Default constructors for new VideoEngine structs.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1543004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4115 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/video_engine/new_include/common.h

@@ -63,6 +63,11 @@ class Transport {
 };
 
 struct RtpStatistics {
+  RtpStatistics()
+      : ssrc(0),
+        fraction_loss(0),
+        cumulative_loss(0),
+        extended_max_sequence_number(0) {}
   uint32_t ssrc;
   int fraction_loss;
   int cumulative_loss;
@@ -70,7 +75,7 @@ struct RtpStatistics {
   std::string c_name;
 };
 
-// RTCP mode to use. Compound mode is described by RFC 4585 and reduced sized
+// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
 // RTCP mode is described by RFC 5506.
 enum RtcpMode {
   kRtcpCompound,
@@ -79,16 +84,18 @@ enum RtcpMode {
 
 // Settings for NACK, see RFC 4585 for details.
 struct NackConfig {
+  NackConfig() : rtp_history_ms(0) {}
   // Send side: the time RTP packets are stored for retransmissions.
   // Receive side: the time the receiver is prepared to wait for
   // retransmissions.
-  // Set to '0' to disable
+  // Set to '0' to disable.
   int rtp_history_ms;
 };
 
 // Settings for forward error correction, see RFC 5109 for details. Set the
 // payload types to '-1' to disable.
 struct FecConfig {
+  FecConfig() : ulpfec_payload_type(-1), red_payload_type(-1) {}
   // Payload type used for ULPFEC packets.
   int ulpfec_payload_type;
 
@@ -98,7 +105,8 @@ struct FecConfig {
 
 // Settings for RTP retransmission payload format, see RFC 4588 for details.
 struct RtxConfig {
-  // SSRC to use for the RTX stream, set to '0' for a random generated SSRC.
+  RtxConfig() : ssrc(0), rtx_payload_type(0), video_payload_type(0) {}
+  // SSRC to use for the RTX stream.
   uint32_t ssrc;
 
   // Payload type to use for the RTX stream.
@@ -110,6 +118,7 @@ struct RtxConfig {
 
 // RTP header extension to use for the video stream, see RFC 5285.
 struct RtpExtension {
+  RtpExtension() : id(0) {}
   // TODO(mflodman) Add API to query supported extensions.
   std::string name;
   int id;

webrtc/video_engine/new_include/video_receive_stream.h

@@ -41,6 +41,7 @@ struct ReceiveStatistics {
 
 // Receive stream specific RTP settings.
 struct RtpReceiveConfig {
+  RtpReceiveConfig() : ssrc(0), nack(NULL), fec(NULL) {}
   // TODO(mflodman) Do we require a set ssrc? What happens if the ssrc changes?
   uint32_t ssrc;
 
@@ -60,6 +61,8 @@ struct RtpReceiveConfig {
 // TODO(mflodman) Move all these settings to VideoDecoder and move the
 // declaration to common_types.h.
 struct ExternalVideoDecoder {
+  ExternalVideoDecoder()
+      : decoder(NULL), payload_type(0), renderer(false), expected_delay_ms(0) {}
   // The actual decoder.
   VideoDecoder* decoder;
 
@@ -78,6 +81,13 @@ struct ExternalVideoDecoder {
 };
 
 struct VideoReceiveStreamConfig {
+  VideoReceiveStreamConfig()
+      : renderer(NULL),
+        render_delay_ms(0),
+        audio_channel_id(0),
+        pre_decode_callback(NULL),
+        post_decode_callback(NULL),
+        target_delay_ms(0) {}
   // Codecs the receive stream
   std::vector<VideoCodec> codecs;
 

webrtc/video_engine/new_include/video_send_stream.h

@@ -27,6 +27,17 @@ namespace newapi {
 struct SendStreamState;
 
 struct SendStatistics {
+  SendStatistics()
+      : input_frame_rate(0),
+        encode_frame(0),
+        key_frames(0),
+        delta_frames(0),
+        video_packets(0),
+        retransmitted_packets(0),
+        fec_packets(0),
+        padding_packets(0),
+        send_bitrate_bps(0),
+        delay_ms(0) {}
   RtpStatistics rtp;
   int input_frame_rate;
   int encode_frame;
@@ -53,6 +64,12 @@ class VideoSendStreamInput {
 };
 
 struct RtpSendConfig {
+  RtpSendConfig()
+      : mode(kRtcpReducedSize),
+        max_packet_size(0),
+        nack(NULL),
+        fec(NULL),
+        rtx(NULL) {}
   RtcpMode mode;
 
   std::vector<uint32_t> ssrcs;
@@ -77,6 +94,15 @@ struct RtpSendConfig {
 };
 
 struct VideoSendStreamConfig {
+  VideoSendStreamConfig()
+      : pre_encode_callback(NULL),
+        encoded_callback(NULL),
+        local_renderer(NULL),
+        render_delay_ms(0),
+        encoder(NULL),
+        internal_source(false),
+        target_delay_ms(0),
+        start_state(NULL) {}
   VideoCodec codec;
 
   RtpSendConfig rtp;