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Diffstat (limited to '')
| -rw-r--r-- | third_party/libwebrtc/api/audio_options.cc | 107 |
1 files changed, 107 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_options.cc b/third_party/libwebrtc/api/audio_options.cc new file mode 100644 index 0000000000..658515062c --- /dev/null +++ b/third_party/libwebrtc/api/audio_options.cc @@ -0,0 +1,107 @@ +/* + * Copyright (c) 2018 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_options.h" + +#include "api/array_view.h" +#include "rtc_base/strings/string_builder.h" + +namespace cricket { +namespace { + +template <class T> +void ToStringIfSet(rtc::SimpleStringBuilder* result, + const char* key, + const absl::optional<T>& val) { + if (val) { + (*result) << key << ": " << *val << ", "; + } +} + +template <typename T> +void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) { + if (o) { + *s = o; + } +} + +} // namespace + +AudioOptions::AudioOptions() = default; +AudioOptions::~AudioOptions() = default; + +void AudioOptions::SetAll(const AudioOptions& change) { + SetFrom(&echo_cancellation, change.echo_cancellation); +#if defined(WEBRTC_IOS) + SetFrom(&ios_force_software_aec_HACK, change.ios_force_software_aec_HACK); +#endif + SetFrom(&auto_gain_control, change.auto_gain_control); + SetFrom(&noise_suppression, change.noise_suppression); + SetFrom(&highpass_filter, change.highpass_filter); + SetFrom(&stereo_swapping, change.stereo_swapping); + SetFrom(&audio_jitter_buffer_max_packets, + change.audio_jitter_buffer_max_packets); + SetFrom(&audio_jitter_buffer_fast_accelerate, + change.audio_jitter_buffer_fast_accelerate); + SetFrom(&audio_jitter_buffer_min_delay_ms, + change.audio_jitter_buffer_min_delay_ms); + SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe); + SetFrom(&audio_network_adaptor, change.audio_network_adaptor); + SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config); + SetFrom(&init_recording_on_send, change.init_recording_on_send); +} + +bool AudioOptions::operator==(const AudioOptions& o) const { + return echo_cancellation == o.echo_cancellation && +#if defined(WEBRTC_IOS) + ios_force_software_aec_HACK == o.ios_force_software_aec_HACK && +#endif + auto_gain_control == o.auto_gain_control && + noise_suppression == o.noise_suppression && + highpass_filter == o.highpass_filter && + stereo_swapping == o.stereo_swapping && + audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets && + audio_jitter_buffer_fast_accelerate == + o.audio_jitter_buffer_fast_accelerate && + audio_jitter_buffer_min_delay_ms == + o.audio_jitter_buffer_min_delay_ms && + combined_audio_video_bwe == o.combined_audio_video_bwe && + audio_network_adaptor == o.audio_network_adaptor && + audio_network_adaptor_config == o.audio_network_adaptor_config && + init_recording_on_send == o.init_recording_on_send; +} + +std::string AudioOptions::ToString() const { + char buffer[1024]; + rtc::SimpleStringBuilder result(buffer); + result << "AudioOptions {"; + ToStringIfSet(&result, "aec", echo_cancellation); +#if defined(WEBRTC_IOS) + ToStringIfSet(&result, "ios_force_software_aec_HACK", + ios_force_software_aec_HACK); +#endif + ToStringIfSet(&result, "agc", auto_gain_control); + ToStringIfSet(&result, "ns", noise_suppression); + ToStringIfSet(&result, "hf", highpass_filter); + ToStringIfSet(&result, "swap", stereo_swapping); + ToStringIfSet(&result, "audio_jitter_buffer_max_packets", + audio_jitter_buffer_max_packets); + ToStringIfSet(&result, "audio_jitter_buffer_fast_accelerate", + audio_jitter_buffer_fast_accelerate); + ToStringIfSet(&result, "audio_jitter_buffer_min_delay_ms", + audio_jitter_buffer_min_delay_ms); + ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe); + ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor); + ToStringIfSet(&result, "init_recording_on_send", init_recording_on_send); + result << "}"; + return result.str(); +} + +} // namespace cricket |