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 892eb37c3f6423424b153c377e57e8b1992e0f6b..cee4aafb434c642383227e13c571645dad336c66 100644 |
--- a/webrtc/modules/audio_coding/test/opus_test.cc |
+++ b/webrtc/modules/audio_coding/test/opus_test.cc |
@@ -263,7 +263,7 @@ 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, |
@@ -348,7 +348,7 @@ 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. |