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https://github.com/mollyim/webrtc.git
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7eb0a5e210
In addition to the 48 kHz that we've always used. Bug: webrtc:10631 Change-Id: If73bf7ff9c1c0d22e0d1caa245128612850f8e41 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138268 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28104}
42 lines
1.4 KiB
C++
42 lines
1.4 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 "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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static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
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Config config,
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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_DECODER_OPUS_H_
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