webrtc/api/audio_codecs/opus/audio_encoder_opus.h
Jonas Oreland 6e2b9e2210 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 5/inf
Add field trials to audio api.

It is added as a pointer with nullptr as default.
It is not (yet) used anywhere.
Usage of field trials comes in subsequent patches.

Bug: webrtc:10335
Change-Id: Icbe22d95c356a6fefde34590f11ea63f005ab09e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255521
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36213}
2022-03-16 09:11:43 +00:00

44 lines
1.6 KiB
C++

/*
* 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_ENCODER_OPUS_H_
#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_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_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
#include "api/webrtc_key_value_config.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
// Opus encoder API for use as a template parameter to
// CreateAudioEncoderFactory<...>().
struct RTC_EXPORT AudioEncoderOpus {
using Config = AudioEncoderOpusConfig;
static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
const SdpAudioFormat& audio_format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const AudioEncoderOpusConfig& config,
int payload_type,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
};
} // namespace webrtc
#endif // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_