From fc433e65463e3a2e3cc43b3f8cb68849b00f381f Mon Sep 17 00:00:00 2001
From: solenberg <solenberg@webrtc.org>
Date: Fri, 9 Sep 2016 11:04:53 -0700
Subject: [PATCH] Don't use VoE legacy APIs in force_mic_volume_max tool.

BUG=webrtc:4690

Committed: https://crrev.com/ae9f2bdcef7f3a338ece6f57744c8c8f74d15483
Review-Url: https://codereview.webrtc.org/2268183007
Cr-Original-Commit-Position: refs/heads/master@{#14135}
Cr-Commit-Position: refs/heads/master@{#14169}
---
 webrtc/tools/BUILD.gn                         | 31 ++++++-----
 webrtc/tools/DEPS                             |  1 +
 .../force_mic_volume_max.cc                   | 54 +++++++++++--------
 webrtc/tools/tools.gyp                        |  2 +-
 4 files changed, 52 insertions(+), 36 deletions(-)

diff --git a/webrtc/tools/BUILD.gn b/webrtc/tools/BUILD.gn
index 37f015c1ea..eba0122338 100644
--- a/webrtc/tools/BUILD.gn
+++ b/webrtc/tools/BUILD.gn
@@ -117,22 +117,25 @@ rtc_executable("frame_editor") {
   ]
 }
 
-rtc_executable("force_mic_volume_max") {
-  sources = [
-    "force_mic_volume_max/force_mic_volume_max.cc",
-  ]
+# It doesn't make sense to build this tool without the ADM enabled.
+if (rtc_include_internal_audio_device) {
+  rtc_executable("force_mic_volume_max") {
+    sources = [
+      "force_mic_volume_max/force_mic_volume_max.cc",
+    ]
 
-  if (is_clang) {
-    # Suppress warnings from the Chromium Clang plugin.
-    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+    if (is_clang) {
+      # Suppress warnings from the Chromium Clang plugin.
+      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+    }
+
+    deps = [
+      "../modules/audio_device:audio_device",
+      "../system_wrappers:system_wrappers_default",
+      "//build/win:default_exe_manifest",
+    ]
   }
-
-  deps = [
-    "../system_wrappers:system_wrappers_default",
-    "../voice_engine",
-    "//build/win:default_exe_manifest",
-  ]
 }
 
 if (rtc_enable_protobuf) {
diff --git a/webrtc/tools/DEPS b/webrtc/tools/DEPS
index 035245095c..cc33de1f96 100644
--- a/webrtc/tools/DEPS
+++ b/webrtc/tools/DEPS
@@ -2,6 +2,7 @@ include_rules = [
   "+webrtc/base",
   "+webrtc/call",
   "+webrtc/common_video",
+  "+webrtc/modules/audio_device",
   "+webrtc/modules/audio_processing",
   "+webrtc/modules/congestion_controller",
   "+webrtc/modules/rtp_rtcp",
diff --git a/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc b/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc
index 2bab2881bb..0a2347d5ba 100644
--- a/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc
+++ b/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc
@@ -12,33 +12,45 @@
 
 #include <stdio.h>
 
-#include "webrtc/test/channel_transport/channel_transport.h"
-#include "webrtc/voice_engine/include/voe_audio_processing.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_volume_control.h"
+#include "webrtc/modules/audio_device/include/audio_device.h"
 
-int main(int argc, char** argv) {
-  webrtc::VoiceEngine* voe = webrtc::VoiceEngine::Create();
-  if (voe == NULL) {
-    fprintf(stderr, "Failed to initialize voice engine.\n");
+using webrtc::AudioDeviceModule;
+
+#if defined(_WIN32)
+#define DEFAULT_INPUT_DEVICE (AudioDeviceModule::kDefaultCommunicationDevice)
+#else
+#define DEFAULT_INPUT_DEVICE (0)
+#endif
+
+int main(int /*argc*/, char** /*argv*/) {
+  // Create and initialize the ADM.
+  rtc::scoped_refptr<AudioDeviceModule> adm(
+      AudioDeviceModule::Create(1, AudioDeviceModule::kPlatformDefaultAudio));
+  if (!adm.get()) {
+    fprintf(stderr, "Failed to create Audio Device Module.\n");
+    return 1;
+  }
+  if (adm->Init() != 0) {
+    fprintf(stderr, "Failed to initialize Audio Device Module.\n");
+    return 1;
+  }
+  if (adm->SetRecordingDevice(DEFAULT_INPUT_DEVICE) != 0) {
+    fprintf(stderr, "Failed to set the default input device.\n");
+    return 1;
+  }
+  if (adm->InitMicrophone() != 0) {
+    fprintf(stderr, "Failed to to initialize the microphone.\n");
     return 1;
   }
 
-  webrtc::VoEBase* base = webrtc::VoEBase::GetInterface(voe);
-  webrtc::VoEVolumeControl* volume_control =
-      webrtc::VoEVolumeControl::GetInterface(voe);
-
-  if (base->Init() != 0) {
-    fprintf(stderr, "Failed to initialize voice engine base.\n");
+  // Set mic volume to max.
+  uint32_t max_vol = 0;
+  if (adm->MaxMicrophoneVolume(&max_vol) != 0) {
+    fprintf(stderr, "Failed to get max volume.\n");
     return 1;
   }
-  // Set to 0 first in case the mic is above 100%.
-  if (volume_control->SetMicVolume(0) != 0) {
-    fprintf(stderr, "Failed set volume to 0.\n");
-    return 1;
-  }
-  if (volume_control->SetMicVolume(255) != 0) {
-    fprintf(stderr, "Failed set volume to 255.\n");
+  if (adm->SetMicrophoneVolume(max_vol) != 0) {
+    fprintf(stderr, "Failed to set mic volume.\n");
     return 1;
   }
 
diff --git a/webrtc/tools/tools.gyp b/webrtc/tools/tools.gyp
index 12eff5de2a..d1ff42bcbe 100644
--- a/webrtc/tools/tools.gyp
+++ b/webrtc/tools/tools.gyp
@@ -90,7 +90,7 @@
       'target_name': 'force_mic_volume_max',
       'type': 'executable',
       'dependencies': [
-        '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
+        '<(webrtc_root)/modules/modules.gyp:audio_device',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
       ],
       'sources': [