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 "webrtc/base/buffer.h" 15 #include "webrtc/base/scoped_ptr.h" 16 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" 17 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h" 18 19 namespace webrtc { 20 21 struct CodecInst; 22 23 class AudioEncoderG722 final : public AudioEncoder { 24 public: 25 struct Config { 26 bool IsOk() const; 27 28 int payload_type = 9; 29 int frame_size_ms = 20; 30 size_t num_channels = 1; 31 }; 32 33 explicit AudioEncoderG722(const Config& config); 34 explicit AudioEncoderG722(const CodecInst& codec_inst); 35 ~AudioEncoderG722() override; 36 37 size_t MaxEncodedBytes() const override; 38 int SampleRateHz() const override; 39 size_t NumChannels() const override; 40 int RtpTimestampRateHz() const override; 41 size_t Num10MsFramesInNextPacket() const override; 42 size_t Max10MsFramesInAPacket() const override; 43 int GetTargetBitrate() const override; 44 EncodedInfo EncodeInternal(uint32_t rtp_timestamp, 45 rtc::ArrayView<const int16_t> audio, 46 size_t max_encoded_bytes, 47 uint8_t* encoded) override; 48 void Reset() override; 49 50 private: 51 // The encoder state for one channel. 52 struct EncoderState { 53 G722EncInst* encoder; 54 rtc::scoped_ptr<int16_t[]> speech_buffer; // Queued up for encoding. 55 rtc::Buffer encoded_buffer; // Already encoded. 56 EncoderState(); 57 ~EncoderState(); 58 }; 59 60 size_t SamplesPerChannel() const; 61 62 const size_t num_channels_; 63 const int payload_type_; 64 const size_t num_10ms_frames_per_packet_; 65 size_t num_10ms_frames_buffered_; 66 uint32_t first_timestamp_in_buffer_; 67 const rtc::scoped_ptr<EncoderState[]> encoders_; 68 rtc::Buffer interleave_buffer_; 69 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722); 70 }; 71 72 } // namespace webrtc 73 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ 74