From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- .../audio_coding/codecs/opus/audio_decoder_opus.h | 64 ++++++++++++++++++++++ 1 file changed, 64 insertions(+) create mode 100644 third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h') diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h new file mode 100644 index 0000000000..e8fd0440bc --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h @@ -0,0 +1,64 @@ +/* + * Copyright (c) 2015 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_DECODER_OPUS_H_ +#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_ + +#include +#include + +#include + +#include "api/audio_codecs/audio_decoder.h" +#include "modules/audio_coding/codecs/opus/opus_interface.h" +#include "rtc_base/buffer.h" + +namespace webrtc { + +class AudioDecoderOpusImpl final : public AudioDecoder { + public: + explicit AudioDecoderOpusImpl(size_t num_channels, + int sample_rate_hz = 48000); + ~AudioDecoderOpusImpl() override; + + AudioDecoderOpusImpl(const AudioDecoderOpusImpl&) = delete; + AudioDecoderOpusImpl& operator=(const AudioDecoderOpusImpl&) = delete; + + std::vector ParsePayload(rtc::Buffer&& payload, + uint32_t timestamp) override; + void Reset() override; + int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override; + int PacketDurationRedundant(const uint8_t* encoded, + size_t encoded_len) const override; + bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override; + int SampleRateHz() const override; + size_t Channels() const override; + + protected: + int DecodeInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) override; + int DecodeRedundantInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) override; + + private: + OpusDecInst* dec_state_; + const size_t channels_; + const int sample_rate_hz_; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_ -- cgit v1.2.3