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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/opus/audio_decoder_opus.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/opus/audio_decoder_opus.cc | 86 |
1 files changed, 86 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.cc new file mode 100644 index 0000000000..efc9a73546 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.cc @@ -0,0 +1,86 @@ +/* + * 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/opus/audio_decoder_opus.h" + +#include <memory> +#include <utility> +#include <vector> + +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h" + +namespace webrtc { + +bool AudioDecoderOpus::Config::IsOk() const { + if (sample_rate_hz != 16000 && sample_rate_hz != 48000) { + // Unsupported sample rate. (libopus supports a few other rates as + // well; we can add support for them when needed.) + return false; + } + if (num_channels != 1 && num_channels != 2) { + return false; + } + return true; +} + +absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig( + const SdpAudioFormat& format) { + const auto num_channels = [&]() -> absl::optional<int> { + auto stereo = format.parameters.find("stereo"); + if (stereo != format.parameters.end()) { + if (stereo->second == "0") { + return 1; + } else if (stereo->second == "1") { + return 2; + } else { + return absl::nullopt; // Bad stereo parameter. + } + } + return 1; // Default to mono. + }(); + if (absl::EqualsIgnoreCase(format.name, "opus") && + format.clockrate_hz == 48000 && format.num_channels == 2 && + num_channels) { + Config config; + config.num_channels = *num_channels; + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return absl::nullopt; + } + return config; + } else { + return absl::nullopt; + } +} + +void AudioDecoderOpus::AppendSupportedDecoders( + std::vector<AudioCodecSpec>* specs) { + AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000}; + opus_info.allow_comfort_noise = false; + opus_info.supports_network_adaption = true; + SdpAudioFormat opus_format( + {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}}); + specs->push_back({std::move(opus_format), opus_info}); +} + +std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder( + Config config, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return nullptr; + } + return std::make_unique<AudioDecoderOpusImpl>(config.num_channels, + config.sample_rate_hz); +} + +} // namespace webrtc |