| Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
|
| diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
|
| index e9672ed120d98c9e7f01d8f78303a6a3185e8ce1..6ea9aede81fec23ee350ced4b58f95ff6ea14649 100644
|
| --- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
|
| +++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
|
| @@ -69,7 +69,6 @@ class AudioEncoderOpus final : public AudioEncoder {
|
| int complexity_threshold_window_bps = 1500;
|
| bool dtx_enabled = false;
|
| std::vector<int> supported_frame_lengths_ms;
|
| - const Clock* clock = Clock::GetRealTimeClock();
|
| int uplink_bandwidth_update_interval_ms = 200;
|
|
|
| private:
|
| @@ -87,8 +86,7 @@ class AudioEncoderOpus final : public AudioEncoder {
|
|
|
| using AudioNetworkAdaptorCreator =
|
| std::function<std::unique_ptr<AudioNetworkAdaptor>(const std::string&,
|
| - RtcEventLog*,
|
| - const Clock*)>;
|
| + RtcEventLog*)>;
|
| AudioEncoderOpus(
|
| const Config& config,
|
| AudioNetworkAdaptorCreator&& audio_network_adaptor_creator = nullptr,
|
| @@ -121,8 +119,7 @@ class AudioEncoderOpus final : public AudioEncoder {
|
| bool SetApplication(Application application) override;
|
| void SetMaxPlaybackRate(int frequency_hz) override;
|
| bool EnableAudioNetworkAdaptor(const std::string& config_string,
|
| - RtcEventLog* event_log,
|
| - const Clock* clock) override;
|
| + RtcEventLog* event_log) override;
|
| void DisableAudioNetworkAdaptor() override;
|
| void OnReceivedUplinkPacketLossFraction(
|
| float uplink_packet_loss_fraction) override;
|
| @@ -169,8 +166,7 @@ class AudioEncoderOpus final : public AudioEncoder {
|
| void ApplyAudioNetworkAdaptor();
|
| std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
|
| const ProtoString& config_string,
|
| - RtcEventLog* event_log,
|
| - const Clock* clock) const;
|
| + RtcEventLog* event_log) const;
|
|
|
| void MaybeUpdateUplinkBandwidth();
|
|
|
|
|