| /* | |
| * Copyright (c) 2019 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_OPUS_AUDIO_CODER_OPUS_COMMON_H_ | |
| #define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_ | |
| #include <cstddef> | |
| #include <cstdint> | |
| #include <optional> | |
| #include <span> | |
| #include <string> | |
| #include <utility> | |
| #include <vector> | |
| #include "absl/strings/string_view.h" | |
| #include "api/audio_codecs/audio_decoder.h" | |
| #include "api/audio_codecs/audio_format.h" | |
| #include "rtc_base/buffer.h" | |
| #include "rtc_base/string_to_number.h" | |
| namespace webrtc { | |
| std::optional<std::string> GetFormatParameter(const SdpAudioFormat& format, | |
| absl::string_view param); | |
| template <typename T> | |
| std::optional<T> GetFormatParameter(const SdpAudioFormat& format, | |
| absl::string_view param) { | |
| return StringToNumber<T>(GetFormatParameter(format, param).value_or("")); | |
| } | |
| template <> | |
| std::optional<std::vector<unsigned char>> GetFormatParameter( | |
| const SdpAudioFormat& format, | |
| absl::string_view param); | |
| class OpusFrame : public AudioDecoder::EncodedAudioFrame { | |
| public: | |
| OpusFrame(AudioDecoder* decoder, Buffer&& payload, bool is_primary_payload) | |
| : decoder_(decoder), | |
| payload_(std::move(payload)), | |
| is_primary_payload_(is_primary_payload) {} | |
| size_t Duration() const override { | |
| int ret; | |
| if (is_primary_payload_) { | |
| ret = decoder_->PacketDuration(payload_.data(), payload_.size()); | |
| } else { | |
| ret = decoder_->PacketDurationRedundant(payload_.data(), payload_.size()); | |
| } | |
| return (ret < 0) ? 0 : static_cast<size_t>(ret); | |
| } | |
| bool IsDtxPacket() const override { return payload_.size() <= 2; } | |
| std::optional<DecodeResult> Decode( | |
| std::span<int16_t> decoded) const override { | |
| AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech; | |
| int ret; | |
| if (is_primary_payload_) { | |
| ret = decoder_->Decode( | |
| payload_.data(), payload_.size(), decoder_->SampleRateHz(), | |
| decoded.size() * sizeof(int16_t), decoded.data(), &speech_type); | |
| } else { | |
| ret = decoder_->DecodeRedundant( | |
| payload_.data(), payload_.size(), decoder_->SampleRateHz(), | |
| decoded.size() * sizeof(int16_t), decoded.data(), &speech_type); | |
| } | |
| if (ret < 0) | |
| return std::nullopt; | |
| return DecodeResult{.num_decoded_samples = static_cast<size_t>(ret), | |
| .speech_type = speech_type}; | |
| } | |
| private: | |
| AudioDecoder* const decoder_; | |
| const Buffer payload_; | |
| const bool is_primary_payload_; | |
| }; | |
| } // namespace webrtc | |
| #endif // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_ |