Merge branch 'support-old-webrtc' into 'master'

echo-cancel: Support old webrtc-audio-processing library

See merge request pulseaudio/pulseaudio!836

meson.build

@@ -726,9 +726,15 @@ if get_option('daemon')
     cdata.set('HAVE_SOXR', 1)
   endif
 
-  webrtc_dep = dependency('webrtc-audio-processing-1', version : '>= 1.0', required : get_option('webrtc-aec'))
+  webrtc_dep = dependency('webrtc-audio-processing-1', version : '>= 1.0', required : false)
   if webrtc_dep.found()
     cdata.set('HAVE_WEBRTC', 1)
+    cdata.set('HAVE_WEBRTC1', 1)
+  else
+    webrtc_dep = dependency('webrtc-audio-processing', version : ['>= 0.2', '< 1.0'], required : get_option('webrtc-aec'))
+    if webrtc_dep.found()
+      cdata.set('HAVE_WEBRTC', 1)
+    endif
   endif
 
   systemd_dep = dependency('systemd', required : get_option('systemd'))

po/ar.po

@@ -9,9 +9,8 @@ msgstr ""
 "Report-Msgid-Bugs-To: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/"
 "issues/new\n"
 "POT-Creation-Date: 2022-06-18 09:49+0300\n"
-"PO-Revision-Date: 2025-04-27 12:48+0000\n"
+"PO-Revision-Date: 2025-05-08 08:25+0000\n"
-"Last-Translator: Weblate Translation Memory <noreply-mt-weblate-translation-"
+"Last-Translator: Zayed Al-Saidi <zayed.alsaidi@gmail.com>\n"
-"memory@weblate.org>\n"
 "Language-Team: Arabic <https://translate.fedoraproject.org/projects/"
 "pulseaudio/pulseaudio/ar/>\n"
 "Language: ar\n"
@@ -20,7 +19,7 @@ msgstr ""
 "Content-Transfer-Encoding: 8bit\n"
 "Plural-Forms: nplurals=6; plural=n==0 ? 0 : n==1 ? 1 : n==2 ? 2 : n%100>=3 "
 "&& n%100<=10 ? 3 : n%100>=11 ? 4 : 5;\n"
-"X-Generator: Weblate 5.11\n"
+"X-Generator: Weblate 5.11.3\n"
 
 #: src/daemon/cmdline.c:113
 #, c-format
@@ -462,7 +461,7 @@ msgstr "نظام صوت بلس أوديو"
 
 #: src/daemon/pulseaudio.desktop.in:5
 msgid "Start the PulseAudio Sound System"
-msgstr ""
+msgstr "ابدأ نظام صوت بلس أوديو"
 
 #: src/modules/alsa/alsa-mixer.c:2708
 msgid "Input"
@@ -525,27 +524,27 @@ msgstr "لا يوجد التحكم في الرفع الصوت آلي"
 
 #: src/modules/alsa/alsa-mixer.c:2722
 msgid "Boost"
-msgstr ""
+msgstr "تعزيز"
 
 #: src/modules/alsa/alsa-mixer.c:2723
 msgid "No Boost"
-msgstr ""
+msgstr "دون تعزيز"
 
 #: src/modules/alsa/alsa-mixer.c:2724
 msgid "Amplifier"
-msgstr ""
+msgstr "مُضَخِّم"
 
 #: src/modules/alsa/alsa-mixer.c:2725
 msgid "No Amplifier"
-msgstr ""
+msgstr "دون مُضَخِّم"
 
 #: src/modules/alsa/alsa-mixer.c:2726
 msgid "Bass Boost"
-msgstr ""
+msgstr "تعزيز القرار"
 
 #: src/modules/alsa/alsa-mixer.c:2727
 msgid "No Bass Boost"
-msgstr ""
+msgstr "دون تعزيز القرار"
 
 #: src/modules/alsa/alsa-mixer.c:2728 src/modules/bluetooth/module-bluez5-device.c:1964
 #: src/utils/pactl.c:333
@@ -1046,7 +1045,7 @@ msgstr ""
 
 #: src/modules/reserve-wrap.c:149
 msgid "PulseAudio Sound Server"
-msgstr ""
+msgstr "خادم صوت بلس أوديو"
 
 #: src/pulse/channelmap.c:105
 msgid "Front Center"
@@ -1074,7 +1073,7 @@ msgstr "خلف يمين"
 
 #: src/pulse/channelmap.c:113
 msgid "Subwoofer"
-msgstr "مضخم صوت"
+msgstr "مضخم الترددات المنخفضة"
 
 #: src/pulse/channelmap.c:115
 msgid "Front Left-of-center"
@@ -1332,7 +1331,7 @@ msgstr "نعم"
 
 #: src/pulsecore/core-util.h:97
 msgid "no"
-msgstr ""
+msgstr "لا"
 
 #: src/pulsecore/lock-autospawn.c:141 src/pulsecore/lock-autospawn.c:227
 msgid "Cannot access autospawn lock."
@@ -1370,7 +1369,7 @@ msgstr "ممنوع الوصول"
 
 #: src/pulse/error.c:40
 msgid "Unknown command"
-msgstr ""
+msgstr "أمر مجهول"
 
 #: src/pulse/error.c:41
 msgid "Invalid argument"
@@ -1402,7 +1401,7 @@ msgstr ""
 
 #: src/pulse/error.c:48
 msgid "Internal error"
-msgstr "خطأ داخلي."
+msgstr "خطأ داخلي"
 
 #: src/pulse/error.c:49
 msgid "Connection terminated"
@@ -1438,7 +1437,7 @@ msgstr ""
 
 #: src/pulse/error.c:57
 msgid "Not supported"
-msgstr "غير معتمد"
+msgstr "غير مدعوم"
 
 #: src/pulse/error.c:58
 msgid "Unknown error code"
@@ -1466,7 +1465,7 @@ msgstr "خطأ في الإدخال/الإخراج"
 
 #: src/pulse/error.c:64
 msgid "Device or resource busy"
-msgstr ""
+msgstr "الجهاز أو المورد مشغول"
 
 #: src/pulse/sample.c:179
 #, c-format

po/el.po

@@ -9,8 +9,8 @@ msgstr ""
 "Report-Msgid-Bugs-To: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/"
 "issues/new\n"
 "POT-Creation-Date: 2022-06-18 09:49+0300\n"
-"PO-Revision-Date: 2024-09-04 16:38+0000\n"
+"PO-Revision-Date: 2025-06-22 10:49+0000\n"
-"Last-Translator: Giannis Antypas <gianni.antypas@gmail.com>\n"
+"Last-Translator: Jim Spentzos <jimspentzos2000@gmail.com>\n"
 "Language-Team: Greek <https://translate.fedoraproject.org/projects/"
 "pulseaudio/pulseaudio/el/>\n"
 "Language: el\n"
@@ -18,7 +18,7 @@ msgstr ""
 "Content-Type: text/plain; charset=UTF-8\n"
 "Content-Transfer-Encoding: 8bit\n"
 "Plural-Forms: nplurals=2; plural=n != 1;\n"
-"X-Generator: Weblate 5.7.1\n"
+"X-Generator: Weblate 5.12.2\n"
 
 #: src/daemon/cmdline.c:113
 #, c-format
@@ -359,7 +359,7 @@ msgstr "Περιγραφή: %s\n"
 #: src/daemon/dumpmodules.c:67
 #, c-format
 msgid "Author: %s\n"
-msgstr "Συγγραφέας: %s\n"
+msgstr "Δημιουργός: %s\n"
 
 #: src/daemon/dumpmodules.c:69
 #, c-format
@@ -2298,9 +2298,7 @@ msgstr "ΠΛΑΙΣΙΑ"
 
 #: src/utils/pacmd.c:80 src/utils/pactl.c:2659
 msgid "RECIPIENT MESSAGE [MESSAGE_PARAMETERS]"
-msgstr ""
+msgstr "ΜΗΝΥΜΑ ΠΑΡΑΛΗΠΤΗ [MESSAGE_PARAMETERS]"
-"ΜΗΝΥΜΑ ΠΑΡΑΛΗΠΤΗ\n"
-"[ΠΑΡΑΜΕΤΡΟΙ_ΜΗΝΥΜΑΤΟΣ]"
 
 #: src/utils/pacmd.c:82
 #, c-format

po/es.po

@@ -14,8 +14,8 @@ msgstr ""
 "Report-Msgid-Bugs-To: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/"
 "issues/new\n"
 "POT-Creation-Date: 2022-06-18 09:49+0300\n"
-"PO-Revision-Date: 2022-12-18 11:19+0000\n"
+"PO-Revision-Date: 2025-07-29 09:34+0000\n"
-"Last-Translator: Toni Estevez <toni.estevez@gmail.com>\n"
+"Last-Translator: \"Fco. Javier F. Serrador\" <fserrador@gmail.com>\n"
 "Language-Team: Spanish <https://translate.fedoraproject.org/projects/"
 "pulseaudio/pulseaudio/es/>\n"
 "Language: es\n"
@@ -23,7 +23,7 @@ msgstr ""
 "Content-Type: text/plain; charset=UTF-8\n"
 "Content-Transfer-Encoding: 8bit\n"
 "Plural-Forms: nplurals=2; plural=n != 1;\n"
-"X-Generator: Weblate 4.15\n"
+"X-Generator: Weblate 5.12.2\n"
 "X-Poedit-Language: Spanish\n"
 
