| 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 {
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| const RTPFragmentationHeader* fragmentation) override;
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|
|
| // Get the next payload with generic payload header.
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| - // buffer is a pointer to where the output will be written.
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| - // bytes_to_send is an output variable that will contain number of bytes
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| - // 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).
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| - // Returns true on success or false if there was no payload to packetize.
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| - bool NextPacket(uint8_t* buffer,
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| - size_t* bytes_to_send,
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| - bool* last_packet) override;
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| + // Write payload and set marker bit of the |packet|.
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| + // The parameter |last_packet| is true for the last packet of the frame, false
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| + // otherwise (i.e., call the function again to get the next packet).
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| + // Returns true on success, false otherwise.
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| + bool NextPacket(RtpPacketToSend* packet, bool* last_packet) override;
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|
|
| ProtectionType GetProtectionType() override;
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|