Index: webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h |
index e72fe310cfcf934a43fe681914f5cdfaf7f0ee21..118166fbf3089bd9b507f0145e20ab49034373a0 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h |
@@ -36,15 +36,11 @@ class RtpPacketizerGeneric : public RtpPacketizer { |
const RTPFragmentationHeader* fragmentation) override; |
// Get the next payload with generic payload header. |
- // buffer is a pointer to where the output will be written. |
- // bytes_to_send is an output variable that will contain number of bytes |
- // written to buffer. The parameter last_packet is true for the last packet of |
- // the frame, false otherwise (i.e., call the function again to get the |
- // next packet). |
- // Returns true on success or false if there was no payload to packetize. |
- bool NextPacket(uint8_t* buffer, |
- size_t* bytes_to_send, |
- bool* last_packet) override; |
+ // Write payload and set marker bit of the |packet|. |
+ // The parameter |last_packet| is true for the last packet of the frame, false |
+ // otherwise (i.e., call the function again to get the next packet). |
+ // Returns true on success, false otherwise. |
+ bool NextPacket(RtpPacketToSend* packet, bool* last_packet) override; |
ProtectionType GetProtectionType() override; |