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Revert "Delete RtpUtility::Payload, and refactor RTPSender to not use it"

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This reverts commit 171df9326200d1e01bce530e2ff01ac5890e6cb7.

Reason for revert: Breaks downstream project

Original change's description:
> Delete RtpUtility::Payload, and refactor RTPSender to not use it
> 
> Replaced by a payload type --> video codec map in RTPSenderVideo,
> where it is used to select the right packetizer.
> 
> Bug: webrtc:6883
> Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
> Reviewed-on: https://webrtc-review.googlesource.com/c/119263
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26380}

TBR=danilchap@webrtc.org,brandtr@webrtc.org,nisse@webrtc.org

Change-Id: I76489c29541827aaba72515a76db54bdb7495e28
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6883
Reviewed-on: https://webrtc-review.googlesource.com/c/119640
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26385}
This commit is contained in:
Artem Titov
2019-01-24 12:01:59 +00:00
committed by Commit Bot
parent 1e27fec293
commit 81d4bf7af6
10 changed files with 214 additions and 57 deletions
Download Patch File Download Diff File Expand all files Collapse all files

41
modules/rtp_rtcp/include/rtp_rtcp_defines.h
View File

@ -41,6 +41,47 @@ const int kBogusRtpRateForAudioRtcp = 8000;
// Minimum RTP header size in bytes.
const uint8_t kRtpHeaderSize = 12;
struct AudioPayload {
SdpAudioFormat format;
uint32_t rate;
};
struct VideoPayload {
VideoCodecType videoCodecType;
// The H264 profile only matters if videoCodecType == kVideoCodecH264.
H264::Profile h264_profile;
};
class PayloadUnion {
public:
explicit PayloadUnion(const AudioPayload& payload);
explicit PayloadUnion(const VideoPayload& payload);
PayloadUnion(const PayloadUnion&);
PayloadUnion(PayloadUnion&&);
~PayloadUnion();
PayloadUnion& operator=(const PayloadUnion&);
PayloadUnion& operator=(PayloadUnion&&);
bool is_audio() const {
return absl::holds_alternative<AudioPayload>(payload_);
}
bool is_video() const {
return absl::holds_alternative<VideoPayload>(payload_);
}
const AudioPayload& audio_payload() const {
return absl::get<AudioPayload>(payload_);
}
const VideoPayload& video_payload() const {
return absl::get<VideoPayload>(payload_);
}
AudioPayload& audio_payload() { return absl::get<AudioPayload>(payload_); }
VideoPayload& video_payload() { return absl::get<VideoPayload>(payload_); }
private:
absl::variant<AudioPayload, VideoPayload> payload_;
};
enum ProtectionType { kUnprotectedPacket, kProtectedPacket };
enum StorageType { kDontRetransmit, kAllowRetransmission };