| 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 ccca73d1a19dc21a4ebd9ef79b20b0065ff69bea..ad39619d3c795fdf49a376241a1e5ada570f6e16 100644
|
| --- a/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
|
| +++ b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
|
| @@ -24,6 +24,9 @@ class AudioDecoderG722 final : public AudioDecoder {
|
| ~AudioDecoderG722() override;
|
| bool HasDecodePlc() const override;
|
| void Reset() override;
|
| + std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
|
| + uint32_t timestamp,
|
| + bool is_primary) override;
|
| int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
|
| int SampleRateHz() const override;
|
| size_t Channels() const override;
|
| @@ -45,6 +48,9 @@ class AudioDecoderG722Stereo final : public AudioDecoder {
|
| AudioDecoderG722Stereo();
|
| ~AudioDecoderG722Stereo() override;
|
| void Reset() override;
|
| + std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
|
| + uint32_t timestamp,
|
| + bool is_primary) override;
|
| int SampleRateHz() const override;
|
| size_t Channels() const override;
|
|
|
|
|