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-rw-r--r--third_party/libwebrtc/api/audio_options.cc107
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
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+++ 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