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_ |