Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.

webrtc/media/base/videoengine_unittest.h

 /*
  *  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.
  */
 
@@ skipping 241 matching lines @@
   }
   int NumRtpPackets() {
     return network_interface_.NumRtpPackets();
   }
   int NumRtpPackets(uint32_t ssrc) {
     return network_interface_.NumRtpPackets(ssrc);
   }
   int NumSentSsrcs() {
     return network_interface_.NumSentSsrcs();
   }
-  const rtc::Buffer* GetRtpPacket(int index) {
+  const rtc::CopyOnWriteBuffer* GetRtpPacket(int index) {
     return network_interface_.GetRtpPacket(index);
   }
   int NumRtcpPackets() {
     return network_interface_.NumRtcpPackets();
   }
-  const rtc::Buffer* GetRtcpPacket(int index) {
+  const rtc::CopyOnWriteBuffer* GetRtcpPacket(int index) {
     return network_interface_.GetRtcpPacket(index);
   }
-  static int GetPayloadType(const rtc::Buffer* p) {
+  static int GetPayloadType(const rtc::CopyOnWriteBuffer* p) {
     int pt = -1;
     ParseRtpPacket(p, NULL, &pt, NULL, NULL, NULL, NULL);
     return pt;
   }
-  static bool ParseRtpPacket(const rtc::Buffer* p,
+  static bool ParseRtpPacket(const rtc::CopyOnWriteBuffer* p,
                              bool* x,
                              int* pt,
                              int* seqnum,
                              uint32_t* tstamp,
                              uint32_t* ssrc,
                              std::string* payload) {
-    rtc::ByteBuffer buf(*p);
+    // TODO(jbauch): avoid copying the buffer data into the ByteBuffer
+    rtc::ByteBuffer buf(p->data<char>(), p->size());
     uint8_t u08 = 0;
     uint16_t u16 = 0;
     uint32_t u32 = 0;
 
     // Read X and CC fields.
     if (!buf.ReadUInt8(&u08)) return false;
     bool extension = ((u08 & 0x10) != 0);
     uint8_t cc = (u08 & 0x0F);
     if (x) *x = extension;
 
@@ skipping 38 matching lines @@
     }
     return true;
   }
 
   // Parse all RTCP packet, from start_index to stop_index, and count how many
   // FIR (PT=206 and FMT=4 according to RFC 5104). If successful, set the count
   // and return true.
   bool CountRtcpFir(int start_index, int stop_index, int* fir_count) {
     int count = 0;
     for (int i = start_index; i < stop_index; ++i) {
-      std::unique_ptr<const rtc::Buffer> p(GetRtcpPacket(i));
-      rtc::ByteBuffer buf(*p);
+      std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtcpPacket(i));
+      // TODO(jbauch): avoid copying the buffer data into the ByteBuffer
+      rtc::ByteBuffer buf(p->data<char>(), p->size());
       size_t total_len = 0;
       // The packet may be a compound RTCP packet.
       while (total_len < p->size()) {
         // Read FMT, type and length.
         uint8_t fmt = 0;
         uint8_t type = 0;
         uint16_t length = 0;
         if (!buf.ReadUInt8(&fmt)) return false;
         fmt &= 0x1F;
         if (!buf.ReadUInt8(&type)) return false;
@@ skipping 43 matching lines @@
     EXPECT_TRUE(SetSend(true));
     EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
     EXPECT_EQ(64 * 1024, network_interface_.recvbuf_size());
   }
   // Tests that we can send frames and the right payload type is used.
   void Send(const cricket::VideoCodec& codec) {
     EXPECT_TRUE(SetOneCodec(codec));
     EXPECT_TRUE(SetSend(true));
     EXPECT_TRUE(SendFrame());
     EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
-    std::unique_ptr<const rtc::Buffer> p(GetRtpPacket(0));
+    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
     EXPECT_EQ(codec.id, GetPayloadType(p.get()));
   }
   // Tests that we can send and receive frames.
   void SendAndReceive(const cricket::VideoCodec& codec) {
     EXPECT_TRUE(SetOneCodec(codec));
     EXPECT_TRUE(SetSend(true));
     EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
     EXPECT_EQ(0, renderer_.num_rendered_frames());
     EXPECT_TRUE(SendFrame());
     EXPECT_FRAME_WAIT(1, codec.width, codec.height, kTimeout);
-    std::unique_ptr<const rtc::Buffer> p(GetRtpPacket(0));
+    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
     EXPECT_EQ(codec.id, GetPayloadType(p.get()));
   }
   void SendReceiveManyAndGetStats(const cricket::VideoCodec& codec,
                                   int duration_sec, int fps) {
     EXPECT_TRUE(SetOneCodec(codec));
     EXPECT_TRUE(SetSend(true));
     EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
     EXPECT_EQ(0, renderer_.num_rendered_frames());
     for (int i = 0; i < duration_sec; ++i) {
       for (int frame = 1; frame <= fps; ++frame) {
         EXPECT_TRUE(WaitAndSendFrame(1000 / fps));
         EXPECT_FRAME_WAIT(frame + i * fps, codec.width, codec.height, kTimeout);
       }
     }
-    std::unique_ptr<const rtc::Buffer> p(GetRtpPacket(0));
+    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
     EXPECT_EQ(codec.id, GetPayloadType(p.get()));
   }
 
