Index: webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h |
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h |
index 8bae160476149c2227f1422c90304c03459a0fd5..43c95b80625e5d7949c04998e116884f338ab316 100644 |
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h |
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h |
@@ -77,7 +77,7 @@ class NetEqQualityTest : public ::testing::Test { |
// 2. save the bit stream to |payload| of |max_bytes| bytes in size, |
// 3. returns the length of the payload (in bytes), |
virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples, |
- uint8_t* payload, size_t max_bytes) = 0; |
+ rtc::Buffer* payload, size_t max_bytes) = 0; |
// PacketLost(...) determines weather a packet sent at an indicated time gets |
// lost or not. |
@@ -128,7 +128,7 @@ class NetEqQualityTest : public ::testing::Test { |
std::unique_ptr<LossModel> loss_model_; |
std::unique_ptr<int16_t[]> in_data_; |
- std::unique_ptr<uint8_t[]> payload_; |
+ rtc::Buffer payload_; |
std::unique_ptr<int16_t[]> out_data_; |
WebRtcRTPHeader rtp_header_; |