Move audioproc_f to rtc_tools.

The motivation in https://webrtc-review.googlesource.com/c/src/+/32340/3 applies here as well. We
would like to use this tool downstream.

Bug: None
Change-Id: Id5b23f792679ab9c07294bfb8e53119c423044b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161681
Commit-Queue: Daniel Johansson <dajo@webrtc.org>
Reviewed-by: Daniel Johansson <dajo@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Ã…hgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30051}
diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index 00528a4..bd4d5ad 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -36,6 +36,7 @@
         deps += [ ":event_log_visualizer" ]
       }
       deps += [
+        ":audioproc_f",
         ":rtp_analyzer",
         ":unpack_aecdump",
         "network_tester",
@@ -413,6 +414,19 @@
   }
 
   if (rtc_enable_protobuf) {
+    rtc_executable("audioproc_f") {
+      testonly = true
+      sources = [
+        "audioproc_f/audioproc_float_main.cc",
+      ]
+      deps = [
+        "../api:audioproc_f_api",
+        "../modules/audio_processing",
+        "../modules/audio_processing:api",
+        "../rtc_base:rtc_base_approved",
+      ]
+    }
+
     copy("rtp_analyzer") {
       sources = [
         "py_event_log_analyzer/misc.py",