Removing unused Opus wrapper API: WebRTCOpus_DecodePlc.
Bug: None
Change-Id: I5b613b4c13ec5f6ad13d8430043d006f6d83c11f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158671
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29664}
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index 7f0bdd2..10644e2 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -299,8 +299,9 @@
opus_mono_decoder_, bitstream, bitstream_len_byte,
&out_audio[decoded_samples * channels], &audio_type);
} else {
- decoded_samples += WebRtcOpus_DecodePlc(
- opus_mono_decoder_, &out_audio[decoded_samples * channels], 1);
+ decoded_samples += WebRtcOpus_Decode(
+ opus_mono_decoder_, NULL, 0,
+ &out_audio[decoded_samples * channels], &audio_type);
}
} else {
if (!lost_packet) {
@@ -308,9 +309,9 @@
opus_stereo_decoder_, bitstream, bitstream_len_byte,
&out_audio[decoded_samples * channels], &audio_type);
} else {
- decoded_samples +=
- WebRtcOpus_DecodePlc(opus_stereo_decoder_,
- &out_audio[decoded_samples * channels], 1);
+ decoded_samples += WebRtcOpus_Decode(
+ opus_stereo_decoder_, NULL, 0,
+ &out_audio[decoded_samples * channels], &audio_type);
}
}