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Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h | 89 |
1 files changed, 89 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h new file mode 100644 index 0000000000..5ebb51b577 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h @@ -0,0 +1,89 @@ +/* + * 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 <string> +#include <utility> +#include <vector> + +#include "absl/strings/string_view.h" +#include "absl/types/optional.h" +#include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_format.h" +#include "rtc_base/string_to_number.h" + +namespace webrtc { + +absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format, + absl::string_view param); + +template <typename T> +absl::optional<T> GetFormatParameter(const SdpAudioFormat& format, + absl::string_view param) { + return rtc::StringToNumber<T>(GetFormatParameter(format, param).value_or("")); +} + +template <> +absl::optional<std::vector<unsigned char>> GetFormatParameter( + const SdpAudioFormat& format, + absl::string_view param); + +class OpusFrame : public AudioDecoder::EncodedAudioFrame { + public: + OpusFrame(AudioDecoder* decoder, + rtc::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; } + + absl::optional<DecodeResult> Decode( + rtc::ArrayView<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 absl::nullopt; + + return DecodeResult{static_cast<size_t>(ret), speech_type}; + } + + private: + AudioDecoder* const decoder_; + const rtc::Buffer payload_; + const bool is_primary_payload_; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_ |