Re-land "Remove <(webrtc_root) from source file entries."
Changes differing from https://webrtc-codereview.appspot.com/37859004:
* I put the include_tests==1 stuff of audio_coding.gypi in its
own audio_coding_tests.gypi file, including the Android and isolate
targets which were incorrectly located in the previous CL
* I moved the bwe utilities in remote_bitrate_estimator.gypi
into include_tests==1 since they depend on test.gyp after I
cleaned up the duplicated inclusion of rtp_file_reader.cc
R=stefan@webrtc.org
TBR=tina.legrand@webrtc.org
TESTED=Passing gyp and compile using:
webrtc/build/gyp_webrtc -Dinclude_tests=1
webrtc/build/gyp_webrtc -Dinclude_tests=0
I also setup a Chromium checkout with my checkout mounted in
third_party/webrtc and ran build/gyp_chromium successfully.
BUG=4185
Review URL: https://webrtc-codereview.appspot.com/33159004
Cr-Commit-Position: refs/heads/master@{#8205}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8205 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.isolate b/webrtc/modules/audio_coding/audio_codec_speed_tests.isolate
similarity index 100%
rename from webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.isolate
rename to webrtc/modules/audio_coding/audio_codec_speed_tests.isolate
diff --git a/webrtc/modules/audio_coding/audio_coding.gypi b/webrtc/modules/audio_coding/audio_coding.gypi
new file mode 100644
index 0000000..0a500d9
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_coding.gypi
@@ -0,0 +1,29 @@
+# Copyright (c) 2015 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.
+
+{
+ 'includes': [
+ '../../build/common.gypi',
+ 'codecs/interfaces.gypi',
+ 'codecs/cng/cng.gypi',
+ 'codecs/g711/g711.gypi',
+ 'codecs/g722/g722.gypi',
+ 'codecs/ilbc/ilbc.gypi',
+ 'codecs/isac/main/source/isac.gypi',
+ 'codecs/isac/fix/source/isacfix.gypi',
+ 'codecs/pcm16b/pcm16b.gypi',
+ 'codecs/red/red.gypi',
+ 'main/acm2/audio_coding_module.gypi',
+ 'neteq/neteq.gypi',
+ ],
+ 'conditions': [
+ ['include_opus==1', {
+ 'includes': ['codecs/opus/opus.gypi',],
+ }],
+ ],
+}
diff --git a/webrtc/modules/audio_coding/audio_coding_tests.gypi b/webrtc/modules/audio_coding/audio_coding_tests.gypi
new file mode 100644
index 0000000..86a92c5
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_coding_tests.gypi
@@ -0,0 +1,72 @@
+# Copyright (c) 2015 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.
+
+{
+ 'includes': [
+ '../../build/common.gypi',
+ 'codecs/isac/isac_test.gypi',
+ 'codecs/isac/isacfix_test.gypi',
+ ],
+ 'targets': [
+ {
+ 'target_name': 'audio_codec_speed_tests',
+ 'type': '<(gtest_target_type)',
+ 'dependencies': [
+ 'audio_processing',
+ 'iSACFix',
+ 'webrtc_opus',
+ '<(DEPTH)/testing/gtest.gyp:gtest',
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/test/test.gyp:test_support_main',
+ ],
+ 'sources': [
+ 'codecs/isac/fix/test/isac_speed_test.cc',
+ 'codecs/opus/opus_speed_test.cc',
+ 'codecs/tools/audio_codec_speed_test.h',
+ 'codecs/tools/audio_codec_speed_test.cc',
+ ],
+ 'conditions': [
+ ['OS=="android"', {
+ 'dependencies': [
+ '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
+ ],
+ }],
+ ],
+ },
+ ],
+ 'conditions': [
+ ['OS=="android"', {
+ 'targets': [
+ {
+ 'target_name': 'audio_codec_speed_tests_apk_target',
+ 'type': 'none',
+ 'dependencies': [
+ '<(apk_tests_path):audio_codec_speed_tests_apk',
+ ],
+ },
+ ],
+ }],
+ ['test_isolation_mode != "noop"', {
+ 'targets': [
+ {
+ 'target_name': 'audio_codec_speed_tests_run',
+ 'type': 'none',
+ 'dependencies': [
+ 'audio_codec_speed_tests',
+ ],
+ 'includes': [
+ '../../build/isolate.gypi',
+ ],
+ 'sources': [
+ 'audio_codec_speed_tests.isolate',
+ ],
+ },
+ ],
+ }],
+ ],
+}
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.gypi b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.gypi
deleted file mode 100644
index e296634..0000000
--- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_tests.gypi
+++ /dev/null
@@ -1,66 +0,0 @@
-# Copyright (c) 2014 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.
-
-{
- 'targets': [
- {
- 'target_name': 'audio_codec_speed_tests',
- 'type': '<(gtest_target_type)',
- 'dependencies': [
- 'audio_processing',
- 'iSACFix',
- 'webrtc_opus',
- '<(DEPTH)/testing/gtest.gyp:gtest',
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/test/test.gyp:test_support_main',
- ],
- 'sources': [
- 'audio_codec_speed_test.h',
- 'audio_codec_speed_test.cc',
- '<(webrtc_root)/modules/audio_coding/codecs/opus/opus_speed_test.cc',
- '<(webrtc_root)/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc',
- ],
- 'conditions': [
- ['OS=="android"', {
- 'dependencies': [
- '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
- ],
- }],
- ],
- }],
- 'conditions': [
- ['OS=="android"', {
- 'targets': [
- {
- 'target_name': 'audio_codec_speed_tests_apk_target',
- 'type': 'none',
- 'dependencies': [
- '<(apk_tests_path):audio_codec_speed_tests_apk',
- ],
- },
- ],
- }],
- ['test_isolation_mode != "noop"', {
- 'targets': [
- {
- 'target_name': 'audio_codec_speed_tests_run',
- 'type': 'none',
- 'dependencies': [
- 'audio_codec_speed_tests',
- ],
- 'includes': [
- '../../../../build/isolate.gypi',
- ],
- 'sources': [
- 'audio_codec_speed_tests.isolate',
- ],
- },
- ],
- }],
- ],
-}
diff --git a/webrtc/modules/audio_processing/agc/test/activity_metric.cc b/webrtc/modules/audio_processing/agc/test/activity_metric.cc
deleted file mode 100644
index 474b553..0000000
--- a/webrtc/modules/audio_processing/agc/test/activity_metric.cc
+++ /dev/null
@@ -1,384 +0,0 @@
-/*
- * Copyright (c) 2012 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 <math.h>
-#include <stdio.h>
-#include <stdlib.h>
-
-#include <algorithm>
-
-#include "gflags/gflags.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_processing/agc/agc.h"
-#include "webrtc/modules/audio_processing/agc/agc_audio_proc.h"
-#include "webrtc/modules/audio_processing/agc/common.h"
-#include "webrtc/modules/audio_processing/agc/histogram.h"
-#include "webrtc/modules/audio_processing/agc/pitch_based_vad.h"
-#include "webrtc/modules/audio_processing/agc/standalone_vad.h"
-#include "webrtc/modules/audio_processing/agc/utility.h"
-#include "webrtc/modules/interface/module_common_types.h"
-
-static const int kAgcAnalWindowSamples = 100;
-static const double kDefaultActivityThreshold = 0.3;
-
-DEFINE_bool(standalone_vad, true, "enable stand-alone VAD");
-DEFINE_string(true_vad, "", "name of a file containing true VAD in 'int'"
- " format");
-DEFINE_string(video_vad, "", "name of a file containing video VAD (activity"
- " probabilities) in double format. One activity per 10ms is"
- " required. If no file is given the video information is not"
- " incorporated. Negative activity is interpreted as video is"
- " not adapted and the statistics are not computed during"
- " the learning phase. Note that the negative video activities"
- " are ONLY allowed at the beginning.");
-DEFINE_string(result, "", "name of a file to write the results. The results"
- " will be appended to the end of the file. This is optional.");
-DEFINE_string(audio_content, "", "name of a file where audio content is written"
- " to, in double format.");
-DEFINE_double(activity_threshold, kDefaultActivityThreshold,
- "Activity threshold");
-
-namespace webrtc {
-
-// TODO(turajs) A new CL will be committed soon where ExtractFeatures will
-// notify the caller of "silence" input, instead of bailing out. We would not
-// need the following function when such a change is made.
-
-// Add some dither to quiet frames. This avoids the ExtractFeatures skip a
-// silence frame. Otherwise true VAD would drift with respect to the audio.
-// We only consider mono inputs.
-static void DitherSilence(AudioFrame* frame) {
- ASSERT_EQ(1, frame->num_channels_);
- const double kRmsSilence = 5;
- const double sum_squared_silence = kRmsSilence * kRmsSilence *
- frame->samples_per_channel_;
- double sum_squared = 0;
- for (int n = 0; n < frame->samples_per_channel_; n++)
- sum_squared += frame->data_[n] * frame->data_[n];
- if (sum_squared <= sum_squared_silence) {
- for (int n = 0; n < frame->samples_per_channel_; n++)
- frame->data_[n] = (rand() & 0xF) - 8;
- }
-}
-
-class AgcStat {
- public:
- AgcStat()
- : video_index_(0),
- activity_threshold_(kDefaultActivityThreshold),
- audio_content_(Histogram::Create(kAgcAnalWindowSamples)),
- audio_processing_(new AgcAudioProc()),
- vad_(new PitchBasedVad()),
- standalone_vad_(StandaloneVad::Create()),
- audio_content_fid_(NULL) {
- for (int n = 0; n < kMaxNumFrames; n++)
- video_vad_[n] = 0.5;
- }
-
- ~AgcStat() {
- if (audio_content_fid_ != NULL) {
- fclose(audio_content_fid_);
- }
- }
-
- void set_audio_content_file(FILE* audio_content_fid) {
- audio_content_fid_ = audio_content_fid;
- }
-
- int AddAudio(const AudioFrame& frame, double p_video,
- int* combined_vad) {
- if (frame.num_channels_ != 1 ||
- frame.samples_per_channel_ !=
- kSampleRateHz / 100 ||
- frame.sample_rate_hz_ != kSampleRateHz)
- return -1;
- video_vad_[video_index_++] = p_video;
- AudioFeatures features;
- audio_processing_->ExtractFeatures(
- frame.data_, frame.samples_per_channel_, &features);
- if (FLAGS_standalone_vad) {
- standalone_vad_->AddAudio(frame.data_,
- frame.samples_per_channel_);
- }
- if (features.num_frames > 0) {
- double p[kMaxNumFrames] = {0.5, 0.5, 0.5, 0.5};
- if (FLAGS_standalone_vad) {
- standalone_vad_->GetActivity(p, kMaxNumFrames);
- }
- // TODO(turajs) combining and limiting are used in the source files as
- // well they can be moved to utility.
