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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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_G722_AUDIO_ENCODER_G722_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ |
12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ | 12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ |
13 | 13 |
14 #include "webrtc/base/buffer.h" | 14 #include "webrtc/base/buffer.h" |
15 #include "webrtc/base/scoped_ptr.h" | 15 #include "webrtc/base/scoped_ptr.h" |
16 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" | 16 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
17 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h" | 17 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h" |
18 | 18 |
19 namespace webrtc { | 19 namespace webrtc { |
20 | 20 |
21 struct CodecInst; | 21 struct CodecInst; |
22 | 22 |
23 class AudioEncoderG722 final : public AudioEncoder { | 23 class AudioEncoderG722 final : public AudioEncoder { |
24 public: | 24 public: |
25 struct Config { | 25 struct Config { |
26 bool IsOk() const; | 26 bool IsOk() const; |
27 | 27 |
28 int payload_type = 9; | 28 int payload_type = 9; |
29 int frame_size_ms = 20; | 29 int frame_size_ms = 20; |
30 int num_channels = 1; | 30 size_t num_channels = 1; |
31 }; | 31 }; |
32 | 32 |
33 explicit AudioEncoderG722(const Config& config); | 33 explicit AudioEncoderG722(const Config& config); |
34 explicit AudioEncoderG722(const CodecInst& codec_inst); | 34 explicit AudioEncoderG722(const CodecInst& codec_inst); |
35 ~AudioEncoderG722() override; | 35 ~AudioEncoderG722() override; |
36 | 36 |
37 size_t MaxEncodedBytes() const override; | 37 size_t MaxEncodedBytes() const override; |
38 int SampleRateHz() const override; | 38 int SampleRateHz() const override; |
39 int NumChannels() const override; | 39 size_t NumChannels() const override; |
40 int RtpTimestampRateHz() const override; | 40 int RtpTimestampRateHz() const override; |
41 size_t Num10MsFramesInNextPacket() const override; | 41 size_t Num10MsFramesInNextPacket() const override; |
42 size_t Max10MsFramesInAPacket() const override; | 42 size_t Max10MsFramesInAPacket() const override; |
43 int GetTargetBitrate() const override; | 43 int GetTargetBitrate() const override; |
44 EncodedInfo EncodeInternal(uint32_t rtp_timestamp, | 44 EncodedInfo EncodeInternal(uint32_t rtp_timestamp, |
45 rtc::ArrayView<const int16_t> audio, | 45 rtc::ArrayView<const int16_t> audio, |
46 size_t max_encoded_bytes, | 46 size_t max_encoded_bytes, |
47 uint8_t* encoded) override; | 47 uint8_t* encoded) override; |
48 void Reset() override; | 48 void Reset() override; |
49 | 49 |
50 private: | 50 private: |
51 // The encoder state for one channel. | 51 // The encoder state for one channel. |
52 struct EncoderState { | 52 struct EncoderState { |
53 G722EncInst* encoder; | 53 G722EncInst* encoder; |
54 rtc::scoped_ptr<int16_t[]> speech_buffer; // Queued up for encoding. | 54 rtc::scoped_ptr<int16_t[]> speech_buffer; // Queued up for encoding. |
55 rtc::Buffer encoded_buffer; // Already encoded. | 55 rtc::Buffer encoded_buffer; // Already encoded. |
56 EncoderState(); | 56 EncoderState(); |
57 ~EncoderState(); | 57 ~EncoderState(); |
58 }; | 58 }; |
59 | 59 |
60 size_t SamplesPerChannel() const; | 60 size_t SamplesPerChannel() const; |
61 | 61 |
62 const int num_channels_; | 62 const size_t num_channels_; |
63 const int payload_type_; | 63 const int payload_type_; |
64 const size_t num_10ms_frames_per_packet_; | 64 const size_t num_10ms_frames_per_packet_; |
65 size_t num_10ms_frames_buffered_; | 65 size_t num_10ms_frames_buffered_; |
66 uint32_t first_timestamp_in_buffer_; | 66 uint32_t first_timestamp_in_buffer_; |
67 const rtc::scoped_ptr<EncoderState[]> encoders_; | 67 const rtc::scoped_ptr<EncoderState[]> encoders_; |
68 rtc::Buffer interleave_buffer_; | 68 rtc::Buffer interleave_buffer_; |
69 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722); | 69 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722); |
70 }; | 70 }; |
71 | 71 |
72 } // namespace webrtc | 72 } // namespace webrtc |
73 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ | 73 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ |
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