9a7c838ec4
This is to prevent size_t from undefined. This does not happen in current WebRTC since the sources that opus_inst.h gets used have proper definitions. But it would be good to add the definition in itself. Review URL: https://codereview.webrtc.org/1446093003 Cr-Commit-Position: refs/heads/master@{#10653}
39 lines
1.2 KiB
C
39 lines
1.2 KiB
C
/*
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* Copyright (c) 2014 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 WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
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#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
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#include <stddef.h>
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#include "opus.h"
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struct WebRtcOpusEncInst {
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OpusEncoder* encoder;
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int channels;
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int in_dtx_mode;
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// When Opus is in DTX mode, we use |zero_counts| to count consecutive zeros
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// to break long zero segment so as to prevent DTX from going wrong. We use
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// one counter for each channel. After each encoding, |zero_counts| contain
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// the remaining zeros from the last frame.
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// TODO(minyue): remove this when Opus gets an internal fix to DTX.
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size_t* zero_counts;
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};
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struct WebRtcOpusDecInst {
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OpusDecoder* decoder;
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int prev_decoded_samples;
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int channels;
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int in_dtx_mode;
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};
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#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
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