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..b5fdd44435f756e875ae2d88a41f7ead22f6e740 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 SetTargetBitrate(int bits_per_second) override; |
EncodedInfo EncodeInternal(uint32_t rtp_timestamp, |
const int16_t* audio, |
size_t max_encoded_bytes, |