4c862e781b
Bug: webrtc:356878416 Change-Id: Iae9369eb6a2ae09a707854a18d909eed453bbfd0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359960 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42874}
58 lines
2.0 KiB
C++
58 lines
2.0 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_DECODER_OPUS_H_
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#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_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_decoder.h"
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#include "api/audio_codecs/audio_format.h"
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#include "api/environment/environment.h"
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#include "api/field_trials_view.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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// Opus decoder API for use as a template parameter to
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// CreateAudioDecoderFactory<...>().
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struct RTC_EXPORT AudioDecoderOpus {
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struct Config {
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bool IsOk() const; // Checks if the values are currently OK.
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int sample_rate_hz = 48000;
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int num_channels = 1;
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};
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static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
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static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
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[[deprecated("bugs.webrtc.org/356878416")]] //
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static std::unique_ptr<AudioDecoder>
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MakeAudioDecoder(
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Config config,
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absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
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const FieldTrialsView* field_trials = nullptr);
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static std::unique_ptr<AudioDecoder> MakeAudioDecoder(const Environment& env,
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Config config);
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static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
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const Environment& env,
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Config config,
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absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
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return MakeAudioDecoder(env, config);
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}
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};
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} // namespace webrtc
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#endif // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
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