| /* |
| * 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 WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_ |
| |
| #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
| #include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| |
| namespace webrtc { |
| |
| class AudioEncoderG722 : public AudioEncoder { |
| public: |
| struct Config { |
| Config() : payload_type(9), frame_size_ms(20), num_channels(1) {} |
| |
| int payload_type; |
| int frame_size_ms; |
| int num_channels; |
| }; |
| |
| explicit AudioEncoderG722(const Config& config); |
| virtual ~AudioEncoderG722(); |
| |
| virtual int sample_rate_hz() const OVERRIDE; |
| virtual int num_channels() const OVERRIDE; |
| virtual int Num10MsFramesInNextPacket() const OVERRIDE; |
| virtual int Max10MsFramesInAPacket() const OVERRIDE; |
| |
| protected: |
| virtual bool EncodeInternal(uint32_t timestamp, |
| const int16_t* audio, |
| size_t max_encoded_bytes, |
| uint8_t* encoded, |
| EncodedInfo* info) OVERRIDE; |
| |
| private: |
| // The encoder state for one channel. |
| struct EncoderState { |
| G722EncInst* encoder; |
| scoped_ptr<int16_t[]> speech_buffer; // Queued up for encoding. |
| scoped_ptr<uint8_t[]> encoded_buffer; // Already encoded. |
| EncoderState(); |
| ~EncoderState(); |
| }; |
| |
| const int num_channels_; |
| const int payload_type_; |
| const int num_10ms_frames_per_packet_; |
| int num_10ms_frames_buffered_; |
| uint32_t first_timestamp_in_buffer_; |
| const scoped_ptr<EncoderState[]> encoders_; |
| const scoped_ptr<uint8_t[]> interleave_buffer_; |
| }; |
| |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_ |