| /* |
| * 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/base/buffer.h" |
| #include "webrtc/base/scoped_ptr.h" |
| #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
| #include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h" |
| |
| namespace webrtc { |
| |
| struct CodecInst; |
| |
| class AudioEncoderG722 final : public AudioEncoder { |
| public: |
| struct Config { |
| bool IsOk() const; |
| |
| int payload_type = 9; |
| int frame_size_ms = 20; |
| int num_channels = 1; |
| }; |
| |
| explicit AudioEncoderG722(const Config& config); |
| explicit AudioEncoderG722(const CodecInst& codec_inst); |
| ~AudioEncoderG722() override; |
| |
| size_t MaxEncodedBytes() const override; |
| int SampleRateHz() const override; |
| int NumChannels() const override; |
| int RtpTimestampRateHz() const override; |
| size_t Num10MsFramesInNextPacket() const override; |
| size_t Max10MsFramesInAPacket() const override; |
| int GetTargetBitrate() const override; |
| EncodedInfo EncodeInternal(uint32_t rtp_timestamp, |
| const int16_t* audio, |
| size_t max_encoded_bytes, |
| uint8_t* encoded) override; |
| void Reset() override; |
| |
| private: |
| // The encoder state for one channel. |
| struct EncoderState { |
| G722EncInst* encoder; |
| rtc::scoped_ptr<int16_t[]> speech_buffer; // Queued up for encoding. |
| rtc::Buffer encoded_buffer; // Already encoded. |
| EncoderState(); |
| ~EncoderState(); |
| }; |
| |
| size_t SamplesPerChannel() const; |
| |
| const int 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 rtc::scoped_ptr<EncoderState[]> encoders_; |
| rtc::Buffer interleave_buffer_; |
| }; |
| |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_INCLUDE_AUDIO_ENCODER_G722_H_ |