Source code
Revision control
Copy as Markdown
Other Tools
/*
* 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