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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/rtc_base/strings/audio_format_to_string.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/rtc_base/strings/audio_format_to_string.cc')
-rw-r--r--third_party/libwebrtc/rtc_base/strings/audio_format_to_string.cc54
1 files changed, 54 insertions, 0 deletions
diff --git a/third_party/libwebrtc/rtc_base/strings/audio_format_to_string.cc b/third_party/libwebrtc/rtc_base/strings/audio_format_to_string.cc
new file mode 100644
index 0000000000..7e91c3b49d
--- /dev/null
+++ b/third_party/libwebrtc/rtc_base/strings/audio_format_to_string.cc
@@ -0,0 +1,54 @@
+/*
+ * Copyright 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 "rtc_base/strings/audio_format_to_string.h"
+
+#include <utility>
+
+#include "rtc_base/strings/string_builder.h"
+
+namespace rtc {
+std::string ToString(const webrtc::SdpAudioFormat& saf) {
+ char sb_buf[1024];
+ rtc::SimpleStringBuilder sb(sb_buf);
+ sb << "{name: " << saf.name;
+ sb << ", clockrate_hz: " << saf.clockrate_hz;
+ sb << ", num_channels: " << saf.num_channels;
+ sb << ", parameters: {";
+ const char* sep = "";
+ for (const auto& kv : saf.parameters) {
+ sb << sep << kv.first << ": " << kv.second;
+ sep = ", ";
+ }
+ sb << "}}";
+ return sb.str();
+}
+std::string ToString(const webrtc::AudioCodecInfo& aci) {
+ char sb_buf[1024];
+ rtc::SimpleStringBuilder sb(sb_buf);
+ sb << "{sample_rate_hz: " << aci.sample_rate_hz;
+ sb << ", num_channels: " << aci.num_channels;
+ sb << ", default_bitrate_bps: " << aci.default_bitrate_bps;
+ sb << ", min_bitrate_bps: " << aci.min_bitrate_bps;
+ sb << ", max_bitrate_bps: " << aci.max_bitrate_bps;
+ sb << ", allow_comfort_noise: " << aci.allow_comfort_noise;
+ sb << ", supports_network_adaption: " << aci.supports_network_adaption;
+ sb << "}";
+ return sb.str();
+}
+std::string ToString(const webrtc::AudioCodecSpec& acs) {
+ char sb_buf[1024];
+ rtc::SimpleStringBuilder sb(sb_buf);
+ sb << "{format: " << ToString(acs.format);
+ sb << ", info: " << ToString(acs.info);
+ sb << "}";
+ return sb.str();
+}
+} // namespace rtc