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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.
*/
#include "api/audio_codecs/g722/audio_encoder_g722.h"
#include <stddef.h>
#include <map>
#include <memory>
#include <optional>
#include <vector>
#include "absl/strings/match.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/audio_codecs/g722/audio_encoder_g722_config.h"
#include "api/field_trials_view.h"
#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/string_to_number.h"
namespace webrtc {
std::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
const SdpAudioFormat& format) {
if (!absl::EqualsIgnoreCase(format.name, "g722") ||
format.clockrate_hz != 8000) {
return std::nullopt;
}
AudioEncoderG722Config config;
config.num_channels = checked_cast<int>(format.num_channels);
auto ptime_iter = format.parameters.find("ptime");
if (ptime_iter != format.parameters.end()) {
auto ptime = StringToNumber<int>(ptime_iter->second);
if (ptime && *ptime > 0) {
const int whole_packets = *ptime / 10;
config.frame_size_ms = SafeClamp<int>(whole_packets * 10, 10, 60);
}
}
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return std::nullopt;
}
return config;
}
void AudioEncoderG722::AppendSupportedEncoders(
std::vector<AudioCodecSpec>* specs) {
const SdpAudioFormat fmt = {"G722", 8000, 1};
const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
specs->push_back({fmt, info});
}
AudioCodecInfo AudioEncoderG722::QueryAudioEncoder(
const AudioEncoderG722Config& config) {
RTC_DCHECK(config.IsOk());
return {16000, dchecked_cast<size_t>(config.num_channels),
64000 * config.num_channels};
}
std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
const AudioEncoderG722Config& config,
int payload_type,
std::optional<AudioCodecPairId> /*codec_pair_id*/,
const FieldTrialsView* /* field_trials */) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return nullptr;
}
return std::make_unique<AudioEncoderG722Impl>(config, payload_type);
}
} // namespace webrtc
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