webrtc/modules/audio_coding/codecs/audio_encoder.h

/*
 *  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_AUDIO_ENCODER_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_

#include <algorithm>
#include <limits>

#include "webrtc/base/checks.h"
#include "webrtc/typedefs.h"

namespace webrtc {

// This is the interface class for encoders in AudioCoding module. Each codec
// codec type must have an implementation of this class.
class AudioEncoder {
 public:
  virtual ~AudioEncoder() {}

  // Accepts one 10 ms block of input audio (i.e., sample_rate_hz() / 100 *
  // num_channels() samples). Multi-channel audio must be sample-interleaved.
  // If successful, the encoder produces zero or more bytes of output in
  // |encoded|, and returns the number of bytes. In case of error, -1 is
  // returned. It is an error for the encoder to attempt to produce more than
  // |max_encoded_bytes| bytes of output.
  ssize_t Encode(uint32_t timestamp,
                 const int16_t* audio,
                 size_t num_samples,
                 size_t max_encoded_bytes,
                 uint8_t* encoded) {
    CHECK_EQ(num_samples,
             static_cast<size_t>(sample_rate_hz() / 100 * num_channels()));
    ssize_t num_bytes = Encode(timestamp, audio, max_encoded_bytes, encoded);
    CHECK_LE(num_bytes,
             static_cast<ssize_t>(std::min(
                 max_encoded_bytes,
                 static_cast<size_t>(std::numeric_limits<ssize_t>::max()))));
    return num_bytes;
  }

  // Returns the input sample rate in Hz, the number of input channels, and the
  // number of 10 ms frames the encoder puts in one output packet. These are
  // constants set at instantiation time.
  virtual int sample_rate_hz() const = 0;
  virtual int num_channels() const = 0;
  virtual int num_10ms_frames_per_packet() const = 0;

 protected:
  virtual ssize_t Encode(uint32_t timestamp,
                         const int16_t* audio,
                         size_t max_encoded_bytes,
                         uint8_t* encoded) = 0;
};

}  // namespace webrtc
#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_