Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.

webrtc/media/engine/webrtcvideoengine2.cc

 /*
  *  Copyright (c) 2014 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/engine/webrtcvideoengine2.h"
 
 #include <algorithm>
 #include <set>
 #include <string>
 
-#include "webrtc/base/buffer.h"
+#include "webrtc/base/copyonwritebuffer.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call.h"
 #include "webrtc/media/base/videocapturer.h"
 #include "webrtc/media/base/videorenderer.h"
 #include "webrtc/media/engine/constants.h"
 #include "webrtc/media/engine/simulcast.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
@@ skipping 1252 matching lines @@
       return false;
     }
     if (!kv->second->SetCapturer(capturer)) {
       return false;
     }
   }
   return true;
 }
 
 void WebRtcVideoChannel2::OnPacketReceived(
-    rtc::Buffer* packet,
+    rtc::CopyOnWriteBuffer* packet,
     const rtc::PacketTime& packet_time) {
   const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                               packet_time.not_before);
   const webrtc::PacketReceiver::DeliveryStatus delivery_result =
       call_->Receiver()->DeliverPacket(
           webrtc::MediaType::VIDEO,
-          reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+          packet->cdata(), packet->size(),
           webrtc_packet_time);
   switch (delivery_result) {
     case webrtc::PacketReceiver::DELIVERY_OK:
       return;
     case webrtc::PacketReceiver::DELIVERY_PACKET_ERROR:
       return;
     case webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC:
       break;
   }
 
   uint32_t ssrc = 0;
-  if (!GetRtpSsrc(packet->data(), packet->size(), &ssrc)) {
+  if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
     return;
   }
 
   int payload_type = 0;
-  if (!GetRtpPayloadType(packet->data(), packet->size(), &payload_type)) {
+  if (!GetRtpPayloadType(packet->cdata(), packet->size(), &payload_type)) {
     return;
   }
 
   // See if this payload_type is registered as one that usually gets its own
   // SSRC (RTX) or at least is safe to drop either way (ULPFEC). If it is, and
   // it wasn't handled above by DeliverPacket, that means we don't know what
   // stream it associates with, and we shouldn't ever create an implicit channel
   // for these.
   for (auto& codec : recv_codecs_) {
     if (payload_type == codec.rtx_payload_type ||
         payload_type == codec.fec.red_rtx_payload_type ||
         payload_type == codec.fec.ulpfec_payload_type) {
       return;
     }
   }
 
   switch (unsignalled_ssrc_handler_->OnUnsignalledSsrc(this, ssrc)) {
     case UnsignalledSsrcHandler::kDropPacket:
       return;
     case UnsignalledSsrcHandler::kDeliverPacket:
       break;
   }
 
   if (call_->Receiver()->DeliverPacket(
           webrtc::MediaType::VIDEO,
-          reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+          packet->cdata(), packet->size(),
           webrtc_packet_time) != webrtc::PacketReceiver::DELIVERY_OK) {
     LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
     return;
   }
 }
 
 void WebRtcVideoChannel2::OnRtcpReceived(
-    rtc::Buffer* packet,
+    rtc::CopyOnWriteBuffer* packet,
     const rtc::PacketTime& packet_time) {
   const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                               packet_time.not_before);
   // TODO(pbos): Check webrtc::PacketReceiver::DELIVERY_OK once we deliver
   // for both audio and video on the same path. Since BundleFilter doesn't
   // filter RTCP anymore incoming RTCP packets could've been going to audio (so
   // logging failures spam the log).
   call_->Receiver()->DeliverPacket(
       webrtc::MediaType::VIDEO,
-      reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+      packet->cdata(), packet->size(),
       webrtc_packet_time);
 }
 
 void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
   LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
   call_->SignalNetworkState(ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
 }
 
 bool WebRtcVideoChannel2::MuteStream(uint32_t ssrc, bool mute) {
   LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
@@ skipping 35 matching lines @@
   // due to lack of socket buffer space, although it's not yet clear what the
   // ideal value should be.
   MediaChannel::SetOption(NetworkInterface::ST_RTP,
                           rtc::Socket::OPT_SNDBUF,
                           kVideoRtpBufferSize);
 }
 
 bool WebRtcVideoChannel2::SendRtp(const uint8_t* data,
                                   size_t len,
                                   const webrtc::PacketOptions& options) {
-  rtc::Buffer packet(data, len, kMaxRtpPacketLen);
+  rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
   rtc::PacketOptions rtc_options;
   rtc_options.packet_id = options.packet_id;
   return MediaChannel::SendPacket(&packet, rtc_options);
 }
 
 bool WebRtcVideoChannel2::SendRtcp(const uint8_t* data, size_t len) {
-  rtc::Buffer packet(data, len, kMaxRtpPacketLen);
+  rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
   return MediaChannel::SendRtcp(&packet, rtc::PacketOptions());
 }
 
 void WebRtcVideoChannel2::StartAllSendStreams() {
   rtc::CritScope stream_lock(&stream_crit_);
   for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
            send_streams_.begin();
        it != send_streams_.end(); ++it) {
     it->second->Start();
   }
@@ skipping 1097 matching lines @@
         rtx_mapping[video_codecs[i].codec.id] !=
             fec_settings.red_payload_type) {
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket