Index: webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
index b958e27a64532442fd0588e662017b71dfd3ae9f..6dbe53f20d4c6be5fbe7c69cdba1ff87763ea681 100644 |
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
@@ -29,6 +29,7 @@ |
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" |
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h" |
#include "webrtc/modules/include/module_common_types.h" |
+#include "webrtc/modules/utility/include/audio_frame_operations.h" |
#include "webrtc/test/gtest.h" |
#include "webrtc/test/testsupport/fileutils.h" |
#include "webrtc/typedefs.h" |
@@ -957,7 +958,7 @@ class NetEqBgnTest : public NetEqDecodingTest { |
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>( |
payload, enc_len_bytes), |
receive_timestamp)); |
- output.Reset(); |
+ AudioFrameOperations::Reset(&output); |
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted)); |
ASSERT_EQ(1u, output.num_channels_); |
ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_); |
@@ -969,7 +970,7 @@ class NetEqBgnTest : public NetEqDecodingTest { |
receive_timestamp += expected_samples_per_channel; |
} |
- output.Reset(); |
+ AudioFrameOperations::Reset(&output); |
// Get audio without inserting packets, expecting PLC and PLC-to-CNG. Pull |
// one frame without checking speech-type. This is the first frame pulled |
@@ -987,7 +988,7 @@ class NetEqBgnTest : public NetEqDecodingTest { |
const int kNumPlcToCngTestFrames = 20; |
bool plc_to_cng = false; |
for (int n = 0; n < kFadingThreshold + kNumPlcToCngTestFrames; ++n) { |
- output.Reset(); |
+ AudioFrameOperations::Reset(&output); |
memset(output.data_, 1, sizeof(output.data_)); // Set to non-zero. |
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted)); |
ASSERT_FALSE(muted); |