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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h | 74 |
1 files changed, 74 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h new file mode 100644 index 0000000000..2ff47a8a53 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h @@ -0,0 +1,74 @@ +/* + * 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_DECODER_MULTI_CHANNEL_OPUS_IMPL_H_ +#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_IMPL_H_ + +#include <stddef.h> + +#include <memory> +#include <vector> + +#include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_format.h" +#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h" +#include "modules/audio_coding/codecs/opus/opus_interface.h" +#include "rtc_base/buffer.h" + +namespace webrtc { + +class AudioDecoderMultiChannelOpusImpl final : public AudioDecoder { + public: + static std::unique_ptr<AudioDecoderMultiChannelOpusImpl> MakeAudioDecoder( + AudioDecoderMultiChannelOpusConfig config); + + ~AudioDecoderMultiChannelOpusImpl() override; + + AudioDecoderMultiChannelOpusImpl(const AudioDecoderMultiChannelOpusImpl&) = + delete; + AudioDecoderMultiChannelOpusImpl& operator=( + const AudioDecoderMultiChannelOpusImpl&) = delete; + + std::vector<ParseResult> 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; + + static absl::optional<AudioDecoderMultiChannelOpusConfig> SdpToConfig( + const SdpAudioFormat& format); + + 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: + AudioDecoderMultiChannelOpusImpl(OpusDecInst* dec_state, + AudioDecoderMultiChannelOpusConfig config); + + OpusDecInst* dec_state_; + const AudioDecoderMultiChannelOpusConfig config_; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_IMPL_H_ |