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}
This commit is contained in:
Niels Möller
2019-01-24 09:44:14 +01:00
committed by Commit Bot
parent 2820d174b5
commit 171df93262
10 changed files with 57 additions and 214 deletions

View File

@ -41,47 +41,6 @@ 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 };