| 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 3d0483b9ca5bbb12b34b40860c543bf05ccc914d..bbed3f4fb48609ad088dc39400267fe90f9bbbfd 100644
|
| --- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
|
| +++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
|
| @@ -16,6 +16,7 @@
|
| #include <string>
|
| #include <vector>
|
|
|
| +#include "webrtc/api/audio_codecs/audio_format.h"
|
| #include "webrtc/base/constructormagic.h"
|
| #include "webrtc/base/optional.h"
|
| #include "webrtc/common_audio/smoothing_filter.h"
|
| @@ -78,6 +79,9 @@ class AudioEncoderOpus final : public AudioEncoder {
|
| #endif
|
| };
|
|
|
| + static Config CreateConfig(int payload_type, const SdpAudioFormat& format);
|
| + static Config CreateConfig(const CodecInst& codec_inst);
|
| +
|
| using AudioNetworkAdaptorCreator =
|
| std::function<std::unique_ptr<AudioNetworkAdaptor>(const std::string&,
|
| RtcEventLog*,
|
| @@ -88,9 +92,14 @@ class AudioEncoderOpus final : public AudioEncoder {
|
| std::unique_ptr<SmoothingFilter> bitrate_smoother = nullptr);
|
|
|
| explicit AudioEncoderOpus(const CodecInst& codec_inst);
|
| -
|
| + AudioEncoderOpus(int payload_type, const SdpAudioFormat& format);
|
| ~AudioEncoderOpus() override;
|
|
|
| + // Static interface for use by BuiltinAudioEncoderFactory.
|
| + static constexpr const char* GetPayloadName() { return "opus"; }
|
| + static rtc::Optional<AudioFormatInfo> QueryAudioFormat(
|
| + const SdpAudioFormat& format);
|
| +
|
| int SampleRateHz() const override;
|
| size_t NumChannels() const override;
|
| size_t Num10MsFramesInNextPacket() const override;
|
|
|