Line data Source code
1 : /*
2 : * Copyright (c) 2014 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_G722_AUDIO_ENCODER_G722_H_
12 : #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
13 :
14 : #include <memory>
15 :
16 : #include "webrtc/base/buffer.h"
17 : #include "webrtc/base/constructormagic.h"
18 : #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
19 : #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
20 :
21 : namespace webrtc {
22 :
23 : struct CodecInst;
24 :
25 0 : class AudioEncoderG722 final : public AudioEncoder {
26 : public:
27 : struct Config {
28 : bool IsOk() const;
29 :
30 : int payload_type = 9;
31 : int frame_size_ms = 20;
32 : size_t num_channels = 1;
33 : };
34 :
35 : explicit AudioEncoderG722(const Config& config);
36 : explicit AudioEncoderG722(const CodecInst& codec_inst);
37 : ~AudioEncoderG722() override;
38 :
39 : int SampleRateHz() const override;
40 : size_t NumChannels() const override;
41 : int RtpTimestampRateHz() const override;
42 : size_t Num10MsFramesInNextPacket() const override;
43 : size_t Max10MsFramesInAPacket() const override;
44 : int GetTargetBitrate() const override;
45 : void Reset() override;
46 :
47 : protected:
48 : EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
49 : rtc::ArrayView<const int16_t> audio,
50 : rtc::Buffer* encoded) override;
51 :
52 : private:
53 : // The encoder state for one channel.
54 : struct EncoderState {
55 : G722EncInst* encoder;
56 : std::unique_ptr<int16_t[]> speech_buffer; // Queued up for encoding.
57 : rtc::Buffer encoded_buffer; // Already encoded.
58 : EncoderState();
59 : ~EncoderState();
60 : };
61 :
62 : size_t SamplesPerChannel() const;
63 :
64 : const size_t num_channels_;
65 : const int payload_type_;
66 : const size_t num_10ms_frames_per_packet_;
67 : size_t num_10ms_frames_buffered_;
68 : uint32_t first_timestamp_in_buffer_;
69 : const std::unique_ptr<EncoderState[]> encoders_;
70 : rtc::Buffer interleave_buffer_;
71 : RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722);
72 : };
73 :
74 : } // namespace webrtc
75 : #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
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