Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(832)

Unified Diff: webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h

Issue 2930243003: Opus implementation of the AudioEncoderFactoryTemplate API (Closed)
Patch Set: rebase Created 3 years, 6 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
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_

Powered by Google App Engine
This is Rietveld 408576698