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Break the ANA build-target into ANA and ANA-config

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This is done to solve a dependency-cycle with the RtcEventLog - now the RtcEventLog can depend on the config part of ANA, and be able to peer inside, while the implementation part of ANA can invoke the RtcEventLog.

BUG=webrtc:8111
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/3010343002
Cr-Commit-Position: refs/heads/master@{#19793}
This commit is contained in:
eladalon 2017-09-12 04:38:25 -07:00 committed by Commit Bot
parent adfc2795bb
commit 1e7dd31001
4 changed files with 54 additions and 20 deletions
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15
webrtc/modules/audio_coding/BUILD.gn
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@ -881,9 +881,18 @@ if (rtc_enable_protobuf) {
}
}
rtc_static_library("audio_network_adaptor_config") {
sources = [
"audio_network_adaptor/audio_network_adaptor_config.cc",
"audio_network_adaptor/include/audio_network_adaptor_config.h",
]
deps = [
"../../api:optional",
]
}
rtc_static_library("audio_network_adaptor") {
sources = [
"audio_network_adaptor/audio_network_adaptor.cc",
"audio_network_adaptor/audio_network_adaptor_impl.cc",
"audio_network_adaptor/audio_network_adaptor_impl.h",
"audio_network_adaptor/bitrate_controller.cc",
@ -910,6 +919,10 @@ rtc_static_library("audio_network_adaptor") {
"audio_network_adaptor/util/threshold_curve.h",
]
public_deps = [
":audio_network_adaptor_config",
]
deps = [
"../..:webrtc_common",
"../../api:optional",

2
webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.cc → webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc
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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
namespace webrtc {

19
webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
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@ -13,27 +13,10 @@
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/optional.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
namespace webrtc {
struct AudioEncoderRuntimeConfig {
AudioEncoderRuntimeConfig();
AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other);
~AudioEncoderRuntimeConfig();
rtc::Optional<int> bitrate_bps;
rtc::Optional<int> frame_length_ms;
// Note: This is what we tell the encoder. It doesn't have to reflect
// the actual NetworkMetrics; it's subject to our decision.
rtc::Optional<float> uplink_packet_loss_fraction;
rtc::Optional<bool> enable_fec;
rtc::Optional<bool> enable_dtx;
// Some encoders can encode fewer channels than the actual input to make
// better use of the bandwidth. |num_channels| sets the number of channels
// to encode.
rtc::Optional<size_t> num_channels;
};
// An AudioNetworkAdaptor optimizes the audio experience by suggesting a
// suitable runtime configuration (bit rate, frame length, FEC, etc.) to the
// encoder based on network metrics.

38
webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h Normal file
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@ -0,0 +1,38 @@
/*
* Copyright (c) 2017 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
#include "webrtc/api/optional.h"
namespace webrtc {
struct AudioEncoderRuntimeConfig {
AudioEncoderRuntimeConfig();
AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other);
~AudioEncoderRuntimeConfig();
rtc::Optional<int> bitrate_bps;
rtc::Optional<int> frame_length_ms;
// Note: This is what we tell the encoder. It doesn't have to reflect
// the actual NetworkMetrics; it's subject to our decision.
rtc::Optional<float> uplink_packet_loss_fraction;
rtc::Optional<bool> enable_fec;
rtc::Optional<bool> enable_dtx;
// Some encoders can encode fewer channels than the actual input to make
// better use of the bandwidth. |num_channels| sets the number of channels
// to encode.
rtc::Optional<size_t> num_channels;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_