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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc | 74 |
1 files changed, 74 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc new file mode 100644 index 0000000000..56a6c4da6a --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc @@ -0,0 +1,74 @@ +/* + * 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 <memory> +#include <vector> + +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h" +#include "rtc_base/numerics/safe_conversions.h" +#include "rtc_base/numerics/safe_minmax.h" +#include "rtc_base/string_to_number.h" + +namespace webrtc { + +absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig( + const SdpAudioFormat& format) { + if (!absl::EqualsIgnoreCase(format.name, "g722") || + format.clockrate_hz != 8000) { + return absl::nullopt; + } + + AudioEncoderG722Config config; + config.num_channels = rtc::checked_cast<int>(format.num_channels); + auto ptime_iter = format.parameters.find("ptime"); + if (ptime_iter != format.parameters.end()) { + auto ptime = rtc::StringToNumber<int>(ptime_iter->second); + if (ptime && *ptime > 0) { + const int whole_packets = *ptime / 10; + config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60); + } + } + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return absl::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, rtc::dchecked_cast<size_t>(config.num_channels), + 64000 * config.num_channels}; +} + +std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder( + const AudioEncoderG722Config& config, + int payload_type, + absl::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 |