#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,
- &g_create_webrtc_audio_processing);
+ &init_diagnostic_logging);
+
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