webrtc/modules/audio_coding/codecs/opus/opus_inst.h
Karl Wiberg a1d1a1e976 WebRTC Opus C interface: Add support for non-48 kHz decode sample rate
Plus tests for 16 kHz.

Bug: webrtc:10631
Change-Id: I2d89bc6d0d9548f0ad7bb1e36d6dfde6b6b31f83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138072
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28099}
2019-05-29 10:33:03 +00:00

39 lines
1 KiB
C

/*
* Copyright (c) 2014 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_OPUS_INST_H_
#define MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
#include <stddef.h>
#include "rtc_base/ignore_wundef.h"
RTC_PUSH_IGNORING_WUNDEF()
#include "opus.h"
#include "opus_multistream.h"
RTC_POP_IGNORING_WUNDEF()
struct WebRtcOpusEncInst {
OpusEncoder* encoder;
OpusMSEncoder* multistream_encoder;
size_t channels;
int in_dtx_mode;
};
struct WebRtcOpusDecInst {
OpusDecoder* decoder;
OpusMSDecoder* multistream_decoder;
int prev_decoded_samples;
size_t channels;
int in_dtx_mode;
int sample_rate_hz;
};
#endif // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_