- Removes voe_conference_test.
- Adds a new AudioStatsTest, with better coverage of the same features, based on call_test.
- Adds an AudioEndToEndTest utility, which AudioStatsTest and LowBandwidthAudioTest uses.
BUG=webrtc:4690
R=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/3008273002 .
Cr-Commit-Position: refs/heads/master@{#19833}
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index 42b74ff..890de51 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -58,6 +58,27 @@
]
}
if (rtc_include_tests) {
+ rtc_source_set("audio_end_to_end_test") {
+ testonly = true
+
+ sources = [
+ "test/audio_end_to_end_test.cc",
+ "test/audio_end_to_end_test.h",
+ ]
+ deps = [
+ ":audio",
+ "../system_wrappers:system_wrappers",
+ "../test:fake_audio_device",
+ "../test:test_common",
+ "../test:test_support",
+ ]
+
+ if (!build_with_chromium && is_clang) {
+ # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+ suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+ }
+ }
+
rtc_source_set("audio_tests") {
testonly = true
@@ -80,6 +101,7 @@
]
deps = [
":audio",
+ ":audio_end_to_end_test",
"../api:mock_audio_mixer",
"../call:rtp_receiver",
"../modules/audio_device:mock_audio_device",
@@ -96,6 +118,11 @@
"//testing/gtest",
]
+ if (!rtc_use_memcheck) {
+ # This test is timing dependent, which rules out running on memcheck bots.
+ sources += [ "test/audio_stats_test.cc" ]
+ }
+
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
@@ -108,10 +135,10 @@
sources = [
"test/low_bandwidth_audio_test.cc",
- "test/low_bandwidth_audio_test.h",
]
deps = [
+ ":audio_end_to_end_test",
"../common_audio",
"../rtc_base:rtc_base_approved",
"../system_wrappers",