 #: src/daemon/cmdline.c:113
@@ -97,7 +97,7 @@ msgid ""
 "\n"
 "  -n                                    Don't load default script file\n"
 msgstr ""
-"%s [opciones]\n"
+"--dl-search-path%s [opciones]\n"
 "\n"
 "ÓRDENES:\n"
 "  -h, --help                            Mostrar esta ayuda\n"
@@ -123,8 +123,8 @@ msgstr ""
 "inicio\n"
 "      --fail[=BOOL]                     Salir cuando falla el inicio\n"
 "      --high-priority[=BOOL]            Intentar asignar una prioridad alta\n"
-"                                        (solo disponible como superusuario, "
+"                                        (solo disponible como admin, con "
-"con SUID\n"
+"SUID\n"
 "                                        o con un valor RLIMIT_NICE elevado)\n"
 "      --realtime[=BOOL]                 Intentar activar la programación en "
 "tiempo real\n"
@@ -136,19 +136,19 @@ msgstr ""
 "                                        del usuario tras el inicio\n"
 "      --disallow-exit[=BOOL]            Denegar la salida a petición del "
 "usuario\n"
-"      --exit-idle-time=SEGUNDOS         Finalizar el servicio cuando esté "
+"      --exit-idle-time=SEGS         Finalizar el servicio cuando esté "
 "inactivo y\n"
 "                                        haya pasado este tiempo\n"
-"      --scache-idle-time=SEGUNDOS       Descargar las muestras cargadas "
+"      --scache-idle-time=SEGS       Descargar las muestras cargadas "
 "automáticamente\n"
-"                                        cuando esté inactivo y  haya pasado "
+"                                        cuando esté inactivo y haya pasado "
 "este tiempo\n"
-"      --log-level[=NIVEL]               Aumentar o configurar el nivel de "
+"      --log-level[=NIVEL]           Aumentar o configurar el nivel de "
 "registro\n"
 "  -v  --verbose                         Aumentar el nivel de registro\n"
 "      --log-target={auto,syslog,stderr,file:RUTA,newfile:RUTA}\n"
 "                                        Especificar el destino del registro\n"
-"      --log-meta[=BOOL]                 Incluir ubicaciones del código en el "
+"      --log-meta[=BOOL]                 Incluir lugares del código en el "
 "registro\n"
 "      --log-time[=BOOL]                 Incluir marcas de tiempo en el "
 "registro\n"
@@ -156,16 +156,19 @@ msgstr ""
 "registro\n"
 "  -p, --dl-search-path=RUTA             Configurar la ruta de búsqueda de "
 "objetos\n"
-"                                        dinámicos compartidos (complementos)"
+"                                                                         "
-"\n"
+"dinámicos compartidos (complementos)\n"
 "      --resample-method=MÉTODO          Usar el método de remuestreo "
 "especificado\n"
-"                                        (Consulte los valores posibles con\n"
+"                                                                                 "
-"                                        --dump-resample-methods)\n"
+"(Consulte los valores posibles con\n"
+"                                                                                     "
+"--dump-resample-methods)\n"
 "      --use-pid-file[=BOOL]             Crear un archivo PID\n"
 "      --no-cpu-limit[=BOOL]             No instalar el limitador de carga de "
 "la CPU\n"
-"                                        en las plataformas compatibles.\n"
+"                                                                     en las "
+"plataformas compatibles.\n"
 "      --disable-shm[=BOOL]              Desactivar el uso de memoria "
 "compartida\n"
 "      --enable-memfd[=BOOL]             Activar el uso de memoria compartida "
@@ -174,10 +177,12 @@ msgstr ""
 "GUION DE INICIO:\n"
 "  -L, --load=\"ARGUMENTO DEL MÓDULO\"     Cargar el módulo del complemento "
 "especificado\n"
-"                                        con el argumento especificado\n"
+"                                                                               "
-"  -F, --file=ARCHIVO                    Ejecutar el guion especificado\n"
+"con el argumento especificado\n"
-"  -C                                    Abrir una línea de órdenes en el TTY "
+"  -F, --file=ARCHIVO                               Ejecutar el guion "
-"en\n"
+"especificado\n"
+"  -C                                                 Abrir una línea de "
+"órdenes en el TTY en\n"
 "                                        ejecución tras el inicio\n"
 "  -n                                    No cargar el archivo de órdenes "
 "predeterminado\n"
@@ -195,9 +200,9 @@ msgid ""
 "--log-level expects log level argument (either numeric in range 0..4 or one "
 "of error, warn, notice, info, debug)."
 msgstr ""
-"--log-level espera un argumento para el nivel de registro (un valor numérico "
+"--log-level espera un argumento para el nivel de bitácora (un valor numérico "
-"en el rango de 0-4 o bien uno de estos valores: error, warn, notice, info, "
+"en el intervalo de 0..4 o bien uno de estos valores: error, warn, notice, "
-"debug)."
+"info, debug)."
 
 #: src/daemon/cmdline.c:277
 msgid "--high-priority expects boolean argument"
@@ -217,7 +222,7 @@ msgstr "--disallow-exit espera un argumento booleano"
 
 #: src/daemon/cmdline.c:309
 msgid "--use-pid-file expects boolean argument"
-msgstr "--use pid-file espera un argumento booleano"
+msgstr "--use-pid-file espera un argumento booleano"
 
 #: src/daemon/cmdline.c:328
 msgid ""
@@ -750,11 +755,11 @@ msgstr "Salida de juego"
 
 #: src/modules/alsa/alsa-mixer.c:2819 src/modules/alsa/alsa-mixer.c:2820
 msgid "Chat Output"
-msgstr "Salida de chat"
+msgstr "Salida de charla"
 
 #: src/modules/alsa/alsa-mixer.c:2821
 msgid "Chat Input"
-msgstr "Entrada de chat"
+msgstr "Entrada de charla"
 
 #: src/modules/alsa/alsa-mixer.c:2822
 msgid "Virtual Surround 7.1"
@@ -1546,10 +1551,10 @@ msgid ""
 "e.g. happen if you try to connect to a non-root PulseAudio as a root user, "
 "over the native protocol. Don't do that.)"
 msgstr ""
-"XDG_RUNTIME_DIR (%s) no es de nuestra propiedad (usuario %d), sino del "
+"XDG_RUNTIME_DIR (%s) no es de nuestra propiedad (uid %d), sino del usuario "
-"usuario %d. (Esto puede pasar, por ejemplo, al intentar conectarse como "
+"%d. (Esto puede pasar, por ejemplo, al intentar conectarse como superusuario "
-"superusuario a un servidor PulseAudio que se ejecuta sin privilegios de "
+"a un servidor PulseAudio que se ejecuta sin privilegios de administrador "
-"administrador mediante el protocolo nativo. No lo haga.)"
+"mediante el protocolo nativo. No lo haga.)"
 
 #: src/pulsecore/core-util.h:97
 msgid "yes"
@@ -1742,7 +1747,7 @@ msgstr "pa_stream_drain(): %s"
 #: src/utils/pacat.c:194 src/utils/pacat.c:543
 #, c-format
 msgid "pa_stream_begin_write() failed: %s"
-msgstr "Ha fallado pa_stream_write(): %s"
+msgstr "Ha fallado pa_stream_begin_write(): %s"
 
 #: src/utils/pacat.c:244 src/utils/pacat.c:274
 #, c-format
@@ -2285,7 +2290,7 @@ msgstr "NOMBRE|NÚMERO PUERTO"
 
 #: src/utils/pacmd.c:74 src/utils/pactl.c:2658
 msgid "CARD-NAME|CARD-#N PORT OFFSET"
-msgstr "NOMBRE-TARJETA|NÚMERO-TARJETA PUERTO COMPENSACIÓN"
+msgstr "NOMBRE-TARJETA|Nº-TARJETA PUERTO COMPENSACIÓN"
 
 #: src/utils/pacmd.c:75
 msgid "TARGET"

po/it.po

@@ -12,7 +12,7 @@ msgstr ""
 "Report-Msgid-Bugs-To: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/"
 "issues/new\n"
 "POT-Creation-Date: 2022-06-18 09:49+0300\n"
-"PO-Revision-Date: 2025-04-18 18:51+0000\n"
+"PO-Revision-Date: 2025-06-08 14:42+0000\n"
 "Last-Translator: Salvatore Cocuzza <info@salvatorecocuzza.it>\n"
 "Language-Team: Italian <https://translate.fedoraproject.org/projects/"
 "pulseaudio/pulseaudio/it/>\n"
@@ -21,7 +21,7 @@ msgstr ""
 "Content-Type: text/plain; charset=UTF-8\n"
 "Content-Transfer-Encoding: 8bit\n"
 "Plural-Forms: nplurals=2; plural=n != 1;\n"
-"X-Generator: Weblate 5.11\n"
+"X-Generator: Weblate 5.11.4\n"
 
 # mamma mia che impressione
 #: src/daemon/cmdline.c:113
@@ -1245,7 +1245,6 @@ msgid "Virtual surround sink"
 msgstr "Sink surround virtuale"
 
