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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: Rebase Created 4 years, 4 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..3a78b0e66dcc15648cd02cc56dfc44ab388adeb4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -31,7 +31,7 @@ namespace webrtc {
class RTPSenderVideo {
public:
- RTPSenderVideo(Clock* clock, RTPSender* rtpSender);
+ RTPSenderVideo(Clock* clock, RTPSender* rtp_sender);
virtual ~RTPSenderVideo();
virtual RtpVideoCodecTypes VideoCodecType() const;
@@ -75,20 +75,10 @@ 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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