| 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|.
|
|
|