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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_encoder_opus_config.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.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 'third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc')
-rw-r--r-- | third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc | 75 |
1 files changed, 75 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc new file mode 100644 index 0000000000..a9ab924b38 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc @@ -0,0 +1,75 @@ +/* + * 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_encoder_opus_config.h" + +namespace webrtc { + +namespace { + +#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) +constexpr int kDefaultComplexity = 5; +#else +constexpr int kDefaultComplexity = 9; +#endif + +constexpr int kDefaultLowRateComplexity = + WEBRTC_OPUS_VARIABLE_COMPLEXITY ? 9 : kDefaultComplexity; + +} // namespace + +constexpr int AudioEncoderOpusConfig::kDefaultFrameSizeMs; +constexpr int AudioEncoderOpusConfig::kMinBitrateBps; +constexpr int AudioEncoderOpusConfig::kMaxBitrateBps; + +AudioEncoderOpusConfig::AudioEncoderOpusConfig() + : frame_size_ms(kDefaultFrameSizeMs), + sample_rate_hz(48000), + num_channels(1), + application(ApplicationMode::kVoip), + bitrate_bps(32000), + fec_enabled(false), + cbr_enabled(false), + max_playback_rate_hz(48000), + complexity(kDefaultComplexity), + low_rate_complexity(kDefaultLowRateComplexity), + complexity_threshold_bps(12500), + complexity_threshold_window_bps(1500), + dtx_enabled(false), + uplink_bandwidth_update_interval_ms(200), + payload_type(-1) {} +AudioEncoderOpusConfig::AudioEncoderOpusConfig(const AudioEncoderOpusConfig&) = + default; +AudioEncoderOpusConfig::~AudioEncoderOpusConfig() = default; +AudioEncoderOpusConfig& AudioEncoderOpusConfig::operator=( + const AudioEncoderOpusConfig&) = default; + +bool AudioEncoderOpusConfig::IsOk() const { + if (frame_size_ms <= 0 || frame_size_ms % 10 != 0) + return false; + if (sample_rate_hz != 16000 && sample_rate_hz != 48000) { + // Unsupported input sample rate. (libopus supports a few other rates as + // well; we can add support for them when needed.) + return false; + } + if (num_channels >= 255) { + return false; + } + if (!bitrate_bps) + return false; + if (*bitrate_bps < kMinBitrateBps || *bitrate_bps > kMaxBitrateBps) + return false; + if (complexity < 0 || complexity > 10) + return false; + if (low_rate_complexity < 0 || low_rate_complexity > 10) + return false; + return true; +} +} // namespace webrtc |