   // Test that stats work properly for a 1-1 call.
   void GetStats() {
     const int kDurationSec = 3;
     const int kFps = 10;
     SendReceiveManyAndGetStats(DefaultCodec(), kDurationSec, kFps);
 
     cricket::VideoMediaInfo info;
@@ skipping 172 matching lines @@
     parameters.max_bandwidth_bps = 128 * 1024;
     EXPECT_TRUE(channel_->SetSendParameters(parameters));
   }
   // Test that we can set the SSRC for the default send source.
   void SetSendSsrc() {
     EXPECT_TRUE(SetDefaultCodec());
     EXPECT_TRUE(SetSend(true));
     EXPECT_TRUE(SendFrame());
     EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
     uint32_t ssrc = 0;
-    std::unique_ptr<const rtc::Buffer> p(GetRtpPacket(0));
+    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
     ParseRtpPacket(p.get(), NULL, NULL, NULL, NULL, &ssrc, NULL);
     EXPECT_EQ(kSsrc, ssrc);
     // Packets are being paced out, so these can mismatch between the first and
     // second call to NumRtpPackets until pending packets are paced out.
     EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(ssrc), kTimeout);
     EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(ssrc), kTimeout);
     EXPECT_EQ(1, NumSentSsrcs());
     EXPECT_EQ(0, NumRtpPackets(kSsrc - 1));
     EXPECT_EQ(0, NumRtpBytes(kSsrc - 1));
   }
   // Test that we can set the SSRC even after codecs are set.
   void SetSendSsrcAfterSetCodecs() {
     // Remove stream added in Setup.
     EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
     EXPECT_TRUE(SetDefaultCodec());
     EXPECT_TRUE(channel_->AddSendStream(
         cricket::StreamParams::CreateLegacy(999)));
     EXPECT_TRUE(channel_->SetCapturer(999u, video_capturer_.get()));
     EXPECT_TRUE(SetSend(true));
     EXPECT_TRUE(WaitAndSendFrame(0));
     EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
     uint32_t ssrc = 0;
-    std::unique_ptr<const rtc::Buffer> p(GetRtpPacket(0));
+    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
     ParseRtpPacket(p.get(), NULL, NULL, NULL, NULL, &ssrc, NULL);
     EXPECT_EQ(999u, ssrc);
     // Packets are being paced out, so these can mismatch between the first and
     // second call to NumRtpPackets until pending packets are paced out.
     EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(ssrc), kTimeout);
     EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(ssrc), kTimeout);
     EXPECT_EQ(1, NumSentSsrcs());
     EXPECT_EQ(0, NumRtpPackets(kSsrc));
     EXPECT_EQ(0, NumRtpBytes(kSsrc));
   }
   // Test that we can set the default video renderer before and after
   // media is received.
   void SetSink() {
     uint8_t data1[] = {
         0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
 
