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/*
* Copyright (c) 2017 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_OPUS_CONFIG_H_
#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
#include <stddef.h>
#include <vector>
#include "absl/types/optional.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
struct RTC_EXPORT AudioEncoderOpusConfig {
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;
AudioEncoderOpusConfig();
AudioEncoderOpusConfig(const AudioEncoderOpusConfig&);
~AudioEncoderOpusConfig();
AudioEncoderOpusConfig& operator=(const AudioEncoderOpusConfig&);
bool IsOk() const; // Checks if the values are currently OK.
int frame_size_ms;
int sample_rate_hz;
size_t num_channels;
enum class ApplicationMode { kVoip, kAudio };
ApplicationMode application;
// NOTE: This member must always be set.
// TODO(kwiberg): Turn it into just an int.
absl::optional<int> bitrate_bps;
bool fec_enabled;
bool cbr_enabled;
int max_playback_rate_hz;
// `complexity` is used when the bitrate goes above
// `complexity_threshold_bps` + `complexity_threshold_window_bps`;
// `low_rate_complexity` is used when the bitrate falls below
// `complexity_threshold_bps` - `complexity_threshold_window_bps`. In the
// interval in the middle, we keep using the most recent of the two
// complexity settings.
int complexity;
int low_rate_complexity;
int complexity_threshold_bps;
int complexity_threshold_window_bps;
bool dtx_enabled;
std::vector<int> supported_frame_lengths_ms;
int uplink_bandwidth_update_interval_ms;
// NOTE: This member isn't necessary, and will soon go away. See
// https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
int payload_type;
};
} // namespace webrtc
#endif // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
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