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",