Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
index 1d6203a68a75f1d7cdc7db95262058478fd691bd..f39309a5d1243eb013e8fb88e5d3e2513a4ec688 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
@@ -166,17 +166,24 @@ class RTPSender : public RTPSenderInterface { |
size_t RtpHeaderExtensionLength() const; |
- uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer, bool marker_bit) const; |
+ uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer, bool marker_bit) const |
+ EXCLUSIVE_LOCKS_REQUIRED(send_critsect_); |
- uint8_t BuildTransmissionTimeOffsetExtension(uint8_t *data_buffer) const; |
- uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const; |
- uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const; |
- uint8_t BuildVideoRotationExtension(uint8_t* data_buffer) const; |
+ uint8_t BuildTransmissionTimeOffsetExtension(uint8_t* data_buffer) const |
+ EXCLUSIVE_LOCKS_REQUIRED(send_critsect_); |
+ uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const |
+ EXCLUSIVE_LOCKS_REQUIRED(send_critsect_); |
+ uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const |
+ EXCLUSIVE_LOCKS_REQUIRED(send_critsect_); |
+ uint8_t BuildVideoRotationExtension(uint8_t* data_buffer) const |
+ EXCLUSIVE_LOCKS_REQUIRED(send_critsect_); |
uint8_t BuildTransportSequenceNumberExtension(uint8_t* data_buffer, |
- uint16_t sequence_number) const; |
+ uint16_t sequence_number) const |
+ EXCLUSIVE_LOCKS_REQUIRED(send_critsect_); |
uint8_t BuildPlayoutDelayExtension(uint8_t* data_buffer, |
uint16_t min_playout_delay_ms, |
- uint16_t max_playout_delay_ms) const; |
+ uint16_t max_playout_delay_ms) const |
+ EXCLUSIVE_LOCKS_REQUIRED(send_critsect_); |
// Verifies that the specified extension is registered, and that it is |
// present in rtp packet. If extension is not registered kNotRegistered is |
@@ -335,7 +342,8 @@ class RTPSender : public RTPSenderInterface { |
bool marker_bit, |
uint32_t timestamp, |
uint16_t sequence_number, |
- const std::vector<uint32_t>& csrcs) const; |
+ const std::vector<uint32_t>& csrcs) const |
+ EXCLUSIVE_LOCKS_REQUIRED(send_critsect_); |
bool PrepareAndSendPacket(uint8_t* buffer, |
size_t length, |
@@ -370,7 +378,8 @@ class RTPSender : public RTPSenderInterface { |
const uint8_t* rtp_packet, |
size_t rtp_packet_length, |
const RTPHeader& rtp_header, |
- size_t* position) const; |
+ size_t* position) const |
+ EXCLUSIVE_LOCKS_REQUIRED(send_critsect_); |
void UpdateTransmissionTimeOffset(uint8_t* rtp_packet, |
size_t rtp_packet_length, |
@@ -381,10 +390,10 @@ class RTPSender : public RTPSenderInterface { |
const RTPHeader& rtp_header, |
int64_t now_ms) const; |
- bool UpdateTransportSequenceNumber(uint16_t sequence_number, |
- uint8_t* rtp_packet, |
+ bool UpdateTransportSequenceNumber(uint8_t* rtp_packet, |
size_t rtp_packet_length, |
- const RTPHeader& rtp_header) const; |
+ const RTPHeader& rtp_header, |
+ int* sequence_number) const; |
void UpdatePlayoutDelayLimits(uint8_t* rtp_packet, |
size_t rtp_packet_length, |
@@ -423,7 +432,7 @@ class RTPSender : public RTPSenderInterface { |
int8_t payload_type_ GUARDED_BY(send_critsect_); |
std::map<int8_t, RtpUtility::Payload*> payload_type_map_; |
- RtpHeaderExtensionMap rtp_header_extension_map_; |
+ RtpHeaderExtensionMap rtp_header_extension_map_ GUARDED_BY(send_critsect_); |
int32_t transmission_time_offset_; |
uint32_t absolute_send_time_; |
VideoRotation rotation_; |