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Delete RtpDepacketizer interface as no longer used

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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}
This commit is contained in:
Danil Chapovalov
2020-01-15 13:38:40 +01:00
committed by Commit Bot
parent 52ccb5e5b6
commit edb80cff01
1 changed files with 0 additions and 24 deletions
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24
modules/rtp_rtcp/source/rtp_format.h
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@ -59,29 +59,5 @@ class RtpPacketizer {
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_