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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/opus/audio_encoder_opus.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/opus/audio_encoder_opus.cc | 44 |
1 files changed, 44 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.cc new file mode 100644 index 0000000000..5b6322da4c --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.cc @@ -0,0 +1,44 @@ +/* + * 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 "api/audio_codecs/opus/audio_encoder_opus.h" + +#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h" + +namespace webrtc { + +absl::optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig( + const SdpAudioFormat& format) { + return AudioEncoderOpusImpl::SdpToConfig(format); +} + +void AudioEncoderOpus::AppendSupportedEncoders( + std::vector<AudioCodecSpec>* specs) { + AudioEncoderOpusImpl::AppendSupportedEncoders(specs); +} + +AudioCodecInfo AudioEncoderOpus::QueryAudioEncoder( + const AudioEncoderOpusConfig& config) { + return AudioEncoderOpusImpl::QueryAudioEncoder(config); +} + +std::unique_ptr<AudioEncoder> AudioEncoderOpus::MakeAudioEncoder( + const AudioEncoderOpusConfig& config, + int payload_type, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return nullptr; + } + return AudioEncoderOpusImpl::MakeAudioEncoder(config, payload_type); +} + +} // namespace webrtc |