- // Combine Video and stand-alone VAD.
- for (int n = 0; n < features.num_frames; n++) {
- double p_active = p[n] * video_vad_[n];
- double p_passive = (1 - p[n]) * (1 - video_vad_[n]);
- p[n] = p_active / (p_active + p_passive);
- // Limit probabilities.
- p[n] = std::min(std::max(p[n], 0.01), 0.99);
- }
- if (vad_->VoicingProbability(features, p) < 0)
- return -1;
- for (int n = 0; n < features.num_frames; n++) {
- audio_content_->Update(features.rms[n], p[n]);
- double ac = audio_content_->AudioContent();
- if (audio_content_fid_ != NULL) {
- fwrite(&ac, sizeof(ac), 1, audio_content_fid_);
- }
- if (ac > kAgcAnalWindowSamples * activity_threshold_) {
- combined_vad[n] = 1;
- } else {
- combined_vad[n] = 0;
- }
- }
- video_index_ = 0;
- }
- return features.num_frames;
- }
-
- void Reset() {
- audio_content_->Reset();
- }
-
- void SetActivityThreshold(double activity_threshold) {
- activity_threshold_ = activity_threshold;
- }
-
- private:
- int video_index_;
- double activity_threshold_;
- double video_vad_[kMaxNumFrames];
- scoped_ptr<Histogram> audio_content_;
- scoped_ptr<AgcAudioProc> audio_processing_;
- scoped_ptr<PitchBasedVad> vad_;
- scoped_ptr<StandaloneVad> standalone_vad_;
-
- FILE* audio_content_fid_;
-};
-
-
-void void_main(int argc, char* argv[]) {
- webrtc::AgcStat agc_stat;
-
- FILE* pcm_fid = fopen(argv[1], "rb");
- ASSERT_TRUE(pcm_fid != NULL) << "Cannot open PCM file " << argv[1];
-
- if (argc < 2) {
- fprintf(stderr, "\nNot Enough arguments\n");
- }
-
- FILE* true_vad_fid = NULL;
- ASSERT_GT(FLAGS_true_vad.size(), 0u) << "Specify the file containing true "
- "VADs using --true_vad flag.";
- true_vad_fid = fopen(FLAGS_true_vad.c_str(), "rb");
- ASSERT_TRUE(true_vad_fid != NULL) << "Cannot open the active list " <<
- FLAGS_true_vad;
-
- FILE* results_fid = NULL;
- if (FLAGS_result.size() > 0) {
- // True if this is the first time writing to this function and we add a
- // header to the beginning of the file.
- bool write_header;
- // Open in the read mode. If it fails, the file doesn't exist and has to
- // write a header for it. Otherwise no need to write a header.
- results_fid = fopen(FLAGS_result.c_str(), "r");
- if (results_fid == NULL) {
- write_header = true;
- } else {
- fclose(results_fid);
- write_header = false;
- }
- // Open in append mode.
- results_fid = fopen(FLAGS_result.c_str(), "a");
- ASSERT_TRUE(results_fid != NULL) << "Cannot open the file, " <<
- FLAGS_result << ", to write the results.";
- // Write the header if required.
- if (write_header) {
- fprintf(results_fid, "%% Total Active, Misdetection, "
- "Total inactive, False Positive, On-sets, Missed segments, "
- "Average response\n");
- }
- }
-
- FILE* video_vad_fid = NULL;
- if (FLAGS_video_vad.size() > 0) {
- video_vad_fid = fopen(FLAGS_video_vad.c_str(), "rb");
- ASSERT_TRUE(video_vad_fid != NULL) << "Cannot open the file, " <<
- FLAGS_video_vad << " to read video-based VAD decisions.\n";
- }
-
- // AgsStat will be the owner of this file and will close it at its
- // destructor.
- FILE* audio_content_fid = NULL;
- if (FLAGS_audio_content.size() > 0) {
- audio_content_fid = fopen(FLAGS_audio_content.c_str(), "wb");
- ASSERT_TRUE(audio_content_fid != NULL) << "Cannot open file, " <<
- FLAGS_audio_content << " to write audio-content.\n";
- agc_stat.set_audio_content_file(audio_content_fid);
- }
-
- webrtc::AudioFrame frame;
- frame.num_channels_ = 1;
- frame.sample_rate_hz_ = 16000;
- frame.samples_per_channel_ = frame.sample_rate_hz_ / 100;
- const size_t kSamplesToRead = frame.num_channels_ *
- frame.samples_per_channel_;
-
- agc_stat.SetActivityThreshold(FLAGS_activity_threshold);
-
- int ret_val = 0;
- int num_frames = 0;
- int agc_vad[kMaxNumFrames];
- uint8_t true_vad[kMaxNumFrames];
- double p_video = 0.5;
- int total_active = 0;
- int total_passive = 0;
- int total_false_positive = 0;
- int total_missed_detection = 0;
- int onset_adaptation = 0;
- int num_onsets = 0;
- bool onset = false;
- uint8_t previous_true_vad = 0;
- int num_not_adapted = 0;
- int true_vad_index = 0;
- bool in_false_positive_region = false;
- int total_false_positive_duration = 0;
- bool video_adapted = false;
- while (kSamplesToRead == fread(frame.data_, sizeof(int16_t),
- kSamplesToRead, pcm_fid)) {
- assert(true_vad_index < kMaxNumFrames);
- ASSERT_EQ(1u, fread(&true_vad[true_vad_index], sizeof(*true_vad), 1,
- true_vad_fid))
- << "Size mismatch between True-VAD and the PCM file.\n";
- if (video_vad_fid != NULL) {
- ASSERT_EQ(1u, fread(&p_video, sizeof(p_video), 1, video_vad_fid)) <<
- "Not enough video-based VAD probabilities.";
- }
-
- // Negative video activity indicates that the video-based VAD is not yet
- // adapted. Disregards the learning phase in statistics.
- if (p_video < 0) {
- if (video_adapted) {
- fprintf(stderr, "Negative video probabilities ONLY allowed at the "
- "beginning of the sequence, not in the middle.\n");
- exit(1);
- }
- continue;
- } else {
- video_adapted = true;
- }
-
- num_frames++;
- uint8_t last_true_vad;
- if (true_vad_index == 0) {
- last_true_vad = previous_true_vad;
- } else {
- last_true_vad = true_vad[true_vad_index - 1];
- }
- if (last_true_vad == 1 && true_vad[true_vad_index] == 0) {
- agc_stat.Reset();
- }
- true_vad_index++;
-
- DitherSilence(&frame);
-
- ret_val = agc_stat.AddAudio(frame, p_video, agc_vad);
- ASSERT_GE(ret_val, 0);
-
- if (ret_val > 0) {
- ASSERT_TRUE(ret_val == true_vad_index);
- for (int n = 0; n < ret_val; n++) {
- if (true_vad[n] == 1) {
- total_active++;
- if (previous_true_vad == 0) {
- num_onsets++;
- onset = true;
- }
- if (agc_vad[n] == 0) {
- total_missed_detection++;
- if (onset)
- onset_adaptation++;
- } else {
- in_false_positive_region = false;
- onset = false;
- }
- } else if (true_vad[n] == 0) {
- // Check if |on_set| flag is still up. If so it means that we totally
- // missed an active region
- if (onset)
- num_not_adapted++;
- onset = false;
-
- total_passive++;
- if (agc_vad[n] == 1) {
- total_false_positive++;
- in_false_positive_region = true;
- }
- if (in_false_positive_region) {
- total_false_positive_duration++;
- }
- } else {
- ASSERT_TRUE(false) << "Invalid value for true-VAD.\n";
- }
- previous_true_vad = true_vad[n];
- }
- true_vad_index = 0;
- }
- }
-
- if (results_fid != NULL) {
- fprintf(results_fid, "%4d %4d %4d %4d %4d %4d %4.0f %4.0f\n",
- total_active,
- total_missed_detection,
- total_passive,
- total_false_positive,
- num_onsets,
- num_not_adapted,
- static_cast<float>(onset_adaptation) / (num_onsets + 1e-12),
- static_cast<float>(total_false_positive_duration) /
- (total_passive + 1e-12));
- }
- fprintf(stdout, "%4d %4d %4d %4d %4d %4d %4.0f %4.0f\n",
- total_active,
- total_missed_detection,
- total_passive,
- total_false_positive,
- num_onsets,
- num_not_adapted,
- static_cast<float>(onset_adaptation) / (num_onsets + 1e-12),
- static_cast<float>(total_false_positive_duration) /
- (total_passive + 1e-12));
-
- fclose(true_vad_fid);
- fclose(pcm_fid);
- if (video_vad_fid != NULL) {
- fclose(video_vad_fid);
- }
- if (results_fid != NULL) {
- fclose(results_fid);
- }
-}
-
-} // namespace webrtc
-
-int main(int argc, char* argv[]) {
- char kUsage[] =
- "\nCompute the number of misdetected and false-positive frames. Not\n"
- " that for each frame of audio (10 ms) there should be one true\n"
- " activity. If any video-based activity is given, there should also be\n"
- " one probability per frame.\n"
- "\nUsage:\n\n"
- "activity_metric input_pcm [options]\n"
- "where 'input_pcm' is the input audio sampled at 16 kHz in 16 bits "
- "format.\n\n";
- google::SetUsageMessage(kUsage);
- google::ParseCommandLineFlags(&argc, &argv, true);
- webrtc::void_main(argc, argv);
- return 0;
-}
diff --git a/webrtc/modules/audio_processing/agc/test/agc_harness.cc b/webrtc/modules/audio_processing/agc/test/agc_harness.cc
deleted file mode 100644
index d7c32b0..0000000
--- a/webrtc/modules/audio_processing/agc/test/agc_harness.cc
+++ /dev/null
@@ -1,286 +0,0 @@
-/*
- * Copyright (c) 2012 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.
- */
-
-// Refer to kUsage below for a description.
