Opus implementation of the AudioEncoderFactoryTemplate API

webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h

+/*
+ *  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 AudioEncoderOpus? They're even in
separate build targets.
It would be neat to have AudioEncoderOpus::Config, instead.

[kwiberg-webrtc, 2017/06/16 12:44:14]

They need to be in separate build targets, because AudioEncoderOpus depends on
AudioEncoderOpusImpl, and both of them depend on AudioEncoderOpusConfig---so if
I put AudioEncoderOpus and AudioEncoderOpusConfig in the same build target, we
have a dependency cycle.

I could break that by not letting AudioEncoderOpusImpl use this config struct,
but then it needs another config struct that'll basically be a copy of this one.
I didn't want to do that.

[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_