blob: 21da1367edb1a7b70ebd04ea55dbaf1ed47d321e (
plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
|
/*
* 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 MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
#define MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
#include <stddef.h>
#include <stdint.h>
#include <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_decoder.h"
#include "rtc_base/buffer.h"
namespace webrtc {
class LegacyEncodedAudioFrame final : public AudioDecoder::EncodedAudioFrame {
public:
LegacyEncodedAudioFrame(AudioDecoder* decoder, rtc::Buffer&& payload);
~LegacyEncodedAudioFrame() override;
static std::vector<AudioDecoder::ParseResult> SplitBySamples(
AudioDecoder* decoder,
rtc::Buffer&& payload,
uint32_t timestamp,
size_t bytes_per_ms,
uint32_t timestamps_per_ms);
size_t Duration() const override;
absl::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_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
|