-
-#include "gflags/gflags.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_processing/agc/test/agc_manager.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
-#include "webrtc/test/testsupport/trace_to_stderr.h"
-#include "webrtc/voice_engine/include/voe_audio_processing.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_external_media.h"
-#include "webrtc/voice_engine/include/voe_file.h"
-#include "webrtc/voice_engine/include/voe_hardware.h"
-#include "webrtc/voice_engine/include/voe_network.h"
-#include "webrtc/voice_engine/include/voe_volume_control.h"
-
-DEFINE_bool(codecs, false, "print out available codecs");
-DEFINE_int32(pt, 103, "codec payload type (defaults to ISAC/16000/1)");
-DEFINE_bool(internal, true, "use the internal AGC in 'serial' mode, or as the "
- "first voice engine's AGC in parallel mode");
-DEFINE_bool(parallel, false, "run internal and public AGCs in parallel, with "
- "left- and right-panning respectively. Not compatible with -aec.");
-DEFINE_bool(devices, false, "print out capture devices and indexes to be used "
- "with the capture flags");
-DEFINE_int32(capture1, 0, "capture device index for the first voice engine");
-DEFINE_int32(capture2, 0, "capture device index for second voice engine");
-DEFINE_int32(render1, 0, "render device index for first voice engine");
-DEFINE_int32(render2, 0, "render device index for second voice engine");
-DEFINE_bool(aec, false, "runs two voice engines in parallel, with the first "
- "playing out a file and sending its captured signal to the second voice "
- "engine. Also enables echo cancellation.");
-DEFINE_bool(ns, true, "enable noise suppression");
-DEFINE_bool(highpass, true, "enable high pass filter");
-DEFINE_string(filename, "", "filename for the -aec mode");
-
-namespace webrtc {
-namespace {
-
-const char kUsage[] =
- "\nWithout additional flags, sets up a simple VoiceEngine loopback call\n"
- "with the default audio devices and runs forever. The internal AGC is\n"
- "enabled and the public disabled.\n\n"
-
- "It can also run the public AGC in parallel with the internal, panned to\n"
- "opposite stereo channels on the default render device. The capture\n"
- "devices for each can be selected (recommended, because otherwise they\n"
- "will fight for the level on the same device).\n\n"
-
- "Lastly, it can be used for local AEC testing. In this mode, the first\n"
- "voice engine plays out a file over the selected render device (normally\n"
- "loudspeakers) and records from the selected capture device. The second\n"
- "voice engine receives the capture signal and plays it out over the\n"
- "selected render device (normally headphones). This allows the user to\n"
- "test an echo scenario with the first voice engine, while monitoring the\n"
- "result with the second.";
-
-class AgcVoiceEngine {
- public:
- enum Pan {
- NoPan,
- PanLeft,
- PanRight
- };
-
- AgcVoiceEngine(bool internal, int tx_port, int rx_port, int capture_idx,
- int render_idx)
- : voe_(VoiceEngine::Create()),
- base_(VoEBase::GetInterface(voe_)),
- hardware_(VoEHardware::GetInterface(voe_)),
- codec_(VoECodec::GetInterface(voe_)),
- manager_(new AgcManager(voe_)),
- channel_(-1),
- capture_idx_(capture_idx),
- render_idx_(render_idx) {
- SetUp(internal, tx_port, rx_port);
- }
-
- ~AgcVoiceEngine() {
- TearDown();
- }
-
- void SetUp(bool internal, int tx_port, int rx_port) {
- ASSERT_TRUE(voe_ != NULL);
- ASSERT_TRUE(base_ != NULL);
- ASSERT_TRUE(hardware_ != NULL);
- ASSERT_TRUE(codec_ != NULL);
- VoEAudioProcessing* audio = VoEAudioProcessing::GetInterface(voe_);
- ASSERT_TRUE(audio != NULL);
- VoENetwork* network = VoENetwork::GetInterface(voe_);
- ASSERT_TRUE(network != NULL);
-
- ASSERT_EQ(0, base_->Init());
- channel_ = base_->CreateChannel();
- ASSERT_NE(-1, channel_);
-
- channel_transport_.reset(
- new test::VoiceChannelTransport(network, channel_));
- ASSERT_EQ(0, channel_transport_->SetSendDestination("127.0.0.1", tx_port));
- ASSERT_EQ(0, channel_transport_->SetLocalReceiver(rx_port));
-
- ASSERT_EQ(0, hardware_->SetRecordingDevice(capture_idx_));
- ASSERT_EQ(0, hardware_->SetPlayoutDevice(render_idx_));
-
- CodecInst codec_params = {0};
- bool codec_found = false;
- for (int i = 0; i < codec_->NumOfCodecs(); i++) {
- ASSERT_EQ(0, codec_->GetCodec(i, codec_params));
- if (FLAGS_pt == codec_params.pltype) {
- codec_found = true;
- break;
- }
- }
- ASSERT_TRUE(codec_found);
- ASSERT_EQ(0, codec_->SetSendCodec(channel_, codec_params));
-
- ASSERT_EQ(0, audio->EnableHighPassFilter(FLAGS_highpass));
- ASSERT_EQ(0, audio->SetNsStatus(FLAGS_ns));
- ASSERT_EQ(0, audio->SetEcStatus(FLAGS_aec));
-
- ASSERT_EQ(0, manager_->Enable(internal));
- ASSERT_EQ(0, audio->SetAgcStatus(!internal));
-
- audio->Release();
- network->Release();
- }
-
- void TearDown() {
- Stop();
- channel_transport_.reset(NULL);
- ASSERT_EQ(0, base_->DeleteChannel(channel_));
- ASSERT_EQ(0, base_->Terminate());
- // Don't test; the manager hasn't released its interfaces.
- hardware_->Release();
- base_->Release();
- codec_->Release();
- delete manager_;
- ASSERT_TRUE(VoiceEngine::Delete(voe_));
- }
-
- void PrintDevices() {
- int num_devices = 0;
- char device_name[128] = {0};
- char guid[128] = {0};
- ASSERT_EQ(0, hardware_->GetNumOfRecordingDevices(num_devices));
- printf("Capture devices:\n");
- for (int i = 0; i < num_devices; i++) {
- ASSERT_EQ(0, hardware_->GetRecordingDeviceName(i, device_name, guid));
- printf("%d: %s\n", i, device_name);
- }
- ASSERT_EQ(0, hardware_->GetNumOfPlayoutDevices(num_devices));
- printf("Render devices:\n");
- for (int i = 0; i < num_devices; i++) {
- ASSERT_EQ(0, hardware_->GetPlayoutDeviceName(i, device_name, guid));
- printf("%d: %s\n", i, device_name);
- }
- }
-
- void PrintCodecs() {
- CodecInst params = {0};
- printf("Codecs:\n");
- for (int i = 0; i < codec_->NumOfCodecs(); i++) {
- ASSERT_EQ(0, codec_->GetCodec(i, params));
- printf("%d %s/%d/%d\n", params.pltype, params.plname, params.plfreq,
- params.channels);
- }
- }
-
- void StartSending() {
- ASSERT_EQ(0, base_->StartSend(channel_));
- }
-
- void StartPlaying(Pan pan, const std::string& filename) {
- VoEVolumeControl* volume = VoEVolumeControl::GetInterface(voe_);
- VoEFile* file = VoEFile::GetInterface(voe_);
- ASSERT_TRUE(volume != NULL);
- ASSERT_TRUE(file != NULL);
- if (pan == PanLeft) {
- volume->SetOutputVolumePan(channel_, 1, 0);
- } else if (pan == PanRight) {
- volume->SetOutputVolumePan(channel_, 0, 1);
- }
- if (filename != "") {
- printf("playing file\n");
- ASSERT_EQ(0, file->StartPlayingFileLocally(channel_, filename.c_str(),
- true, kFileFormatPcm16kHzFile, 1.0, 0, 0));
- }
- ASSERT_EQ(0, base_->StartReceive(channel_));
- ASSERT_EQ(0, base_->StartPlayout(channel_));
- volume->Release();
- file->Release();
- }
-
- void Stop() {
- ASSERT_EQ(0, base_->StopSend(channel_));
- ASSERT_EQ(0, base_->StopPlayout(channel_));
- }
-
- private:
- VoiceEngine* voe_;
- VoEBase* base_;
- VoEHardware* hardware_;
- VoECodec* codec_;
- AgcManager* manager_;
- int channel_;
- int capture_idx_;
- int render_idx_;
- scoped_ptr<test::VoiceChannelTransport> channel_transport_;
-};
-
-void RunHarness() {
- scoped_ptr<AgcVoiceEngine> voe1(new AgcVoiceEngine(FLAGS_internal,
- 2000,
- 2000,
- FLAGS_capture1,
- FLAGS_render1));
- scoped_ptr<AgcVoiceEngine> voe2;
- if (FLAGS_parallel) {
- voe2.reset(new AgcVoiceEngine(!FLAGS_internal, 3000, 3000, FLAGS_capture2,
- FLAGS_render2));
- voe1->StartPlaying(AgcVoiceEngine::PanLeft, "");
- voe1->StartSending();
- voe2->StartPlaying(AgcVoiceEngine::PanRight, "");
- voe2->StartSending();
- } else if (FLAGS_aec) {
- voe1.reset(new AgcVoiceEngine(FLAGS_internal, 2000, 4242, FLAGS_capture1,
- FLAGS_render1));
- voe2.reset(new AgcVoiceEngine(!FLAGS_internal, 4242, 2000, FLAGS_capture2,
- FLAGS_render2));
- voe1->StartPlaying(AgcVoiceEngine::NoPan, FLAGS_filename);
- voe1->StartSending();
- voe2->StartPlaying(AgcVoiceEngine::NoPan, "");
- } else {
- voe1->StartPlaying(AgcVoiceEngine::NoPan, "");
- voe1->StartSending();
- }
-
- // Run forever...
