Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.

webrtc/media/base/rtpdataengine.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/media/base/rtpdataengine.h"
 
-#include "webrtc/base/buffer.h"
+#include "webrtc/base/copyonwritebuffer.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/ratelimiter.h"
 #include "webrtc/base/timing.h"
 #include "webrtc/media/base/codec.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/rtputils.h"
 #include "webrtc/media/base/streamparams.h"
 
 namespace cricket {
@@ skipping 174 matching lines @@
                << "' with ssrc=" << stream.first_ssrc();
   return true;
 }
 
 bool RtpDataMediaChannel::RemoveRecvStream(uint32_t ssrc) {
   RemoveStreamBySsrc(&recv_streams_, ssrc);
   return true;
 }
 
 void RtpDataMediaChannel::OnPacketReceived(
-    rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
+    rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
   RtpHeader header;
-  if (!GetRtpHeader(packet->data(), packet->size(), &header)) {
+  if (!GetRtpHeader(packet->cdata(), packet->size(), &header)) {
     // Don't want to log for every corrupt packet.
     // LOG(LS_WARNING) << "Could not read rtp header from packet of length "
     //                 << packet->length() << ".";
     return;
   }
 
   size_t header_length;
-  if (!GetRtpHeaderLen(packet->data(), packet->size(), &header_length)) {
+  if (!GetRtpHeaderLen(packet->cdata(), packet->size(), &header_length)) {
     // Don't want to log for every corrupt packet.
     // LOG(LS_WARNING) << "Could not read rtp header"
     //                 << length from packet of length "
     //                 << packet->length() << ".";
     return;
   }
   const char* data =
-      packet->data<char>() + header_length + sizeof(kReservedSpace);
+      packet->cdata<char>() + header_length + sizeof(kReservedSpace);
   size_t data_len = packet->size() - header_length - sizeof(kReservedSpace);
 
   if (!receiving_) {
     LOG(LS_WARNING) << "Not receiving packet "
                     << header.ssrc << ":" << header.seq_num
                     << " before SetReceive(true) called.";
     return;
   }
 
   DataCodec codec;
@@ skipping 32 matching lines @@
   if (bps <= 0) {
     bps = kDataMaxBandwidth;
   }
   send_limiter_.reset(new rtc::RateLimiter(bps / 8, 1.0));
   LOG(LS_INFO) << "RtpDataMediaChannel::SetSendBandwidth to " << bps << "bps.";
   return true;
 }
 
 bool RtpDataMediaChannel::SendData(
     const SendDataParams& params,
-    const rtc::Buffer& payload,
+    const rtc::CopyOnWriteBuffer& payload,
     SendDataResult* result) {
   if (result) {
     // If we return true, we'll set this to SDR_SUCCESS.
     *result = SDR_ERROR;
   }
   if (!sending_) {
     LOG(LS_WARNING) << "Not sending packet with ssrc=" << params.ssrc
                     << " len=" << payload.size() << " before SetSend(true).";
     return false;
   }
@@ skipping 32 matching lines @@
                     << "/" << send_limiter_->max_per_period();
     return false;
   }
 
   RtpHeader header;
   header.payload_type = found_codec.id;
   header.ssrc = params.ssrc;
   rtp_clock_by_send_ssrc_[header.ssrc]->Tick(
       now, &header.seq_num, &header.timestamp);
 
-  rtc::Buffer packet(kMinRtpPacketLen, packet_len);
+  rtc::CopyOnWriteBuffer packet(kMinRtpPacketLen, packet_len);
   if (!SetRtpHeader(packet.data(), packet.size(), header)) {
     return false;
   }
   packet.AppendData(kReservedSpace);
   packet.AppendData(payload);
 
   LOG(LS_VERBOSE) << "Sent RTP data packet: "
                   << " stream=" << found_stream->id << " ssrc=" << header.ssrc
                   << ", seqnum=" << header.seq_num
                   << ", timestamp=" << header.timestamp
                   << ", len=" << payload.size();
 
   MediaChannel::SendPacket(&packet, rtc::PacketOptions());
   send_limiter_->Use(packet_len, now);
   if (result) {
     *result = SDR_SUCCESS;
   }
   return true;
 }
 
 }  // namespace cricket