Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(257)

Unified Diff: talk/media/webrtc/fakewebrtcvoiceengine.h

Issue 1413483003: Added option to specify a maximum file size when recording an AEC dump. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added function to avoid breaking Chromium. Created 5 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « talk/media/base/mediaengine.h ('k') | talk/media/webrtc/webrtcvoiceengine.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: talk/media/webrtc/fakewebrtcvoiceengine.h
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index b7a7262af74ccb39e59c34b705f031406294619e..d45880763f53b5b473e4d30b16445dbd72638ba0 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -112,8 +112,9 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed));
WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset));
WEBRTC_STUB_CONST(delay_offset_ms, ());
- WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize]));
- WEBRTC_STUB(StartDebugRecording, (FILE* handle));
+ WEBRTC_STUB(StartDebugRecording,
+ (const char filename[kMaxFilenameSize], int64_t max_size_bytes));
+ WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes));
WEBRTC_STUB(StopDebugRecording, ());
WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ());
webrtc::EchoCancellation* echo_cancellation() const override { return NULL; }
« no previous file with comments | « talk/media/base/mediaengine.h ('k') | talk/media/webrtc/webrtcvoiceengine.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698