blob: 500cfd1797bc951ec75bacb751daa3cfecf59c04 [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.
*/
#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/string_to_number.h"
namespace webrtc {
rtc::Optional<AudioEncoderIsacFloat::Config> AudioEncoderIsacFloat::SdpToConfig(
const SdpAudioFormat& format) {
if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
(format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
format.num_channels == 1) {
Config config;
config.sample_rate_hz = format.clockrate_hz;
if (config.sample_rate_hz == 16000) {
// For sample rate 16 kHz, optionally use 60 ms frames, instead of the
// default 30 ms.
const auto ptime_iter = format.parameters.find("ptime");
if (ptime_iter != format.parameters.end()) {
const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
if (ptime && *ptime >= 60) {
config.frame_size_ms = 60;
}
}
}
return rtc::Optional<Config>(config);
} else {
return rtc::Optional<Config>();
}
}
void AudioEncoderIsacFloat::AppendSupportedEncoders(
std::vector<AudioCodecSpec>* specs) {
for (int sample_rate_hz : {16000, 32000}) {
const SdpAudioFormat fmt = {"ISAC", sample_rate_hz, 1};
const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
specs->push_back({fmt, info});
}
}
AudioCodecInfo AudioEncoderIsacFloat::QueryAudioEncoder(
const AudioEncoderIsacFloat::Config& config) {
RTC_DCHECK(config.IsOk());
constexpr int min_bitrate = 10000;
const int max_bitrate = config.sample_rate_hz == 16000 ? 32000 : 56000;
const int default_bitrate = max_bitrate;
return {config.sample_rate_hz, 1, default_bitrate, min_bitrate, max_bitrate};
}
std::unique_ptr<AudioEncoder> AudioEncoderIsacFloat::MakeAudioEncoder(
const AudioEncoderIsacFloat::Config& config,
int payload_type) {
RTC_DCHECK(config.IsOk());
AudioEncoderIsacFloatImpl::Config c;
c.sample_rate_hz = config.sample_rate_hz;
c.frame_size_ms = config.frame_size_ms;
c.payload_type = payload_type;
return rtc::MakeUnique<AudioEncoderIsacFloatImpl>(c);
}
} // namespace webrtc