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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_G722_AUDIO_ENCODER_G722_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
#include "webrtc/base/buffer.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
namespace webrtc {
struct CodecInst;
class AudioEncoderG722 final : public AudioEncoder {
public:
struct Config {
bool IsOk() const;
int payload_type = 9;
int frame_size_ms = 20;
size_t num_channels = 1;
};
explicit AudioEncoderG722(const Config& config);
explicit AudioEncoderG722(const CodecInst& codec_inst);
~AudioEncoderG722() override;
size_t MaxEncodedBytes() const override;
int SampleRateHz() const override;
size_t NumChannels() const override;
int RtpTimestampRateHz() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
size_t max_encoded_bytes,
uint8_t* encoded) override;
void Reset() override;
private:
// The encoder state for one channel.
struct EncoderState {
G722EncInst* encoder;
rtc::scoped_ptr<int16_t[]> speech_buffer; // Queued up for encoding.
rtc::Buffer encoded_buffer; // Already encoded.
EncoderState();
~EncoderState();
};
size_t SamplesPerChannel() const;
const size_t num_channels_;
const int payload_type_;
const size_t num_10ms_frames_per_packet_;
size_t num_10ms_frames_buffered_;
uint32_t first_timestamp_in_buffer_;
const rtc::scoped_ptr<EncoderState[]> encoders_;
rtc::Buffer interleave_buffer_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
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