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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 <vector>
#include "webrtc/typedefs.h"
namespace webrtc {
// This is the interface class for encoders in AudioCoding module. Each codec
// type must have an implementation of this class.
class AudioEncoder {
public:
struct EncodedInfoLeaf {
EncodedInfoLeaf()
: encoded_bytes(0),
encoded_timestamp(0),
payload_type(0),
send_even_if_empty(false),
speech(true) {}
size_t encoded_bytes;
uint32_t encoded_timestamp;
int payload_type;
bool send_even_if_empty;
bool speech;
};
// This is the main struct for auxiliary encoding information. Each encoded
// packet should be accompanied by one EncodedInfo struct, containing the
// total number of |encoded_bytes|, the |encoded_timestamp| and the
// |payload_type|. If the packet contains redundant encodings, the |redundant|
// vector will be populated with EncodedInfoLeaf structs. Each struct in the
// vector represents one encoding; the order of structs in the vector is the
// same as the order in which the actual payloads are written to the byte
// stream. When EncoderInfoLeaf structs are present in the vector, the main
// struct's |encoded_bytes| will be the sum of all the |encoded_bytes| in the
// vector.
struct EncodedInfo : public EncodedInfoLeaf {
EncodedInfo();
~EncodedInfo();
std::vector<EncodedInfoLeaf> redundant;
};
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.
// The encoder produces zero or more bytes of output in |encoded|,
// and provides additional encoding information in |info|.
// The caller is responsible for making sure that |max_encoded_bytes| is
// not smaller than the number of bytes actually produced by the encoder.
void Encode(uint32_t rtp_timestamp,
const int16_t* audio,
size_t num_samples_per_channel,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info);
// Return the input sample rate in Hz and the number of input channels.
// These are constants set at instantiation time.
virtual int SampleRateHz() const = 0;
virtual int NumChannels() const = 0;
// Return the maximum number of bytes that can be produced by the encoder
// at each Encode() call. The caller can use the return value to determine
// the size of the buffer that needs to be allocated. This value is allowed
// to depend on encoder parameters like bitrate, frame size etc., so if
// any of these change, the caller of Encode() is responsible for checking
// that the buffer is large enough by calling MaxEncodedBytes() again.
virtual size_t MaxEncodedBytes() const = 0;
// Returns the rate with which the RTP timestamps are updated. By default,
// this is the same as sample_rate_hz().
virtual int RtpTimestampRateHz() const;
// 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;
// Returns the maximum value that can be returned by
// Num10MsFramesInNextPacket().
virtual int Max10MsFramesInAPacket() const = 0;
// Changes the target bitrate. The implementation is free to alter this value,
// e.g., if the desired value is outside the valid range.
virtual void SetTargetBitrate(int bits_per_second) {}
// Tells the implementation what the projected packet loss rate is. The rate
// is in the range [0.0, 1.0]. This rate is typically used to adjust channel
// coding efforts, such as FEC.
virtual void SetProjectedPacketLossRate(double fraction) {}
protected:
virtual void EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) = 0;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
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