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https://github.com/mollyim/webrtc.git
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7eb0a5e210
In addition to the 48 kHz that we've always used. Bug: webrtc:10631 Change-Id: If73bf7ff9c1c0d22e0d1caa245128612850f8e41 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138268 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28104}
62 lines
2.2 KiB
C++
62 lines
2.2 KiB
C++
/*
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* Copyright (c) 2015 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_DECODER_OPUS_H_
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#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <vector>
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#include "api/audio_codecs/audio_decoder.h"
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#include "modules/audio_coding/codecs/opus/opus_interface.h"
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#include "rtc_base/buffer.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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class AudioDecoderOpusImpl final : public AudioDecoder {
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public:
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explicit AudioDecoderOpusImpl(size_t num_channels,
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int sample_rate_hz = 48000);
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~AudioDecoderOpusImpl() override;
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std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
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uint32_t timestamp) override;
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void Reset() override;
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int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
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int PacketDurationRedundant(const uint8_t* encoded,
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size_t encoded_len) const override;
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bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
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int SampleRateHz() const override;
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size_t Channels() const override;
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protected:
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int DecodeInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type) override;
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int DecodeRedundantInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type) override;
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private:
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OpusDecInst* dec_state_;
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const size_t channels_;
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const int sample_rate_hz_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderOpusImpl);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
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