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