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Unified Diff: third_party/libjingle/overrides/init_webrtc.cc

Issue 717203002: Revert of Reland 597923002: Fix the way how we create webrtc::AudioProcessing in Chrome (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: fix build/android/pylib/utils/isolator.py Created 6 years, 1 month ago
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Index: third_party/libjingle/overrides/init_webrtc.cc
diff --git a/third_party/libjingle/overrides/init_webrtc.cc b/third_party/libjingle/overrides/init_webrtc.cc
index b160cbb1bf1ed37128f436982476b5ba425355c4..041fb20b05c1caa00fbf41aff20fc6dbeb96e51c 100644
--- a/third_party/libjingle/overrides/init_webrtc.cc
+++ b/third_party/libjingle/overrides/init_webrtc.cc
@@ -12,8 +12,6 @@
#include "base/metrics/histogram.h"
#include "base/native_library.h"
#include "base/path_service.h"
-#include "third_party/webrtc/common.h"
-#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/logging.h"
@@ -82,13 +80,6 @@ bool InitializeWebRtcModule() {
return true;
}
-webrtc::AudioProcessing* CreateWebRtcAudioProcessing(
- const webrtc::Config& config) {
- // libpeerconnection is being compiled as a static lib, use
- // webrtc::AudioProcessing directly.
- return webrtc::AudioProcessing::Create(config);
-}
-
#else // !LIBPEERCONNECTION_LIB
// When being compiled as a shared library, we need to bridge the gap between
@@ -98,7 +89,6 @@ webrtc::AudioProcessing* CreateWebRtcAudioProcessing(
// Global function pointers to the factory functions in the shared library.
CreateWebRtcMediaEngineFunction g_create_webrtc_media_engine = NULL;
DestroyWebRtcMediaEngineFunction g_destroy_webrtc_media_engine = NULL;
-CreateWebRtcAudioProcessingFunction g_create_webrtc_audio_processing = NULL;
// Returns the full or relative path to the libpeerconnection module depending
// on what platform we're on.
@@ -175,8 +165,8 @@ bool InitializeWebRtcModule() {
&AddTraceEvent,
&g_create_webrtc_media_engine,
&g_destroy_webrtc_media_engine,
- &init_diagnostic_logging,
- &g_create_webrtc_audio_processing);
+ &init_diagnostic_logging);
+
if (init_ok)
rtc::SetExtraLoggingInit(init_diagnostic_logging);
return init_ok;
@@ -200,12 +190,4 @@ void DestroyWebRtcMediaEngine(cricket::MediaEngineInterface* media_engine) {
g_destroy_webrtc_media_engine(media_engine);
}
-webrtc::AudioProcessing* CreateWebRtcAudioProcessing(
- const webrtc::Config& config) {
- // The same as CreateWebRtcMediaEngine(), we call InitializeWebRtcModule here
- // for convenience of tests.
- InitializeWebRtcModule();
- return g_create_webrtc_audio_processing(config);
-}
-
#endif // LIBPEERCONNECTION_LIB
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