 #: src/modules/module-virtual-surround-sink.c:54
-#, fuzzy
 msgid ""
 "sink_name=<name for the sink> sink_properties=<properties for the sink> "
 "master=<name of sink to filter> sink_master=<name of sink to filter> "
@@ -2418,7 +2417,7 @@ msgstr "Recupero delle informazioni del server non riuscito: %s"
 #: src/utils/pactl.c:224 src/utils/pactl.c:236
 #, c-format
 msgid "%s\n"
-msgstr ""
+msgstr "%s\n"
 
 #: src/utils/pactl.c:281
 #, c-format
@@ -2489,7 +2488,6 @@ msgid "Mic"
 msgstr "Mic"
 
 #: src/utils/pactl.c:338
-#, fuzzy
 msgid "Handset"
 msgstr "Cuffie con microfono"
 
@@ -2584,7 +2582,6 @@ msgid "\t\t%s: %s (type: %s, priority: %u%s%s, %s)\n"
 msgstr "\t\t%s: %s (tipo: %s, priorità: %u%s%s, %s)\n"
 
 #: src/utils/pactl.c:710 src/utils/pactl.c:894 src/utils/pactl.c:1256
-#, fuzzy
 msgid ", availability group: "
 msgstr ", gruppo disponibilità: "
 

po/ka.po

@@ -9,7 +9,7 @@ msgstr ""
 "Report-Msgid-Bugs-To: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/"
 "issues/new\n"
 "POT-Creation-Date: 2022-06-18 09:49+0300\n"
-"PO-Revision-Date: 2025-04-13 20:52+0000\n"
+"PO-Revision-Date: 2025-07-21 10:49+0000\n"
 "Last-Translator: Temuri Doghonadze <temuri.doghonadze@gmail.com>\n"
 "Language-Team: Georgian <https://translate.fedoraproject.org/projects/"
 "pulseaudio/pulseaudio/ka/>\n"
@@ -18,7 +18,7 @@ msgstr ""
 "Content-Type: text/plain; charset=UTF-8\n"
 "Content-Transfer-Encoding: 8bit\n"
 "Plural-Forms: nplurals=2; plural=n != 1;\n"
-"X-Generator: Weblate 5.10.4\n"
+"X-Generator: Weblate 5.12.2\n"
 
 #: src/daemon/cmdline.c:113
 #, c-format
@@ -328,7 +328,7 @@ msgstr ""
 #: src/daemon/main.c:287 src/daemon/main.c:292
 #, c-format
 msgid "Failed to create '%s': %s"
-msgstr "შეცდომა %s-ის გახსნისას: %s"
+msgstr "'%s'-ის შექმნა ჩავარდა: %s"
 
 #: src/daemon/main.c:299
 #, c-format
@@ -554,11 +554,11 @@ msgstr "გამაძლიერებლის გარეშე"
 
 #: src/modules/alsa/alsa-mixer.c:2726
 msgid "Bass Boost"
-msgstr "Bass-ის გაძლიერება"
+msgstr "ბასის გაძლიერება"
 
 #: src/modules/alsa/alsa-mixer.c:2727
 msgid "No Bass Boost"
-msgstr "Bass-ის გაძლიერების გარეშე"
+msgstr "ბასის გაძლიერების გარეშე"
 
 #: src/modules/alsa/alsa-mixer.c:2728
 #: src/modules/bluetooth/module-bluez5-device.c:1964 src/utils/pactl.c:333
@@ -1338,7 +1338,7 @@ msgstr "xcb_connect() ჩავარდა"
 
 #: src/pulse/client-conf-x11.c:66 src/utils/pax11publish.c:102
 msgid "xcb_connection_has_error() returned true"
-msgstr "xcb_connection_has_error()-მა 1 დააბრუნა"
+msgstr "xcb_connection_has_error()-მა დააბრუნა ჭეშმარიტი მნისვნელობა"
 
 #: src/pulse/client-conf-x11.c:102
 msgid "Failed to parse cookie data"
@@ -2112,8 +2112,8 @@ msgstr "სტატისტიკის მიღების შეცდო
 #, c-format
 msgid "Currently in use: %u block containing %s bytes total.\n"
 msgid_plural "Currently in use: %u blocks containing %s bytes total.\n"
-msgstr[0] ""
+msgstr[0] "ამჟამად გამოიყენება: %u ბლოკი, რომელიც შეიცავს სულ %s ბაიტს.\n"
-msgstr[1] ""
+msgstr[1] "ამჟამად გამოიყენება: %u ბლოკი, რომელიც შეიცავს სულ %s ბაიტს.\n"
 
 #: src/utils/pactl.c:205
 #, c-format
@@ -2121,7 +2121,11 @@ msgid "Allocated during whole lifetime: %u block containing %s bytes total.\n"
 msgid_plural ""
 "Allocated during whole lifetime: %u blocks containing %s bytes total.\n"
 msgstr[0] ""
+"გამოყოფილია მთელი სიცოცხლის განმავლობაში: %u ბლოკი, რომელიც სულ %s ბაიტს "
+"შეიცავს.\n"
 msgstr[1] ""
+"გამოყოფილია მთელი სიცოცხლის განმავლობაში: %u ბლოკი, რომელიც სულ %s ბაიტს "
+"შეიცავს.\n"
 
 #: src/utils/pactl.c:211
 #, c-format
@@ -2560,7 +2564,11 @@ msgid_plural ""
 "Failed to set volume: You tried to set volumes for %d channels, whereas "
 "channel(s) supported = %d\n"
 msgstr[0] ""
+"ხმის დაყენაბა ჩავარდა: სცადეთ, დაგეყენებინათ ხმა %d არხზე მაშინ, როცა "
+"მხარდაჭერილი არხები = %d\n"
 msgstr[1] ""
+"ხმის დაყენაბა ჩავარდა: სცადეთ, დაგეყენებინათ ხმა %d არხზე მაშინ, როცა "
+"მხარდაჭერილი არხები = %d\n"
 
 #: src/utils/pactl.c:2107
 #, c-format

po/ko.po

@@ -7,7 +7,7 @@ msgstr ""
 "Report-Msgid-Bugs-To: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/"
 "issues/new\n"
 "POT-Creation-Date: 2022-06-18 09:49+0300\n"
-"PO-Revision-Date: 2024-08-04 19:41+0000\n"
+"PO-Revision-Date: 2025-06-14 13:17+0000\n"
 "Last-Translator: 김인수 <simmon@nplob.com>\n"
 "Language-Team: Korean <https://translate.fedoraproject.org/projects/"
 "pulseaudio/pulseaudio/ko/>\n"
@@ -16,7 +16,7 @@ msgstr ""
 "Content-Type: text/plain; charset=UTF-8\n"
 "Content-Transfer-Encoding: 8bit\n"
 "Plural-Forms: nplurals=1; plural=0;\n"
-"X-Generator: Weblate 5.6.2\n"
+"X-Generator: Weblate 5.11.4\n"
 
 #: src/daemon/cmdline.c:113
 #, c-format
@@ -230,7 +230,7 @@ msgstr "--disable-shm 에는 부울 인자 값이 필요합니다"
 
 #: src/daemon/cmdline.c:397
 msgid "--enable-memfd expects boolean argument"
-msgstr "--enable-memfd는 부울 인수가 필요합니다"
+msgstr "--enable-memfd 는 부울 인수가 예상됩니다"
 
 #: src/daemon/daemon-conf.c:270
 #, c-format
@@ -423,9 +423,8 @@ msgstr "명령어 행 분석 실패."
 msgid ""
 "System mode refused for non-root user. Only starting the D-Bus server lookup "
 "service."
-msgstr ""
+msgstr "non-root 사용자에게 시스템 방식이 거부되었습니다. D-Bus 서버 검색 서비스만 "
-"비 루트 사용자의 시스템 모드 전환을 거부했습니다. D-Bus 서버 검색 서비스만 시"
+"시작합니다."
-"작합니다."
 
 #: src/daemon/main.c:788
 #, c-format
@@ -436,9 +435,8 @@ msgstr "데몬 종료 실패: %s"
 msgid ""
 "This program is not intended to be run as root (unless --system is "
 "specified)."
-msgstr ""
+msgstr "이와 같은 프로그램은 root로 동작하도록 의도되지 않았습니다 (--system 이 "
-"이 프로그램은 루트 계정으로 실행하도록 만들지 않았습니다. (실행하려면 --"
+"지정되지 않은 경우)."
-"system을 명기하십시오)."
 
 #: src/daemon/main.c:820
 msgid "Root privileges required."
@@ -451,24 +449,22 @@ msgstr "--start는 시스템 인스턴스에 대해 지원되지 않습니다."
 #: src/daemon/main.c:867
 #, c-format
 msgid "User-configured server at %s, refusing to start/autospawn."
-msgstr "%s에서 사용자 설정한 서버, start/autospawn을 거부하고 있습니다."
+msgstr "%s에 User-configured 서버, start/autospawn을 거부하고 있습니다."
 
 #: src/daemon/main.c:873
 #, c-format
 msgid ""
 "User-configured server at %s, which appears to be local. Probing deeper."
-msgstr "%s에 사용자가 설정한 서버, 이는 로컬에 있습니다. 상세히 조사합니다."
+msgstr "%s에 User-configured 서버, 이는 로컬에 있습니다. 상세히 조사합니다."
 
