1 files changed, 75 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
new file mode 100644
index 0000000000..a9ab924b38
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
@@ -0,0 +1,75 @@
+/*
+ *  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 "api/audio_codecs/opus/audio_encoder_opus_config.h"
+
+namespace webrtc {
+
+namespace {
+
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+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),
+      sample_rate_hz(48000),
+      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 (sample_rate_hz != 16000 && sample_rate_hz != 48000) {
+    // Unsupported input sample rate. (libopus supports a few other rates as
+    // well; we can add support for them when needed.)
+    return false;
+  }
+  if (num_channels >= 255) {
+    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