Index: webrtc/media/base/testutils.h |
diff --git a/webrtc/media/base/testutils.h b/webrtc/media/base/testutils.h |
index 966a25be97f5bd62779e6da1dcdb0d404e9c110b..b0aacd57464f908d3a113f41015f6a7f7232346d 100644 |
--- a/webrtc/media/base/testutils.h |
+++ b/webrtc/media/base/testutils.h |
@@ -79,43 +79,6 @@ struct RawRtcpPacket { |
char payload[16]; |
}; |
-class RtpTestUtility { |
- public: |
- static size_t GetTestPacketCount(); |
- |
- // Write the first count number of kTestRawRtcpPackets or kTestRawRtpPackets, |
- // depending on the flag rtcp. If it is RTP, use the specified SSRC. Return |
- // true if successful. |
- static bool WriteTestPackets(size_t count, |
- bool rtcp, |
- uint32_t rtp_ssrc, |
- RtpDumpWriter* writer); |
- |
- // Loop read the first count number of packets from the specified stream. |
- // Verify the elapsed time of the dump packets increase monotonically. If the |
- // stream is a RTP stream, verify the RTP sequence number, timestamp, and |
- // payload. If the stream is a RTCP stream, verify the RTCP header and |
- // payload. |
- static bool VerifyTestPacketsFromStream(size_t count, |
- rtc::StreamInterface* stream, |
- uint32_t ssrc); |
- |
- // Verify the dump packet is the same as the raw RTP packet. |
- static bool VerifyPacket(const RtpDumpPacket* dump, |
- const RawRtpPacket* raw, |
- bool header_only); |
- |
- static const uint32_t kDefaultSsrc = 1; |
- static const uint32_t kRtpTimestampIncrease = 90; |
- static const uint32_t kDefaultTimeIncrease = 30; |
- static const uint32_t kElapsedTimeInterval = 10; |
- static const RawRtpPacket kTestRawRtpPackets[]; |
- static const RawRtcpPacket kTestRawRtcpPackets[]; |
- |
- private: |
- RtpTestUtility() {} |
-}; |
- |
// Test helper for testing VideoCapturer implementations. |
class VideoCapturerListener |
: public sigslot::has_slots<>, |