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Unified Diff: webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc

Issue 2334613002: Adding BitrateController to audio network adaptor. (Closed)
Patch Set: adding a TODO Created 4 years, 3 months ago
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Index: webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
new file mode 100644
index 0000000000000000000000000000000000000000..4cc49e8c0d7717f943d9b297ff2d85a8ce764443
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
@@ -0,0 +1,68 @@
+/*
+ * 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 "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+namespace {
+// TODO(minyue): consider passing this from a higher layer through
+// SetConstraints().
+// L2(14B) + IPv4(20B) + UDP(8B) + RTP(12B) + SRTP_AUTH(10B) = 64B = 512 bits
+constexpr int kPacketOverheadBits = 512;
+}
+
+BitrateController::Config::Config(int initial_bitrate_bps,
+ int initial_frame_length_ms)
+ : initial_bitrate_bps(initial_bitrate_bps),
+ initial_frame_length_ms(initial_frame_length_ms) {}
+
+BitrateController::Config::~Config() = default;
+
+BitrateController::BitrateController(const Config& config)
+ : config_(config),
+ bitrate_bps_(config_.initial_bitrate_bps),
+ overhead_rate_bps_(kPacketOverheadBits * 1000 /
+ config_.initial_frame_length_ms) {
+ RTC_DCHECK_GT(bitrate_bps_, 0);
+ RTC_DCHECK_GT(overhead_rate_bps_, 0);
+}
+
+void BitrateController::MakeDecision(
+ const NetworkMetrics& metrics,
+ AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
+ // Decision on |bitrate_bps| should not have been made.
+ RTC_DCHECK(!config->bitrate_bps);
+
+ if (metrics.target_audio_bitrate_bps) {
+ int overhead_rate =
+ config->frame_length_ms
+ ? kPacketOverheadBits * 1000 / *config->frame_length_ms
+ : overhead_rate_bps_;
+ // If |metrics.target_audio_bitrate_bps| had included overhead, we would
+ // simply do:
+ // bitrate_bps_ = metrics.target_audio_bitrate_bps - overhead_rate;
+ // Follow https://bugs.chromium.org/p/webrtc/issues/detail?id=6315 to track
+ // progress regarding this.
+ // Now we assume that |metrics.target_audio_bitrate_bps| can handle the
+ // overhead of most recent packets.
+ bitrate_bps_ = std::max(0, *metrics.target_audio_bitrate_bps +
+ overhead_rate_bps_ - overhead_rate);
+ // TODO(minyue): apply a smoothing on the |overhead_rate_bps_|.
+ overhead_rate_bps_ = overhead_rate;
+ }
+ config->bitrate_bps = rtc::Optional<int>(bitrate_bps_);
+}
+
+} // namespace webrtc

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