Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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.
  */
 
@@ skipping 2212 matching lines @@
   }
   if (duration < kMinTelephoneEventDuration ||
       duration > kMaxTelephoneEventDuration) {
     LOG(LS_WARNING) << "DTMF event duration " << duration << " out of range.";
     return false;
   }
   return it->second->SendTelephoneEvent(*dtmf_payload_type_, event, duration);
 }
 
 void WebRtcVoiceMediaChannel::OnPacketReceived(
-    rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
+    rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
 
   uint32_t ssrc = 0;
-  if (!GetRtpSsrc(packet->data(), packet->size(), &ssrc)) {
+  if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
     return;
   }
 
   // If we don't have a default channel, and the SSRC is unknown, create a
   // default channel.
   if (default_recv_ssrc_ == -1 && GetReceiveChannelId(ssrc) == -1) {
     StreamParams sp;
     sp.ssrcs.push_back(ssrc);
     LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
     if (!AddRecvStream(sp)) {
       LOG(LS_WARNING) << "Could not create default receive stream.";
       return;
     }
     default_recv_ssrc_ = ssrc;
     SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
     if (default_sink_) {
       std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
           new ProxySink(default_sink_.get()));
       SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
     }
   }
 
   // Forward packet to Call. If the SSRC is unknown we'll return after this.
   const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                               packet_time.not_before);
   webrtc::PacketReceiver::DeliveryStatus delivery_result =
       call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
-          reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
-          webrtc_packet_time);
+          packet->cdata(), packet->size(), webrtc_packet_time);
   if (webrtc::PacketReceiver::DELIVERY_OK != delivery_result) {
     // If the SSRC is unknown here, route it to the default channel, if we have
     // one. See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
     if (default_recv_ssrc_ == -1) {
       return;
     } else {
       ssrc = default_recv_ssrc_;
     }
   }
 
   // Find the channel to send this packet to. It must exist since webrtc::Call
   // was able to demux the packet.
   int channel = GetReceiveChannelId(ssrc);
   RTC_DCHECK(channel != -1);
 
   // Pass it off to the decoder.
   engine()->voe()->network()->ReceivedRTPPacket(
-      channel, packet->data(), packet->size(), webrtc_packet_time);
+      channel, packet->cdata(), packet->size(), webrtc_packet_time);
 }
 
 void WebRtcVoiceMediaChannel::OnRtcpReceived(
-    rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
+    rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
 
   // Forward packet to Call as well.
   const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                               packet_time.not_before);
   call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
-      reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
-      webrtc_packet_time);
+      packet->cdata(), packet->size(), webrtc_packet_time);
 
   // Sending channels need all RTCP packets with feedback information.
   // Even sender reports can contain attached report blocks.
   // Receiving channels need sender reports in order to create
   // correct receiver reports.
   int type = 0;
-  if (!GetRtcpType(packet->data(), packet->size(), &type)) {
+  if (!GetRtcpType(packet->cdata(), packet->size(), &type)) {
     LOG(LS_WARNING) << "Failed to parse type from received RTCP packet";
     return;
   }
 
   // If it is a sender report, find the receive channel that is listening.
   if (type == kRtcpTypeSR) {
     uint32_t ssrc = 0;
-    if (!GetRtcpSsrc(packet->data(), packet->size(), &ssrc)) {
+    if (!GetRtcpSsrc(packet->cdata(), packet->size(), &ssrc)) {
       return;
     }
     int recv_channel_id = GetReceiveChannelId(ssrc);
     if (recv_channel_id != -1) {
       engine()->voe()->network()->ReceivedRTCPPacket(
-          recv_channel_id, packet->data(), packet->size());
+          recv_channel_id, packet->cdata(), packet->size());
     }
   }
 
   // SR may continue RR and any RR entry may correspond to any one of the send
   // channels. So all RTCP packets must be forwarded all send channels. VoE
   // will filter out RR internally.
   for (const auto& ch : send_streams_) {
     engine()->voe()->network()->ReceivedRTCPPacket(
-        ch.second->channel(), packet->data(), packet->size());
+        ch.second->channel(), packet->cdata(), packet->size());
   }
 }
 
 bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   int channel = GetSendChannelId(ssrc);
   if (channel == -1) {
     LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
     return false;
   }
@@ skipping 198 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE