blob: 6850926c27925b421fa47d732fb295b8c2d8bf4e [file] [log] [blame]
/*
* 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 "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
#include <algorithm>
#include "rtc_base/checks.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace audio_network_adaptor {
BitrateController::Config::Config(int initial_bitrate_bps,
int initial_frame_length_ms,
int fl_increase_overhead_offset,
int fl_decrease_overhead_offset)
: initial_bitrate_bps(initial_bitrate_bps),
initial_frame_length_ms(initial_frame_length_ms),
fl_increase_overhead_offset(fl_increase_overhead_offset),
fl_decrease_overhead_offset(fl_decrease_overhead_offset) {}
BitrateController::Config::~Config() = default;
BitrateController::BitrateController(const Config& config)
: config_(config),
bitrate_bps_(config_.initial_bitrate_bps),
frame_length_ms_(config_.initial_frame_length_ms) {
RTC_DCHECK_GT(bitrate_bps_, 0);
RTC_DCHECK_GT(frame_length_ms_, 0);
}
BitrateController::~BitrateController() = default;
void BitrateController::UpdateNetworkMetrics(
const NetworkMetrics& network_metrics) {
if (network_metrics.target_audio_bitrate_bps)
target_audio_bitrate_bps_ = network_metrics.target_audio_bitrate_bps;
if (network_metrics.overhead_bytes_per_packet)
overhead_bytes_per_packet_ = network_metrics.overhead_bytes_per_packet;
}
void BitrateController::MakeDecision(AudioEncoderRuntimeConfig* config) {
// Decision on |bitrate_bps| should not have been made.
RTC_DCHECK(!config->bitrate_bps);
if (target_audio_bitrate_bps_ && overhead_bytes_per_packet_) {
// Current implementation of BitrateController can only work when
// |metrics.target_audio_bitrate_bps| includes overhead is enabled. This is
// currently governed by the following field trial.
RTC_DCHECK(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead"));
if (config->frame_length_ms)
frame_length_ms_ = *config->frame_length_ms;
int offset = config->last_fl_change_increase
? config_.fl_increase_overhead_offset
: config_.fl_decrease_overhead_offset;
// Check that
// -(*overhead_bytes_per_packet_) <= offset <= (*overhead_bytes_per_packet_)
RTC_DCHECK_GE(*overhead_bytes_per_packet_, -offset);
RTC_DCHECK_LE(offset, *overhead_bytes_per_packet_);
int overhead_rate_bps = static_cast<int>(
(*overhead_bytes_per_packet_ + offset) * 8 * 1000 / frame_length_ms_);
bitrate_bps_ = std::max(0, *target_audio_bitrate_bps_ - overhead_rate_bps);
}
config->bitrate_bps = bitrate_bps_;
}
} // namespace audio_network_adaptor
} // namespace webrtc