| Index: webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
|
| diff --git a/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc b/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..7d29883b391433d3666bd8f876b3dde18a3ee6d6
|
| --- /dev/null
|
| +++ b/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
|
| @@ -0,0 +1,70 @@
|
| +/*
|
| + * 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/opus/audio_encoder_opus_config.h"
|
| +
|
| +namespace webrtc {
|
| +
|
| +namespace {
|
| +
|
| +#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
|
| +// If we are on Android, iOS and/or ARM, use a lower complexity setting by
|
| +// default, to save encoder complexity.
|
| +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),
|
| + 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 (num_channels != 1 && num_channels != 2)
|
| + 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
|
|
|