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

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This is a reland of 171df9326200d1e01bce530e2ff01ac5890e6cb7

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
Bug: webrtc:6883
Change-Id: I30771b86bbe50de609353e23e80dc532dc884ad4
Reviewed-on: https://webrtc-review.googlesource.com/c/119661
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26394}
This commit is contained in:
Niels Möller
2019-01-24 09:44:14 +01:00
committed by Commit Bot
parent 5affbf2327
commit 3ea55d56eb
10 changed files with 57 additions and 214 deletions
Download Patch File Download Diff File Expand all files Collapse all files

9
modules/rtp_rtcp/include/rtp_rtcp_defines.cc
View File

@ -53,15 +53,6 @@ void StreamId::Set(const char* data, size_t size) {
// and thus assume trivial destructibility.
static_assert(std::is_trivially_destructible<StreamId>::value, "");
PayloadUnion::PayloadUnion(const AudioPayload& payload) : payload_(payload) {}
PayloadUnion::PayloadUnion(const VideoPayload& payload) : payload_(payload) {}
PayloadUnion::PayloadUnion(const PayloadUnion&) = default;
PayloadUnion::PayloadUnion(PayloadUnion&&) = default;
PayloadUnion::~PayloadUnion() = default;
PayloadUnion& PayloadUnion::operator=(const PayloadUnion&) = default;
PayloadUnion& PayloadUnion::operator=(PayloadUnion&&) = default;
PacketFeedback::PacketFeedback(int64_t arrival_time_ms,
uint16_t sequence_number)
: PacketFeedback(-1,

41
modules/rtp_rtcp/include/rtp_rtcp_defines.h
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 };