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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_RED_AUDIO_ENCODER_COPY_RED_H_
#define MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
#include <stddef.h>
#include <stdint.h>
#include <list>
#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 as described in
// https://tools.ietf.org/html/rfc2198
// The class object will have an underlying AudioEncoder object that performs
// the actual encodings. The current class will gather the N latest encodings
// from the underlying codec into one packet. Currently N is hard-coded to 2.
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 GetDtx() const override;
bool SetApplication(Application application) override;
void SetMaxPlaybackRate(int frequency_hz) override;
bool EnableAudioNetworkAdaptor(const std::string& config_string,
RtcEventLog* event_log) override;
void DisableAudioNetworkAdaptor() 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 OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;
void OnReceivedRtt(int rtt_ms) override;
void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
void SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms) override;
ANAStats GetANAStats() const override;
absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
const override;
rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
override;
protected:
EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
rtc::ArrayView<const int16_t> audio,
rtc::Buffer* encoded) override;
private:
std::unique_ptr<AudioEncoder> speech_encoder_;
rtc::Buffer primary_encoded_;
size_t max_packet_length_;
int red_payload_type_;
std::list<std::pair<EncodedInfo, rtc::Buffer>> redundant_encodings_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCopyRed);
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
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