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b3bb13c572
potentially allowing distances of more than 2. BUG=webrtc:11640 Change-Id: I0d8c831218285d57cf07f0a8e5829810afd4ab3f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188383 Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33913}
101 lines
3.4 KiB
C++
101 lines
3.4 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
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#define MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <list>
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#include <memory>
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#include <utility>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/audio_codecs/audio_encoder.h"
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#include "api/units/time_delta.h"
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#include "rtc_base/buffer.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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// This class implements redundant audio coding as described in
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// https://tools.ietf.org/html/rfc2198
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// The class object will have an underlying AudioEncoder object that performs
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// the actual encodings. The current class will gather the N latest encodings
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// from the underlying codec into one packet. Currently N is hard-coded to 2.
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class AudioEncoderCopyRed final : public AudioEncoder {
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public:
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struct Config {
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Config();
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Config(Config&&);
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~Config();
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int payload_type;
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std::unique_ptr<AudioEncoder> speech_encoder;
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};
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explicit AudioEncoderCopyRed(Config&& config);
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~AudioEncoderCopyRed() override;
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int SampleRateHz() const override;
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size_t NumChannels() const override;
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int RtpTimestampRateHz() const override;
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size_t Num10MsFramesInNextPacket() const override;
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size_t Max10MsFramesInAPacket() const override;
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int GetTargetBitrate() const override;
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void Reset() override;
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bool SetFec(bool enable) override;
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bool SetDtx(bool enable) override;
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bool GetDtx() const override;
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bool SetApplication(Application application) override;
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void SetMaxPlaybackRate(int frequency_hz) override;
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bool EnableAudioNetworkAdaptor(const std::string& config_string,
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RtcEventLog* event_log) override;
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void DisableAudioNetworkAdaptor() override;
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void OnReceivedUplinkPacketLossFraction(
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float uplink_packet_loss_fraction) override;
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void OnReceivedUplinkBandwidth(
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int target_audio_bitrate_bps,
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absl::optional<int64_t> bwe_period_ms) override;
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void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;
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void OnReceivedRtt(int rtt_ms) override;
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void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
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void SetReceiverFrameLengthRange(int min_frame_length_ms,
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int max_frame_length_ms) override;
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ANAStats GetANAStats() const override;
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absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
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const override;
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rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
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override;
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protected:
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EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded) override;
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private:
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std::unique_ptr<AudioEncoder> speech_encoder_;
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rtc::Buffer primary_encoded_;
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size_t max_packet_length_;
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int red_payload_type_;
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std::list<std::pair<EncodedInfo, rtc::Buffer>> redundant_encodings_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCopyRed);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
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