Index: webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h |
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h |
index d0856a819ce94b21b7c6cb7d8b3d3d13f3814eac..cd7d7e874c82192fcee7d8aac9e379b54cac7566 100644 |
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h |
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h |
@@ -34,6 +34,10 @@ class RtpFileSource : public PacketSource { |
// opened, or has the wrong format, NULL will be returned. |
static RtpFileSource* Create(const std::string& file_name); |
+ // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file. |
+ static bool ValidRtpDump(const std::string& file_name); |
+ static bool ValidPcap(const std::string& file_name); |
+ |
virtual ~RtpFileSource(); |
// Registers an RTP header extension and binds it to |id|. |