Index: webrtc/media/engine/webrtcvoiceengine_unittest.cc |
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc |
index 4fd5918efd1b90b072a819b815e4e354926f9f94..b538c6ff752f2916e05a3d869279be57eb7c106e 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc |
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc |
@@ -113,7 +113,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test { |
EXPECT_FALSE(call_.GetAudioSendStream(kSsrc1)); |
} |
void DeliverPacket(const void* data, int len) { |
- rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len); |
+ rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len); |
channel_->OnPacketReceived(&packet, rtc::PacketTime()); |
} |
void TearDown() override { |
@@ -3081,14 +3081,14 @@ TEST_F(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) { |
TEST_F(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) { |
// Test that packets are forwarded to the Call when configured accordingly. |
const uint32_t kAudioSsrc = 1; |
- rtc::Buffer kPcmuPacket(kPcmuFrame, sizeof(kPcmuFrame)); |
+ rtc::CopyOnWriteBuffer 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::Buffer kRtcpPacket(kRtcp, sizeof(kRtcp)); |
+ rtc::CopyOnWriteBuffer kRtcpPacket(kRtcp, sizeof(kRtcp)); |
EXPECT_TRUE(SetupEngineWithSendStream()); |
cricket::WebRtcVoiceMediaChannel* media_channel = |