- SleepMs(0x7fffffff);
-}
-
-void PrintDevices() {
- AgcVoiceEngine device_voe(false, 4242, 4242, 0, 0);
- device_voe.PrintDevices();
-}
-
-void PrintCodecs() {
- AgcVoiceEngine codec_voe(false, 4242, 4242, 0, 0);
- codec_voe.PrintCodecs();
-}
-
-} // namespace
-} // namespace webrtc
-
-int main(int argc, char** argv) {
- google::SetUsageMessage(webrtc::kUsage);
- google::ParseCommandLineFlags(&argc, &argv, true);
- webrtc::test::TraceToStderr trace_to_stderr;
-
- if (FLAGS_parallel && FLAGS_aec) {
- printf("-parallel and -aec are not compatible\n");
- return 1;
- }
- if (FLAGS_devices) {
- webrtc::PrintDevices();
- }
- if (FLAGS_codecs) {
- webrtc::PrintCodecs();
- }
- if (!FLAGS_devices && !FLAGS_codecs) {
- webrtc::RunHarness();
- }
- return 0;
-}
diff --git a/webrtc/modules/audio_processing/agc/test/agc_manager.cc b/webrtc/modules/audio_processing/agc/test/agc_manager.cc
deleted file mode 100644
index a741e64..0000000
--- a/webrtc/modules/audio_processing/agc/test/agc_manager.cc
+++ /dev/null
@@ -1,252 +0,0 @@
-/*
- * Copyright (c) 2012 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_processing/agc/test/agc_manager.h"
-
-#include <assert.h>
-
-#include "webrtc/modules/audio_processing/agc/agc.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/voice_engine/include/voe_external_media.h"
-#include "webrtc/voice_engine/include/voe_volume_control.h"
-
-namespace webrtc {
-
-class AgcManagerVolume : public VolumeCallbacks {
- public:
- // AgcManagerVolume acquires ownership of |volume|.
- explicit AgcManagerVolume(VoEVolumeControl* volume)
- : volume_(volume) {
- }
-
- ~AgcManagerVolume() {
- if (volume_) {
- volume_->Release();
- }
- }
-
- virtual void SetMicVolume(int volume) {
- if (volume_->SetMicVolume(volume) != 0) {
- LOG_FERR1(LS_WARNING, SetMicVolume, volume);
- }
- }
-
- int GetMicVolume() {
- unsigned int volume = 0;
- if (volume_->GetMicVolume(volume) != 0) {
- LOG_FERR0(LS_WARNING, GetMicVolume);
- return -1;
- }
- return volume;
- }
-
- private:
- VoEVolumeControl* volume_;
-};
-
-class MediaCallback : public VoEMediaProcess {
- public:
- MediaCallback(AgcManagerDirect* direct, AudioProcessing* audioproc,
- CriticalSectionWrapper* crit)
- : direct_(direct),
- audioproc_(audioproc),
- crit_(crit),
- frame_() {
- }
-
- protected:
- virtual void Process(const int channel, const ProcessingTypes type,
- int16_t audio[], const int samples_per_channel,
- const int sample_rate_hz, const bool is_stereo) {
- CriticalSectionScoped cs(crit_);
- if (direct_->capture_muted()) {
- return;
- }
-
- // Extract the first channel.
- const int kMaxSampleRateHz = 48000;
- const int kMaxSamplesPerChannel = kMaxSampleRateHz / 100;
- assert(samples_per_channel < kMaxSamplesPerChannel &&
- sample_rate_hz < kMaxSampleRateHz);
- int16_t mono[kMaxSamplesPerChannel];
- int16_t* mono_ptr = audio;
- if (is_stereo) {
- for (int n = 0; n < samples_per_channel; n++) {
- mono[n] = audio[n * 2];
- }
- mono_ptr = mono;
- }
-
- direct_->Process(mono_ptr, samples_per_channel, sample_rate_hz);
-
- // TODO(ajm): It's unfortunate we have to memcpy to this frame here, but
- // it's needed for use with AudioProcessing.
- frame_.num_channels_ = is_stereo ? 2 : 1;
- frame_.samples_per_channel_ = samples_per_channel;
- frame_.sample_rate_hz_ = sample_rate_hz;
- const int length_samples = frame_.num_channels_ * samples_per_channel;
- memcpy(frame_.data_, audio, length_samples * sizeof(int16_t));
-
- // Apply compression to the audio.
- if (audioproc_->ProcessStream(&frame_) != 0) {
- LOG_FERR0(LS_ERROR, ProcessStream);
- }
-
- // Copy the compressed audio back to voice engine's array.
- memcpy(audio, frame_.data_, length_samples * sizeof(int16_t));
- }
-
- private:
- AgcManagerDirect* direct_;
- AudioProcessing* audioproc_;
- CriticalSectionWrapper* crit_;
- AudioFrame frame_;
-};
-
-class PreprocCallback : public VoEMediaProcess {
- public:
- PreprocCallback(AgcManagerDirect* direct, CriticalSectionWrapper* crit)
- : direct_(direct),
- crit_(crit) {
- }
-
- protected:
- virtual void Process(const int channel, const ProcessingTypes type,
- int16_t audio[], const int samples_per_channel,
- const int sample_rate_hz, const bool is_stereo) {
- CriticalSectionScoped cs(crit_);
- if (direct_->capture_muted()) {
- return;
- }
- direct_->AnalyzePreProcess(audio, is_stereo ? 2 : 1, samples_per_channel);
- }
-
- private:
- AgcManagerDirect* direct_;
- CriticalSectionWrapper* crit_;
-};
-
-AgcManager::AgcManager(VoiceEngine* voe)
- : media_(VoEExternalMedia::GetInterface(voe)),
- volume_callbacks_(new AgcManagerVolume(VoEVolumeControl::GetInterface(
- voe))),
- crit_(CriticalSectionWrapper::CreateCriticalSection()),
- enabled_(false),
- initialized_(false) {
- Config config;
- config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
- audioproc_.reset(AudioProcessing::Create(config));
- direct_.reset(new AgcManagerDirect(audioproc_->gain_control(),
- volume_callbacks_.get()));
- media_callback_.reset(new MediaCallback(direct_.get(),
- audioproc_.get(),
- crit_.get()));
- preproc_callback_.reset(new PreprocCallback(direct_.get(), crit_.get()));
-}
-
-AgcManager::AgcManager(VoEExternalMedia* media, VoEVolumeControl* volume,
- Agc* agc, AudioProcessing* audioproc)
- : media_(media),
- volume_callbacks_(new AgcManagerVolume(volume)),
- crit_(CriticalSectionWrapper::CreateCriticalSection()),
- audioproc_(audioproc),
- direct_(new AgcManagerDirect(agc,
- audioproc_->gain_control(),
- volume_callbacks_.get())),
- media_callback_(new MediaCallback(direct_.get(),
- audioproc_.get(),
- crit_.get())),
- preproc_callback_(new PreprocCallback(direct_.get(), crit_.get())),
- enabled_(false),
- initialized_(false) {
-}
-
-AgcManager::AgcManager()
- : media_(NULL),
- enabled_(false),
- initialized_(false) {
-}
-
-AgcManager::~AgcManager() {
- if (media_) {
- if (enabled_) {
- DeregisterCallbacks();
- }
- media_->Release();
- }
-}
-
-int AgcManager::Enable(bool enable) {
- if (enable == enabled_) {
- return 0;
- }
- if (!initialized_) {
- CriticalSectionScoped cs(crit_.get());
- if (audioproc_->gain_control()->Enable(true) != 0) {
- LOG_FERR1(LS_ERROR, gain_control()->Enable, true);
- return -1;
- }
- if (direct_->Initialize() != 0) {
- assert(false);
- return -1;
- }
- initialized_ = true;
- }
-
- if (enable) {
- if (media_->RegisterExternalMediaProcessing(0, kRecordingAllChannelsMixed,
- *media_callback_) != 0) {
- LOG(LS_ERROR) << "Failed to register postproc callback";
- return -1;
- }
- if (media_->RegisterExternalMediaProcessing(0, kRecordingPreprocessing,
- *preproc_callback_) != 0) {
- LOG(LS_ERROR) << "Failed to register preproc callback";
- return -1;
- }
- } else {
- if (DeregisterCallbacks() != 0)
- return -1;
- }
- enabled_ = enable;
- return 0;
-}
-
-void AgcManager::CaptureDeviceChanged() {
- CriticalSectionScoped cs(crit_.get());
- direct_->Initialize();
-}
-
-void AgcManager::SetCaptureMuted(bool muted) {
- CriticalSectionScoped cs(crit_.get());
- direct_->SetCaptureMuted(muted);
-}
-
-int AgcManager::DeregisterCallbacks() {
- // DeRegister shares a lock with the Process() callback. This call will block
- // until the callback is finished and it's safe to continue teardown.
- int err = 0;
- if (media_->DeRegisterExternalMediaProcessing(0,
- kRecordingAllChannelsMixed) != 0) {
- LOG(LS_ERROR) << "Failed to deregister postproc callback";
- err = -1;
- }
- if (media_->DeRegisterExternalMediaProcessing(0,
- kRecordingPreprocessing) != 0) {
- LOG(LS_ERROR) << "Failed to deregister preproc callback";
- err = -1;
- }
- return err;
-}
-
-} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/agc/test/agc_manager.h b/webrtc/modules/audio_processing/agc/test/agc_manager.h
deleted file mode 100644
index ec8161c..0000000
--- a/webrtc/modules/audio_processing/agc/test/agc_manager.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * Copyright (c) 2012 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_PROCESSING_AGC_TEST_AGC_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_AGC_MANAGER_H_
-
-#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
-
-namespace webrtc {
-
-class Agc;
-class AudioProcessing;
-class CriticalSectionWrapper;
-class MediaCallback;
-class PreprocCallback;
-class VoEExternalMedia;
-class VoEVolumeControl;
-class VoiceEngine;
-class VolumeCallbacks;
-
-// Handles the interaction between VoiceEngine and the internal AGC. It hooks
-// into the capture stream through VoiceEngine's external media interface and
-// sends the audio to the AGC for analysis. It forwards requests for a capture
-// volume change from the AGC to the VoiceEngine volume interface.
-class AgcManager {
- public:
- explicit AgcManager(VoiceEngine* voe);
- // Dependency injection for testing. Don't delete |agc| or |audioproc| as the
- // memory is owned by the manager. If |media| or |volume| are non-fake
- // reference counted classes, don't release them as this is handled by the
- // manager.
- AgcManager(VoEExternalMedia* media, VoEVolumeControl* volume, Agc* agc,
- AudioProcessing* audioproc);
- virtual ~AgcManager();
-
- // When enabled, registers external media processing callbacks with
- // VoiceEngine to hook into the capture stream. Disabling deregisters the
- // callbacks.
- virtual int Enable(bool enable);
- virtual bool enabled() const { return enabled_; }
-
- // Call when the capture device has changed. This will trigger a retrieval of
- // the initial capture volume on the next audio frame.
- virtual void CaptureDeviceChanged();
-
- // Call when the capture stream has been muted/unmuted. This causes the
- // manager to disregard all incoming audio; chances are good it's background
- // noise to which we'd like to avoid adapting.
