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 8307b83864de53302f9bf5cbe2973ad568c01073..5a580f9cf3d6e500e2b99a97b6a4f6c8971cbfa5 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, RTPSenderInterface* rtpSender); |
+ RTPSenderVideo(Clock* clock, RTPSender* rtpSender); |
virtual ~RTPSenderVideo(); |
virtual RtpVideoCodecTypes VideoCodecType() const; |
@@ -77,24 +77,14 @@ class RTPSenderVideo { |
void SetSelectiveRetransmissions(uint8_t settings); |
private: |
- void SendVideoPacket(uint8_t* dataBuffer, |
- const size_t payloadLength, |
- const size_t rtpHeaderLength, |
- uint16_t seq_num, |
- const uint32_t capture_timestamp, |
- int64_t capture_time_ms, |
+ void SendVideoPacket(std::unique_ptr<RtpPacketToSend> packet, |
StorageType storage); |
- void SendVideoPacketAsRed(uint8_t* dataBuffer, |
- const size_t payloadLength, |
- const size_t rtpHeaderLength, |
- uint16_t video_seq_num, |
- const uint32_t capture_timestamp, |
- int64_t capture_time_ms, |
+ void SendVideoPacketAsRed(std::unique_ptr<RtpPacketToSend> packet, |
StorageType media_packet_storage, |
bool protect); |
- RTPSenderInterface& _rtpSender; |
+ RTPSender& _rtpSender; |
// Should never be held when calling out of this class. |
const rtc::CriticalSection crit_; |