| 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;
|
|
|