 #: src/daemon/main.c:878
 msgid "Running in system mode, but --disallow-exit not set."
-msgstr ""
+msgstr "시스템 방식에서 실행 중이지만, --disallow-exit 를 설정하지 않았습니다."
-"시스템 모드에서 실행 중입니다. 하지만 --disallow-exit을 설정하지 않았습니다."
 
 #: src/daemon/main.c:881
 msgid "Running in system mode, but --disallow-module-loading not set."
-msgstr ""
+msgstr "시스템 방식에서 실행 중이지만, --disallow-module-loading 를 설정하지 "
-"시스템 모드에서 실행 중입니다. 하지만 --disallow-module-loading을 설정하지 않"
+"않았습니다."
-"았습니다."
 
 #: src/daemon/main.c:884
 msgid "Running in system mode, forcibly disabling SHM mode."
@@ -1014,11 +1010,11 @@ msgstr "전화기"
 
 #: src/modules/bluetooth/module-bluez5-device.c:2042
 msgid "High Fidelity Playback (A2DP Sink)"
-msgstr "Hi-Fi 재생 (A2DP Sink)"
+msgstr "고음질 재생 (A2DP Sink)"
 
 #: src/modules/bluetooth/module-bluez5-device.c:2054
 msgid "High Fidelity Capture (A2DP Source)"
-msgstr "Hi-Fi 캡쳐 (A2DP Source)"
+msgstr "고음질 캡쳐 (A2DP Source)"
 
 #: src/modules/bluetooth/module-bluez5-device.c:2066
 msgid "Headset Head Unit (HSP)"
@@ -1479,9 +1475,9 @@ msgid ""
 "e.g. happen if you try to connect to a non-root PulseAudio as a root user, "
 "over the native protocol. Don't do that.)"
 msgstr ""
-"XDG_RUNTIME_DIR (%s)은 우리(uid %d)가 아니라 uid %d가 소유합니다! (자체 프로"
+"XDG_RUNTIME_DIR (%s)는 우리(uid %d)가 아니라 uid %d가 소유합니다! (자체 "
-"토콜로 비 루트 펄스오디오 사용자가 루트 사용자 권한으로 연결할 때 이 문제가 "
+"통신규약을 통해 root 사용자로 root가 아닌 PluseAudio에 연결을 시도 할 때에 "
-"일어납니다. 그렇게 하지 마십시오.)"
+"예시로 발생 할 수 있습니다. 그렇게 하지 않습니다.)"
 
 #: src/pulsecore/core-util.h:97
 msgid "yes"

po/pt_BR.po

@@ -11,7 +11,7 @@ msgstr ""
 "Report-Msgid-Bugs-To: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/"
 "issues/new\n"
 "POT-Creation-Date: 2022-06-18 09:49+0300\n"
-"PO-Revision-Date: 2024-10-11 16:01+0000\n"
+"PO-Revision-Date: 2025-05-10 22:55+0000\n"
 "Last-Translator: Rafael Fontenelle <rafaelff@gnome.org>\n"
 "Language-Team: Portuguese (Brazil) <https://translate.fedoraproject.org/"
 "projects/pulseaudio/pulseaudio/pt_BR/>\n"
@@ -20,7 +20,7 @@ msgstr ""
 "Content-Type: text/plain; charset=UTF-8\n"
 "Content-Transfer-Encoding: 8bit\n"
 "Plural-Forms: nplurals=2; plural=n > 1;\n"
-"X-Generator: Weblate 5.7.2\n"
+"X-Generator: Weblate 5.11.3\n"
 
 #: src/daemon/cmdline.c:113
 #, c-format
@@ -1908,7 +1908,7 @@ msgid "pa_stream_update_timing_info() failed: %s"
 msgstr "pa_stream_update_timing_info() falhou: %s"
 
 #: src/utils/pacat.c:676
-#, fuzzy, c-format
+#, c-format
 msgid ""
 "%s [options]\n"
 "%s\n"
@@ -1987,62 +1987,58 @@ msgstr ""
 "  -v, --verbose                         Habilita operações no modo "
 "detalhado\n"
 "\n"
-"  -s, --server=SERVIDOR                 O nome do servidor a conectar-se\n"
+"  -s, --server=SERVIDOR                 O nome do servidor para se conectar\n"
-"  -d, --device=DISPOSITIVO              O nome do destino/fonte a conectar-"
+"  -d, --device=DISPOSITIVO              O nome do destino/fonte para se "
-"se\n"
+"conectar\n"
 "  -n, --client-name=NOME                Como chamar este cliente no "
 "servidor\n"
 "      --stream-name=NOME                Como chamar este fluxo no servidor\n"
-"      --volume=VOLUME                   Especifica a faixa (linear) inicial\n"
+"      --volume=VOLUME                   Especifica o volume (linear) inicial "
-"                                        de volume no intervalo 0...65536\n"
+"no intervalo 0...65536\n"
-"      --rate=TAXA_DE_AMOSTRAGEM         Taxa de amostragem, Hz (padrão "
+"      --rate=TAXA_DE_AMOSTRAGEM         Taxa de amostragem em Hz (padrão: "
 "44100)\n"
-"      --format=FORMATO_DE_AMOSTRAGEM    Tipo de amostragem, veja\n"
+"      --format=FORMATO_DE_AMOSTRAGEM    Formato da amostragem, veja\n"
 "                                        https://www.freedesktop.org/wiki/"
 "Software/PulseAudio/Documentation/User/SupportedAudioFormats/\n"
 "                                        para valores possíveis (padrão: "
 "s16ne)\n"
-"      --channels=CANAIS                 O número de canais, 1 para mono,\n"
+"      --channels=CANAIS                 O número de canais, 1 para mono, 2 "
-"                                        2 para estéreo (padrão: 2)\n"
+"para estéreo\n"
-"      --channel-map=MAPA_DE_CANAIS      Mapeamento de canais a ser usado no\n"
+"                                        (padrão: 2)\n"
-"                                        lugar do padrão\n"
+"      --channel-map=MAPA_DE_CANAIS      Mapeamento de canais para usar em "
-"      --fix-format                      Obtém o formato da amostragem do\n"
+"vez do padrão\n"
-"                                        destino/fonte onde o fluxo está\n"
+"      --fix-format                      Obtém o formato da amostragem do "
-"                                        sendo conectado.\n"
+"destino/fonte onde\n"
-"      --fix-rate                        Obtém a taxa de amostragem do\n"
+"                                        o fluxo está sendo conectado.\n"
-"                                        destino/fonte onde o fluxo está\n"
+"      --fix-rate                        Obtém a taxa de amostragem do "
-"                                        sendo conectado.\n"
+"destino/fonte onde\n"
-"      --fix-channels                    Obtém o número de canais e o mapa "
+"                                        o fluxo está sendo conectado.\n"
-"de\n"
+"      --fix-channels                    Obtém o número de canais e o mapa de "
-"                                        canais do destino onde o fluxo está\n"
+"canais do destino\n"
-"                                        sendo conectado.\n"
+"                                        onde o fluxo está sendo conectado.\n"
-"      --no-remix                        Não faz upmix nem downmix dos "
+"      --no-remix                        Não faz upmix nem downmix dos canais."
-"canais.\n"
+"\n"
-"      --no-remap                        Mapeia os canais por índice em vez\n"
+"      --no-remap                        Mapeia os canais por índice em vez "
-"                                        de nome\n"
+"de nome.\n"
-"      --latency=BYTES                   Requisita a latência especificada "
+"      --latency=BYTES                   Requisita a latência especificada em "
-"em\n"
-"                                        bytes.\n"
-"      --process-time=BYTES              Requisita o tempo de processo\n"
-"                                        especificado por requisições em "
 "bytes.\n"
-"      --latency-msec=MSEGUNDOS          Requisita a latência especificada "
+"      --process-time=BYTES              Requisita o tempo de processo por "
-"em\n"
+"requisições em bytes.\n"
-"                                        milissegundos.\n"
+"      --latency-msec=MSEGUNDOS          Requisita a latência especificada em "
-"      --process-time-msec=MSEGUNDOS     Requisita a o tempo do processo por\n"
+"milissegundos.\n"
-"                                        requisição em milissegundos.\n"
+"      --process-time-msec=MSEGUNDOS     Requisita o tempo de processo por "
+"requisições em milissegundos.\n"
 "      --property=PROPRIEDADE=VALOR      Define a propriedade especificada "
-"para\n"
+"para o valor especificado.\n"
-"                                        o valor especificado.\n"
 "      --raw                             Grava/reproduz dados PCM não "
 "tratados.\n"
 "      --passthrough                     Dados para conversão.\n"
-"      --file-format[=FORMATO_ARQUIVO]   Grava/reproduz dados PCM "
+"      --file-format[=FORMATO_ARQUIVO]   Grava/reproduz dados PCM formatados."
-"formatados.\n"
+"\n"
 "      --list-file-formats               Lista formatos de arquivo "
 "disponíveis.\n"
 "      --monitor-stream=ÍNDICE           Grava da entrada do destino com "
-"índice.\n"
+"índice ÍNDICE.\n"
 
