diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -338,33 +338,37 @@ rtc_library("audio_device_impl") {
       defines += [ "WEBRTC_DUMMY_FILE_DEVICES" ]
     } else {
       if (is_linux || is_chromeos) {
-        sources += [
-          "linux/alsasymboltable_linux.cc",
-          "linux/alsasymboltable_linux.h",
-          "linux/audio_device_alsa_linux.cc",
-          "linux/audio_device_alsa_linux.h",
-          "linux/audio_mixer_manager_alsa_linux.cc",
-          "linux/audio_mixer_manager_alsa_linux.h",
-          "linux/latebindingsymboltable_linux.cc",
-          "linux/latebindingsymboltable_linux.h",
-        ]
-        defines += [ "WEBRTC_ENABLE_LINUX_ALSA" ]
         libs = [ "dl" ]
         if (rtc_use_x11) {
           libs += [ "X11" ]
           defines += [ "WEBRTC_USE_X11" ]
         }
+        if (rtc_include_alsa_audio) {
+          configs += [ "//third_party/alsa:headers" ]
+          sources += [
+            "linux/alsasymboltable_linux.cc",
+            "linux/alsasymboltable_linux.h",
+            "linux/audio_device_alsa_linux.cc",
+            "linux/audio_device_alsa_linux.h",
+            "linux/audio_mixer_manager_alsa_linux.cc",
+            "linux/audio_mixer_manager_alsa_linux.h",
+            "linux/latebindingsymboltable_linux.cc",
+            "linux/latebindingsymboltable_linux.h",
+          ]
+          defines += [ "WEBRTC_ENABLE_LINUX_ALSA" ]
+        }
         if (rtc_include_pulse_audio) {
+          configs += [ "//third_party/pulseaudio:headers" ]
           defines += [ "WEBRTC_ENABLE_LINUX_PULSE" ]
+          sources += [
+            "linux/audio_device_pulse_linux.cc",
+            "linux/audio_device_pulse_linux.h",
+            "linux/audio_mixer_manager_pulse_linux.cc",
+            "linux/audio_mixer_manager_pulse_linux.h",
+            "linux/pulseaudiosymboltable_linux.cc",
+            "linux/pulseaudiosymboltable_linux.h",
+          ]
         }
-        sources += [
-          "linux/audio_device_pulse_linux.cc",
-          "linux/audio_device_pulse_linux.h",
-          "linux/audio_mixer_manager_pulse_linux.cc",
-          "linux/audio_mixer_manager_pulse_linux.h",
-          "linux/pulseaudiosymboltable_linux.cc",
-          "linux/pulseaudiosymboltable_linux.h",
-        ]
       }
       if (is_mac) {
         sources += [
diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc
--- a/modules/audio_device/audio_device_impl.cc
+++ b/modules/audio_device/audio_device_impl.cc
@@ -191,40 +191,41 @@ AudioDeviceModuleImpl::CreatePlatformSpecificObjects(const Environment& env) {
 #endif
 
 // Linux ADM implementation.
-// Note that, WEBRTC_ENABLE_LINUX_ALSA is always defined by default when
-// WEBRTC_LINUX is defined. WEBRTC_ENABLE_LINUX_PULSE depends on the
-// 'rtc_include_pulse_audio' build flag.
-// TODO(bugs.webrtc.org/9127): improve support and make it more clear that
-// PulseAudio is the default selection.
+// WEBRTC_ENABLE_LINUX_ALSA depends on the 'rtc_include_alsa_audio' build flag.
+// WEBRTC_ENABLE_LINUX_PULSE depends on the 'rtc_include_pulse_audio' build
+// flag. PulseAudio remains the default selection when both backends are
+// enabled.
 #if !defined(WEBRTC_ANDROID) && defined(WEBRTC_LINUX)
-#if !defined(WEBRTC_ENABLE_LINUX_PULSE)
-  // Build flag 'rtc_include_pulse_audio' is set to false. In this mode:
-  // - kPlatformDefaultAudio => ALSA, and
-  // - kLinuxAlsaAudio => ALSA, and
-  // - kLinuxPulseAudio => Invalid selection.
-  RTC_LOG(LS_WARNING) << "PulseAudio is disabled using build flag.";
-  if ((audio_layer == kLinuxAlsaAudio) ||
-      (audio_layer == kPlatformDefaultAudio)) {
-    audio_device_.reset(new AudioDeviceLinuxALSA());
-    RTC_LOG(LS_INFO) << "Linux ALSA APIs will be utilized.";
-  }
-#else
-  // Build flag 'rtc_include_pulse_audio' is set to true (default). In this
-  // mode:
-  // - kPlatformDefaultAudio => PulseAudio, and
-  // - kLinuxPulseAudio => PulseAudio, and
-  // - kLinuxAlsaAudio => ALSA (supported but not default).
+#if defined(WEBRTC_ENABLE_LINUX_PULSE)
   RTC_LOG(LS_INFO) << "PulseAudio support is enabled.";
   if ((audio_layer == kLinuxPulseAudio) ||
       (audio_layer == kPlatformDefaultAudio)) {
-    // Linux PulseAudio implementation is default.
     audio_device_.reset(new AudioDeviceLinuxPulse());
     RTC_LOG(LS_INFO) << "Linux PulseAudio APIs will be utilized";
-  } else if (audio_layer == kLinuxAlsaAudio) {
+  }
+#endif
+
+#if defined(WEBRTC_ENABLE_LINUX_ALSA)
+  if (!audio_device_ &&
+      ((audio_layer == kLinuxAlsaAudio) ||
+       (audio_layer == kPlatformDefaultAudio))) {
     audio_device_.reset(new AudioDeviceLinuxALSA());
-    RTC_LOG(LS_WARNING) << "Linux ALSA APIs will be utilized.";
+    RTC_LOG(LS_INFO) << "Linux ALSA APIs will be utilized.";
   }
-#endif  // #if !defined(WEBRTC_ENABLE_LINUX_PULSE)
+#endif
+
+#if !defined(WEBRTC_ENABLE_LINUX_ALSA) && !defined(WEBRTC_ENABLE_LINUX_PULSE)
+  RTC_LOG(LS_WARNING)
+      << "Linux audio backends are disabled; falling back to dummy audio.";
+  if (audio_layer == kPlatformDefaultAudio) {
+    audio_device_.reset(new AudioDeviceDummy());
+    RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized.";
+  }
+#elif !defined(WEBRTC_ENABLE_LINUX_PULSE)
+  RTC_LOG(LS_WARNING) << "PulseAudio is disabled using build flag.";
+#elif !defined(WEBRTC_ENABLE_LINUX_ALSA)
+  RTC_LOG(LS_WARNING) << "ALSA is disabled using build flag.";
+#endif
 #endif  // #if defined(WEBRTC_LINUX)
 
 // iOS ADM implementation.
diff --git a/webrtc.gni b/webrtc.gni
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -274,6 +274,9 @@ declare_args() {
   rtc_build_opus = !build_with_mozilla
   rtc_build_ssl = !build_with_mozilla
 
+  # Excluded in Chromium since its prerequisites don't require ALSA.
+  rtc_include_alsa_audio = !build_with_chromium
+
   # Excluded in Chromium since its prerequisites don't require Pulse Audio.
   rtc_include_pulse_audio = !build_with_chromium