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b602123a5a
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_coding'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}
77 lines
2.6 KiB
C++
77 lines
2.6 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 <memory>
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#include <vector>
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#include "api/audio_codecs/audio_encoder.h"
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#include "rtc_base/buffer.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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// This class implements redundant audio coding. The class object will have an
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// underlying AudioEncoder object that performs the actual encodings. The
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// current class will gather the two latest encodings from the underlying codec
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// into one packet.
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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 SetApplication(Application application) override;
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void SetMaxPlaybackRate(int frequency_hz) override;
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rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
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override;
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void OnReceivedUplinkPacketLossFraction(
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float uplink_packet_loss_fraction) override;
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void OnReceivedUplinkRecoverablePacketLossFraction(
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float uplink_recoverable_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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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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int red_payload_type_;
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rtc::Buffer secondary_encoded_;
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EncodedInfoLeaf secondary_info_;
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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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