + +-Include.hs.File "/codecs/rt5640/HeadPhones.conf" ++If.hp { ++ Condition { Type String Empty "" } ++ True.Include.hs.File "/codecs/rt5640/HeadPhones.conf" ++} + + If.dmic1 { + Condition { +@@ -63,4 +66,7 @@ If.in3 { + } + } + +-Include.hsmic.File "/codecs/rt5640/HeadsetMic.conf" ++If.hsmic { ++ Condition { Type String Empty "" } ++ True.Include.hsmic.File "/codecs/rt5640/HeadsetMic.conf" ++} +diff --git a/ucm2/bytcr-rt5640/HiFi-LongName.conf b/ucm2/bytcr-rt5640/HiFi-LongName.conf +index 6cb5556..34acef8 100644 +--- a/ucm2/bytcr-rt5640/HiFi-LongName.conf ++++ b/ucm2/bytcr-rt5640/HiFi-LongName.conf +@@ -25,7 +25,10 @@ If.mono { + } + } + +-Include.hs.File "/codecs/rt5640/HeadPhones.conf" ++If.hp { ++ Condition { Type String Empty "" } ++ True.Include.hs.File "/codecs/rt5640/HeadPhones.conf" ++} + + If.dmic1 { + Condition { +@@ -63,4 +66,7 @@ If.in3 { + } + } + +-Include.hsmic.File "/codecs/rt5640/HeadsetMic.conf" ++If.hsmic { ++ Condition { Type String Empty "" } ++ True.Include.hsmic.File "/codecs/rt5640/HeadsetMic.conf" ++} +-- +2.29.2 + + +From 09e5271401480297c2c9c0b753df9705123a1af1 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Thu, 14 Jan 2021 11:03:56 +0100 +Subject: [PATCH 16/24] ucm2: add initial configuration for TRX40 Gigabyte + Aorus Master Audio + +BugLink: https://github.com/alsa-project/alsa-ucm-conf/pull/25 +Signed-off-by: Jaroslav Kysela +--- + ...Gigabyte-Aorus-Master-Front-Headphone.conf | 1 + + ...Gigabyte-Aorus-Master-Main-Audio-HiFi.conf | 61 +++++++++++++++++++ + .../Gigabyte-Aorus-Master-Main-Audio.conf | 6 ++ + 3 files changed, 68 insertions(+) + create mode 120000 ucm2/USB-Audio/Gigabyte-Aorus-Master-Front-Headphone.conf + create mode 100644 ucm2/USB-Audio/Gigabyte-Aorus-Master-Main-Audio-HiFi.conf + create mode 100644 ucm2/USB-Audio/Gigabyte-Aorus-Master-Main-Audio.conf + +diff --git a/ucm2/USB-Audio/Gigabyte-Aorus-Master-Front-Headphone.conf b/ucm2/USB-Audio/Gigabyte-Aorus-Master-Front-Headphone.conf +new file mode 120000 +index 0000000..1376437 +--- /dev/null ++++ b/ucm2/USB-Audio/Gigabyte-Aorus-Master-Front-Headphone.conf +@@ -0,0 +1 @@ ++../module/lib/linked.conf +\ No newline at end of file +diff --git a/ucm2/USB-Audio/Gigabyte-Aorus-Master-Main-Audio-HiFi.conf b/ucm2/USB-Audio/Gigabyte-Aorus-Master-Main-Audio-HiFi.conf +new file mode 100644 +index 0000000..5c24165 +--- /dev/null ++++ b/ucm2/USB-Audio/Gigabyte-Aorus-Master-Main-Audio-HiFi.conf +@@ -0,0 +1,61 @@ ++Define.SecondaryCardId "$${CardIdByName:Aorus Master Front Headphone}" ++ ++SectionDevice."Speaker" { ++ Comment "Speakers" ++ Value { ++ PlaybackChannels 8 ++ PlaybackPriority 200 ++ PlaybackPCM "hw:${CardId}" ++ JackControl "Line Out Jack" ++ PlaybackMixerElem "Line Out" ++ } ++} ++ ++SectionDevice."Headphones" { ++ Comment "Front Headphones" ++ Value { ++ PlaybackPriority 300 ++ PlaybackPCM "hw:${var:SecondaryCardId}" ++ JackCTL "hw:${var:SecondaryCardId}" ++ JackControl "Headphone - Output Jack" ++ } ++} ++ ++SectionDevice."SPDIF" { ++ Comment "S/PDIF Out" ++ Value { ++ PlaybackPriority 100 ++ PlaybackPCM "hw:${CardId},1" ++ # PlaybackMixerElem "IEC958" ++ } ++} ++ ++SectionDevice."Line" { ++ Comment "Line In" ++ Value { ++ CapturePriority 100 ++ CapturePCM "hw:${CardId}" ++ JackControl "Line Jack" ++ CaptureMixerElem "Line" ++ } ++} ++ ++SectionDevice."Mic1" { ++ Comment "Microphone" ++ Value { ++ CapturePriority 200 ++ CapturePCM "hw:${CardId},1" ++ JackControl "Mic Jack" ++ CaptureMixerElem "Mic" ++ } ++} ++ ++SectionDevice."Mic2" { ++ Comment "Front Microphone" ++ Value { ++ CapturePriority 300 ++ CapturePCM "hw:${CardId},2" ++ JackControl "Front Mic Jack" ++ CaptureMixerElem "Front Mic" ++ } ++} +diff --git a/ucm2/USB-Audio/Gigabyte-Aorus-Master-Main-Audio.conf b/ucm2/USB-Audio/Gigabyte-Aorus-Master-Main-Audio.conf +new file mode 100644 +index 0000000..656b989 +--- /dev/null ++++ b/ucm2/USB-Audio/Gigabyte-Aorus-Master-Main-Audio.conf +@@ -0,0 +1,6 @@ ++Syntax 3 ++Comment "Gigabyte Aorus Integrated Audio" ++SectionUseCase."HiFi" { ++ File "Gigabyte-Aorus-Master-Main-Audio-HiFi.conf" ++ Comment "Default Alsa Profile" ++} +-- +2.29.2 + + +From 062e4710c524283c5d11cbf99931b24c6d963f98 Mon Sep 17 00:00:00 2001 +From: Marijn Suijten +Date: Thu, 14 Jan 2021 14:11:50 +0100 +Subject: [PATCH 17/24] USB-Audio/ALC1220: Bump analog Speaker priority over + S/PDIF + +The `S/PDIF` port does not have any jack sensing, and with a priority +higher than the `Speakers` it will always be selected by default instead +of the `Speakers` even if unplugged. + +Swapping the priorities around allows analog `Speakers` to be selected +first, _if_ they are plugged in. Otherwise `S/PDIF` is used. + +Signed-off-by: Marijn Suijten +Signed-off-by: Jaroslav Kysela +--- + ucm2/USB-Audio/Realtek-ALC1220-VB-Desktop-HiFi.conf | 4 ++-- + 1 file changed, 2 insertions(+), 2 deletions(-) + +diff --git a/ucm2/USB-Audio/Realtek-ALC1220-VB-Desktop-HiFi.conf b/ucm2/USB-Audio/Realtek-ALC1220-VB-Desktop-HiFi.conf +index 344db83..cbf7154 100644 +--- a/ucm2/USB-Audio/Realtek-ALC1220-VB-Desktop-HiFi.conf ++++ b/ucm2/USB-Audio/Realtek-ALC1220-VB-Desktop-HiFi.conf +@@ -2,7 +2,7 @@ SectionDevice."Speaker" { + Comment "Speakers" + Value { + PlaybackChannels 8 +- PlaybackPriority 100 ++ PlaybackPriority 200 + PlaybackPCM "hw:${CardId}" + JackControl "Speaker Jack" + PlaybackMixerElem "Speaker" +@@ -22,7 +22,7 @@ SectionDevice."Headphones" { + SectionDevice."SPDIF" { + Comment "S/PDIF Out" + Value { +- PlaybackPriority 200 ++ PlaybackPriority 100 + PlaybackPCM "hw:${CardId},2" + PlaybackMixerElem "IEC958" + } +-- +2.29.2 + + +From 8e875c9467cc72e53e53104b07d5515f3ef0a779 Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Fri, 15 Jan 2021 14:03:35 +0100 +Subject: [PATCH 18/24] USB-Audio/ALC1220: fix indentation for Speaker device + +Signed-off-by: Jaroslav Kysela +--- + ucm2/USB-Audio/Realtek-ALC1220-VB-Desktop-HiFi.conf | 10 +++++----- + 1 file changed, 5 insertions(+), 5 deletions(-) + +diff --git a/ucm2/USB-Audio/Realtek-ALC1220-VB-Desktop-HiFi.conf b/ucm2/USB-Audio/Realtek-ALC1220-VB-Desktop-HiFi.conf +index cbf7154..0d75384 100644 +--- a/ucm2/USB-Audio/Realtek-ALC1220-VB-Desktop-HiFi.conf ++++ b/ucm2/USB-Audio/Realtek-ALC1220-VB-Desktop-HiFi.conf +@@ -1,11 +1,11 @@ + SectionDevice."Speaker" { + Comment "Speakers" + Value { +- PlaybackChannels 8 +- PlaybackPriority 200 +- PlaybackPCM "hw:${CardId}" +- JackControl "Speaker Jack" +- PlaybackMixerElem "Speaker" ++ PlaybackChannels 8 ++ PlaybackPriority 200 ++ PlaybackPCM "hw:${CardId}" ++ JackControl "Speaker Jack" ++ PlaybackMixerElem "Speaker" + } + } + +-- +2.29.2 + + +From 5503703ab95d5ed40b1da138016777bb5429190e Mon Sep 17 00:00:00 2001 +From: Jaroslav Kysela +Date: Fri, 15 Jan 2021 14:55:53 +0100 +Subject: [PATCH 19/24] USB-Audio: fix indentation in + Gigabyte-Aorus-Master-Main-Audio-HiFi.conf + +Signed-off-by: Jaroslav Kysela +--- + .../Gigabyte-Aorus-Master-Main-Audio-HiFi.conf | 10 +++++----- + 1 file changed, 5 insertions(+), 5 deletions(-) + +diff --git a/ucm2/USB-Audio/Gigabyte-Aorus-Master-Main-Audio-HiFi.conf b/ucm2/USB-Audio/Gigabyte-Aorus-Master-Main-Audio-HiFi.conf +index 5c24165..37800fb 100644 +--- a/ucm2/USB-Audio/Gigabyte-Aorus-Master-Main-Audio-HiFi.conf ++++ b/ucm2/USB-Audio/Gigabyte-Aorus-Master-Main-Audio-HiFi.conf +@@ -3,11 +3,11 @@ Define.SecondaryCardId "$${CardIdByName:Aorus Master Front Headphone}" + SectionDevice."Speaker" { + Comment "Speakers" + Value { +- PlaybackChannels 8 +- PlaybackPriority 200 +- PlaybackPCM "hw:${CardId}" +- JackControl "Line Out Jack" +- PlaybackMixerElem "Line Out" ++ PlaybackChannels 8 ++ PlaybackPriority 200 ++ PlaybackPCM "hw:${CardId}" ++ JackControl "Line Out Jack" ++ PlaybackMixerElem "Line Out" + } + } + +-- +2.29.2 + + +From eaa5eacd3e068471537888dee627751dc35e714a Mon Sep 17 00:00:00 2001 +From: Hans de Goede +Date: Wed, 16 Dec 2020 16:38:37 +0100 +Subject: [PATCH 20/24] chtnau8824: Add a SST define/variable + +Add a SST define which gets set to "yes" when using the SST driver +and to "" when not using the SST driver; + +And decide if platforms/bytcr/PlatformEnableSeq.conf should be included +or not based on