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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.h | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.h | 74 |
1 files changed, 74 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.h new file mode 100644 index 0000000000..d5d7256c70 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.h @@ -0,0 +1,74 @@ +/* + * 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_ |