blob: 47509849deb53d7c301b4b08bf7e72bde0447f77 [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_L16_AUDIO_ENCODER_L16_H_
#define API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_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/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<Config> SdpToConfig(const SdpAudioFormat& audio_format);
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
static AudioCodecInfo QueryAudioEncoder(const Config& config);
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
const Config& config,
int payload_type,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc
#endif // API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_