blob: ec0f61d5bbd17807d9e100b6d689844ab6048fac [file] [log] [blame]
/*
* 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_OPUS_AUDIO_DECODER_OPUS_H_
#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
#include <memory>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
// Opus decoder API for use as a template parameter to
// CreateAudioDecoderFactory<...>().
struct RTC_EXPORT AudioDecoderOpus {
struct Config {
bool IsOk() const; // Checks if the values are currently OK.
int sample_rate_hz = 48000;
int num_channels = 1;
};
static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
Config config,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
};
} // namespace webrtc
#endif // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_