Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
index 7c72e5917c8ab2feb8acdac1c119cd3e25b8b5c6..a6c349a8c90578501e705dd62682fb5f4f465e76 100644 |
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
@@ -40,45 +40,47 @@ class RtpRtcp : public Module { |
struct Configuration { |
Configuration(); |
- /* id - Unique identifier of this RTP/RTCP module object |
- * audio - True for a audio version of the RTP/RTCP module |
- * object false will create a video version |
- * clock - The clock to use to read time. If NULL object |
- * will be using the system clock. |
- * incoming_data - Callback object that will receive the incoming |
- * data. May not be NULL; default callback will do |
- * nothing. |
- * incoming_messages - Callback object that will receive the incoming |
- * RTP messages. May not be NULL; default callback |
- * will do nothing. |
- * outgoing_transport - Transport object that will be called when packets |
- * are ready to be sent out on the network |
- * intra_frame_callback - Called when the receiver request a intra frame. |
- * bandwidth_callback - Called when we receive a changed estimate from |
- * the receiver of out stream. |
- * remote_bitrate_estimator - Estimates the bandwidth available for a set of |
- * streams from the same client. |
- * paced_sender - Spread any bursts of packets into smaller |
- * bursts to minimize packet loss. |
- */ |
- bool audio; |
- bool receiver_only; |
- Clock* clock; |
+ // True for a audio version of the RTP/RTCP module object false will create |
+ // a video version. |
+ bool audio = false; |
+ bool receiver_only = false; |
+ |
+ // The clock to use to read time. If nullptr then system clock will be used. |
+ Clock* clock = nullptr; |
+ |
ReceiveStatistics* receive_statistics; |
- Transport* outgoing_transport; |
- RtcpIntraFrameObserver* intra_frame_callback; |
- RtcpBandwidthObserver* bandwidth_callback; |
- TransportFeedbackObserver* transport_feedback_callback; |
- RtcpRttStats* rtt_stats; |
- RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer; |
- RemoteBitrateEstimator* remote_bitrate_estimator; |
- RtpPacketSender* paced_sender; |
- TransportSequenceNumberAllocator* transport_sequence_number_allocator; |
- BitrateStatisticsObserver* send_bitrate_observer; |
- FrameCountObserver* send_frame_count_observer; |
- SendSideDelayObserver* send_side_delay_observer; |
- RtcEventLog* event_log; |
- SendPacketObserver* send_packet_observer; |
+ |
+ // Transport object that will be called when packets are ready to be sent |
+ // out on the network. |
+ Transport* outgoing_transport = nullptr; |
+ |
+ // Called when the receiver request a intra frame. |
+ RtcpIntraFrameObserver* intra_frame_callback = nullptr; |
+ |
+ // Called when we receive a changed estimate from the receiver of out |
+ // stream. |
+ RtcpBandwidthObserver* bandwidth_callback = nullptr; |
+ |
+ TransportFeedbackObserver* transport_feedback_callback = nullptr; |
+ RtcpRttStats* rtt_stats = nullptr; |
+ RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr; |
+ |
+ // Estimates the bandwidth available for a set of streams from the same |
+ // client. |
+ RemoteBitrateEstimator* remote_bitrate_estimator = nullptr; |
+ |
+ // Spread any bursts of packets into smaller bursts to minimize packet loss. |
+ RtpPacketSender* paced_sender = nullptr; |
+ |
+ TransportSequenceNumberAllocator* transport_sequence_number_allocator = |
+ nullptr; |
+ BitrateStatisticsObserver* send_bitrate_observer = nullptr; |
+ FrameCountObserver* send_frame_count_observer = nullptr; |
+ SendSideDelayObserver* send_side_delay_observer = nullptr; |
+ RtcEventLog* event_log = nullptr; |
+ SendPacketObserver* send_packet_observer = nullptr; |
+ |
+ private: |
RTC_DISALLOW_COPY_AND_ASSIGN(Configuration); |
}; |
@@ -173,11 +175,11 @@ class RtpRtcp : public Module { |
/* |
* Unregister a send payload |
* |
- * payloadType - payload type of codec |
+ * payload_type - payload type of codec |
* |
* return -1 on failure else 0 |
*/ |
- virtual int32_t DeRegisterSendPayload(int8_t payloadType) = 0; |
+ virtual int32_t DeRegisterSendPayload(int8_t payload_type) = 0; |
/* |
* (De)register RTP header extension type and id. |
