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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
53 lines
1.5 KiB
C++
53 lines
1.5 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 <stddef.h>
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#include <stdint.h>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/audio_codecs/audio_decoder.h"
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#include "rtc_base/buffer.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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absl::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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