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/*
* 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 API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_
#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_
#include <stddef.h>
#include <vector>
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
struct RTC_EXPORT AudioEncoderMultiChannelOpusConfig {
static constexpr int kDefaultFrameSizeMs = 20;
// Opus API allows a min bitrate of 500bps, but Opus documentation suggests
// bitrate should be in the range of 6000 to 510000, inclusive.
static constexpr int kMinBitrateBps = 6000;
static constexpr int kMaxBitrateBps = 510000;
AudioEncoderMultiChannelOpusConfig();
AudioEncoderMultiChannelOpusConfig(const AudioEncoderMultiChannelOpusConfig&);
~AudioEncoderMultiChannelOpusConfig();
AudioEncoderMultiChannelOpusConfig& operator=(
const AudioEncoderMultiChannelOpusConfig&);
int frame_size_ms;
size_t num_channels;
enum class ApplicationMode { kVoip, kAudio };
ApplicationMode application;
int bitrate_bps;
bool fec_enabled;
bool cbr_enabled;
bool dtx_enabled;
int max_playback_rate_hz;
std::vector<int> supported_frame_lengths_ms;
int complexity;
// Number of mono/stereo Opus streams.
int num_streams;
// Number of channel pairs coupled together, see RFC 7845 section
// 5.1.1. Has to be less than the number of streams
int coupled_streams;
// Channel mapping table, defines the mapping from encoded streams to input
// channels. See RFC 7845 section 5.1.1.
std::vector<unsigned char> channel_mapping;
bool IsOk() const;
};
} // namespace webrtc
#endif // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_