-    rtc::Buffer packet1(data1, sizeof(data1));
+    rtc::CopyOnWriteBuffer packet1(data1, sizeof(data1));
     rtc::SetBE32(packet1.data() + 8, kSsrc);
     channel_->SetSink(kDefaultReceiveSsrc, NULL);
     EXPECT_TRUE(SetDefaultCodec());
     EXPECT_TRUE(SetSend(true));
     EXPECT_EQ(0, renderer_.num_rendered_frames());
     channel_->OnPacketReceived(&packet1, rtc::PacketTime());
     EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
     EXPECT_TRUE(SendFrame());
     EXPECT_FRAME_WAIT(1, DefaultCodec().width, DefaultCodec().height, kTimeout);
   }
@@ skipping 12 matching lines @@
   // Tests setting up and configuring a send stream.
   void AddRemoveSendStreams() {
     EXPECT_TRUE(SetOneCodec(DefaultCodec()));
     EXPECT_TRUE(SetSend(true));
     EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
     EXPECT_TRUE(SendFrame());
     EXPECT_FRAME_WAIT(1, DefaultCodec().width, DefaultCodec().height, kTimeout);
     EXPECT_GT(NumRtpPackets(), 0);
     uint32_t ssrc = 0;
     size_t last_packet = NumRtpPackets() - 1;
-    std::unique_ptr<const rtc::Buffer>
+    std::unique_ptr<const rtc::CopyOnWriteBuffer>
         p(GetRtpPacket(static_cast<int>(last_packet)));
     ParseRtpPacket(p.get(), NULL, NULL, NULL, NULL, &ssrc, NULL);
     EXPECT_EQ(kSsrc, ssrc);
 
     // Remove the send stream that was added during Setup.
     EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
     int rtp_packets = NumRtpPackets();
 
     EXPECT_TRUE(channel_->AddSendStream(
         cricket::StreamParams::CreateLegacy(789u)));
@@ skipping 29 matching lines @@
     std::vector<uint32_t> ssrcs;
     ssrcs.push_back(1);
     ssrcs.push_back(2);
     network_interface_.SetConferenceMode(true, ssrcs);
     EXPECT_TRUE(SendFrame());
     EXPECT_FRAME_ON_RENDERER_WAIT(
         renderer1, 1, DefaultCodec().width, DefaultCodec().height, kTimeout);
     EXPECT_FRAME_ON_RENDERER_WAIT(
         renderer2, 1, DefaultCodec().width, DefaultCodec().height, kTimeout);
 
-    std::unique_ptr<const rtc::Buffer> p(GetRtpPacket(0));
+    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
     EXPECT_EQ(DefaultCodec().id, GetPayloadType(p.get()));
     EXPECT_EQ(DefaultCodec().width, renderer1.width());
     EXPECT_EQ(DefaultCodec().height, renderer1.height());
     EXPECT_EQ(DefaultCodec().width, renderer2.width());
     EXPECT_EQ(DefaultCodec().height, renderer2.height());
     EXPECT_TRUE(channel_->RemoveRecvStream(2));
     EXPECT_TRUE(channel_->RemoveRecvStream(1));
   }
 
   // Tests that we can add and remove capturers and frames are sent out properly
@@ skipping 244 matching lines @@
     cricket::VideoCodec codec(DefaultCodec());
     int frame_count = 0;
     // The capturer runs at 30 fps. The channel requires 30 fps.
     EXPECT_TRUE(SetOneCodec(codec));
     EXPECT_TRUE(SetSend(true));
     EXPECT_EQ(frame_count, renderer_.num_rendered_frames());
     EXPECT_TRUE(WaitAndSendFrame(0));  // Should be rendered.
     EXPECT_TRUE(WaitAndSendFrame(30));  // Should be rendered.
     frame_count += 2;
     EXPECT_FRAME_WAIT(frame_count, codec.width, codec.height, kTimeout);
-    std::unique_ptr<const rtc::Buffer> p(GetRtpPacket(0));
+    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
     EXPECT_EQ(codec.id, GetPayloadType(p.get()));
 
     // The channel requires 15 fps.
     codec.framerate = 15;
     EXPECT_TRUE(SetOneCodec(codec));
     EXPECT_TRUE(WaitAndSendFrame(0));  // Should be rendered.
     EXPECT_TRUE(WaitAndSendFrame(30));  // Should be dropped.
     EXPECT_TRUE(WaitAndSendFrame(30));  // Should be rendered.
     frame_count += 2;
     EXPECT_EQ_WAIT(frame_count, renderer_.num_rendered_frames(), kTimeout);
@@ skipping 161 matching lines @@
   std::unique_ptr<C> channel_;
   cricket::FakeNetworkInterface network_interface_;
   cricket::FakeVideoRenderer renderer_;
   cricket::VideoMediaChannel::Error media_error_;
 
   // Used by test cases where 2 streams are run on the same channel.
   cricket::FakeVideoRenderer renderer2_;
 };
 
 #endif  // WEBRTC_MEDIA_BASE_VIDEOENGINE_UNITTEST_H_  NOLINT