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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 MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
#define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
#include <stddef.h>
#include <stdint.h>
#include <utility>
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
#include "api/units/time_delta.h"
#include "modules/audio_coding/codecs/ilbc/ilbc.h"
namespace webrtc {
class AudioEncoderIlbcImpl final : public AudioEncoder {
public:
AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type);
~AudioEncoderIlbcImpl() override;
AudioEncoderIlbcImpl(const AudioEncoderIlbcImpl&) = delete;
AudioEncoderIlbcImpl& operator=(const AudioEncoderIlbcImpl&) = delete;
int SampleRateHz() const override;
size_t NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) override;
void Reset() override;
absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const override;
private:
size_t RequiredOutputSizeBytes() const;
static constexpr size_t kMaxSamplesPerPacket = 480;
const int frame_size_ms_;
const int payload_type_;
const size_t num_10ms_frames_per_packet_;
size_t num_10ms_frames_buffered_;
uint32_t first_timestamp_in_buffer_;
int16_t input_buffer_[kMaxSamplesPerPacket];
IlbcEncoderInstance* encoder_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
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