Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 73 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
new file mode 100644
index 0000000000..88ca38d074
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
@@ -0,0 +1,73 @@
+/*
+ *  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) {
+    RTC_DCHECK_GT(*network_metrics.overhead_bytes_per_packet, 0);
+    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_) {
+    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