Index: webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h |
diff --git a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h |
index 4de4e26273f98d3fa077c286eb008d185711f6cb..babed8ad95501201811300445c9cbf4899ec605a 100644 |
--- a/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h |
+++ b/webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h |
@@ -52,12 +52,14 @@ class AudioEncoderOpus final : public AudioEncoder { |
size_t MaxEncodedBytes() const override; |
int Num10MsFramesInNextPacket() const override; |
int Max10MsFramesInAPacket() const override; |
- void SetTargetBitrate(int bits_per_second) override; |
+ int SetTargetBitrate(int bits_per_second) override; |
void SetProjectedPacketLossRate(double fraction) override; |
double packet_loss_rate() const { return packet_loss_rate_; } |
ApplicationMode application() const { return application_; } |
bool dtx_enabled() const { return dtx_enabled_; } |
+ int bitrate_bps() const { return bitrate_bps_; } |
+ |
EncodedInfo EncodeInternal(uint32_t rtp_timestamp, |
const int16_t* audio, |
size_t max_encoded_bytes, |
@@ -104,6 +106,10 @@ class AudioEncoderMutableOpus |
CriticalSectionScoped cs(encoder_lock_.get()); |
return encoder()->dtx_enabled(); |
} |
+ int bitrate_bps() const { |
+ CriticalSectionScoped cs(encoder_lock_.get()); |
+ return encoder()->bitrate_bps(); |
+ } |
}; |
} // namespace webrtc |