Made send pad data generic (audio and video)

Review URL: http://webrtc-codereview.appspot.com/343001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1346 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/rtp_rtcp/source/rtp_sender.cc

@@ -759,6 +759,63 @@ RTPSender::SendOutgoingData(const FrameType frameType,
     }
 }
 
+WebRtc_Word32 RTPSender::SendPadData(WebRtc_Word8 payload_type,
+                                     WebRtc_UWord32 capture_timestamp,
+                                     WebRtc_Word32 bytes) {
+  // Drop this packet if we're not sending media packets
+  if (!_sendingMedia) {
+    return 0;
+  }
+  // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
+  int max_length = 224;
+  WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
+
+  for (; bytes > 0; bytes -= max_length) {
+    WebRtc_Word32 header_length;
+    {
+      // Correct seq num, timestamp and payload type.
+      header_length = BuildRTPheader(data_buffer,
+                                     payload_type,
+                                     false,  // No markerbit.
+                                     capture_timestamp,
+                                     true,  // Timestamp provided.
+                                     true);  // Increment sequence number.
+    }
+    data_buffer[0] |= 0x20;  // Set padding bit.
+    WebRtc_Word32* data =
+        reinterpret_cast<WebRtc_Word32*>(&(data_buffer[header_length]));
+
+    int padding_bytes_in_packet = max_length;
+    if (bytes < max_length) {
+      padding_bytes_in_packet = (bytes + 16) & 0xffe0;  // Keep our modulus 32.
+    }
+    if (padding_bytes_in_packet < 32) {
+       // Sanity don't send empty packets.
+       break;
+    }
+    // Fill data buffer with random data.
+    for(int j = 0; j < (padding_bytes_in_packet >> 2); j++) {
+      data[j] = rand();
+    }
+    // Set number of padding bytes in the last byte of the packet.
+    data_buffer[header_length + padding_bytes_in_packet - 1] =
+        padding_bytes_in_packet;
+    // Send the packet
+    if (0 > SendToNetwork(data_buffer,
+                          padding_bytes_in_packet,
+                          header_length,
+                          kDontRetransmit)) {
+      // Error sending the packet.
+      break;
+    }
+  }
+  if (bytes > 31) {  // 31 due to our modulus 32.
+    // We did not manage to send all bytes.
+    return -1;
+  }
+  return 0;
+}
+
 WebRtc_Word32
 RTPSender::SetStorePacketsStatus(const bool enable, const WebRtc_UWord16 numberToStore)
 {

src/modules/rtp_rtcp/source/rtp_sender.h

@@ -147,6 +147,9 @@ public:
                      VideoCodecInformation* codecInfo = NULL,
                      const RTPVideoTypeHeader* rtpTypeHdr = NULL);
 
+    WebRtc_Word32 SendPadData(WebRtc_Word8 payload_type,
+                              WebRtc_UWord32 capture_timestamp,
+                              WebRtc_Word32 bytes);
     /*
     * RTP header extension
     */

src/modules/rtp_rtcp/source/rtp_sender_video.cc

@@ -549,55 +549,6 @@ RTPSenderVideo::SendGeneric(const WebRtc_Word8 payloadType,
     return 0;
 }
 
-void RTPSenderVideo::SendPadData(WebRtc_Word8 payload_type,
-                                 WebRtc_UWord32 capture_timestamp,
-                                 WebRtc_Word32 bytes) {
-  // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
-  int max_length = 224;
-  WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
-
-  for (; bytes > 0; bytes -= max_length) {
-    WebRtc_Word32 header_length;
-    {
-      CriticalSectionScoped cs(_sendVideoCritsect);
-
-      // Correct seq num, timestamp and payload type.
-      header_length = _rtpSender.BuildRTPheader(
-          data_buffer,
-          payload_type,
-          false,  // No markerbit.
-          capture_timestamp,
-          true,  // Timestamp provided.
-          true);  // Increment sequence number.
-
-    }
-    data_buffer[0] |= 0x20;  // Set padding bit.
-    WebRtc_Word32* data =
-        reinterpret_cast<WebRtc_Word32*>(&(data_buffer[header_length]));
-
-    int padding_bytes_in_packet = max_length;
-    if (bytes < max_length) {
-      padding_bytes_in_packet = (bytes + 16) & 0xffe0;  // Keep our modulus 32.
-    }
-    if (padding_bytes_in_packet < 32) {
-       // Sanity don't send empty packets.
-       return;
-    }
-    // Fill data buffer with random data.
-    for(int j = 0; j < (padding_bytes_in_packet >> 2); j++) {
-      data[j] = rand();
-    }
-    // Set number of padding bytes in the last byte of the packet.
-    data_buffer[header_length + padding_bytes_in_packet - 1] =
-        padding_bytes_in_packet;
-    // Send the packet
-    _rtpSender.SendToNetwork(data_buffer,
-                             padding_bytes_in_packet,
-                             header_length,
-                             kDontRetransmit);
-  }
-}
-
 WebRtc_Word32
 RTPSenderVideo::SendMPEG4(const FrameType frameType,
                           const WebRtc_Word8 payloadType,

src/modules/rtp_rtcp/source/rtp_sender_video.h

@@ -68,10 +68,6 @@ public:
 
     WebRtc_UWord32 MaxConfiguredBitrateVideo() const;
 
-    void SendPadData(WebRtc_Word8 payload_type,
-                     WebRtc_UWord32 capture_timestamp,
-                     WebRtc_Word32 bytes);
-
     // FEC
     WebRtc_Word32 SetGenericFECStatus(const bool enable,
                                     const WebRtc_UWord8 payloadTypeRED,