| 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;
|
|
|
|
|