blob: 244e59b11d4b683f9ed8c7678844231849c89611 [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 WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
#define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
namespace webrtc {
// iSAC encoder API (floating-point implementation) for use as a template
// parameter to CreateAudioEncoderFactory<...>().
//
// NOTE: This struct is still under development and may change without notice.
struct AudioEncoderIsacFloat {
struct Config {
bool IsOk() const {
return (sample_rate_hz == 16000 &&
(frame_size_ms == 30 || frame_size_ms == 60)) ||
(sample_rate_hz == 32000 && frame_size_ms == 30);
}
int sample_rate_hz = 16000;
int frame_size_ms = 30;
};
static rtc::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);
};
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_