OLD | NEW |
(Empty) | |
| 1 /* |
| 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| 3 * |
| 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ |
| 10 |
| 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_ |
| 12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_ |
| 13 |
| 14 #include <vector> |
| 15 |
| 16 #include "webrtc/base/array_view.h" |
| 17 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" |
| 18 |
| 19 namespace webrtc { |
| 20 |
| 21 class LegacyEncodedAudioFrame final : public AudioDecoder::EncodedAudioFrame { |
| 22 public: |
| 23 LegacyEncodedAudioFrame(AudioDecoder* decoder, |
| 24 rtc::Buffer&& payload, |
| 25 bool is_primary_payload); |
| 26 ~LegacyEncodedAudioFrame() override; |
| 27 |
| 28 static std::vector<AudioDecoder::ParseResult> SplitBySamples( |
| 29 AudioDecoder* decoder, |
| 30 rtc::Buffer&& payload, |
| 31 uint32_t timestamp, |
| 32 bool is_primary, |
| 33 size_t bytes_per_ms, |
| 34 uint32_t timestamps_per_ms); |
| 35 |
| 36 size_t Duration() const override; |
| 37 |
| 38 rtc::Optional<DecodeResult> Decode( |
| 39 rtc::ArrayView<int16_t> decoded) const override; |
| 40 |
| 41 // For testing: |
| 42 const rtc::Buffer& payload() const { return payload_; } |
| 43 |
| 44 private: |
| 45 AudioDecoder* const decoder_; |
| 46 const rtc::Buffer payload_; |
| 47 const bool is_primary_payload_; |
| 48 }; |
| 49 |
| 50 } // namespace webrtc |
| 51 |
| 52 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_ |
OLD | NEW |