| Index: webrtc/media/engine/webrtcvoiceengine_unittest.cc
|
| diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
|
| index aab140c6f47ad964ac09ac17e1ccc6218a553253..1fe9fc1ab6001f3bbbd1ff16b2b1544c1a4e9b7c 100644
|
| --- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
|
| +++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
|
| @@ -113,7 +113,7 @@
|
| EXPECT_FALSE(call_.GetAudioSendStream(kSsrc1));
|
| }
|
| void DeliverPacket(const void* data, int len) {
|
| - rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len);
|
| + rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len);
|
| channel_->OnPacketReceived(&packet, rtc::PacketTime());
|
| }
|
| void TearDown() override {
|
| @@ -3060,14 +3060,14 @@
|
| TEST_F(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) {
|
| // Test that packets are forwarded to the Call when configured accordingly.
|
| const uint32_t kAudioSsrc = 1;
|
| - rtc::CopyOnWriteBuffer kPcmuPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
| + rtc::Buffer kPcmuPacket(kPcmuFrame, sizeof(kPcmuFrame));
|
| static const unsigned char kRtcp[] = {
|
| 0x80, 0xc9, 0x00, 0x01, 0x00, 0x00, 0x00, 0x02,
|
| 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00,
|
| 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
|
| 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
|
| };
|
| - rtc::CopyOnWriteBuffer kRtcpPacket(kRtcp, sizeof(kRtcp));
|
| + rtc::Buffer kRtcpPacket(kRtcp, sizeof(kRtcp));
|
|
|
| EXPECT_TRUE(SetupEngineWithSendStream());
|
| cricket::WebRtcVoiceMediaChannel* media_channel =
|
|
|