| 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.
|
|
|