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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_ILBC_INTERFACE_AUDIO_ENCODER_ILBC_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INTERFACE_AUDIO_ENCODER_ILBC_H_
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
class AudioEncoderIlbc : public AudioEncoder {
public:
struct Config {
Config() : payload_type(102), frame_size_ms(30) {}
int payload_type;
int frame_size_ms; // Valid values are 20, 30, 40, and 60 ms.
// Note that frame size 40 ms produces encodings with two 20 ms frames in
// them, and frame size 60 ms consists of two 30 ms frames.
};
explicit AudioEncoderIlbc(const Config& config);
virtual ~AudioEncoderIlbc();
virtual int SampleRateHz() const OVERRIDE;
virtual int NumChannels() const OVERRIDE;
virtual int Num10MsFramesInNextPacket() const OVERRIDE;
virtual int Max10MsFramesInAPacket() const OVERRIDE;
protected:
virtual bool EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) OVERRIDE;
private:
static const int kMaxSamplesPerPacket = 480;
const int payload_type_;
const int num_10ms_frames_per_packet_;
int num_10ms_frames_buffered_;
uint32_t first_timestamp_in_buffer_;
int16_t input_buffer_[kMaxSamplesPerPacket];
IlbcEncoderInstance* encoder_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_INTERFACE_AUDIO_ENCODER_ILBC_H_