 #: src/utils/pacat.c:793
 msgid "Play back encoded audio files on a PulseAudio sound server."
@@ -2875,11 +2871,12 @@ msgstr "mensagem list-handlers falhou: %s"
 
 #: src/utils/pactl.c:1711 src/utils/pactl.c:1760
 msgid "list-handlers message response could not be parsed correctly"
-msgstr "a resposta da mensagem list-handlers não pôde ser tratada corretamente"
+msgstr ""
+"a resposta da mensagem de list-handlers não pôde ser tratada corretamente"
 
 #: src/utils/pactl.c:1718
 msgid "list-handlers message response is not a JSON array"
-msgstr "a resposta da mensagem list-handlers não é um array JSON"
+msgstr "a resposta da mensagem de list-handlers não é um array JSON"
 
 #: src/utils/pactl.c:1729
 #, c-format
@@ -3051,7 +3048,7 @@ msgstr ""
 "padrão.\n"
 
 #: src/utils/pactl.c:2664
-#, fuzzy, c-format
+#, c-format
 msgid ""
 "\n"
 "  -h, --help                            Show this help\n"
@@ -3068,7 +3065,9 @@ msgstr ""
 "  -h, --help                            Mostra esta ajuda\n"
 "      --version                         Mostra a versão\n"
 "\n"
-"  -s, --server=SERVIDOR                 Nome do servidor a ser conectado\n"
+"  -f, --format=FORMATO                  O formato da saída. \"normal\" ou "
+"\"json\"\n"
+"  -s, --server=SERVIDOR                 Nome do servidor para se conectar\n"
 "  -n, --client-name=NOME                Como chamar este cliente no "
 "servidor\n"
 
@@ -3084,9 +3083,9 @@ msgstr ""
 "Vinculado com libpulse %s\n"
 
 #: src/utils/pactl.c:2751
-#, fuzzy, c-format
+#, c-format
 msgid "Invalid format value '%s'"
-msgstr "Nome do fluxo “%s” inválido"
+msgstr "Valor de formato “%s” inválido"
 
 #: src/utils/pactl.c:2778
 #, c-format

po/ro.po

@@ -9,7 +9,7 @@ msgstr ""
 "Report-Msgid-Bugs-To: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/"
 "issues/new\n"
 "POT-Creation-Date: 2022-06-18 09:49+0300\n"
-"PO-Revision-Date: 2025-02-23 20:53+0000\n"
+"PO-Revision-Date: 2025-05-11 11:49+0000\n"
 "Last-Translator: Remus-Gabriel Chelu <remusgabriel.chelu@disroot.org>\n"
 "Language-Team: Romanian <https://translate.fedoraproject.org/projects/"
 "pulseaudio/pulseaudio/ro/>\n"
@@ -19,7 +19,7 @@ msgstr ""
 "Content-Transfer-Encoding: 8bit\n"
 "Plural-Forms: nplurals=3; plural=n==1 ? 0 : (n==0 || (n%100 > 0 && n%100 < "
 "20)) ? 1 : 2;\n"
-"X-Generator: Weblate 5.10\n"
+"X-Generator: Weblate 5.11.3\n"
 
 #: src/daemon/cmdline.c:113
 #, c-format
@@ -502,7 +502,7 @@ msgstr ""
 
 #: src/daemon/main.c:922
 msgid "Failed to acquire stdio."
-msgstr "Nu s-a reușit să se achizționeze stdio."
+msgstr "Nu s-a reușit să se achiziționeze stdio."
 
 #: src/daemon/main.c:928 src/daemon/main.c:999
 #, c-format
@@ -2135,7 +2135,7 @@ msgstr "Nu s-a putut determina specificația eșantionului din fișier."
 
 #: src/utils/pacat.c:1100
 msgid "Warning: Failed to determine channel map from file."
-msgstr "Avertisment: Nu s-a reușit determinarea schemei canalelor din fișier."
+msgstr "Avertisment: Nu s-a reușit să se determine schema canalelor din fișier."
 
 #: src/utils/pacat.c:1111
 msgid "Channel map doesn't match sample specification"

po/ru.po

@@ -10,8 +10,8 @@ msgstr ""
 "Report-Msgid-Bugs-To: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/"
 "issues/new\n"
 "POT-Creation-Date: 2022-06-18 09:49+0300\n"
-"PO-Revision-Date: 2023-09-12 13:35+0000\n"
+"PO-Revision-Date: 2025-05-11 00:38+0000\n"
-"Last-Translator: \"Sergey A.\" <sw@atrus.ru>\n"
+"Last-Translator: \"Sergey A.\" <Ser82-png@yandex.ru>\n"
 "Language-Team: Russian <https://translate.fedoraproject.org/projects/"
 "pulseaudio/pulseaudio/ru/>\n"
 "Language: ru\n"
@@ -20,7 +20,7 @@ msgstr ""
 "Content-Transfer-Encoding: 8bit\n"
 "Plural-Forms: nplurals=3; plural=n%10==1 && n%100!=11 ? 0 : n%10>=2 && "
 "n%10<=4 && (n%100<10 || n%100>=20) ? 1 : 2;\n"
-"X-Generator: Weblate 5.0.1\n"
+"X-Generator: Weblate 5.11.3\n"
 
 #: src/daemon/cmdline.c:113
 #, c-format
@@ -397,7 +397,7 @@ msgstr ""
 
 #: src/daemon/ltdl-bind-now.c:144
 msgid "Failed to add bind-now-loader."
-msgstr "Не удалось добавить новый загрузчик bind-now."
+msgstr "Не удалось добавить bind-now-loader."
 
 #: src/daemon/main.c:265
 #, c-format

po/sl.po

@@ -9,7 +9,7 @@ msgstr ""
 "Report-Msgid-Bugs-To: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/"
 "issues/new\n"
 "POT-Creation-Date: 2022-06-18 09:49+0300\n"
-"PO-Revision-Date: 2024-08-17 18:38+0000\n"
+"PO-Revision-Date: 2025-07-25 04:53+0000\n"
 "Last-Translator: Martin Srebotnjak <miles@filmsi.net>\n"
 "Language-Team: Slovenian <https://translate.fedoraproject.org/projects/"
 "pulseaudio/pulseaudio/sl/>\n"
@@ -19,7 +19,7 @@ msgstr ""
 "Content-Transfer-Encoding: 8bit\n"
 "Plural-Forms: nplurals=4; plural=n%100==1 ? 0 : n%100==2 ? 1 : n%100==3 || "
 "n%100==4 ? 2 : 3;\n"
-"X-Generator: Weblate 5.6.2\n"
+"X-Generator: Weblate 5.12.2\n"
 
 #: src/daemon/cmdline.c:113
 #, c-format
@@ -447,7 +447,7 @@ msgstr "Ni uspelo ubiti zalednega procesa: %s"
 
 #: src/daemon/main.c:817
 msgid "This program is not intended to be run as root (unless --system is specified)."
-msgstr "Ta program ni namenjen zagonu kot root (razen če je določeno --sistem)."
+msgstr "Ta program ni namenjen zagonu kot root (razen če je določeno --system)."
 
 #: src/daemon/main.c:820
 msgid "Root privileges required."
@@ -1526,7 +1526,7 @@ msgid ""
 "happen if you try to connect to a non-root PulseAudio as a root user, over the native "
 "protocol. Don't do that.)"
 msgstr ""
-"XDG_RUNTIME_DIR (%s) ni v naši lasti (uiid %d), ampak v lasti uid %d (to se "
+"XDG_RUNTIME_DIR (%s) ni v naši lasti (uid %d), ampak v lasti uid %d (to se "
 "lahko npr. zgodi, če se poskušate povezati z nekorenskim PulseAudio kot "
 "korenski uporabnik prek izvornega protokola; ne počnite tega)!"
 
@@ -1966,7 +1966,7 @@ msgstr ""
 "uporabiti namesto privzete\n"
 "      --fix-format                      Vzemite obliko vzorca iz ponora/"
 "vira, s katerim je tok povezan.\n"
-"      —fix-rate                        Hitrost vzorčenja vzemite iz ponora/"
+"      --fix-rate                        Hitrost vzorčenja vzemite iz ponora/"
 "vira, s katerim je tok povezan.\n"
 "      --fix-channels                    Vzame število kanalov in preslikavo "
 "kanalov\n"

src/modules/echo-cancel/meson.build

@@ -9,9 +9,15 @@
 
 add_languages('cpp')
 
