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https://github.com/mollyim/webrtc.git
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e482ff8f70
These things are no longer brand new, so it makes even less sense than it once did to warn users that they may change at any time. Bug: none Change-Id: I43a6915d9e00fbfef30fdb89869873b129297c8d No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/106980 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25283}
42 lines
1.5 KiB
C++
42 lines
1.5 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
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#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
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#include <memory>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/audio_codecs/audio_codec_pair_id.h"
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#include "api/audio_codecs/audio_encoder.h"
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#include "api/audio_codecs/audio_format.h"
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#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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// Opus encoder API for use as a template parameter to
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// CreateAudioEncoderFactory<...>().
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struct RTC_EXPORT AudioEncoderOpus {
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using Config = AudioEncoderOpusConfig;
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static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
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const SdpAudioFormat& audio_format);
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static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
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static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
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static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
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const AudioEncoderOpusConfig& config,
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int payload_type,
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absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
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};
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} // namespace webrtc
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#endif // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
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