Index: webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc b/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc |
index 2fb723e3f916e45704cdecb06a23c62b71dceefe..904a9e2562b50ef6c8393b22c709572188578030 100644 |
--- a/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc |
+++ b/webrtc/modules/rtp_rtcp/source/h264_sps_parser.cc |
@@ -34,7 +34,7 @@ bool H264SpsParser::Parse() { |
// First, parse out rbsp, which is basically the source buffer minus emulation |
// bytes (the last byte of a 0x00 0x00 0x03 sequence). RBSP is defined in |
// section 7.3.1 of the H.264 standard. |
- rtc::ByteBuffer rbsp_buffer; |
+ rtc::ByteBufferWriter rbsp_buffer; |
for (size_t i = 0; i < byte_length_;) { |
// Be careful about over/underflow here. byte_length_ - 3 can underflow, and |
// i + 3 can overflow, but byte_length_ - i can't, because i < byte_length_ |