- virtual void SetCaptureMuted(bool muted);
- virtual bool capture_muted() const { return direct_->capture_muted(); }
-
- protected:
- // Provide a default constructor for testing.
- AgcManager();
-
- private:
- int DeregisterCallbacks();
- int CheckVolumeAndReset();
-
- VoEExternalMedia* media_;
- scoped_ptr<VolumeCallbacks> volume_callbacks_;
- scoped_ptr<CriticalSectionWrapper> crit_;
- scoped_ptr<AudioProcessing> audioproc_;
- scoped_ptr<AgcManagerDirect> direct_;
- scoped_ptr<MediaCallback> media_callback_;
- scoped_ptr<PreprocCallback> preproc_callback_;
- bool enabled_;
- bool initialized_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_AGC_MANAGER_H_
diff --git a/webrtc/modules/audio_processing/agc/test/agc_manager_integrationtest.cc b/webrtc/modules/audio_processing/agc/test/agc_manager_integrationtest.cc
deleted file mode 100644
index 9dbbc22..0000000
--- a/webrtc/modules/audio_processing/agc/test/agc_manager_integrationtest.cc
+++ /dev/null
@@ -1,123 +0,0 @@
-/*
- * Copyright (c) 2012 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_processing/agc/test/agc_manager.h"
-
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_processing/agc/mock_agc.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
-#include "webrtc/system_wrappers/interface/sleep.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_external_media.h"
-#include "webrtc/voice_engine/include/voe_network.h"
-#include "webrtc/voice_engine/include/voe_volume_control.h"
-
-using ::testing::_;
-using ::testing::AtLeast;
-using ::testing::Mock;
-using ::testing::Return;
-
-namespace webrtc {
-
-class AgcManagerTest : public ::testing::Test {
- protected:
- AgcManagerTest()
- : voe_(VoiceEngine::Create()),
- base_(VoEBase::GetInterface(voe_)),
- agc_(new MockAgc()),
- manager_(new AgcManager(VoEExternalMedia::GetInterface(voe_),
- VoEVolumeControl::GetInterface(voe_),
- agc_,
- AudioProcessing::Create())),
- channel_(-1) {
- }
-
- virtual void SetUp() {
- ASSERT_TRUE(voe_ != NULL);
- ASSERT_TRUE(base_ != NULL);
- ASSERT_EQ(0, base_->Init());
- channel_ = base_->CreateChannel();
- ASSERT_NE(-1, channel_);
-
- VoENetwork* network = VoENetwork::GetInterface(voe_);
- ASSERT_TRUE(network != NULL);
- channel_transport_.reset(
- new test::VoiceChannelTransport(network, channel_));
- ASSERT_EQ(0, channel_transport_->SetSendDestination("127.0.0.1", 1234));
- network->Release();
- }
-
- virtual void TearDown() {
- channel_transport_.reset(NULL);
- ASSERT_EQ(0, base_->DeleteChannel(channel_));
- ASSERT_EQ(0, base_->Terminate());
- delete manager_;
- // Test that the manager has released all VoE interfaces. The last
- // reference is released in VoiceEngine::Delete.
- EXPECT_EQ(1, base_->Release());
- ASSERT_TRUE(VoiceEngine::Delete(voe_));
- }
-
- VoiceEngine* voe_;
- VoEBase* base_;
- MockAgc* agc_;
- scoped_ptr<test::VoiceChannelTransport> channel_transport_;
- // We use a pointer for the manager, so we can tear it down and test
- // base_->Release() in the destructor.
- AgcManager* manager_;
- int channel_;
-};
-
-TEST_F(AgcManagerTest, DISABLED_ON_ANDROID(EnableSucceeds)) {
- EXPECT_EQ(0, manager_->Enable(true));
- EXPECT_TRUE(manager_->enabled());
- EXPECT_EQ(0, manager_->Enable(false));
- EXPECT_FALSE(manager_->enabled());
-}
-
-TEST_F(AgcManagerTest, DISABLED_ON_ANDROID(ProcessIsNotCalledByDefault)) {
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).Times(0);
- EXPECT_CALL(*agc_, Process(_, _, _)).Times(0);
- EXPECT_CALL(*agc_, GetRmsErrorDb(_)).Times(0);
- ASSERT_EQ(0, base_->StartSend(channel_));
- SleepMs(100);
- ASSERT_EQ(0, base_->StopSend(channel_));
-}
-
-TEST_F(AgcManagerTest, DISABLED_ProcessIsCalledOnlyWhenEnabled) {
- EXPECT_CALL(*agc_, Reset());
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .Times(AtLeast(1))
- .WillRepeatedly(Return(0));
- EXPECT_CALL(*agc_, Process(_, _, _))
- .Times(AtLeast(1))
- .WillRepeatedly(Return(0));
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .Times(AtLeast(1))
- .WillRepeatedly(Return(false));
- EXPECT_EQ(0, manager_->Enable(true));
- ASSERT_EQ(0, base_->StartSend(channel_));
- SleepMs(100);
- EXPECT_EQ(0, manager_->Enable(false));
- SleepMs(100);
- Mock::VerifyAndClearExpectations(agc_);
-
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).Times(0);
- EXPECT_CALL(*agc_, Process(_, _, _)).Times(0);
- EXPECT_CALL(*agc_, GetRmsErrorDb(_)).Times(0);
- SleepMs(100);
- ASSERT_EQ(0, base_->StopSend(channel_));
-}
-
-} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/agc/test/agc_manager_unittest.cc b/webrtc/modules/audio_processing/agc/test/agc_manager_unittest.cc
deleted file mode 100644
index 92464ef..0000000
--- a/webrtc/modules/audio_processing/agc/test/agc_manager_unittest.cc
+++ /dev/null
@@ -1,736 +0,0 @@
-/*
- * Copyright (c) 2013 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_processing/agc/test/agc_manager.h"
-
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_processing/agc/mock_agc.h"
-#include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/voice_engine/include/mock/fake_voe_external_media.h"
-#include "webrtc/voice_engine/include/mock/mock_voe_volume_control.h"
-#include "webrtc/test/testsupport/trace_to_stderr.h"
-
-using ::testing::_;
-using ::testing::DoAll;
-using ::testing::Eq;
-using ::testing::Mock;
-using ::testing::Return;
-using ::testing::SetArgPointee;
-using ::testing::SetArgReferee;
-
-namespace webrtc {
-namespace {
-
-const int kSampleRateHz = 32000;
-const int kNumChannels = 1;
-const int kSamplesPerChannel = kSampleRateHz / 100;
-const float kAboveClippedThreshold = 0.2f;
-
-} // namespace
-
-class AgcManagerUnitTest : public ::testing::Test {
- protected:
- AgcManagerUnitTest()
- : media_(),
- volume_(),
- agc_(new MockAgc),
- audioproc_(new MockAudioProcessing),
- gctrl_(audioproc_->gain_control()),
- manager_(&media_, &volume_, agc_, audioproc_) {
- EXPECT_CALL(*gctrl_, Enable(true));
- ExpectInitialize();
- manager_.Enable(true);
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(Return(false));
- // TODO(bjornv): Find a better solution that adds an initial volume here
- // instead of applying SetVolumeAndProcess(128u) in each test, but at the
- // same time can test a too low initial value.
- }
-
- void SetInitialVolume(unsigned int volume) {
- ExpectInitialize();
- manager_.CaptureDeviceChanged();
- ExpectCheckVolumeAndReset(volume);
- EXPECT_CALL(*agc_, GetRmsErrorDb(_)).WillOnce(Return(false));
- PostProcCallback(1);
- }
-
- void SetVolumeAndProcess(unsigned int volume) {
- // Volume is checked on first process call.
