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Unified Diff: webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h

Issue 1725143003: Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added more fixes for override hiding in AudioEncoder implementations. Created 4 years, 10 months ago
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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_;

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