Index: webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h |
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h |
index ac3c5b433425c96df72ac83fed0a92d9df3d59b8..ecfb1d70e88aa0137939a99d4cd128a956424cfa 100644 |
--- a/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h |
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h |
@@ -25,8 +25,7 @@ class AudioDecoderG722 final : public AudioDecoder { |
bool HasDecodePlc() const override; |
void Reset() override; |
std::vector<ParseResult> ParsePayload(rtc::Buffer* payload, |
- uint32_t timestamp, |
- bool is_primary) override; |
+ uint32_t timestamp) override; |
int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override; |
int SampleRateHz() const override; |
size_t Channels() const override; |
@@ -49,8 +48,7 @@ class AudioDecoderG722Stereo final : public AudioDecoder { |
~AudioDecoderG722Stereo() override; |
void Reset() override; |
std::vector<ParseResult> ParsePayload(rtc::Buffer* payload, |
- uint32_t timestamp, |
- bool is_primary) override; |
+ uint32_t timestamp) override; |
int SampleRateHz() const override; |
size_t Channels() const override; |