mirror of
https://github.com/mollyim/webrtc.git
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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}
59 lines
2 KiB
C++
59 lines
2 KiB
C++
/*
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* Copyright (c) 2015 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_ISAC_AUDIO_DECODER_ISAC_T_H_
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#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/audio_codecs/audio_decoder.h"
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#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/scoped_ref_ptr.h"
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namespace webrtc {
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template <typename T>
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class AudioDecoderIsacT final : public AudioDecoder {
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public:
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explicit AudioDecoderIsacT(int sample_rate_hz);
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AudioDecoderIsacT(int sample_rate_hz,
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const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
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~AudioDecoderIsacT() override;
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bool HasDecodePlc() const override;
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size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
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void Reset() override;
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int IncomingPacket(const uint8_t* payload,
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size_t payload_len,
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uint16_t rtp_sequence_number,
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uint32_t rtp_timestamp,
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uint32_t arrival_timestamp) override;
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int ErrorCode() override;
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int SampleRateHz() const override;
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size_t Channels() const override;
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int DecodeInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type) override;
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private:
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typename T::instance_type* isac_state_;
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int sample_rate_hz_;
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rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
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