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Add application_data field(s) to RtpPacketToSend and PacketOptions.

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Pass pointer to application_data from RtpPacketToSend arriving via RtpSender::SendToNetwork through to Transport::SendRtp, in PacketOptions.

Bug: webrtc:8906
Change-Id: Ie75013ed472710f4efcfbcc160e46a6119a1f41d
Reviewed-on: https://webrtc-review.googlesource.com/55600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dino Radaković <dinor@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22174}
This commit is contained in:
Dino Radaković
2018-02-22 14:18:06 +01:00
committed by Commit Bot
parent f35c6667d6
commit 1807d57ab8
7 changed files with 82 additions and 6 deletions
Download Patch File Download Diff File Expand all files Collapse all files

26
modules/rtp_rtcp/source/rtp_packet_to_send.h
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@ -10,6 +10,9 @@
#ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
#include <vector>
#include "api/array_view.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
@ -17,19 +20,29 @@ namespace webrtc {
// Class to hold rtp packet with metadata for sender side.
class RtpPacketToSend : public RtpPacket {
public:
explicit RtpPacketToSend(const ExtensionManager* extensions)
: RtpPacket(extensions) {}
RtpPacketToSend(const RtpPacketToSend& packet) = default;
RtpPacketToSend(const ExtensionManager* extensions, size_t capacity)
: RtpPacket(extensions, capacity) {}
explicit RtpPacketToSend(const ExtensionManager* extensions);
RtpPacketToSend(const RtpPacketToSend& packet);
RtpPacketToSend(const ExtensionManager* extensions, size_t capacity);
RtpPacketToSend& operator=(const RtpPacketToSend& packet) = default;
RtpPacketToSend& operator=(const RtpPacketToSend& packet);
~RtpPacketToSend();
// Time in local time base as close as it can to frame capture time.
int64_t capture_time_ms() const { return capture_time_ms_; }
void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; }
// Additional data bound to the RTP packet for use in application code,
// outside of WebRTC.
rtc::ArrayView<const uint8_t> application_data() const {
return application_data_;
}
void set_application_data(rtc::ArrayView<const uint8_t> data) {
application_data_.assign(data.begin(), data.end());
}
void set_packetization_finish_time_ms(int64_t time) {
SetExtension<VideoTimingExtension>(
VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time),
@ -56,6 +69,7 @@ class RtpPacketToSend : public RtpPacket {
private:
int64_t capture_time_ms_ = 0;
std::vector<uint8_t> application_data_;
};
} // namespace webrtc