| 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,
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| - bool rtcp,
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| - 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,
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| - bool header_only);
|
| -
|
| - static const uint32_t kDefaultSsrc = 1;
|
| - static const uint32_t kRtpTimestampIncrease = 90;
|
| - static const uint32_t kDefaultTimeIncrease = 30;
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| - static const uint32_t kElapsedTimeInterval = 10;
|
| - static const RawRtpPacket kTestRawRtpPackets[];
|
| - static const RawRtcpPacket kTestRawRtcpPackets[];
|
| -
|
| - private:
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| - RtpTestUtility() {}
|
| -};
|
| -
|
| // Test helper for testing VideoCapturer implementations.
|
| class VideoCapturerListener
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| : public sigslot::has_slots<>,
|
|
|