Index: webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc |
diff --git a/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc b/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..7d29883b391433d3666bd8f876b3dde18a3ee6d6 |
--- /dev/null |
+++ b/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc |
@@ -0,0 +1,70 @@ |
+/* |
+ * 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. |
+ */ |
+ |
+#include "webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h" |
+ |
+namespace webrtc { |
+ |
+namespace { |
+ |
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM) |
+// If we are on Android, iOS and/or ARM, use a lower complexity setting by |
+// default, to save encoder complexity. |
+constexpr int kDefaultComplexity = 5; |
+#else |
+constexpr int kDefaultComplexity = 9; |
+#endif |
+ |
+constexpr int kDefaultLowRateComplexity = |
+ WEBRTC_OPUS_VARIABLE_COMPLEXITY ? 9 : kDefaultComplexity; |
+ |
+} // namespace |
+ |
+constexpr int AudioEncoderOpusConfig::kDefaultFrameSizeMs; |
+constexpr int AudioEncoderOpusConfig::kMinBitrateBps; |
+constexpr int AudioEncoderOpusConfig::kMaxBitrateBps; |
+ |
+AudioEncoderOpusConfig::AudioEncoderOpusConfig() |
+ : frame_size_ms(kDefaultFrameSizeMs), |
+ num_channels(1), |
+ application(ApplicationMode::kVoip), |
+ bitrate_bps(32000), |
+ fec_enabled(false), |
+ cbr_enabled(false), |
+ max_playback_rate_hz(48000), |
+ complexity(kDefaultComplexity), |
+ low_rate_complexity(kDefaultLowRateComplexity), |
+ complexity_threshold_bps(12500), |
+ complexity_threshold_window_bps(1500), |
+ dtx_enabled(false), |
+ uplink_bandwidth_update_interval_ms(200), |
+ payload_type(-1) {} |
+AudioEncoderOpusConfig::AudioEncoderOpusConfig(const AudioEncoderOpusConfig&) = |
+ default; |
+AudioEncoderOpusConfig::~AudioEncoderOpusConfig() = default; |
+AudioEncoderOpusConfig& AudioEncoderOpusConfig::operator=( |
+ const AudioEncoderOpusConfig&) = default; |
+ |
+bool AudioEncoderOpusConfig::IsOk() const { |
+ if (frame_size_ms <= 0 || frame_size_ms % 10 != 0) |
+ return false; |
+ if (num_channels != 1 && num_channels != 2) |
+ return false; |
+ if (!bitrate_bps) |
+ return false; |
+ if (*bitrate_bps < kMinBitrateBps || *bitrate_bps > kMaxBitrateBps) |
+ return false; |
+ if (complexity < 0 || complexity > 10) |
+ return false; |
+ if (low_rate_complexity < 0 || low_rate_complexity > 10) |
+ return false; |
+ return true; |
+} |
+} // namespace webrtc |