1 files changed, 107 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_options.cc b/third_party/libwebrtc/api/audio_options.cc
new file mode 100644
index 0000000000..658515062c
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_options.cc
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2018 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_options.h"
+
+#include "api/array_view.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace cricket {
+namespace {
+
+template <class T>
+void ToStringIfSet(rtc::SimpleStringBuilder* result,
+                   const char* key,
+                   const absl::optional<T>& val) {
+  if (val) {
+    (*result) << key << ": " << *val << ", ";
+  }
+}
+
+template <typename T>
+void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
+  if (o) {
+    *s = o;
+  }
+}
+
+}  // namespace
+
+AudioOptions::AudioOptions() = default;
+AudioOptions::~AudioOptions() = default;
+
+void AudioOptions::SetAll(const AudioOptions& change) {
+  SetFrom(&echo_cancellation, change.echo_cancellation);
+#if defined(WEBRTC_IOS)
+  SetFrom(&ios_force_software_aec_HACK, change.ios_force_software_aec_HACK);
+#endif
+  SetFrom(&auto_gain_control, change.auto_gain_control);
+  SetFrom(&noise_suppression, change.noise_suppression);
+  SetFrom(&highpass_filter, change.highpass_filter);
+  SetFrom(&stereo_swapping, change.stereo_swapping);
+  SetFrom(&audio_jitter_buffer_max_packets,
+          change.audio_jitter_buffer_max_packets);
+  SetFrom(&audio_jitter_buffer_fast_accelerate,
+          change.audio_jitter_buffer_fast_accelerate);
+  SetFrom(&audio_jitter_buffer_min_delay_ms,
+          change.audio_jitter_buffer_min_delay_ms);
+  SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
+  SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
+  SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
+  SetFrom(&init_recording_on_send, change.init_recording_on_send);
+}
+
+bool AudioOptions::operator==(const AudioOptions& o) const {
+  return echo_cancellation == o.echo_cancellation &&
+#if defined(WEBRTC_IOS)
+         ios_force_software_aec_HACK == o.ios_force_software_aec_HACK &&
+#endif
+         auto_gain_control == o.auto_gain_control &&
+         noise_suppression == o.noise_suppression &&
+         highpass_filter == o.highpass_filter &&
+         stereo_swapping == o.stereo_swapping &&
+         audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
+         audio_jitter_buffer_fast_accelerate ==
+             o.audio_jitter_buffer_fast_accelerate &&
+         audio_jitter_buffer_min_delay_ms ==
+             o.audio_jitter_buffer_min_delay_ms &&
+         combined_audio_video_bwe == o.combined_audio_video_bwe &&
+         audio_network_adaptor == o.audio_network_adaptor &&
+         audio_network_adaptor_config == o.audio_network_adaptor_config &&
+         init_recording_on_send == o.init_recording_on_send;
+}
+
+std::string AudioOptions::ToString() const {
+  char buffer[1024];
+  rtc::SimpleStringBuilder result(buffer);
+  result << "AudioOptions {";
+  ToStringIfSet(&result, "aec", echo_cancellation);
+#if defined(WEBRTC_IOS)
+  ToStringIfSet(&result, "ios_force_software_aec_HACK",
+                ios_force_software_aec_HACK);
+#endif
+  ToStringIfSet(&result, "agc", auto_gain_control);
+  ToStringIfSet(&result, "ns", noise_suppression);
+  ToStringIfSet(&result, "hf", highpass_filter);
+  ToStringIfSet(&result, "swap", stereo_swapping);
+  ToStringIfSet(&result, "audio_jitter_buffer_max_packets",
+                audio_jitter_buffer_max_packets);
+  ToStringIfSet(&result, "audio_jitter_buffer_fast_accelerate",
+                audio_jitter_buffer_fast_accelerate);
+  ToStringIfSet(&result, "audio_jitter_buffer_min_delay_ms",
+                audio_jitter_buffer_min_delay_ms);
+  ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe);
+  ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor);
+  ToStringIfSet(&result, "init_recording_on_send", init_recording_on_send);
+  result << "}";
+  return result.str();
+}
+
+}  // namespace cricket