| Index: webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
|
| diff --git a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
|
| index f5e0a9899f7e20394fef463ad5e694a1217f5fd9..9b57fbe625225f69df275223629355e99c00e37c 100644
|
| --- a/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
|
| +++ b/webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h
|
| @@ -19,7 +19,7 @@
|
|
|
| namespace webrtc {
|
|
|
| -class AudioEncoderG722 : public AudioEncoder {
|
| +class AudioEncoderG722 final : public AudioEncoder {
|
| public:
|
| struct Config {
|
| Config() : payload_type(9), frame_size_ms(20), num_channels(1) {}
|
| @@ -39,6 +39,7 @@ class AudioEncoderG722 : public AudioEncoder {
|
| int RtpTimestampRateHz() const override;
|
| int Num10MsFramesInNextPacket() const override;
|
| int Max10MsFramesInAPacket() const override;
|
| + int GetTargetBitrate() const override;
|
| EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
|
| const int16_t* audio,
|
| size_t max_encoded_bytes,
|
|
|