Index: webrtc/modules/audio_coding/test/opus_test.cc |
diff --git a/webrtc/modules/audio_coding/test/opus_test.cc b/webrtc/modules/audio_coding/test/opus_test.cc |
index a558f1c767eb89c5d934c6411de552f2e4f3aecd..9f5720b961fc9bb811225433956c2793a0bf5a1f 100644 |
--- a/webrtc/modules/audio_coding/test/opus_test.cc |
+++ b/webrtc/modules/audio_coding/test/opus_test.cc |
@@ -262,7 +262,7 @@ void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate, |
// If input audio is sampled at 32 kHz, resampling to 48 kHz is required. |
EXPECT_EQ(480, |
- resampler_.Resample10Msec(audio_frame.data_, |
+ resampler_.Resample10Msec(audio_frame.data(), |
audio_frame.sample_rate_hz_, |
48000, |
channels, |
@@ -347,7 +347,7 @@ void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate, |
// Write output speech to file. |
out_file_.Write10MsData( |
- audio_frame.data_, |
+ audio_frame.data(), |
audio_frame.samples_per_channel_ * audio_frame.num_channels_); |
// Write stand-alone speech to file. |