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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/g711/audio_encoder_g711.h | |
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/g711/audio_encoder_g711.h | 54 |
1 files changed, 54 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.h b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.h new file mode 100644 index 0000000000..4b3eb845e0 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.h @@ -0,0 +1,54 @@ +/* + * 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. + */ + +#ifndef API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_ +#define API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_ + +#include <memory> +#include <vector> + +#include "absl/types/optional.h" +#include "api/audio_codecs/audio_codec_pair_id.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_format.h" +#include "api/field_trials_view.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// G711 encoder API for use as a template parameter to +// CreateAudioEncoderFactory<...>(). +struct RTC_EXPORT AudioEncoderG711 { + struct Config { + enum class Type { kPcmU, kPcmA }; + bool IsOk() const { + return (type == Type::kPcmU || type == Type::kPcmA) && + frame_size_ms > 0 && frame_size_ms % 10 == 0 && + num_channels >= 1 && + num_channels <= AudioEncoder::kMaxNumberOfChannels; + } + Type type = Type::kPcmU; + int num_channels = 1; + int frame_size_ms = 20; + }; + static absl::optional<AudioEncoderG711::Config> SdpToConfig( + const SdpAudioFormat& audio_format); + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); + static AudioCodecInfo QueryAudioEncoder(const Config& config); + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + const Config& config, + int payload_type, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr); +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_ |