@@ -292,23 +294,23 @@ class RtpRtcp : public Module { |
* Used by the codec module to deliver a video or audio frame for |
* packetization. |
* |
- * frameType - type of frame to send |
- * payloadType - payload type of frame to send |
+ * frame_type - type of frame to send |
+ * payload_type - payload type of frame to send |
* timestamp - timestamp of frame to send |
- * payloadData - payload buffer of frame to send |
- * payloadSize - size of payload buffer to send |
+ * payload_data - payload buffer of frame to send |
+ * payload_size - size of payload buffer to send |
* fragmentation - fragmentation offset data for fragmented frames such |
* as layers or RED |
* |
* return -1 on failure else 0 |
*/ |
virtual int32_t SendOutgoingData( |
- FrameType frameType, |
- int8_t payloadType, |
+ FrameType frame_type, |
+ int8_t payload_type, |
uint32_t timeStamp, |
int64_t capture_time_ms, |
- const uint8_t* payloadData, |
- size_t payloadSize, |
+ const uint8_t* payload_data, |
+ size_t payload_size, |
const RTPFragmentationHeader* fragmentation = NULL, |
const RTPVideoHeader* rtpVideoHdr = NULL) = 0; |
@@ -356,7 +358,7 @@ class RtpRtcp : public Module { |
* |
* return -1 on failure else 0 |
*/ |
- virtual int32_t RemoteCNAME(uint32_t remoteSSRC, |
+ virtual int32_t RemoteCNAME(uint32_t remote_ssrc, |
char cName[RTCP_CNAME_SIZE]) const = 0; |
/* |
@@ -390,11 +392,11 @@ class RtpRtcp : public Module { |
* |
* return -1 on failure else 0 |
*/ |
- virtual int32_t RTT(uint32_t remoteSSRC, |
- int64_t* RTT, |
- int64_t* avgRTT, |
- int64_t* minRTT, |
- int64_t* maxRTT) const = 0; |
+ virtual int32_t RTT(uint32_t remote_ssrc, |
+ int64_t* rtt, |
+ int64_t* avg_rtt, |
+ int64_t* min_rtt, |
+ int64_t* max_rtt) const = 0; |
/* |
* Force a send of a RTCP packet |
@@ -402,7 +404,7 @@ class RtpRtcp : public Module { |
* |
* return -1 on failure else 0 |
*/ |
- virtual int32_t SendRTCP(RTCPPacketType rtcpPacketType) = 0; |
+ virtual int32_t SendRTCP(RTCPPacketType rtcp_packet_type) = 0; |
/* |
* Force a send of a RTCP packet with more than one packet type. |
@@ -411,7 +413,7 @@ class RtpRtcp : public Module { |
* return -1 on failure else 0 |
*/ |
virtual int32_t SendCompoundRTCP( |
- const std::set<RTCPPacketType>& rtcpPacketTypes) = 0; |
+ const std::set<RTCPPacketType>& rtcp_packet_types) = 0; |
/* |
* Good state of RTP receiver inform sender |
@@ -453,7 +455,7 @@ class RtpRtcp : public Module { |
* |
* return -1 on failure else 0 |
*/ |
- virtual int32_t RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0; |
+ virtual int32_t RemoteRTCPStat(RTCPSenderInfo* sender_info) = 0; |
/* |
* Get received RTCP report block |
@@ -461,14 +463,14 @@ class RtpRtcp : public Module { |
* return -1 on failure else 0 |
*/ |
virtual int32_t RemoteRTCPStat( |
- std::vector<RTCPReportBlock>* receiveBlocks) const = 0; |
+ std::vector<RTCPReportBlock>* receive_blocks) const = 0; |
/* |
* (APP) Application specific data |
* |
* return -1 on failure else 0 |
*/ |
- virtual int32_t SetRTCPApplicationSpecificData(uint8_t subType, |
+ virtual int32_t SetRTCPApplicationSpecificData(uint8_t sub_type, |
uint32_t name, |
const uint8_t* data, |
uint16_t length) = 0; |
@@ -536,7 +538,7 @@ class RtpRtcp : public Module { |
// TODO(philipel): Deprecate this and start using SendNack instead, |
// mostly because we want a function that actually send |
// NACK for the specified packets. |
- virtual int32_t SendNACK(const uint16_t* nackList, uint16_t size) = 0; |
+ virtual int32_t SendNACK(const uint16_t* nack_list, uint16_t size) = 0; |
/* |
* Send NACK for the packets specified. |
@@ -575,7 +577,7 @@ class RtpRtcp : public Module { |
* |
* return -1 on failure else 0 |
*/ |
- virtual int32_t SetAudioPacketSize(uint16_t packetSizeSamples) = 0; |
+ virtual int32_t SetAudioPacketSize(uint16_t packet_size_samples) = 0; |
/* |
* Send a TelephoneEvent tone using RFC 2833 (4733) |
@@ -591,7 +593,7 @@ class RtpRtcp : public Module { |
* |
* return -1 on failure else 0 |
*/ |
- virtual int32_t SetSendREDPayloadType(int8_t payloadType) = 0; |
+ virtual int32_t SetSendREDPayloadType(int8_t payload_type) = 0; |
/* |
* Get payload type for Redundant Audio Data RFC 2198 |