- ExpectCheckVolumeAndReset(volume);
- PostProcCallback(1);
- }
-
- void ExpectCheckVolumeAndReset(unsigned int volume) {
- EXPECT_CALL(volume_, GetMicVolume(_))
- .WillOnce(DoAll(SetArgReferee<0>(volume), Return(0)));
- EXPECT_CALL(*agc_, Reset());
- }
-
- void ExpectVolumeChange(unsigned int current_volume,
- unsigned int new_volume) {
- EXPECT_CALL(volume_, GetMicVolume(_))
- .WillOnce(DoAll(SetArgReferee<0>(current_volume), Return(0)));
- EXPECT_CALL(volume_, SetMicVolume(Eq(new_volume))).WillOnce(Return(0));
- }
-
- void ExpectInitialize() {
- EXPECT_CALL(*gctrl_, set_mode(GainControl::kFixedDigital));
- EXPECT_CALL(*gctrl_, set_target_level_dbfs(2));
- EXPECT_CALL(*gctrl_, set_compression_gain_db(7));
- EXPECT_CALL(*gctrl_, enable_limiter(true));
- }
-
- void PreProcCallback(int num_calls) {
- for (int i = 0; i < num_calls; ++i) {
- media_.CallProcess(kRecordingPreprocessing, NULL, kSamplesPerChannel,
- kSampleRateHz, kNumChannels);
- }
- }
-
- void PostProcCallback(int num_calls) {
- for (int i = 0; i < num_calls; ++i) {
- EXPECT_CALL(*agc_, Process(_, _, _)).WillOnce(Return(0));
- EXPECT_CALL(*audioproc_, ProcessStream(_)).WillOnce(Return(0));
- media_.CallProcess(kRecordingAllChannelsMixed, NULL, kSamplesPerChannel,
- kSampleRateHz, kNumChannels);
- }
- }
-
- ~AgcManagerUnitTest() {
- EXPECT_CALL(volume_, Release()).WillOnce(Return(0));
- }
-
- FakeVoEExternalMedia media_;
- MockVoEVolumeControl volume_;
- MockAgc* agc_;
- MockAudioProcessing* audioproc_;
- MockGainControl* gctrl_;
- AgcManager manager_;
- test::TraceToStderr trace_to_stderr;
-};
-
-TEST_F(AgcManagerUnitTest, MicVolumeResponseToRmsError) {
- SetVolumeAndProcess(128u);
- // Compressor default; no residual error.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)));
- PostProcCallback(1);
-
- // Inside the compressor's window; no change of volume.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)));
- PostProcCallback(1);
-
- // Above the compressor's window; volume should be increased.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
- ExpectVolumeChange(128u, 130u);
- PostProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
- ExpectVolumeChange(130u, 168u);
- PostProcCallback(1);
-
- // Inside the compressor's window; no change of volume.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)));
- PostProcCallback(1);
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)));
- PostProcCallback(1);
-
- // Below the compressor's window; volume should be decreased.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
- ExpectVolumeChange(168u, 167u);
- PostProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
- ExpectVolumeChange(167u, 163u);
- PostProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-9), Return(true)));
- ExpectVolumeChange(163u, 129u);
- PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, MicVolumeIsLimited) {
- SetVolumeAndProcess(128u);
- // Maximum upwards change is limited.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
- ExpectVolumeChange(128u, 183u);
- PostProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
- ExpectVolumeChange(183u, 243u);
- PostProcCallback(1);
-
- // Won't go higher than the maximum.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
- ExpectVolumeChange(243u, 255u);
- PostProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
- ExpectVolumeChange(255u, 254u);
- PostProcCallback(1);
-
- // Maximum downwards change is limited.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
- ExpectVolumeChange(254u, 194u);
- PostProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
- ExpectVolumeChange(194u, 137u);
- PostProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
- ExpectVolumeChange(137u, 88u);
- PostProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
- ExpectVolumeChange(88u, 54u);
- PostProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
- ExpectVolumeChange(54u, 33u);
- PostProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
- ExpectVolumeChange(33u, 18u);
- PostProcCallback(1);
-
- // Won't go lower than the minimum.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-40), Return(true)));
- ExpectVolumeChange(18u, 12u);
- PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, CompressorStepsTowardsTarget) {
- SetVolumeAndProcess(128u);
- // Compressor default; no call to set_compression_gain_db.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)))
- .WillRepeatedly(Return(false));
- EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
- PostProcCallback(20);
-
- // Moves slowly upwards.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(9), Return(true)))
- .WillRepeatedly(Return(false));
- EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
- PostProcCallback(19);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
- PostProcCallback(1);
-
- EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
- PostProcCallback(19);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
- PostProcCallback(1);
-
- EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
- PostProcCallback(20);
-
- // Moves slowly downward, then reverses before reaching the original target.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(5), Return(true)))
- .WillRepeatedly(Return(false));
- EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
- PostProcCallback(19);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
- PostProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(9), Return(true)))
- .WillRepeatedly(Return(false));
- EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
- PostProcCallback(19);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
- PostProcCallback(1);
-
- EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
- PostProcCallback(20);
-}
-
-TEST_F(AgcManagerUnitTest, CompressorErrorIsDeemphasized) {
- SetVolumeAndProcess(128u);
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
- .WillRepeatedly(Return(false));
- PostProcCallback(19);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
- PostProcCallback(1);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
- PostProcCallback(20);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
- .WillRepeatedly(Return(false));
- PostProcCallback(19);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(7)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(6)).WillOnce(Return(0));
- PostProcCallback(1);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(_)).Times(0);
- PostProcCallback(20);
-}
-
-TEST_F(AgcManagerUnitTest, CompressorReachesMaximum) {
- SetVolumeAndProcess(128u);
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
- .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
- .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
- .WillOnce(DoAll(SetArgPointee<0>(10), Return(true)))
- .WillRepeatedly(Return(false));
- PostProcCallback(19);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(10)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(11)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(12)).WillOnce(Return(0));
- PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, CompressorReachesMinimum) {
- SetVolumeAndProcess(128u);
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
- .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
- .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
- .WillOnce(DoAll(SetArgPointee<0>(0), Return(true)))
- .WillRepeatedly(Return(false));
- PostProcCallback(19);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(6)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(5)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(4)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(3)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(2)).WillOnce(Return(0));
- PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, NoActionWhileMuted) {
- SetVolumeAndProcess(128u);
- manager_.SetCaptureMuted(true);
- media_.CallProcess(kRecordingAllChannelsMixed, NULL, kSamplesPerChannel,
- kSampleRateHz, kNumChannels);
-}
-
-TEST_F(AgcManagerUnitTest, UnmutingChecksVolumeWithoutRaising) {
- SetVolumeAndProcess(128u);
- manager_.SetCaptureMuted(true);
- manager_.SetCaptureMuted(false);
- ExpectCheckVolumeAndReset(127u);
- // SetMicVolume should not be called.
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(Return(false));
- PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, UnmutingRaisesTooLowVolume) {
- SetVolumeAndProcess(128u);
- manager_.SetCaptureMuted(true);
- manager_.SetCaptureMuted(false);
- ExpectCheckVolumeAndReset(11u);
- EXPECT_CALL(volume_, SetMicVolume(Eq(12u))).WillOnce(Return(0));
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(Return(false));
- PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, ChangingDevicesChecksVolume) {
- SetVolumeAndProcess(128u);
- ExpectInitialize();
- manager_.CaptureDeviceChanged();
- ExpectCheckVolumeAndReset(128u);
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(Return(false));
- PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, LowInitialVolumeIsRaised) {
- ExpectCheckVolumeAndReset(11u);
- // Should set MicVolume to kMinInitMicLevel = 85.
- EXPECT_CALL(volume_, SetMicVolume(Eq(85u))).WillOnce(Return(0));
- PostProcCallback(1);
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(Return(false));
- PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, ManualLevelChangeResultsInNoSetMicCall) {
- SetVolumeAndProcess(128u);
- // Change outside of compressor's range, which would normally trigger a call
- // to SetMicVolume.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
- // GetMicVolume returns a value outside of the quantization slack, indicating
- // a manual volume change.
- EXPECT_CALL(volume_, GetMicVolume(_))
- .WillOnce(DoAll(SetArgReferee<0>(154u), Return(0)));
- // SetMicVolume should not be called.
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- EXPECT_CALL(*agc_, Reset()).Times(1);
- PostProcCallback(1);
-
- // Do the same thing, except downwards now.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
- EXPECT_CALL(volume_, GetMicVolume(_))
- .WillOnce(DoAll(SetArgReferee<0>(100u), Return(0)));
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- EXPECT_CALL(*agc_, Reset()).Times(1);
- PostProcCallback(1);
-
- // And finally verify the AGC continues working without a manual change.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
- ExpectVolumeChange(100u, 99u);
- PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, RecoveryAfterManualLevelChangeFromMax) {
- SetVolumeAndProcess(128u);
- // Force the mic up to max volume. Takes a few steps due to the residual
- // gain limitation.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
- ExpectVolumeChange(128u, 183u);
- PostProcCallback(1);
- ExpectVolumeChange(183u, 243u);
- PostProcCallback(1);
- ExpectVolumeChange(243u, 255u);
- PostProcCallback(1);
-
- // Manual change does not result in SetMicVolume call.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
- EXPECT_CALL(volume_, GetMicVolume(_))
- .WillOnce(DoAll(SetArgReferee<0>(50u), Return(0)));
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- EXPECT_CALL(*agc_, Reset()).Times(1);
- PostProcCallback(1);
-
- // Continues working as usual afterwards.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
- ExpectVolumeChange(50u, 69u);
- PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, RecoveryAfterManualLevelChangeBelowMin) {
- SetVolumeAndProcess(128u);
- // Manual change below min.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
- // Don't set to zero, which will cause AGC to take no action.
- EXPECT_CALL(volume_, GetMicVolume(_))
- .WillOnce(DoAll(SetArgReferee<0>(1u), Return(0)));
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- EXPECT_CALL(*agc_, Reset()).Times(1);
- PostProcCallback(1);
-
- // Continues working as usual afterwards.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
- ExpectVolumeChange(1u, 2u);
- PostProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
- ExpectVolumeChange(2u, 11u);
- PostProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(20), Return(true)));
- ExpectVolumeChange(11u, 18u);
- PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, NoClippingHasNoImpact) {
- SetVolumeAndProcess(128u);
- EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- EXPECT_CALL(*agc_, Reset()).Times(0);
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).WillRepeatedly(Return(0));
- PreProcCallback(100);
-}
-
-TEST_F(AgcManagerUnitTest, ClippingUnderThresholdHasNoImpact) {
- SetVolumeAndProcess(128u);
- EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- EXPECT_CALL(*agc_, Reset()).Times(0);
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).WillOnce(Return(0.099));
- PreProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, ClippingLowersVolume) {
- SetVolumeAndProcess(128u);
- SetInitialVolume(255u);
-
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).WillOnce(Return(0.101));
- EXPECT_CALL(*agc_, Reset()).Times(1);
- ExpectVolumeChange(255u, 240u);
- PreProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, WaitingPeriodBetweenClippingChecks) {
- SetVolumeAndProcess(128u);
- SetInitialVolume(255u);
-
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillOnce(Return(kAboveClippedThreshold));
- EXPECT_CALL(*agc_, Reset()).Times(1);
- ExpectVolumeChange(255u, 240u);
- PreProcCallback(1);
-
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillRepeatedly(Return(kAboveClippedThreshold));
- EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- EXPECT_CALL(*agc_, Reset()).Times(0);
- PreProcCallback(300);
-
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillOnce(Return(kAboveClippedThreshold));
- EXPECT_CALL(*agc_, Reset()).Times(1);
- ExpectVolumeChange(240u, 225u);
- PreProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, ClippingLoweringIsLimited) {
- SetVolumeAndProcess(128u);
- SetInitialVolume(180u);
-
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillOnce(Return(kAboveClippedThreshold));
- EXPECT_CALL(*agc_, Reset()).Times(1);
- ExpectVolumeChange(180u, 170u);
- PreProcCallback(1);
-
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillRepeatedly(Return(kAboveClippedThreshold));
- EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- EXPECT_CALL(*agc_, Reset()).Times(0);
- PreProcCallback(1000);
-}
-
-TEST_F(AgcManagerUnitTest, ClippingMaxIsRespectedWhenEqualToLevel) {
- SetVolumeAndProcess(128u);
- SetInitialVolume(255u);
-
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillOnce(Return(kAboveClippedThreshold));
- EXPECT_CALL(*agc_, Reset()).Times(1);
- ExpectVolumeChange(255u, 240u);
- PreProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
- EXPECT_CALL(volume_, GetMicVolume(_))
- .WillRepeatedly(DoAll(SetArgReferee<0>(240u), Return(0)));
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- PostProcCallback(10);
-}
-
-TEST_F(AgcManagerUnitTest, ClippingMaxIsRespectedWhenHigherThanLevel) {
- SetVolumeAndProcess(128u);
- SetInitialVolume(200u);
-
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillOnce(Return(kAboveClippedThreshold));
- EXPECT_CALL(*agc_, Reset()).Times(1);
- ExpectVolumeChange(200u, 185u);
- PreProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillRepeatedly(DoAll(SetArgPointee<0>(40), Return(true)));
- ExpectVolumeChange(185u, 240u);
- PostProcCallback(1);
- EXPECT_CALL(volume_, GetMicVolume(_))
- .WillRepeatedly(DoAll(SetArgReferee<0>(240u), Return(0)));
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- PostProcCallback(10);
-}
-
-TEST_F(AgcManagerUnitTest, MaxCompressionIsIncreasedAfterClipping) {
- SetVolumeAndProcess(128u);
- SetInitialVolume(210u);
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillOnce(Return(kAboveClippedThreshold));
- EXPECT_CALL(*agc_, Reset()).Times(1);
- ExpectVolumeChange(210u, 195u);
- PreProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
- .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
- .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
- .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
- .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)))
- .WillRepeatedly(Return(false));
- PostProcCallback(19);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(8)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(9)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(10)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(11)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(12)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(13)).WillOnce(Return(0));
- PostProcCallback(1);
-
- // Continue clipping until we hit the maximum surplus compression.
