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/*
* 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 "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 provides the number of encoded bytes in |encoded_bytes|.
// In case of error, false is returned, otherwise true. It is an error for the
// encoder to attempt to produce more than |max_encoded_bytes| bytes of
// output.
bool Encode(uint32_t timestamp,
const int16_t* audio,
size_t num_samples_per_channel,
size_t max_encoded_bytes,
uint8_t* encoded,
size_t* encoded_bytes,
uint32_t* encoded_timestamp) {
CHECK_EQ(num_samples_per_channel,
static_cast<size_t>(sample_rate_hz() / 100));
bool ret = Encode(timestamp,
audio,
max_encoded_bytes,
encoded,
encoded_bytes,
encoded_timestamp);
CHECK_LE(*encoded_bytes, max_encoded_bytes);
return ret;
}
// Return the input sample rate in Hz and the number of input channels.
// These are constants set at instantiation time.
virtual int sample_rate_hz() const = 0;
virtual int num_channels() const = 0;
// Returns the number of 10 ms frames the encoder will put in the next
// packet. This value may only change when Encode() outputs a packet; i.e.,
// the encoder may vary the number of 10 ms frames from packet to packet, but
// it must decide the length of the next packet no later than when outputting
// the preceding packet.
virtual int Num10MsFramesInNextPacket() const = 0;
protected:
virtual bool Encode(uint32_t timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
size_t* encoded_bytes,
uint32_t* encoded_timestamp) = 0;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
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