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Diffstat (limited to 'third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h')
-rw-r--r-- | third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h | 43 |
1 files changed, 43 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h new file mode 100644 index 0000000000..c1c4db3577 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h @@ -0,0 +1,43 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_ +#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_ + +#include <memory> +#include <vector> + +#include "absl/types/optional.h" +#include "api/audio_codecs/audio_codec_pair_id.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_format.h" +#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h" +#include "api/field_trials_view.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// Opus encoder API for use as a template parameter to +// CreateAudioEncoderFactory<...>(). +struct RTC_EXPORT AudioEncoderMultiChannelOpus { + using Config = AudioEncoderMultiChannelOpusConfig; + static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format); + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); + static AudioCodecInfo QueryAudioEncoder(const Config& config); + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + const Config& config, + int payload_type, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr); +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_ |