- PreProcCallback(300);
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillOnce(Return(kAboveClippedThreshold));
- EXPECT_CALL(*agc_, Reset()).Times(1);
- ExpectVolumeChange(195u, 180u);
- PreProcCallback(1);
-
- PreProcCallback(300);
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillOnce(Return(kAboveClippedThreshold));
- EXPECT_CALL(*agc_, Reset()).Times(1);
- ExpectVolumeChange(180u, 170u);
- PreProcCallback(1);
-
- // Current level is now at the minimum, but the maximum allowed level still
- // has more to decrease.
- PreProcCallback(300);
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillOnce(Return(kAboveClippedThreshold));
- PreProcCallback(1);
-
- PreProcCallback(300);
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillOnce(Return(kAboveClippedThreshold));
- PreProcCallback(1);
-
- PreProcCallback(300);
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillOnce(Return(kAboveClippedThreshold));
- PreProcCallback(1);
-
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
- .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
- .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
- .WillOnce(DoAll(SetArgPointee<0>(16), Return(true)))
- .WillRepeatedly(Return(false));
- PostProcCallback(19);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(14)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(15)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(16)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(17)).WillOnce(Return(0));
- PostProcCallback(20);
- EXPECT_CALL(*gctrl_, set_compression_gain_db(18)).WillOnce(Return(0));
- PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, UserCanRaiseVolumeAfterClipping) {
- SetVolumeAndProcess(128u);
- SetInitialVolume(225u);
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillOnce(Return(kAboveClippedThreshold));
- EXPECT_CALL(*agc_, Reset()).Times(1);
- ExpectVolumeChange(225u, 210u);
- PreProcCallback(1);
-
- // High enough error to trigger a volume check.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(14), Return(true)));
- // User changed the volume.
- EXPECT_CALL(volume_, GetMicVolume(_))
- .WillOnce(DoAll(SetArgReferee<0>(250u), Return(0)));
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- EXPECT_CALL(*agc_, Reset()).Times(1);
- PostProcCallback(1);
-
- // Move down...
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(-10), Return(true)));
- ExpectVolumeChange(250u, 210u);
- PostProcCallback(1);
- // And back up to the new max established by the user.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(40), Return(true)));
- ExpectVolumeChange(210u, 250u);
- PostProcCallback(1);
- // Will not move above new maximum.
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillOnce(DoAll(SetArgPointee<0>(30), Return(true)));
- EXPECT_CALL(volume_, GetMicVolume(_))
- .WillRepeatedly(DoAll(SetArgReferee<0>(250u), Return(0)));
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- PostProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, ClippingDoesNotPullLowVolumeBackUp) {
- SetVolumeAndProcess(128u);
- SetInitialVolume(80u);
- EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
- .WillOnce(Return(kAboveClippedThreshold));
- EXPECT_CALL(volume_, GetMicVolume(_)).Times(0);
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- EXPECT_CALL(*agc_, Reset()).Times(0);
- PreProcCallback(1);
-}
-
-TEST_F(AgcManagerUnitTest, TakesNoActionOnZeroMicVolume) {
- SetVolumeAndProcess(128u);
- EXPECT_CALL(*agc_, GetRmsErrorDb(_))
- .WillRepeatedly(DoAll(SetArgPointee<0>(30), Return(true)));
- EXPECT_CALL(volume_, GetMicVolume(_))
- .WillRepeatedly(DoAll(SetArgReferee<0>(0), Return(0)));
- EXPECT_CALL(volume_, SetMicVolume(_)).Times(0);
- PostProcCallback(10);
-}
-
-} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/agc/test/agc_test.cc b/webrtc/modules/audio_processing/agc/test/agc_test.cc
deleted file mode 100644
index 413b3b0..0000000
--- a/webrtc/modules/audio_processing/agc/test/agc_test.cc
+++ /dev/null
@@ -1,155 +0,0 @@
-/*
- * Copyright (c) 2012 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 <cmath>
-#include <cstdio>
-
-#include <algorithm>
-
-#include "gflags/gflags.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_processing/agc/agc.h"
-#include "webrtc/modules/audio_processing/agc/test/agc_manager.h"
-#include "webrtc/modules/audio_processing/agc/test/test_utils.h"
-#include "webrtc/modules/audio_processing/agc/utility.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/logging.h"
-#include "webrtc/test/testsupport/trace_to_stderr.h"
-#include "webrtc/voice_engine/include/mock/fake_voe_external_media.h"
-#include "webrtc/voice_engine/include/mock/mock_voe_volume_control.h"
-
-DEFINE_string(in, "in.pcm", "input filename");
-DEFINE_string(out, "out.pcm", "output filename");
-DEFINE_int32(rate, 16000, "sample rate in Hz");
-DEFINE_int32(channels, 1, "number of channels");
-DEFINE_int32(level, -18, "target level in RMS dBFs [-100, 0]");
-DEFINE_bool(limiter, true, "enable a limiter for the compression stage");
-DEFINE_int32(cmp_level, 2, "target level in dBFs for the compression stage");
-DEFINE_int32(mic_gain, 80, "range of gain provided by the virtual mic in dB");
-DEFINE_int32(gain_offset, 0,
- "an amount (in dB) to add to every entry in the gain map");
-DEFINE_string(gain_file, "",
- "filename providing a mic gain mapping. The file should be text containing "
- "a (floating-point) gain entry in dBFs per line corresponding to levels "
- "from 0 to 255.");
-
-using ::testing::_;
-using ::testing::ByRef;
-using ::testing::DoAll;
-using ::testing::Mock;
-using ::testing::Return;
-using ::testing::SaveArg;
-using ::testing::SetArgReferee;
-
-namespace webrtc {
-namespace {
-
-const char kUsage[] = "\nProcess an audio file to simulate an analog agc.";
-
-void ReadGainMapFromFile(FILE* file, int offset, int gain_map[256]) {
- for (int i = 0; i < 256; ++i) {
- float gain = 0;
- ASSERT_EQ(1, fscanf(file, "%f", &gain));
- gain_map[i] = std::floor(gain + 0.5);
- }
-
- // Adjust from dBFs to gain in dB. We assume that level 127 provides 0 dB
- // gain. This corresponds to the interpretation in MicLevel2Gain().
- const int midpoint = gain_map[127];
- printf("Gain map\n");
- for (int i = 0; i < 256; ++i) {
- gain_map[i] += offset - midpoint;
- if (i % 5 == 0) {
- printf("%d: %d dB\n", i, gain_map[i]);
- }
- }
-}
-
-void CalculateGainMap(int gain_range_db, int offset, int gain_map[256]) {
- printf("Gain map\n");
- for (int i = 0; i < 256; ++i) {
- gain_map[i] = std::floor(MicLevel2Gain(gain_range_db, i) + 0.5) + offset;
- if (i % 5 == 0) {
- printf("%d: %d dB\n", i, gain_map[i]);
- }
- }
-}
-
-void RunAgc() {
- test::TraceToStderr trace_to_stderr(true);
- FILE* in_file = fopen(FLAGS_in.c_str(), "rb");
- ASSERT_TRUE(in_file != NULL);
- FILE* out_file = fopen(FLAGS_out.c_str(), "wb");
- ASSERT_TRUE(out_file != NULL);
-
- int gain_map[256];
- if (FLAGS_gain_file != "") {
- FILE* gain_file = fopen(FLAGS_gain_file.c_str(), "rt");
- ASSERT_TRUE(gain_file != NULL);
- ReadGainMapFromFile(gain_file, FLAGS_gain_offset, gain_map);
- fclose(gain_file);
- } else {
- CalculateGainMap(FLAGS_mic_gain, FLAGS_gain_offset, gain_map);
- }
-
- FakeVoEExternalMedia media;
- MockVoEVolumeControl volume;
- Agc* agc = new Agc;
- AudioProcessing* audioproc = AudioProcessing::Create();
- ASSERT_TRUE(audioproc != NULL);
- AgcManager manager(&media, &volume, agc, audioproc);
-
- int mic_level = 128;
- int last_mic_level = mic_level;
- EXPECT_CALL(volume, GetMicVolume(_))
- .WillRepeatedly(DoAll(SetArgReferee<0>(ByRef(mic_level)), Return(0)));
- EXPECT_CALL(volume, SetMicVolume(_))
- .WillRepeatedly(DoAll(SaveArg<0>(&mic_level), Return(0)));
-
- manager.Enable(true);
- ASSERT_EQ(0, agc->set_target_level_dbfs(FLAGS_level));
- const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
- GainControl* gctrl = audioproc->gain_control();
- ASSERT_EQ(kNoErr, gctrl->set_target_level_dbfs(FLAGS_cmp_level));
- ASSERT_EQ(kNoErr, gctrl->enable_limiter(FLAGS_limiter));
-
- AudioFrame frame;
- frame.num_channels_ = FLAGS_channels;
- frame.sample_rate_hz_ = FLAGS_rate;
- frame.samples_per_channel_ = FLAGS_rate / 100;
- const size_t frame_length = frame.samples_per_channel_ * FLAGS_channels;
- size_t sample_count = 0;
- while (fread(frame.data_, sizeof(int16_t), frame_length, in_file) ==
- frame_length) {
- SimulateMic(gain_map, mic_level, last_mic_level, &frame);
- last_mic_level = mic_level;
- media.CallProcess(kRecordingAllChannelsMixed, frame.data_,
- frame.samples_per_channel_, FLAGS_rate, FLAGS_channels);
- ASSERT_EQ(frame_length,
- fwrite(frame.data_, sizeof(int16_t), frame_length, out_file));
- sample_count += frame_length;
- trace_to_stderr.SetTimeSeconds(static_cast<float>(sample_count) /
- FLAGS_channels / FLAGS_rate);
- }
- fclose(in_file);
- fclose(out_file);
- EXPECT_CALL(volume, Release());
-}
-
-} // namespace
-} // namespace webrtc
-
-int main(int argc, char* argv[]) {
- google::SetUsageMessage(webrtc::kUsage);
- google::ParseCommandLineFlags(&argc, &argv, true);
- webrtc::RunAgc();
- return 0;
-}
diff --git a/webrtc/modules/audio_processing/agc/test/fake_agc.h b/webrtc/modules/audio_processing/agc/test/fake_agc.h
deleted file mode 100644
index e2aabd8..0000000
--- a/webrtc/modules/audio_processing/agc/test/fake_agc.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2012 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_PROCESSING_AGC_TEST_FAKE_AGC_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_FAKE_AGC_H_
-
-#include "webrtc/modules/audio_processing/agc/agc.h"
-
-namespace webrtc {
-
-class FakeAgc : public Agc {
- public:
- FakeAgc()
- : counter_(0),
- volume_(kMaxVolume / 2) {
- }
-
- virtual int Process(const AudioFrame& audio_frame) {
- const int kUpdateIntervalFrames = 10;
- const int kMaxVolume = 255;
- if (counter_ % kUpdateIntervalFrames == 0) {
- volume_ = (++volume_) % kMaxVolume;
- }
- counter_++;
- return 0;
- }
-
- virtual int FakeAgc::MicVolume() {
- return volume_;
- }
-
- private:
- int counter_;
- int volume_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_FAKE_AGC_H_
diff --git a/webrtc/modules/audio_processing/agc/test/test_utils.cc b/webrtc/modules/audio_processing/agc/test/test_utils.cc
deleted file mode 100644
index e7c884b..0000000
--- a/webrtc/modules/audio_processing/agc/test/test_utils.cc
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- * Copyright (c) 2012 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_processing/agc/test/test_utils.h"
-
-#include <cmath>
-
-#include <algorithm>
-
-#include "webrtc/modules/interface/module_common_types.h"
-
-namespace webrtc {
-
-float MicLevel2Gain(int gain_range_db, int level) {
- return (level - 127.0f) / 128.0f * gain_range_db / 2;
-}
-
-float Db2Linear(float db) {
- return powf(10.0f, db / 20.0f);
-}
-
-void ApplyGainLinear(float gain, float last_gain, AudioFrame* frame) {
- const int frame_length = frame->samples_per_channel_ * frame->num_channels_;
- // Smooth the transition between gain levels across the frame.