this. + +This is a preparation patch for adding SOF support. + +Signed-off-by: Hans de Goede +Signed-off-by: Jaroslav Kysela +--- + ucm2/chtnau8824/HiFi.conf | 19 +++++++++++++++---- + 1 file changed, 15 insertions(+), 4 deletions(-) + +diff --git a/ucm2/chtnau8824/HiFi.conf b/ucm2/chtnau8824/HiFi.conf +index 105f360..b36588f 100644 +--- a/ucm2/chtnau8824/HiFi.conf ++++ b/ucm2/chtnau8824/HiFi.conf +@@ -1,5 +1,16 @@ + Define.Speaker "Speaker" + Define.Mic "InternalMic" ++Define.SST "yes" ++ ++If.Controls { ++ Condition { ++ Type ControlExists ++ Control "name='media0_in Gain 0 Switch'" ++ } ++ False { ++ Define.SST "" ++ } ++} + + If.cfg-mspk { + Condition { +@@ -33,12 +44,12 @@ SectionVerb { + + Include.e.File "/codecs/nau8824/EnableSeq.conf" + +- If.Controls { ++ If.SST { + Condition { +- Type ControlExists +- Control "name='media0_in Gain 0 Switch'" ++ Type String ++ Empty "${var:SST}" + } +- True { ++ False { + Include.pe { + File "/platforms/bytcr/PlatformEnableSeq.conf" + Before.EnableSequence "0" +-- +2.29.2 + + +From 4e0369b4fcebde195a38d5b90c73b08f28e0047a Mon Sep 17 00:00:00 2001 +From: Takashi Iwai +Date: Thu, 21 Jan 2021 11:15:02 +0100 +Subject: [PATCH 21/24] kblrt5660: Fix file permissions + +The exec bit was set wrongly on those files. Fix them. + +Signed-off-by: Takashi Iwai +--- + ucm2/kblrt5660/Hdmi1.conf | 0 + ucm2/kblrt5660/Hdmi2.conf | 0 + ucm2/kblrt5660/HiFi.conf | 0 + ucm2/kblrt5660/kblrt5660.conf | 0 + 4 files changed, 0 insertions(+), 0 deletions(-) + mode change 100755 => 100644 ucm2/kblrt5660/Hdmi1.conf + mode change 100755 => 100644 ucm2/kblrt5660/Hdmi2.conf + mode change 100755 => 100644 ucm2/kblrt5660/HiFi.conf + mode change 100755 => 100644 ucm2/kblrt5660/kblrt5660.conf + +diff --git a/ucm2/kblrt5660/Hdmi1.conf b/ucm2/kblrt5660/Hdmi1.conf +old mode 100755 +new mode 100644 +diff --git a/ucm2/kblrt5660/Hdmi2.conf b/ucm2/kblrt5660/Hdmi2.conf +old mode 100755 +new mode 100644 +diff --git a/ucm2/kblrt5660/HiFi.conf b/ucm2/kblrt5660/HiFi.conf +old mode 100755 +new mode 100644 +diff --git a/ucm2/kblrt5660/kblrt5660.conf b/ucm2/kblrt5660/kblrt5660.conf +old mode 100755 +new mode 100644 +-- +2.29.2 + + +From 96e1c19a6bcc1e353f677966d4d93399204af0ec Mon Sep 17 00:00:00 2001 +From: Hans de Goede +Date: Wed, 16 Dec 2020 16:38:38 +0100 +Subject: [PATCH 22/24] chtnau8824: Add support for using the SOF driver + +The old (and currently the default) SST driver uses TDM 4 slots 24 bit +as wire format to the codec. Where as the new SOF driver uses standard +I2S 2 channel 24 bit. + +Normally this should not impact the UCM settings, but on the NAU8824 +the "DAC Right Channel Source" mixer setting must be set to 1 when +using TDM 4 slots and to 0 when using