#include "base/metrics/field_trial.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"
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
// 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.
&AddTraceEvent,
&g_create_webrtc_media_engine,
&g_destroy_webrtc_media_engine,
- &init_diagnostic_logging);
-
+ &init_diagnostic_logging,
+ &g_create_webrtc_audio_processing);
if (init_ok)
rtc::SetExtraLoggingInit(init_diagnostic_logging);
return init_ok;
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