- float smoothed_gain = last_gain;
- float gain_step = (gain - last_gain) / (frame_length - 1);
- for (int i = 0; i < frame_length; ++i) {
- smoothed_gain += gain_step;
- float sample = std::floor(frame->data_[i] * smoothed_gain + 0.5);
- sample = std::max(std::min(32767.0f, sample), -32768.0f);
- frame->data_[i] = static_cast<int16_t>(sample);
- }
-}
-
-void ApplyGain(float gain_db, float last_gain_db, AudioFrame* frame) {
- ApplyGainLinear(Db2Linear(gain_db), Db2Linear(last_gain_db), frame);
-}
-
-void SimulateMic(int gain_range_db, int mic_level, int last_mic_level,
- AudioFrame* frame) {
- assert(mic_level >= 0 && mic_level <= 255);
- assert(last_mic_level >= 0 && last_mic_level <= 255);
- ApplyGain(MicLevel2Gain(gain_range_db, mic_level),
- MicLevel2Gain(gain_range_db, last_mic_level),
- frame);
-}
-
-void SimulateMic(int gain_map[255], int mic_level, int last_mic_level,
- AudioFrame* frame) {
- assert(mic_level >= 0 && mic_level <= 255);
- assert(last_mic_level >= 0 && last_mic_level <= 255);
- ApplyGain(gain_map[mic_level], gain_map[last_mic_level], frame);
-}
-
-} // namespace webrtc
-
diff --git a/webrtc/modules/audio_processing/agc/test/test_utils.h b/webrtc/modules/audio_processing/agc/test/test_utils.h
deleted file mode 100644
index 25dc496..0000000
--- a/webrtc/modules/audio_processing/agc/test/test_utils.h
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright (c) 2012 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_PROCESSING_TEST_TEST_UTILS_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
-namespace webrtc {
-
-class AudioFrame;
-
-float MicLevel2Gain(int gain_range_db, int level);
-float Db2Linear(float db);
-void ApplyGainLinear(float gain, float last_gain, AudioFrame* frame);
-void ApplyGain(float gain_db, float last_gain_db, AudioFrame* frame);
-void SimulateMic(int gain_range_db, int mic_level, int last_mic_level,
- AudioFrame* frame);
-void SimulateMic(int gain_map[255], int mic_level, int last_mic_level,
- AudioFrame* frame);
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 691f308..23ee29f 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -9,17 +9,7 @@
{
'includes': [
'../build/common.gypi',
- 'audio_coding/codecs/interfaces.gypi',
- 'audio_coding/codecs/cng/cng.gypi',
- 'audio_coding/codecs/g711/g711.gypi',
- 'audio_coding/codecs/g722/g722.gypi',
- 'audio_coding/codecs/ilbc/ilbc.gypi',
- 'audio_coding/codecs/isac/main/source/isac.gypi',
- 'audio_coding/codecs/isac/fix/source/isacfix.gypi',
- 'audio_coding/codecs/pcm16b/pcm16b.gypi',
- 'audio_coding/codecs/red/red.gypi',
- 'audio_coding/main/acm2/audio_coding_module.gypi',
- 'audio_coding/neteq/neteq.gypi',
+ 'audio_coding/audio_coding.gypi',
'audio_conference_mixer/source/audio_conference_mixer.gypi',
'audio_device/audio_device.gypi',
'audio_processing/audio_processing.gypi',
@@ -37,14 +27,9 @@
'video_render/video_render.gypi',
],
'conditions': [
- ['include_opus==1', {
- 'includes': ['audio_coding/codecs/opus/opus.gypi',],
- }],
['include_tests==1', {
'includes': [
- 'audio_coding/codecs/isac/isac_test.gypi',
- 'audio_coding/codecs/isac/isacfix_test.gypi',
- 'audio_coding/codecs/tools/audio_codec_speed_tests.gypi',
+ 'audio_coding/audio_coding_tests.gypi',
'audio_processing/audio_processing_tests.gypi',
'rtp_rtcp/test/testFec/test_fec.gypi',
'video_coding/main/source/video_coding_test.gypi',
@@ -101,6 +86,7 @@
'<(webrtc_root)/test/test.gyp:frame_generator',
'<(webrtc_root)/test/test.gyp:rtp_test_utils',
'<(webrtc_root)/test/test.gyp:test_support_main',
+ '<(webrtc_root)/tools/tools.gyp:agc_test_utils',
],
'sources': [
'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc',
@@ -175,7 +161,6 @@
'audio_processing/agc/pitch_internal_unittest.cc',
'audio_processing/agc/pole_zero_filter_unittest.cc',
'audio_processing/agc/standalone_vad_unittest.cc',
- 'audio_processing/agc/test/test_utils.cc',
'audio_processing/beamformer/complex_matrix_unittest.cc',
'audio_processing/beamformer/covariance_matrix_generator_unittest.cc',
'audio_processing/beamformer/matrix_unittest.cc',
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
index 6580281..c3853f6 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi
@@ -40,63 +40,67 @@
'test/bwe_test_logging.h',
], # source
},
- {
- 'target_name': 'bwe_tools_util',
- 'type': 'static_library',
- 'dependencies': [
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
- 'rtp_rtcp',
- ],
- 'sources': [
- 'tools/bwe_rtp.cc',
- 'tools/bwe_rtp.h',
- ],
- },
- {
- 'target_name': 'bwe_rtp_to_text',
- 'type': 'executable',
- 'includes': [
- '../rtp_rtcp/source/rtp_rtcp.gypi',
- ],
- 'dependencies': [
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
- 'bwe_tools_util',
- 'rtp_rtcp',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'tools/rtp_to_text.cc',
- '<(webrtc_root)/test/rtp_file_reader.cc',
- '<(webrtc_root)/test/rtp_file_reader.h',
- ], # source
- },
- {
- 'target_name': 'bwe_rtp_play',
- 'type': 'executable',
- 'includes': [
- '../rtp_rtcp/source/rtp_rtcp.gypi',
- ],
- 'dependencies': [
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
- 'bwe_tools_util',
- 'rtp_rtcp',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'tools/bwe_rtp_play.cc',
- '<(webrtc_root)/test/rtp_file_reader.cc',
- '<(webrtc_root)/test/rtp_file_reader.h',
- ], # source
- },
], # targets
+ 'conditions': [
+ ['include_tests==1', {
+ 'targets': [
+ {
+ 'target_name': 'bwe_tools_util',
+ 'type': 'static_library',
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+ 'rtp_rtcp',
+ ],
+ 'sources': [
+ 'tools/bwe_rtp.cc',
+ 'tools/bwe_rtp.h',
+ ],
+ },
+ {
+ 'target_name': 'bwe_rtp_to_text',
+ 'type': 'executable',
+ 'includes': [
+ '../rtp_rtcp/source/rtp_rtcp.gypi',
+ ],
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+ '<(webrtc_root)/test/test.gyp:rtp_test_utils',
+ 'bwe_tools_util',
+ 'rtp_rtcp',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'tools/rtp_to_text.cc',
+ ], # source
+ },
+ {
+ 'target_name': 'bwe_rtp_play',
+ 'type': 'executable',
+ 'includes': [
+ '../rtp_rtcp/source/rtp_rtcp.gypi',
+ ],
+ 'dependencies': [
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+ '<(webrtc_root)/test/test.gyp:rtp_test_utils',
+ 'bwe_tools_util',
+ 'rtp_rtcp',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ ],
+ },
+ 'sources': [
+ 'tools/bwe_rtp_play.cc',
+ ], # source
+ },
+ ],
+ }], # include_tests==1
+ ],
}