blob: 9806772ba495e2d6b3636549d4bd61eef6e1edbf [file] [log] [blame]
/*
* 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_RED_AUDIO_ENCODER_COPY_RED_H_
#define MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
#include <stddef.h>
#include <stdint.h>
#include <memory>
#include <utility>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/units/time_delta.h"
#include "rtc_base/buffer.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
// This class implements redundant audio coding. The class object will have an
// underlying AudioEncoder object that performs the actual encodings. The
// current class will gather the two latest encodings from the underlying codec
// into one packet.
class AudioEncoderCopyRed final : public AudioEncoder {
public:
struct Config {
Config();
Config(Config&&);
~Config();
int payload_type;
std::unique_ptr<AudioEncoder> speech_encoder;
};
explicit AudioEncoderCopyRed(Config&& config);
~AudioEncoderCopyRed() 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;
void Reset() override;
bool SetFec(bool enable) override;
bool SetDtx(bool enable) override;
bool SetApplication(Application application) override;
void SetMaxPlaybackRate(int frequency_hz) override;
rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
override;
void OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction) override;
void OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
absl::optional<int64_t> bwe_period_ms) override;
void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const override;
protected:
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) override;
private:
size_t CalculateHeaderLength(size_t encoded_bytes) const;
std::unique_ptr<AudioEncoder> speech_encoder_;
size_t max_packet_length_;
int red_payload_type_;
rtc::Buffer secondary_encoded_;
EncodedInfoLeaf secondary_info_;
rtc::Buffer tertiary_encoded_;
EncodedInfoLeaf tertiary_info_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCopyRed);
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_