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 eed1195975704aea781d564eefca5a9953b164d4..aa8c941c9891a1f8f4f3d8f295e9e8d3ccda34a0 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1259,7 +1259,8 @@ bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm) {
   return true;
 }
 
-bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) {
+bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
+                                     int64_t max_size_bytes) {
   FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
   if (!aec_dump_file_stream) {
     LOG(LS_ERROR) << "Could not open AEC dump file stream.";
@@ -1268,7 +1269,8 @@ bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) {
     return false;
   }
   StopAecDump();
-  if (voe_wrapper_->processing()->StartDebugRecording(aec_dump_file_stream) !=
+  if (voe_wrapper_->base()->audio_processing()->StartDebugRecording(
+          aec_dump_file_stream, max_size_bytes) !=
       webrtc::AudioProcessing::kNoError) {
     LOG_RTCERR0(StartDebugRecording);
     fclose(aec_dump_file_stream);
@@ -1281,8 +1283,8 @@ bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) {
 void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
   if (!is_dumping_aec_) {
     // Start dumping AEC when we are not dumping.
-    if (voe_wrapper_->processing()->StartDebugRecording(
-        filename.c_str()) != webrtc::AudioProcessing::kNoError) {
+    if (voe_wrapper_->base()->audio_processing()->StartDebugRecording(
+            filename.c_str(), -1) != webrtc::AudioProcessing::kNoError) {
       LOG_RTCERR1(StartDebugRecording, filename.c_str());
     } else {
       is_dumping_aec_ = true;
@@ -1293,7 +1295,7 @@ void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
 void WebRtcVoiceEngine::StopAecDump() {
   if (is_dumping_aec_) {
     // Stop dumping AEC when we are dumping.
-    if (voe_wrapper_->processing()->StopDebugRecording() !=
+    if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() !=
         webrtc::AudioProcessing::kNoError) {
       LOG_RTCERR0(StopDebugRecording);
     }