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/*
* 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_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
#define API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_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"
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 {
switch (sample_rate_hz) {
case 16000:
if (frame_size_ms != 30 && frame_size_ms != 60) {
return false;
}
if (bit_rate < 10000 || bit_rate > 32000) {
return false;
}
return true;
case 32000:
if (frame_size_ms != 30) {
return false;
}
if (bit_rate < 10000 || bit_rate > 56000) {
return false;
}
return true;
default:
return false;
}
}
int sample_rate_hz = 16000;
int frame_size_ms = 30;
int bit_rate = 32000; // Limit on short-term average bit rate, in bits/s.
};
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);
};
} // namespace webrtc
#endif // API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_