LCOV - code coverage report
Current view: top level - media/webrtc/trunk/webrtc/modules/audio_coding/codecs/g722 - audio_encoder_g722.h (source / functions) Hit Total Coverage
Test: output.info Lines: 0 1 0.0 %
Date: 2017-07-14 16:53:18 Functions: 0 2 0.0 %
Legend: Lines: hit not hit

          Line data    Source code
       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 <memory>
      15             : 
      16             : #include "webrtc/base/buffer.h"
      17             : #include "webrtc/base/constructormagic.h"
      18             : #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
      19             : #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
      20             : 
      21             : namespace webrtc {
      22             : 
      23             : struct CodecInst;
      24             : 
      25           0 : class AudioEncoderG722 final : public AudioEncoder {
      26             :  public:
      27             :   struct Config {
      28             :     bool IsOk() const;
      29             : 
      30             :     int payload_type = 9;
      31             :     int frame_size_ms = 20;
      32             :     size_t num_channels = 1;
      33             :   };
      34             : 
      35             :   explicit AudioEncoderG722(const Config& config);
      36             :   explicit AudioEncoderG722(const CodecInst& codec_inst);
      37             :   ~AudioEncoderG722() override;
      38             : 
      39             :   int SampleRateHz() const override;
      40             :   size_t NumChannels() const override;
      41             :   int RtpTimestampRateHz() const override;
      42             :   size_t Num10MsFramesInNextPacket() const override;
      43             :   size_t Max10MsFramesInAPacket() const override;
      44             :   int GetTargetBitrate() const override;
      45             :   void Reset() override;
      46             : 
      47             :  protected:
      48             :   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
      49             :                          rtc::ArrayView<const int16_t> audio,
      50             :                          rtc::Buffer* encoded) override;
      51             : 
      52             :  private:
      53             :   // The encoder state for one channel.
      54             :   struct EncoderState {
      55             :     G722EncInst* encoder;
      56             :     std::unique_ptr<int16_t[]> speech_buffer;   // Queued up for encoding.
      57             :     rtc::Buffer encoded_buffer;                 // Already encoded.
      58             :     EncoderState();
      59             :     ~EncoderState();
      60             :   };
      61             : 
      62             :   size_t SamplesPerChannel() const;
      63             : 
      64             :   const size_t num_channels_;
      65             :   const int payload_type_;
      66             :   const size_t num_10ms_frames_per_packet_;
      67             :   size_t num_10ms_frames_buffered_;
      68             :   uint32_t first_timestamp_in_buffer_;
      69             :   const std::unique_ptr<EncoderState[]> encoders_;
      70             :   rtc::Buffer interleave_buffer_;
      71             :   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722);
      72             : };
      73             : 
      74             : }  // namespace webrtc
      75             : #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_

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