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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/logging/rtc_event_log/rtc_stream_config.cc
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/logging/rtc_event_log/rtc_stream_config.cc')
-rw-r--r--third_party/libwebrtc/logging/rtc_event_log/rtc_stream_config.cc49
1 files changed, 49 insertions, 0 deletions
diff --git a/third_party/libwebrtc/logging/rtc_event_log/rtc_stream_config.cc b/third_party/libwebrtc/logging/rtc_event_log/rtc_stream_config.cc
new file mode 100644
index 0000000000..aa107c80bc
--- /dev/null
+++ b/third_party/libwebrtc/logging/rtc_event_log/rtc_stream_config.cc
@@ -0,0 +1,49 @@
+/*
+ * 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 "logging/rtc_event_log/rtc_stream_config.h"
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace rtclog {
+
+StreamConfig::StreamConfig() {}
+
+StreamConfig::~StreamConfig() {}
+
+StreamConfig::StreamConfig(const StreamConfig& other) = default;
+
+bool StreamConfig::operator==(const StreamConfig& other) const {
+ return local_ssrc == other.local_ssrc && remote_ssrc == other.remote_ssrc &&
+ rtx_ssrc == other.rtx_ssrc && rsid == other.rsid &&
+ remb == other.remb && rtcp_mode == other.rtcp_mode &&
+ rtp_extensions == other.rtp_extensions && codecs == other.codecs;
+}
+
+bool StreamConfig::operator!=(const StreamConfig& other) const {
+ return !(*this == other);
+}
+
+StreamConfig::Codec::Codec(absl::string_view payload_name,
+ int payload_type,
+ int rtx_payload_type)
+ : payload_name(payload_name),
+ payload_type(payload_type),
+ rtx_payload_type(rtx_payload_type) {}
+
+bool StreamConfig::Codec::operator==(const Codec& other) const {
+ return payload_name == other.payload_name &&
+ payload_type == other.payload_type &&
+ rtx_payload_type == other.rtx_payload_type;
+}
+
+} // namespace rtclog
+} // namespace webrtc