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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
48 lines
1.4 KiB
C++
48 lines
1.4 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
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#define MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
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#include <vector>
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#include "api/array_view.h"
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#include "api/audio_codecs/audio_decoder.h"
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namespace webrtc {
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class LegacyEncodedAudioFrame final : public AudioDecoder::EncodedAudioFrame {
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public:
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LegacyEncodedAudioFrame(AudioDecoder* decoder, rtc::Buffer&& payload);
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~LegacyEncodedAudioFrame() override;
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static std::vector<AudioDecoder::ParseResult> SplitBySamples(
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AudioDecoder* decoder,
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rtc::Buffer&& payload,
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uint32_t timestamp,
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size_t bytes_per_ms,
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uint32_t timestamps_per_ms);
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size_t Duration() const override;
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rtc::Optional<DecodeResult> Decode(
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rtc::ArrayView<int16_t> decoded) const override;
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// For testing:
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const rtc::Buffer& payload() const { return payload_; }
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private:
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AudioDecoder* const decoder_;
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const rtc::Buffer payload_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
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