-libwebrtc_util_sources = [
+if cdata.has('HAVE_WEBRTC1')
-  'webrtc.cc'
+  libwebrtc_util_sources = [
-]
+    'webrtc.cc'
+  ]
+else
+  libwebrtc_util_sources = [
+    'webrtc-old.cc'
+  ]
+endif
 
 if host_machine.system() == 'darwin'
   ignore_unresolved_symbols_link_args = ['-Wl,-undefined,dynamic_lookup']

src/modules/echo-cancel/webrtc-old.cc

@@ -0,0 +1,594 @@
+/***
+    This file is part of PulseAudio.
+
+    Copyright 2011 Collabora Ltd.
+              2015 Aldebaran SoftBank Group
+
+    Contributor: Arun Raghavan <mail@arunraghavan.net>
+
+    PulseAudio is free software; you can redistribute it and/or modify
+    it under the terms of the GNU Lesser General Public License as published
+    by the Free Software Foundation; either version 2.1 of the License,
+    or (at your option) any later version.
+
+    PulseAudio is distributed in the hope that it will be useful, but
+    WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+    General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public License
+    along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <pulse/cdecl.h>
+
+PA_C_DECL_BEGIN
+#include <pulsecore/core-util.h>
+#include <pulsecore/modargs.h>
+
+#include <pulse/timeval.h>
+#include "echo-cancel.h"
+PA_C_DECL_END
+
+#include <webrtc/modules/audio_processing/include/audio_processing.h>
+#include <webrtc/modules/interface/module_common_types.h>
+#include <webrtc/system_wrappers/include/trace.h>
+
+#define BLOCK_SIZE_US 10000
+
+#define DEFAULT_HIGH_PASS_FILTER true
+#define DEFAULT_NOISE_SUPPRESSION true
+#define DEFAULT_ANALOG_GAIN_CONTROL true
+#define DEFAULT_DIGITAL_GAIN_CONTROL false
+#define DEFAULT_MOBILE false
+#define DEFAULT_ROUTING_MODE "speakerphone"
+#define DEFAULT_COMFORT_NOISE true
+#define DEFAULT_DRIFT_COMPENSATION false
+#define DEFAULT_VAD true
+#define DEFAULT_EXTENDED_FILTER false
+#define DEFAULT_INTELLIGIBILITY_ENHANCER false
+#define DEFAULT_EXPERIMENTAL_AGC false
+#define DEFAULT_AGC_START_VOLUME 85
+#define DEFAULT_BEAMFORMING false
+#define DEFAULT_TRACE false
+
+#define WEBRTC_AGC_MAX_VOLUME 255
+
+static const char* const valid_modargs[] = {
+    "high_pass_filter",
+    "noise_suppression",
+    "analog_gain_control",
+    "digital_gain_control",
+    "mobile",
+    "routing_mode",
+    "comfort_noise",
+    "drift_compensation",
+    "voice_detection",
+    "extended_filter",
+    "intelligibility_enhancer",
+    "experimental_agc",
+    "agc_start_volume",
+    "beamforming",
+    "mic_geometry", /* documented in parse_mic_geometry() */
+    "target_direction", /* documented in parse_mic_geometry() */
+    "trace",
+    NULL
+};
+
+static int routing_mode_from_string(const char *rmode) {
+    if (pa_streq(rmode, "quiet-earpiece-or-headset"))
+        return webrtc::EchoControlMobile::kQuietEarpieceOrHeadset;
+    else if (pa_streq(rmode, "earpiece"))
+        return webrtc::EchoControlMobile::kEarpiece;
+    else if (pa_streq(rmode, "loud-earpiece"))
+        return webrtc::EchoControlMobile::kLoudEarpiece;
+    else if (pa_streq(rmode, "speakerphone"))
+        return webrtc::EchoControlMobile::kSpeakerphone;
+    else if (pa_streq(rmode, "loud-speakerphone"))
+        return webrtc::EchoControlMobile::kLoudSpeakerphone;
+    else
+        return -1;
+}
+
+class PaWebrtcTraceCallback : public webrtc::TraceCallback {
+    void Print(webrtc::TraceLevel level, const char *message, int length)
+    {
+        if (level & webrtc::kTraceError || level & webrtc::kTraceCritical)
+            pa_log("%s", message);
+        else if (level & webrtc::kTraceWarning)
+            pa_log_warn("%s", message);
+        else if (level & webrtc::kTraceInfo)
+            pa_log_info("%s", message);
+        else
+            pa_log_debug("%s", message);
+    }
+};
+
+static int webrtc_volume_from_pa(pa_volume_t v)
+{
+    return (v * WEBRTC_AGC_MAX_VOLUME) / PA_VOLUME_NORM;
+}
+
+static pa_volume_t webrtc_volume_to_pa(int v)
+{
+    return (v * PA_VOLUME_NORM) / WEBRTC_AGC_MAX_VOLUME;
+}
+
+static void webrtc_ec_fixate_spec(pa_sample_spec *rec_ss, pa_channel_map *rec_map,
+                                  pa_sample_spec *play_ss, pa_channel_map *play_map,
+                                  pa_sample_spec *out_ss, pa_channel_map *out_map,
+                                  bool beamforming)
+{
+    rec_ss->format = PA_SAMPLE_FLOAT32NE;
+    play_ss->format = PA_SAMPLE_FLOAT32NE;
+
+    /* AudioProcessing expects one of the following rates */
+    if (rec_ss->rate >= 48000)
+        rec_ss->rate = 48000;
+    else if (rec_ss->rate >= 32000)
+        rec_ss->rate = 32000;
+    else if (rec_ss->rate >= 16000)
+        rec_ss->rate = 16000;
+    else
+        rec_ss->rate = 8000;
+
+    *out_ss = *rec_ss;
+    *out_map = *rec_map;
+
+    if (beamforming) {
+        /* The beamformer gives us a single channel */
+        out_ss->channels = 1;
+        pa_channel_map_init_mono(out_map);
+    }
+
+    /* Playback stream rate needs to be the same as capture */
+    play_ss->rate = rec_ss->rate;
+}
+
+static bool parse_point(const char **point, float (&f)[3]) {
+    int ret, length;
+
+    ret = sscanf(*point, "%g,%g,%g%n", &f[0], &f[1], &f[2], &length);
+    if (ret != 3)
+        return false;
+
+    /* Consume the bytes we've read so far */
+    *point += length;
+
+    return true;
+}
+
+static bool parse_mic_geometry(const char **mic_geometry, std::vector<webrtc::Point>& geometry) {
+    /* The microphone geometry is expressed as cartesian point form:
+     *   x1,y1,z1,x2,y2,z2,...
+     *
+     * Where x1,y1,z1 is the position of the first microphone with regards to
+     * the array's "center", x2,y2,z2 the position of the second, and so on.
+     *
+     * 'x' is the horizontal coordinate, with positive values being to the
+     * right from the mic array's perspective.
+     *
+     * 'y' is the depth coordinate, with positive values being in front of the
+     * array.
+     *
+     * 'z' is the vertical coordinate, with positive values being above the
+     * array.
+     *
+     * All distances are in meters.
+     */
+
+    /* The target direction is expected to be in spherical point form:
+     *   a,e,r
+     *
+     * Where 'a' is the azimuth of the target point relative to the center of
+     * the array, 'e' its elevation, and 'r' the radius.
+     *
+     * 0 radians azimuth is to the right of the array, and positive angles
+     * move in a counter-clockwise direction.
+     *
+     * 0 radians elevation is horizontal w.r.t. the array, and positive
+     * angles go upwards.
+     *
+     * radius is distance from the array center in meters.
+     */
+
+    long unsigned int i;
+    float f[3];
+
+    for (i = 0; i < geometry.size(); i++) {
+        if (!parse_point(mic_geometry, f)) {
+            pa_log("Failed to parse channel %lu in mic_geometry", i);
+            return false;
+        }
+
+        /* Except for the last point, we should have a trailing comma */
+        if (i != geometry.size() - 1) {
+            if (**mic_geometry != ',') {
+                pa_log("Failed to parse channel %lu in mic_geometry", i);
+                return false;
+            }
+
+            (*mic_geometry)++;
+        }
+
+        pa_log_debug("Got mic #%lu position: (%g, %g, %g)", i, f[0], f[1], f[2]);
+
+        geometry[i].c[0] = f[0];
+        geometry[i].c[1] = f[1];
+        geometry[i].c[2] = f[2];
+    }
+
+    if (**mic_geometry != '\0') {
+        pa_log("Failed to parse mic_geometry value: more parameters than expected");
+        return false;
+    }
+
+    return true;
+}
+
+bool pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
+                       pa_sample_spec *rec_ss, pa_channel_map *rec_map,
+                       pa_sample_spec *play_ss, pa_channel_map *play_map,
+                       pa_sample_spec *out_ss, pa_channel_map *out_map,
+                       uint32_t *nframes, const char *args) {
+    webrtc::AudioProcessing *apm = NULL;
+    webrtc::ProcessingConfig pconfig;
+    webrtc::Config config;
+    bool hpf, ns, agc, dgc, mobile, cn, vad, ext_filter, intelligibility, experimental_agc, beamforming;
+    int rm = -1, i;
+    uint32_t agc_start_volume;
+    pa_modargs *ma;
+    bool trace = false;
+
+    if (!(ma = pa_modargs_new(args, valid_modargs))) {
+        pa_log("Failed to parse submodule arguments.");
+        goto fail;
+    }
+
+    hpf = DEFAULT_HIGH_PASS_FILTER;
+    if (pa_modargs_get_value_boolean(ma, "high_pass_filter", &hpf) < 0) {
+        pa_log("Failed to parse high_pass_filter value");
+        goto fail;
+    }
+
+    ns = DEFAULT_NOISE_SUPPRESSION;
+    if (pa_modargs_get_value_boolean(ma, "noise_suppression", &ns) < 0) {
+        pa_log("Failed to parse noise_suppression value");
+        goto fail;
+    }
+
+    agc = DEFAULT_ANALOG_GAIN_CONTROL;
+    if (pa_modargs_get_value_boolean(ma, "analog_gain_control", &agc) < 0) {
+        pa_log("Failed to parse analog_gain_control value");
+        goto fail;
+    }
+
+    dgc = agc ? false : DEFAULT_DIGITAL_GAIN_CONTROL;
+    if (pa_modargs_get_value_boolean(ma, "digital_gain_control", &dgc) < 0) {
+        pa_log("Failed to parse digital_gain_control value");
+        goto fail;
+    }
+
+    if (agc && dgc) {
+        pa_log("You must pick only one between analog and digital gain control");
+        goto fail;
+    }
+
+    mobile = DEFAULT_MOBILE;
+    if (pa_modargs_get_value_boolean(ma, "mobile", &mobile) < 0) {
+        pa_log("Failed to parse mobile value");
+        goto fail;
+    }
+
+    ec->params.drift_compensation = DEFAULT_DRIFT_COMPENSATION;
+    if (pa_modargs_get_value_boolean(ma, "drift_compensation", &ec->params.drift_compensation) < 0) {
+        pa_log("Failed to parse drift_compensation value");
+        goto fail;
+    }
+
+    if (mobile) {
+        if (ec->params.drift_compensation) {
+            pa_log("Can't use drift_compensation in mobile mode");
+            goto fail;
+        }
+
+        if ((rm = routing_mode_from_string(pa_modargs_get_value(ma, "routing_mode", DEFAULT_ROUTING_MODE))) < 0) {
+            pa_log("Failed to parse routing_mode value");
+            goto fail;
+        }
+
+        cn = DEFAULT_COMFORT_NOISE;
+        if (pa_modargs_get_value_boolean(ma, "comfort_noise", &cn) < 0) {
+            pa_log("Failed to parse cn value");
+            goto fail;
+        }
+    } else {
+        if (pa_modargs_get_value(ma, "comfort_noise", NULL) || pa_modargs_get_value(ma, "routing_mode", NULL)) {
+            pa_log("The routing_mode and comfort_noise options are only valid with mobile=true");
+            goto fail;
+        }
+    }
+
+    vad = DEFAULT_VAD;
+    if (pa_modargs_get_value_boolean(ma, "voice_detection", &vad) < 0) {
