/* * 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_fix.h" #include "webrtc/common_types.h" #include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h" #include "webrtc/rtc_base/ptr_util.h" #include "webrtc/rtc_base/string_to_number.h" namespace webrtc { rtc::Optional AudioEncoderIsacFix::SdpToConfig( const SdpAudioFormat& format) { if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 && format.clockrate_hz == 16000 && format.num_channels == 1) { Config config; const auto ptime_iter = format.parameters.find("ptime"); if (ptime_iter != format.parameters.end()) { const auto ptime = rtc::StringToNumber(ptime_iter->second); if (ptime && *ptime >= 60) { config.frame_size_ms = 60; } } return rtc::Optional(config); } else { return rtc::Optional(); } } void AudioEncoderIsacFix::AppendSupportedEncoders( std::vector* specs) { const SdpAudioFormat fmt = {"ISAC", 16000, 1}; const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt)); specs->push_back({fmt, info}); } AudioCodecInfo AudioEncoderIsacFix::QueryAudioEncoder( AudioEncoderIsacFix::Config config) { RTC_DCHECK(config.IsOk()); return {16000, 1, 32000, 10000, 32000}; } std::unique_ptr AudioEncoderIsacFix::MakeAudioEncoder( AudioEncoderIsacFix::Config config, int payload_type) { RTC_DCHECK(config.IsOk()); AudioEncoderIsacFixImpl::Config c; c.frame_size_ms = config.frame_size_ms; c.payload_type = payload_type; return rtc::MakeUnique(c); } } // namespace webrtc