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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_video.h

Issue 2217383002: Use RtpPacketToSend in RtpSenderVideo (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Ported PlayoutDelay extension support. Created 4 years, 3 months ago
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Index: webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
index 59224cc7d9d3130c822f3b1e6ba658274f53135f..0ac1630a0e67c110628a890609c448e024c2a4cd 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -12,6 +12,7 @@
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
#include <list>
+#include <memory>
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/onetimeevent.h"
@@ -75,20 +76,10 @@ class RTPSenderVideo {
void SetSelectiveRetransmissions(uint8_t settings);
private:
- void SendVideoPacket(uint8_t* data_buffer,
- size_t payload_length,
- size_t rtp_header_length,
- uint16_t seq_num,
- uint32_t capture_timestamp,
- int64_t capture_time_ms,
+ void SendVideoPacket(std::unique_ptr<RtpPacketToSend> packet,
StorageType storage);
- void SendVideoPacketAsRed(uint8_t* data_buffer,
- size_t payload_length,
- size_t rtp_header_length,
- uint16_t video_seq_num,
- uint32_t capture_timestamp,
- int64_t capture_time_ms,
+ void SendVideoPacketAsRed(std::unique_ptr<RtpPacketToSend> packet,
StorageType media_packet_storage,
bool protect);
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