+        pa_log("Failed to parse voice_detection value");
+        goto fail;
+    }
+
+    ext_filter = DEFAULT_EXTENDED_FILTER;
+    if (pa_modargs_get_value_boolean(ma, "extended_filter", &ext_filter) < 0) {
+        pa_log("Failed to parse extended_filter value");
+        goto fail;
+    }
+
+    intelligibility = DEFAULT_INTELLIGIBILITY_ENHANCER;
+    if (pa_modargs_get_value_boolean(ma, "intelligibility_enhancer", &intelligibility) < 0) {
+        pa_log("Failed to parse intelligibility_enhancer value");
+        goto fail;
+    }
+
+    experimental_agc = DEFAULT_EXPERIMENTAL_AGC;
+    if (pa_modargs_get_value_boolean(ma, "experimental_agc", &experimental_agc) < 0) {
+        pa_log("Failed to parse experimental_agc value");
+        goto fail;
+    }
+
+    agc_start_volume = DEFAULT_AGC_START_VOLUME;
+    if (pa_modargs_get_value_u32(ma, "agc_start_volume", &agc_start_volume) < 0) {
+        pa_log("Failed to parse agc_start_volume value");
+        goto fail;
+    }
+    if (agc_start_volume > WEBRTC_AGC_MAX_VOLUME) {
+        pa_log("AGC start volume must not exceed %u", WEBRTC_AGC_MAX_VOLUME);
+        goto fail;
+    }
+    ec->params.webrtc.agc_start_volume = agc_start_volume;
+
+    beamforming = DEFAULT_BEAMFORMING;
+    if (pa_modargs_get_value_boolean(ma, "beamforming", &beamforming) < 0) {
+        pa_log("Failed to parse beamforming value");
+        goto fail;
+    }
+
+    if (ext_filter)
+        config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
+    if (intelligibility)
+        pa_log_warn("The intelligibility enhancer is not currently supported");
+    if (experimental_agc)
+        config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(true, ec->params.webrtc.agc_start_volume));
+
+    trace = DEFAULT_TRACE;
+    if (pa_modargs_get_value_boolean(ma, "trace", &trace) < 0) {
+        pa_log("Failed to parse trace value");
+        goto fail;
+    }
+
+    if (trace) {
+        webrtc::Trace::CreateTrace();
+        webrtc::Trace::set_level_filter(webrtc::kTraceAll);
+        ec->params.webrtc.trace_callback = new PaWebrtcTraceCallback();
+        webrtc::Trace::SetTraceCallback((PaWebrtcTraceCallback *) ec->params.webrtc.trace_callback);
+    }
+
+    webrtc_ec_fixate_spec(rec_ss, rec_map, play_ss, play_map, out_ss, out_map, beamforming);
+
+    /* We do this after fixate because we need the capture channel count */
+    if (beamforming) {
+        std::vector<webrtc::Point> geometry(rec_ss->channels);
+        webrtc::SphericalPointf direction(0.0f, 0.0f, 0.0f);
+        const char *mic_geometry, *target_direction;
+
+        if (!(mic_geometry = pa_modargs_get_value(ma, "mic_geometry", NULL))) {
+            pa_log("mic_geometry must be set if beamforming is enabled");
+            goto fail;
+        }
+
+        if (!parse_mic_geometry(&mic_geometry, geometry)) {
+            pa_log("Failed to parse mic_geometry value");
+            goto fail;
+        }
+
+        if ((target_direction = pa_modargs_get_value(ma, "target_direction", NULL))) {
+            float f[3];
+
+            if (!parse_point(&target_direction, f)) {
+                pa_log("Failed to parse target_direction value");
+                goto fail;
+            }
+
+            if (*target_direction != '\0') {
+                pa_log("Failed to parse target_direction value: more parameters than expected");
+                goto fail;
+            }
+
+#define IS_ZERO(f) ((f) < 0.000001 && (f) > -0.000001)
+
+            if (!IS_ZERO(f[1]) || !IS_ZERO(f[2])) {
+                pa_log("The beamformer currently only supports targeting along the azimuth");
+                goto fail;
+            }
+
+            direction.s[0] = f[0];
+            direction.s[1] = f[1];
+            direction.s[2] = f[2];
+        }
+
+        if (!target_direction)
+            config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry));
+        else
+            config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry, direction));
+    }
+
+    apm = webrtc::AudioProcessing::Create(config);
+
+    pconfig = {
+        webrtc::StreamConfig(rec_ss->rate, rec_ss->channels, false), /* input stream */
+        webrtc::StreamConfig(out_ss->rate, out_ss->channels, false), /* output stream */
+        webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* reverse input stream */
+        webrtc::StreamConfig(play_ss->rate, play_ss->channels, false), /* reverse output stream */
+    };
+    if (apm->Initialize(pconfig) != webrtc::AudioProcessing::kNoError) {
+        pa_log("Error initialising audio processing module");
+        goto fail;
+    }
+
+    if (hpf)
+        apm->high_pass_filter()->Enable(true);
+
+    if (!mobile) {
+        apm->echo_cancellation()->enable_drift_compensation(ec->params.drift_compensation);
+        apm->echo_cancellation()->Enable(true);
+    } else {
+        apm->echo_control_mobile()->set_routing_mode(static_cast<webrtc::EchoControlMobile::RoutingMode>(rm));
+        apm->echo_control_mobile()->enable_comfort_noise(cn);
+        apm->echo_control_mobile()->Enable(true);
+    }
+
+    if (ns) {
+        apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
+        apm->noise_suppression()->Enable(true);
+    }
+
+    if (agc || dgc) {
+        if (mobile && rm <= webrtc::EchoControlMobile::kEarpiece) {
+            /* Maybe this should be a knob, but we've got a lot of knobs already */
+            apm->gain_control()->set_mode(webrtc::GainControl::kFixedDigital);
+            ec->params.webrtc.agc = false;
+        } else if (dgc) {
+            apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
+            ec->params.webrtc.agc = false;
+        } else {
+            apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
+            if (apm->gain_control()->set_analog_level_limits(0, WEBRTC_AGC_MAX_VOLUME) !=
+                    webrtc::AudioProcessing::kNoError) {
+                pa_log("Failed to initialise AGC");
+                goto fail;
+            }
+            ec->params.webrtc.agc = true;
+        }
+
+        apm->gain_control()->Enable(true);
+    }
+
+    if (vad)
+        apm->voice_detection()->Enable(true);
+
+    ec->params.webrtc.apm = apm;
+    ec->params.webrtc.rec_ss = *rec_ss;
+    ec->params.webrtc.play_ss = *play_ss;
+    ec->params.webrtc.out_ss = *out_ss;
+    ec->params.webrtc.blocksize = (uint64_t) out_ss->rate * BLOCK_SIZE_US / PA_USEC_PER_SEC;
+    *nframes = ec->params.webrtc.blocksize;
+    ec->params.webrtc.first = true;
+
+    for (i = 0; i < rec_ss->channels; i++)
+        ec->params.webrtc.rec_buffer[i] = pa_xnew(float, *nframes);
+    for (i = 0; i < play_ss->channels; i++)
+        ec->params.webrtc.play_buffer[i] = pa_xnew(float, *nframes);
+
+    pa_modargs_free(ma);
+    return true;
+
+fail:
+    if (ma)
+        pa_modargs_free(ma);
+    if (ec->params.webrtc.trace_callback) {
+        webrtc::Trace::ReturnTrace();
+        delete ((PaWebrtcTraceCallback *) ec->params.webrtc.trace_callback);
+    } if (apm)
+        delete apm;
+
+    return false;
+}
+
+void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
+    webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
+    const pa_sample_spec *ss = &ec->params.webrtc.play_ss;
+    int n = ec->params.webrtc.blocksize;
+    float **buf = ec->params.webrtc.play_buffer;
+    webrtc::StreamConfig config(ss->rate, ss->channels, false);
+
+    pa_deinterleave(play, (void **) buf, ss->channels, pa_sample_size(ss), n);
+
+    pa_assert_se(apm->ProcessReverseStream(buf, config, config, buf) == webrtc::AudioProcessing::kNoError);
+
+    /* FIXME: If ProcessReverseStream() makes any changes to the audio, such as
+     * applying intelligibility enhancement, those changes don't have any
+     * effect. This function is called at the source side, but the processing
+     * would have to be done in the sink to be able to feed the processed audio
+     * to speakers. */
+}
+
+void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out) {
+    webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
+    const pa_sample_spec *rec_ss = &ec->params.webrtc.rec_ss;
+    const pa_sample_spec *out_ss = &ec->params.webrtc.out_ss;
+    float **buf = ec->params.webrtc.rec_buffer;
+    int n = ec->params.webrtc.blocksize;
+    int old_volume, new_volume;
+    webrtc::StreamConfig rec_config(rec_ss->rate, rec_ss->channels, false);
+    webrtc::StreamConfig out_config(out_ss->rate, out_ss->channels, false);
+
+    pa_deinterleave(rec, (void **) buf, rec_ss->channels, pa_sample_size(rec_ss), n);
+
+    if (ec->params.webrtc.agc) {
+        pa_volume_t v = pa_echo_canceller_get_capture_volume(ec);
+        old_volume = webrtc_volume_from_pa(v);
+        apm->gain_control()->set_stream_analog_level(old_volume);
+    }
+
+    apm->set_stream_delay_ms(0);
+    pa_assert_se(apm->ProcessStream(buf, rec_config, out_config, buf) == webrtc::AudioProcessing::kNoError);
+
+    if (ec->params.webrtc.agc) {
+        if (PA_UNLIKELY(ec->params.webrtc.first)) {
+            /* We start at a sane default volume (taken from the Chromium
+             * condition on the experimental AGC in audio_processing.h). This is
+             * needed to make sure that there's enough energy in the capture
+             * signal for the AGC to work */
+            ec->params.webrtc.first = false;
+            new_volume = ec->params.webrtc.agc_start_volume;
+        } else {
+            new_volume = apm->gain_control()->stream_analog_level();
+        }
+
+        if (old_volume != new_volume)
+            pa_echo_canceller_set_capture_volume(ec, webrtc_volume_to_pa(new_volume));
+    }
+
+    pa_interleave((const void **) buf, out_ss->channels, out, pa_sample_size(out_ss), n);
+}
+
+void pa_webrtc_ec_set_drift(pa_echo_canceller *ec, float drift) {
+    webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.webrtc.apm;
+
+    apm->echo_cancellation()->set_stream_drift_samples(drift * ec->params.webrtc.blocksize);
+}
+
+void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
+    pa_webrtc_ec_play(ec, play);
+    pa_webrtc_ec_record(ec, rec, out);
+}
+
+void pa_webrtc_ec_done(pa_echo_canceller *ec) {
+    int i;
+
+    if (ec->params.webrtc.trace_callback) {
+        webrtc::Trace::ReturnTrace();
+        delete ((PaWebrtcTraceCallback *) ec->params.webrtc.trace_callback);
+    }
+
+    if (ec->params.webrtc.apm) {
+        delete (webrtc::AudioProcessing*)ec->params.webrtc.apm;
+        ec->params.webrtc.apm = NULL;
+    }
+
+    for (i = 0; i < ec->params.webrtc.rec_ss.channels; i++)
+        pa_xfree(ec->params.webrtc.rec_buffer[i]);
+    for (i = 0; i < ec->params.webrtc.play_ss.channels; i++)
+        pa_xfree(ec->params.webrtc.play_buffer[i]);
+}

