Revert of Added option to specify a maximum file size when recording an AEC dump.

talk/media/webrtc/webrtcvoiceengine.cc

Index: talk/media/webrtc/webrtcvoiceengine.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index f3a479b6c5fe089eddd7f9a1a98d71644e5911da..2634a24630741224d85c1a5006811b297ecfa394 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1016,8 +1016,7 @@
   return true;
 }
 
-bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
-                                     int64_t max_size_bytes) {
+bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
   if (!aec_dump_file_stream) {
@@ -1027,8 +1026,7 @@
     return false;
   }
   StopAecDump();
-  if (voe_wrapper_->base()->audio_processing()->StartDebugRecording(
-          aec_dump_file_stream, max_size_bytes) !=
+  if (voe_wrapper_->processing()->StartDebugRecording(aec_dump_file_stream) !=
       webrtc::AudioProcessing::kNoError) {
     LOG_RTCERR0(StartDebugRecording);
     fclose(aec_dump_file_stream);
@@ -1042,8 +1040,8 @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (!is_dumping_aec_) {
     // Start dumping AEC when we are not dumping.
-    if (voe_wrapper_->base()->audio_processing()->StartDebugRecording(
-            filename.c_str(), -1) != webrtc::AudioProcessing::kNoError) {
+    if (voe_wrapper_->processing()->StartDebugRecording(
+        filename.c_str()) != webrtc::AudioProcessing::kNoError) {
       LOG_RTCERR1(StartDebugRecording, filename.c_str());
     } else {
       is_dumping_aec_ = true;
@@ -1055,7 +1053,7 @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (is_dumping_aec_) {
     // Stop dumping AEC when we are dumping.
-    if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() !=
+    if (voe_wrapper_->processing()->StopDebugRecording() !=
         webrtc::AudioProcessing::kNoError) {
       LOG_RTCERR0(StopDebugRecording);
     }