| /* |
| * 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 |