| 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 8f5bba6e8bc41cd9f597e808f03deea01d647f6b..3d3597bfd9d4789e64faf7cbf141ee85d73c1b0c 100644
|
| --- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
|
| +++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
|
| @@ -16,12 +16,13 @@
|
| #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"
|
| #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
|
| -#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
|
| #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
|
| +#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
|
|
|
| namespace webrtc {
|
|
|
| @@ -50,7 +51,8 @@ class AudioEncoderOpus final : public AudioEncoder {
|
| // current bitrate is above or below complexity_threshold_bps.
|
| rtc::Optional<int> GetNewComplexity() const;
|
|
|
| - int frame_size_ms = 20;
|
| + static constexpr int kDefaultFrameSizeMs = 20;
|
| + int frame_size_ms = kDefaultFrameSizeMs;
|
| size_t num_channels = 1;
|
| int payload_type = 120;
|
| ApplicationMode application = kVoip;
|
| @@ -79,6 +81,9 @@ 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*,
|
| @@ -89,9 +94,14 @@ 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<AudioCodecInfo> QueryAudioEncoder(
|
| + const SdpAudioFormat& format);
|
| +
|
| int SampleRateHz() const override;
|
| size_t NumChannels() const override;
|
| size_t Num10MsFramesInNextPacket() const override;
|
|
|