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Cleanup RtpPacketizer interface

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merge construction and call to SetPayloadData
Add NumPackets instead of SetPayloadData
Remove virtual ToString() as unused
move CHECK(rtp_video_header) from RtpPacketizer::Create to RtpSenderVideo::SendVideo

Bug: webrtc:9680
Change-Id: I074644e048c797eb836f79979df363fe1ea0075e
Reviewed-on: https://webrtc-review.googlesource.com/96543
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24474}
This commit is contained in:
Danil Chapovalov
2018-08-28 19:45:31 +02:00
committed by Commit Bot
parent 2a4906532f
commit f7f8a1f176
12 changed files with 127 additions and 110 deletions
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56
modules/rtp_rtcp/source/rtp_format.cc
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@ -12,40 +12,56 @@
#include <utility>
#include "absl/memory/memory.h"
#include "modules/rtp_rtcp/source/rtp_format_h264.h"
#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
namespace webrtc {
RtpPacketizer* RtpPacketizer::Create(VideoCodecType type,
size_t max_payload_len,
size_t last_packet_reduction_len,
const RTPVideoHeader* rtp_video_header,
FrameType frame_type) {
RTC_CHECK(rtp_video_header);
std::unique_ptr<RtpPacketizer> RtpPacketizer::Create(
VideoCodecType type,
rtc::ArrayView<const uint8_t> payload,
PayloadSizeLimits limits,
// Codec-specific details.
const RTPVideoHeader& rtp_video_header,
FrameType frame_type,
const RTPFragmentationHeader* fragmentation) {
switch (type) {
case kVideoCodecH264: {
const auto& h264 =
absl::get<RTPVideoHeaderH264>(rtp_video_header->video_type_header);
return new RtpPacketizerH264(max_payload_len, last_packet_reduction_len,
h264.packetization_mode);
absl::get<RTPVideoHeaderH264>(rtp_video_header.video_type_header);
auto packetizer = absl::make_unique<RtpPacketizerH264>(
limits.max_payload_len, limits.last_packet_reduction_len,
h264.packetization_mode);
packetizer->SetPayloadData(payload.data(), payload.size(), fragmentation);
return std::move(packetizer);
}
case kVideoCodecVP8: {
const auto& vp8 =
absl::get<RTPVideoHeaderVP8>(rtp_video_header.video_type_header);
auto packetizer = absl::make_unique<RtpPacketizerVp8>(
vp8, limits.max_payload_len, limits.last_packet_reduction_len);
packetizer->SetPayloadData(payload.data(), payload.size(), nullptr);
return std::move(packetizer);
}
case kVideoCodecVP8:
return new RtpPacketizerVp8(rtp_video_header->vp8(), max_payload_len,
last_packet_reduction_len);
case kVideoCodecVP9: {
const auto& vp9 =
absl::get<RTPVideoHeaderVP9>(rtp_video_header->video_type_header);
return new RtpPacketizerVp9(vp9, max_payload_len,
last_packet_reduction_len);
absl::get<RTPVideoHeaderVP9>(rtp_video_header.video_type_header);
auto packetizer = absl::make_unique<RtpPacketizerVp9>(
vp9, limits.max_payload_len, limits.last_packet_reduction_len);
packetizer->SetPayloadData(payload.data(), payload.size(), nullptr);
return std::move(packetizer);
}
default: {
auto packetizer = absl::make_unique<RtpPacketizerGeneric>(
rtp_video_header, frame_type, limits.max_payload_len,
limits.last_packet_reduction_len);
packetizer->SetPayloadData(payload.data(), payload.size(), nullptr);
return std::move(packetizer);
}
default:
return new RtpPacketizerGeneric(*rtp_video_header, frame_type,
max_payload_len,
last_packet_reduction_len);
}
return nullptr;
}
RtpDepacketizer* RtpDepacketizer::Create(VideoCodecType type) {