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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/rtc_base/strings/audio_format_to_string.cc | |
parent | Initial commit. (diff) | |
download | firefox-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.cc | 54 |
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 |