Delete RtpDepacketizer interface as no longer used
Bug: webrtc:11152
Change-Id: I0c5f2167ba39c22f4491d2e34f3462b9ecb9bf2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166160
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30276}
diff --git a/modules/rtp_rtcp/source/rtp_format.h b/modules/rtp_rtcp/source/rtp_format.h
index 144ae76..dca8285 100644
--- a/modules/rtp_rtcp/source/rtp_format.h
+++ b/modules/rtp_rtcp/source/rtp_format.h
@@ -59,29 +59,5 @@
static std::vector<int> SplitAboutEqually(int payload_len,
const PayloadSizeLimits& limits);
};
-
-// TODO(bugs.webrtc.org/11152): Update the depacketizer to return a copy
-// of the parsed payload, rather than just a pointer into the incoming buffer.
-// This way we can move some parsing out from the jitter buffer into here, and
-// the jitter buffer can just store that pointer rather than doing a copy there.
-class RtpDepacketizer {
- public:
- struct ParsedPayload {
- RTPVideoHeader& video_header() { return video; }
- const RTPVideoHeader& video_header() const { return video; }
-
- RTPVideoHeader video;
-
- const uint8_t* payload;
- size_t payload_length;
- };
-
- virtual ~RtpDepacketizer() {}
-
- // Parses the RTP payload, parsed result will be saved in |parsed_payload|.
- virtual bool Parse(ParsedPayload* parsed_payload,
- const uint8_t* payload_data,
- size_t payload_data_length) = 0;
-};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