I2S 2 channel mode. + +Getting this wrong (in either case) results in the right channel not +outputting any sound. + +This commit introduces a RightOutputChannel variable which gets +set to 0/1 depending on the driver and then uses that for the +"DAC Right Channel Source" mixer setting so that we do the right +thing depending on the driver. + +This has been tested on the following devices: + +Medion E2215T: Stereo speakers, analog mic +Medion E2228T: Stereo speakers, stereo digital mics +Cube iWork 8 Air: Mono speaker, analog mic + +Signed-off-by: Hans de Goede +Signed-off-by: Jaroslav Kysela +--- + ucm2/chtnau8824/HiFi.conf | 2 ++ + ucm2/codecs/nau8824/EnableSeq.conf | 2 +- + 2 files changed, 3 insertions(+), 1 deletion(-) + +diff --git a/ucm2/chtnau8824/HiFi.conf b/ucm2/chtnau8824/HiFi.conf +index b36588f..c3d830d 100644 +--- a/ucm2/chtnau8824/HiFi.conf ++++ b/ucm2/chtnau8824/HiFi.conf +@@ -1,6 +1,7 @@ + Define.Speaker "Speaker" + Define.Mic "InternalMic" + Define.SST "yes" ++Define.RightOutputChannel 1 + + If.Controls { + Condition { +@@ -9,6 +10,7 @@ If.Controls { + } + False { + Define.SST "" ++ Define.RightOutputChannel 0 + } + } + +diff --git a/ucm2/codecs/nau8824/EnableSeq.conf b/ucm2/codecs/nau8824/EnableSeq.conf +index 3650d06..660856f 100644 +--- a/ucm2/codecs/nau8824/EnableSeq.conf ++++ b/ucm2/codecs/nau8824/EnableSeq.conf +@@ -1,7 +1,7 @@ + EnableSequence [ + # Playback TDM configuration + cset "name='DACL Channel Source' 0" +- cset "name='DACR Channel Source' 1" ++ cset "name='DACR Channel Source' ${var:RightOutputChannel}" + # Input Configuration + cset "name='DMIC1 Enable Switch' off" + cset "name='DMIC2 Enable Switch' off" +-- +2.29.2 + + +From 3bea51c3188257c5bc0eaaed922d1f714e3508c7 Mon Sep 17 00:00:00 2001 +From: Pierre-Louis Bossart +Date: Thu, 21 Jan 2021 14:38:26 -0600 +Subject: [PATCH 23/24] sof-hda-dsp: update handling of cfg-dmics strings + +Existing platforms can have 1,2,3 or 4 microphones. The SOF firmware +will generate 2 or 4 channels. Since by default CaptureChannels is 2, +we need to add the case for cfg-dmics = 3. + +Signed-off-by: Pierre-Louis Bossart +Signed-off-by: Jaroslav Kysela +--- + ucm2/sof-hda-dsp/HiFi.conf | 6 +++--- + 1 file changed, 3 insertions(+), 3 deletions(-) + +diff --git a/ucm2/sof-hda-dsp/HiFi.conf b/ucm2/sof-hda-dsp/HiFi.conf +index a250331..2c02c15 100644 +--- a/ucm2/sof-hda-dsp/HiFi.conf ++++ b/ucm2/sof-hda-dsp/HiFi.conf +@@ -16,9 +16,9 @@ SectionDevice."Mic1" { + CapturePCM "hw:${CardId},6" + If.chn { + Condition { +- Type String +- Haystack "${CardComponents}" +- Needle "cfg-dmics:4" ++ Type RegexMatch ++ Regex "cfg-dmics:[34]" ++ String "${CardComponents}" + } + True { + CaptureChannels 4 +-- +2.29.2 + + +From 889c5b9809f1635005d77871bb9abc09df6a3cc3 Mon Sep 17 00:00:00 2001 +From: Pierre-Louis Bossart +Date: Thu, 21 Jan 2021 14:44:05 -0600 +Subject: [PATCH 24/24] sof-soundwire: add support for dmics + +the pinmux allows for SoundWire to be used on e.g. link0 and link1, +and the rest of the pins can be assigned for dmic usages. + +We currently don't track this capability which means users don't see a +capture device, e.g. on HP Spectre x360. + +The sof_sdw machine driver was modified to report the cfg-dmics +string, possible values are 0..4. + +BugLink: https://github.com/thesofproject/linux/issues/2700 +BugLink: https://github.com/alsa-project/alsa-ucm-conf/pull/75 +Signed-off-by: Pierre-Louis Bossart +Signed-off-by: Jaroslav Kysela +--- + ucm2/sof-soundwire/dmic.conf | 21 +++++++++++++++++++++ + ucm2/sof-soundwire/sof-soundwire.conf | 5 +++++ + 2 files changed, 26 insertions(+) + create mode 100644 ucm2/sof-soundwire/dmic.conf + +diff --git a/ucm2/sof-soundwire/dmic.conf b/ucm2/sof-soundwire/dmic.conf +new file mode 100644 +index 0000000..1f704a5 +--- /dev/null ++++ b/ucm2/sof-soundwire/dmic.conf +@@ -0,0 +1,21 @@ ++SectionDevice."Mic" { ++ Comment "Digital Microphone" ++ ++ Value { ++ CapturePriority 100 ++ CapturePCM "hw:${CardId},3" ++ If.chn { ++ Condition { ++ Type RegexMatch ++ Regex "[34]" ++ String "${var:Mics1}" ++ } ++ True { ++ CaptureChannels 4 ++ } ++ } ++ CaptureMixerElem "Dmic0" ++ CaptureVolume "Dmic0 Capture Volume" ++ CaptureSwitch "Dmic0 Capture Switch" ++ } ++} +diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf +index 2ba44be..ec8b01e 100644 +--- a/ucm2/sof-soundwire/sof-soundwire.conf ++++ b/ucm2/sof-soundwire/sof-soundwire.conf +@@ -11,6 +11,7 @@ Define { + SpeakerAmps1 "0" + HeadsetCodec1 "" + MicCodec1 "" ++ Mics1 "0" + } + + DefineRegex { +@@ -34,6 +35,10 @@ DefineRegex { + Regex " mic:([a-z0-9]+)" + String "${CardComponents}" + } ++ Mics { ++ Regex " cfg-mics:([1-9][0-9]*)" ++ String "${CardComponents}" ++ } + } + + If.hs_init { +-- +2.29.2 + diff --git a/SPECS/alsa-lib.spec b/SPECS/alsa-lib.spec index 5d5c9f3..16a7798 100644 --- a/SPECS/alsa-lib.spec +++ b/SPECS/alsa-lib.spec @@ -9,7 +9,7 @@ Summary: The Advanced Linux Sound Architecture (ALSA) library Name: alsa-lib Version: %{version_alsa_lib} -Release: 4%{?prever_dot}%{?dist} +Release: 5%{?prever_dot}%{?dist} License: LGPLv2+ Group: System Environment/Libraries URL: http://www.alsa-project.org/ @@ -163,7 +163,7 @@ rm %{buildroot}/%{_includedir}/asoundlib.h %{_datadir}/alsa/topology %changelog -* Fri Jan 8 2021 Jaroslav Kysela - 1.2.4-4 +* Mon Feb 1 2021 Jaroslav Kysela - 1.2.4-5 - Apply fixes from upstream (alsa-lib, alsa-ucm-conf) * Mon Oct 19 2020 Jaroslav Kysela - 1.2.4-3