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Add option to configure raw RTP packetization per payload type.

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Bug: webrtc:10625
Change-Id: I699f61af29656827eccb3c4ed507b4229dee972a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137803
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28036}
This commit is contained in:
Mirta Dvornicic
2019-05-23 13:21:12 +02:00
committed by Commit Bot
parent a352248c43
commit fe68daab97
17 changed files with 175 additions and 60 deletions
Download Patch File Download Diff File Expand all files Collapse all files

19
modules/rtp_rtcp/source/rtp_format.cc
View File

@ -24,14 +24,19 @@
namespace webrtc {
std::unique_ptr<RtpPacketizer> RtpPacketizer::Create(
VideoCodecType type,
absl::optional<VideoCodecType> type,
rtc::ArrayView<const uint8_t> payload,
PayloadSizeLimits limits,
// Codec-specific details.
const RTPVideoHeader& rtp_video_header,
VideoFrameType frame_type,
const RTPFragmentationHeader* fragmentation) {
switch (type) {
if (!type) {
// Use raw packetizer.
return absl::make_unique<RtpPacketizerGeneric>(payload, limits);
}
switch (*type) {
case kVideoCodecH264: {
RTC_CHECK(fragmentation);
const auto& h264 =
@ -133,8 +138,13 @@ std::vector<int> RtpPacketizer::SplitAboutEqually(
return result;
}
RtpDepacketizer* RtpDepacketizer::Create(VideoCodecType type) {
switch (type) {
RtpDepacketizer* RtpDepacketizer::Create(absl::optional<VideoCodecType> type) {
if (!type) {
// Use raw depacketizer.
return new RtpDepacketizerGeneric(/*generic_header_enabled=*/false);
}
switch (*type) {
case kVideoCodecH264:
return new RtpDepacketizerH264();
case kVideoCodecVP8:
@ -145,4 +155,5 @@ RtpDepacketizer* RtpDepacketizer::Create(VideoCodecType type) {
return new RtpDepacketizerGeneric(/*generic_header_enabled=*/true);
}
}
} // namespace webrtc