Chromium Code Reviews| Index: webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h |
| diff --git a/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h b/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..1726c69d6e413863b7437c1d4b5d13b674ba1eb0 |
| --- /dev/null |
| +++ b/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h |
| @@ -0,0 +1,63 @@ |
| +/* |
| + * 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 WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_ |
| +#define WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_ |
| + |
| +#include <stddef.h> |
| + |
| +#include <vector> |
| + |
| +namespace webrtc { |
| + |
| +// NOTE: This struct is still under development and may change without notice. |
| +struct AudioEncoderOpusConfig { |
|
ossu
2017/06/16 09:53:53
Why is AudioEncoderOpusConfig separate from AudioE
kwiberg-webrtc
2017/06/16 12:44:14
They need to be in separate build targets, because
ossu
2017/06/16 13:18:27
Ah, right!
|
| + 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; |
| + size_t num_channels; |
| + enum class ApplicationMode { kVoip, kAudio }; |
| + ApplicationMode application; |
| + 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; |
| +}; |
| + |
| +} // namespace webrtc |
| + |
| +#endif // WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_ |