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; |