| Index: webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc
|
| diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc b/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..e06b46f3a06852b32779b28ffd0a4a01da02fe17
|
| --- /dev/null
|
| +++ b/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc
|
| @@ -0,0 +1,56 @@
|
| +/*
|
| + * Copyright (c) 2016 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 <algorithm>
|
| +
|
| +#include "webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h"
|
| +#include "webrtc/base/checks.h"
|
| +
|
| +namespace webrtc {
|
| +
|
| +ChannelController::Config::Config(size_t num_encoder_channels,
|
| + size_t intial_channels_to_encode,
|
| + int channel_1_to_2_bandwidth_bps,
|
| + int channel_2_to_1_bandwidth_bps)
|
| + : num_encoder_channels(num_encoder_channels),
|
| + intial_channels_to_encode(intial_channels_to_encode),
|
| + channel_1_to_2_bandwidth_bps(channel_1_to_2_bandwidth_bps),
|
| + channel_2_to_1_bandwidth_bps(channel_2_to_1_bandwidth_bps) {}
|
| +
|
| +ChannelController::ChannelController(const Config& config)
|
| + : config_(config), channels_to_encode_(config_.intial_channels_to_encode) {
|
| + RTC_DCHECK_GT(config_.intial_channels_to_encode, 0lu);
|
| + // Currently, we require |intial_channels_to_encode| to be <= 2.
|
| + RTC_DCHECK_LE(config_.intial_channels_to_encode, 2lu);
|
| + RTC_DCHECK_GE(config_.num_encoder_channels,
|
| + config_.intial_channels_to_encode);
|
| +}
|
| +
|
| +void ChannelController::MakeDecision(
|
| + const NetworkMetrics& metrics,
|
| + AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
|
| + // Decision on |num_channels| should not have been made.
|
| + RTC_DCHECK(!config->num_channels);
|
| +
|
| + if (metrics.uplink_bandwidth_bps) {
|
| + if (channels_to_encode_ == 2 &&
|
| + *metrics.uplink_bandwidth_bps <= config_.channel_2_to_1_bandwidth_bps) {
|
| + channels_to_encode_ = 1;
|
| + } else if (channels_to_encode_ == 1 &&
|
| + *metrics.uplink_bandwidth_bps >=
|
| + config_.channel_1_to_2_bandwidth_bps) {
|
| + channels_to_encode_ =
|
| + std::min(static_cast<size_t>(2), config_.num_encoder_channels);
|
| + }
|
| + }
|
| + config->num_channels = rtc::Optional<size_t>(channels_to_encode_);
|
| +}
|
| +
|
| +} // namespace webrtc
|
|
|