Index: webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc |
index 2d847c17a608ba5f64dae5f938b3ca4b7689693f..e461e8143c8bac6bd1e3690c84fdf23110278a40 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc |
@@ -21,11 +21,10 @@ static const size_t kGenericHeaderLength = 1; |
RtpPacketizerGeneric::RtpPacketizerGeneric(FrameType frame_type, |
size_t max_payload_len) |
- : payload_data_(NULL), |
+ : payload_data_(nullptr), |
payload_size_(0), |
max_payload_len_(max_payload_len - kGenericHeaderLength), |
- frame_type_(frame_type) { |
-} |
+ frame_type_(frame_type) {} |
RtpPacketizerGeneric::~RtpPacketizerGeneric() { |
} |
@@ -92,7 +91,7 @@ std::string RtpPacketizerGeneric::ToString() { |
bool RtpDepacketizerGeneric::Parse(ParsedPayload* parsed_payload, |
const uint8_t* payload_data, |
size_t payload_data_length) { |
- assert(parsed_payload != NULL); |
+ assert(parsed_payload != nullptr); |
if (payload_data_length == 0) { |
LOG(LS_ERROR) << "Empty payload."; |
return false; |