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      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