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714e3cbb48
Bug: webrtc:13579 Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37573}
89 lines
2.9 KiB
C++
89 lines
2.9 KiB
C++
/*
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* Copyright (c) 2019 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_OPUS_AUDIO_CODER_OPUS_COMMON_H_
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#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
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#include <string>
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#include <utility>
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#include <vector>
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#include "absl/strings/string_view.h"
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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 "api/audio_codecs/audio_format.h"
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#include "rtc_base/string_to_number.h"
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namespace webrtc {
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absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
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absl::string_view param);
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template <typename T>
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absl::optional<T> GetFormatParameter(const SdpAudioFormat& format,
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absl::string_view param) {
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return rtc::StringToNumber<T>(GetFormatParameter(format, param).value_or(""));
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}
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template <>
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absl::optional<std::vector<unsigned char>> GetFormatParameter(
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const SdpAudioFormat& format,
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absl::string_view param);
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class OpusFrame : public AudioDecoder::EncodedAudioFrame {
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public:
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OpusFrame(AudioDecoder* decoder,
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rtc::Buffer&& payload,
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bool is_primary_payload)
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: decoder_(decoder),
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payload_(std::move(payload)),
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is_primary_payload_(is_primary_payload) {}
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size_t Duration() const override {
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int ret;
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if (is_primary_payload_) {
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ret = decoder_->PacketDuration(payload_.data(), payload_.size());
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} else {
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ret = decoder_->PacketDurationRedundant(payload_.data(), payload_.size());
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}
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return (ret < 0) ? 0 : static_cast<size_t>(ret);
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}
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bool IsDtxPacket() const override { return payload_.size() <= 2; }
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absl::optional<DecodeResult> Decode(
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rtc::ArrayView<int16_t> decoded) const override {
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AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
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int ret;
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if (is_primary_payload_) {
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ret = decoder_->Decode(
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payload_.data(), payload_.size(), decoder_->SampleRateHz(),
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decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
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} else {
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ret = decoder_->DecodeRedundant(
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payload_.data(), payload_.size(), decoder_->SampleRateHz(),
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decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
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}
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if (ret < 0)
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return absl::nullopt;
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return DecodeResult{static_cast<size_t>(ret), speech_type};
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}
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private:
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AudioDecoder* const decoder_;
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const rtc::Buffer payload_;
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const bool is_primary_payload_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
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