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rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
54 lines
1.9 KiB
C++
54 lines
1.9 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_L16_AUDIO_ENCODER_L16_H_
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#define API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_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/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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// L16 encoder API for use as a template parameter to
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// CreateAudioEncoderFactory<...>().
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struct RTC_EXPORT AudioEncoderL16 {
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struct Config {
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bool IsOk() const {
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return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
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sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
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num_channels >= 1 &&
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num_channels <= AudioEncoder::kMaxNumberOfChannels &&
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frame_size_ms > 0 && frame_size_ms <= 120 &&
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frame_size_ms % 10 == 0;
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}
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int sample_rate_hz = 8000;
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int num_channels = 1;
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int frame_size_ms = 10;
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};
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static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
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static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
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static AudioCodecInfo QueryAudioEncoder(const Config& config);
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static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
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const Config& config,
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int payload_type,
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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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};
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} // namespace webrtc
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#endif // API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
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