From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- .../api/audio_codecs/L16/audio_encoder_L16.h | 54 ++++++++++++++++++++++ 1 file changed, 54 insertions(+) create mode 100644 third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h (limited to 'third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h') diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h new file mode 100644 index 0000000000..47509849de --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h @@ -0,0 +1,54 @@ +/* + * 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_L16_AUDIO_ENCODER_L16_H_ +#define API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_ + +#include +#include + +#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/field_trials_view.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// L16 encoder API for use as a template parameter to +// CreateAudioEncoderFactory<...>(). +struct RTC_EXPORT AudioEncoderL16 { + struct Config { + bool IsOk() const { + return (sample_rate_hz == 8000 || sample_rate_hz == 16000 || + sample_rate_hz == 32000 || sample_rate_hz == 48000) && + num_channels >= 1 && + num_channels <= AudioEncoder::kMaxNumberOfChannels && + frame_size_ms > 0 && frame_size_ms <= 120 && + frame_size_ms % 10 == 0; + } + int sample_rate_hz = 8000; + int num_channels = 1; + int frame_size_ms = 10; + }; + static absl::optional SdpToConfig(const SdpAudioFormat& audio_format); + static void AppendSupportedEncoders(std::vector* specs); + static AudioCodecInfo QueryAudioEncoder(const Config& config); + static std::unique_ptr MakeAudioEncoder( + const Config& config, + int payload_type, + absl::optional codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr); +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_ -- cgit v1.2.3