src/utils/pactl.c

@@ -1841,8 +1841,18 @@ static void get_sink_mute_callback(pa_context *c, const pa_sink_info *i, int is_
 
     pa_assert(i);
 
-    printf(("Mute: %s\n"),
+    if (format == JSON) {
-           pa_yes_no_localised(i->mute));
+        pa_json_encoder *encoder = pa_json_encoder_new();
+        pa_json_encoder_begin_element_object(encoder);
+        pa_json_encoder_add_member_bool(encoder, "mute", i->mute);
+        pa_json_encoder_end_object(encoder);
+        char* json_str = pa_json_encoder_to_string_free(encoder);
+        printf("%s\n", json_str);
+        pa_xfree(json_str);
+    } else {
+        printf(("Mute: %s\n"),
+               pa_yes_no_localised(i->mute));
+    }
 
     complete_action();
 }
@@ -1860,10 +1870,21 @@ static void get_sink_volume_callback(pa_context *c, const pa_sink_info *i, int i
     pa_assert(i);
 
     char cv[PA_CVOLUME_SNPRINT_VERBOSE_MAX];
-    printf(("Volume: %s\n"
+    if (format == JSON) {
-            "        balance %0.2f\n"),
+        pa_json_encoder *encoder = pa_json_encoder_new();
-           pa_cvolume_snprint_verbose(cv, sizeof(cv), &i->volume, &i->channel_map, true),
+        pa_json_encoder_begin_element_object(encoder);
-           pa_cvolume_get_balance(&i->volume, &i->channel_map));
+        pa_json_encoder_add_member_raw_json(encoder, "volume", pa_cvolume_to_json_object(&i->volume, &i->channel_map, i->flags & PA_SINK_DECIBEL_VOLUME));
+        pa_json_encoder_add_member_double(encoder, "balance", pa_cvolume_get_balance(&i->volume, &i->channel_map), 2);
+        pa_json_encoder_end_object(encoder);
+        char* json_str = pa_json_encoder_to_string_free(encoder);
+        printf("%s\n", json_str);
+        pa_xfree(json_str);
+    } else {
+        printf(("Volume: %s\n"
+                "        balance %0.2f\n"),
+               pa_cvolume_snprint_verbose(cv, sizeof(cv), &i->volume, &i->channel_map, true),
+               pa_cvolume_get_balance(&i->volume, &i->channel_map));
+    }
 
     complete_action();
 }
@@ -1907,8 +1928,18 @@ static void get_source_mute_callback(pa_context *c, const pa_source_info *i, int
 
     pa_assert(i);
 
-    printf(("Mute: %s\n"),
+    if (format == JSON) {
-           pa_yes_no_localised(i->mute));
+        pa_json_encoder *encoder = pa_json_encoder_new();
+        pa_json_encoder_begin_element_object(encoder);
+        pa_json_encoder_add_member_bool(encoder, "mute", i->mute);
+        pa_json_encoder_end_object(encoder);
+        char* json_str = pa_json_encoder_to_string_free(encoder);
+        printf("%s\n", json_str);
+        pa_xfree(json_str);
+    } else {
+        printf(("Mute: %s\n"),
+               pa_yes_no_localised(i->mute));
+    }
 
     complete_action();
 }
@@ -1926,10 +1957,21 @@ static void get_source_volume_callback(pa_context *c, const pa_source_info *i, i
     pa_assert(i);
 
     char cv[PA_CVOLUME_SNPRINT_VERBOSE_MAX];
-    printf(("Volume: %s\n"
+    if (format == JSON) {
-            "        balance %0.2f\n"),
+        pa_json_encoder *encoder = pa_json_encoder_new();
-           pa_cvolume_snprint_verbose(cv, sizeof(cv), &i->volume, &i->channel_map, true),
+        pa_json_encoder_begin_element_object(encoder);
-           pa_cvolume_get_balance(&i->volume, &i->channel_map));
+        pa_json_encoder_add_member_raw_json(encoder, "volume", pa_cvolume_to_json_object(&i->volume, &i->channel_map, i->flags & PA_SINK_DECIBEL_VOLUME));
+        pa_json_encoder_add_member_double(encoder, "balance", pa_cvolume_get_balance(&i->volume, &i->channel_map), 2);
+        pa_json_encoder_end_object(encoder);
+        char* json_str = pa_json_encoder_to_string_free(encoder);
+        printf("%s\n", json_str);
+        pa_xfree(json_str);
+    } else {
+        printf(("Volume: %s\n"
+                "        balance %0.2f\n"),
+               pa_cvolume_snprint_verbose(cv, sizeof(cv), &i->volume, &i->channel_map, true),
+               pa_cvolume_get_balance(&i->volume, &i->channel_map));
+    }
 
     complete_action();
 }