/* * Copyright (c) 2016 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 MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_ #define MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_ #include #include #include #include "absl/types/optional.h" #include "api/array_view.h" #include "api/audio_codecs/audio_decoder.h" #include "rtc_base/buffer.h" namespace webrtc { class LegacyEncodedAudioFrame final : public AudioDecoder::EncodedAudioFrame { public: LegacyEncodedAudioFrame(AudioDecoder* decoder, rtc::Buffer&& payload); ~LegacyEncodedAudioFrame() override; static std::vector SplitBySamples( AudioDecoder* decoder, rtc::Buffer&& payload, uint32_t timestamp, size_t bytes_per_ms, uint32_t timestamps_per_ms); size_t Duration() const override; absl::optional Decode( rtc::ArrayView decoded) const override; // For testing: const rtc::Buffer& payload() const { return payload_; } private: AudioDecoder* const decoder_; const rtc::Buffer payload_; }; } // namespace webrtc #endif // MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_