blob: a932aa8b7d0fb628b105f30fa9e6b08b1297ba66 (
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
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
|
/*
* Copyright (c) 2014 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_G722_AUDIO_ENCODER_G722_H_
#define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
#include <memory>
#include <utility>
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
#include "api/units/time_delta.h"
#include "modules/audio_coding/codecs/g722/g722_interface.h"
#include "rtc_base/buffer.h"
namespace webrtc {
class AudioEncoderG722Impl final : public AudioEncoder {
public:
AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type);
~AudioEncoderG722Impl() override;
AudioEncoderG722Impl(const AudioEncoderG722Impl&) = delete;
AudioEncoderG722Impl& operator=(const AudioEncoderG722Impl&) = delete;
int SampleRateHz() const override;
size_t NumChannels() const override;
int RtpTimestampRateHz() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
void Reset() override;
absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const override;
protected:
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) override;
private:
// The encoder state for one channel.
struct EncoderState {
G722EncInst* encoder;
std::unique_ptr<int16_t[]> speech_buffer; // Queued up for encoding.
rtc::Buffer encoded_buffer; // Already encoded.
EncoderState();
~EncoderState();
};
size_t SamplesPerChannel() const;
const size_t num_channels_;
const int payload_type_;
const size_t num_10ms_frames_per_packet_;
size_t num_10ms_frames_buffered_;
uint32_t first_timestamp_in_buffer_;
const std::unique_ptr<EncoderState[]> encoders_;
rtc::Buffer interleave_buffer_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
|