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https://github.com/mollyim/webrtc.git
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e482ff8f70
These things are no longer brand new, so it makes even less sense than it once did to warn users that they may change at any time. Bug: none Change-Id: I43a6915d9e00fbfef30fdb89869873b129297c8d No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/106980 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25283}
73 lines
2.3 KiB
C++
73 lines
2.3 KiB
C++
/*
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* Copyright (c) 2017 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 API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
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#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
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#include <stddef.h>
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#include <vector>
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#include "absl/types/optional.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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struct RTC_EXPORT AudioEncoderOpusConfig {
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static constexpr int kDefaultFrameSizeMs = 20;
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// Opus API allows a min bitrate of 500bps, but Opus documentation suggests
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// bitrate should be in the range of 6000 to 510000, inclusive.
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static constexpr int kMinBitrateBps = 6000;
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static constexpr int kMaxBitrateBps = 510000;
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AudioEncoderOpusConfig();
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AudioEncoderOpusConfig(const AudioEncoderOpusConfig&);
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~AudioEncoderOpusConfig();
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AudioEncoderOpusConfig& operator=(const AudioEncoderOpusConfig&);
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bool IsOk() const; // Checks if the values are currently OK.
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int frame_size_ms;
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size_t num_channels;
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enum class ApplicationMode { kVoip, kAudio };
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ApplicationMode application;
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// NOTE: This member must always be set.
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// TODO(kwiberg): Turn it into just an int.
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absl::optional<int> bitrate_bps;
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bool fec_enabled;
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bool cbr_enabled;
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int max_playback_rate_hz;
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// |complexity| is used when the bitrate goes above
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// |complexity_threshold_bps| + |complexity_threshold_window_bps|;
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// |low_rate_complexity| is used when the bitrate falls below
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// |complexity_threshold_bps| - |complexity_threshold_window_bps|. In the
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// interval in the middle, we keep using the most recent of the two
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// complexity settings.
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int complexity;
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int low_rate_complexity;
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int complexity_threshold_bps;
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int complexity_threshold_window_bps;
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bool dtx_enabled;
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std::vector<int> supported_frame_lengths_ms;
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int uplink_bandwidth_update_interval_ms;
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// NOTE: This member isn't necessary, and will soon go away. See
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// https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
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int payload_type;
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};
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} // namespace webrtc
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#endif // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
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