diff --git a/modules/rtp_rtcp/source/rtp_sender_video.h b/modules/rtp_rtcp/source/rtp_sender_video.h index 57f8fcc7ac..e8cba5073d 100644 --- a/modules/rtp_rtcp/source/rtp_sender_video.h +++ b/modules/rtp_rtcp/source/rtp_sender_video.h @@ -32,7 +32,6 @@ #include "modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h" #include "modules/rtp_rtcp/source/rtp_video_header.h" #include "modules/rtp_rtcp/source/video_fec_generator.h" -#include "rtc_base/deprecation.h" #include "rtc_base/one_time_event.h" #include "rtc_base/race_checker.h" #include "rtc_base/rate_statistics.h" @@ -42,7 +41,6 @@ namespace webrtc { -class RTPFragmentationHeader; class FrameEncryptorInterface; class RtpPacketizer; class RtpPacketToSend; @@ -91,19 +89,6 @@ class RTPSenderVideo { virtual ~RTPSenderVideo(); - RTC_DEPRECATED - bool SendVideo(int payload_type, - absl::optional codec_type, - uint32_t rtp_timestamp, - int64_t capture_time_ms, - rtc::ArrayView payload, - const RTPFragmentationHeader* /*fragmentation*/, - RTPVideoHeader video_header, - absl::optional expected_retransmission_time_ms) { - return SendVideo(payload_type, codec_type, rtp_timestamp, capture_time_ms, - payload, video_header, expected_retransmission_time_ms); - } - // expected_retransmission_time_ms.has_value() -> retransmission allowed. // Calls to this method is assumed to be externally serialized. bool SendVideo(int payload_type,