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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/audio_codecs/audio_format.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/audio_codecs/audio_format.cc | 86 |
1 files changed, 86 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_format.cc b/third_party/libwebrtc/api/audio_codecs/audio_format.cc new file mode 100644 index 0000000000..2a529a49ee --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_format.cc @@ -0,0 +1,86 @@ +/* + * Copyright (c) 2016 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_codecs/audio_format.h" + +#include <utility> + +#include "absl/strings/match.h" + +namespace webrtc { + +SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default; +SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default; + +SdpAudioFormat::SdpAudioFormat(absl::string_view name, + int clockrate_hz, + size_t num_channels) + : name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {} + +SdpAudioFormat::SdpAudioFormat(absl::string_view name, + int clockrate_hz, + size_t num_channels, + const Parameters& param) + : name(name), + clockrate_hz(clockrate_hz), + num_channels(num_channels), + parameters(param) {} + +SdpAudioFormat::SdpAudioFormat(absl::string_view name, + int clockrate_hz, + size_t num_channels, + Parameters&& param) + : name(name), + clockrate_hz(clockrate_hz), + num_channels(num_channels), + parameters(std::move(param)) {} + +bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const { + return absl::EqualsIgnoreCase(name, o.name) && + clockrate_hz == o.clockrate_hz && num_channels == o.num_channels; +} + +SdpAudioFormat::~SdpAudioFormat() = default; +SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default; +SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default; + +bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) { + return absl::EqualsIgnoreCase(a.name, b.name) && + a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels && + a.parameters == b.parameters; +} + +AudioCodecInfo::AudioCodecInfo(int sample_rate_hz, + size_t num_channels, + int bitrate_bps) + : AudioCodecInfo(sample_rate_hz, + num_channels, + bitrate_bps, + bitrate_bps, + bitrate_bps) {} + +AudioCodecInfo::AudioCodecInfo(int sample_rate_hz, + size_t num_channels, + int default_bitrate_bps, + int min_bitrate_bps, + int max_bitrate_bps) + : sample_rate_hz(sample_rate_hz), + num_channels(num_channels), + default_bitrate_bps(default_bitrate_bps), + min_bitrate_bps(min_bitrate_bps), + max_bitrate_bps(max_bitrate_bps) { + RTC_DCHECK_GT(sample_rate_hz, 0); + RTC_DCHECK_GT(num_channels, 0); + RTC_DCHECK_GE(min_bitrate_bps, 0); + RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps); + RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps); +} + +} // namespace webrtc |