| Index: webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
|
| diff --git a/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h b/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..1c9c3f542a8049f27e864bf8730a0bfaa066f165
|
| --- /dev/null
|
| +++ b/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
|
| @@ -0,0 +1,52 @@
|
| +/*
|
| + * 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 WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
|
| +#define WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
|
| +
|
| +#include <vector>
|
| +
|
| +#include "webrtc/base/array_view.h"
|
| +#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
|
| +
|
| +namespace webrtc {
|
| +
|
| +class LegacyEncodedAudioFrame final : public AudioDecoder::EncodedAudioFrame {
|
| + public:
|
| + LegacyEncodedAudioFrame(AudioDecoder* decoder,
|
| + rtc::Buffer&& payload,
|
| + bool is_primary_payload);
|
| + ~LegacyEncodedAudioFrame() override;
|
| +
|
| + static std::vector<AudioDecoder::ParseResult> SplitBySamples(
|
| + AudioDecoder* decoder,
|
| + rtc::Buffer&& payload,
|
| + uint32_t timestamp,
|
| + bool is_primary,
|
| + size_t bytes_per_ms,
|
| + uint32_t timestamps_per_ms);
|
| +
|
| + size_t Duration() const override;
|
| +
|
| + rtc::Optional<DecodeResult> Decode(
|
| + rtc::ArrayView<int16_t> decoded) const override;
|
| +
|
| + // For testing:
|
| + const rtc::Buffer& payload() const { return payload_; }
|
| +
|
| + private:
|
| + AudioDecoder* const decoder_;
|
| + const rtc::Buffer payload_;
|
| + const bool is_primary_payload_;
|
| +};
|
| +
|
| +} // namespace webrtc
|
| +
|
| +#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
|
|
|