Change existing aec dump tests to use webrtc::AecDump.
Currently the debug dump functionality of WebRTC (a log of all
AudioProcessing operations) was tested by the following tests:
1. ApmTest.VerifyDebugDump* which configures and runs AudioProcessing
from a debug dump, and verifies that the same debug dump is
recorded.
2. DebugDumpTest.* which is a comprehensive test of the debug dump
operations. AudioProcessing configuration is changed, and the dump
is scanned for the change.
3. ApmTest::{DebugDump, DebugDumpFromFileHandle} that verify that
debug dumping can be started and files written.
This CL replaces the debug dump mechanism in all these tests to
webrtc::AecDump. Some of the tests are adapted to the chenges of the
new API to AecDump {Start,Stop}DebugRecording: the old functions
signal errors when a file cannot be opened. With AecDump, the
AecDumpFactory instead returns a nullptr.
The CL also changes audioproc_f to use AecDump.
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2864373002
Cr-Commit-Position: refs/heads/master@{#18605}
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index f24abbb..613a3b7 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -595,6 +595,9 @@
":audioproc_debug_proto",
":audioproc_protobuf_utils",
":audioproc_unittest_proto",
+ "../../base:rtc_task_queue",
+ "aec_dump",
+ "aec_dump:aec_dump_unittests",
]
sources += [
"aec3/adaptive_fir_filter_unittest.cc",
@@ -745,10 +748,13 @@
":audioproc_test_utils",
"../../base:protobuf_utils",
"../../base:rtc_base_approved",
+ "../../base:rtc_task_queue",
"../../common_audio:common_audio",
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
+ "aec_dump",
+ "aec_dump:aec_dump_impl",
"//testing/gtest",
"//third_party/gflags:gflags",
]
diff --git a/webrtc/modules/audio_processing/audio_processing_unittest.cc b/webrtc/modules/audio_processing/audio_processing_unittest.cc
index 799063d..11ce917 100644
--- a/webrtc/modules/audio_processing/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_unittest.cc
@@ -22,10 +22,13 @@
#include "webrtc/base/ignore_wundef.h"
#include "webrtc/base/protobuf_utils.h"
#include "webrtc/base/safe_minmax.h"
+#include "webrtc/base/task_queue.h"
+#include "webrtc/base/thread.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "webrtc/modules/audio_processing/audio_processing_impl.h"
#include "webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h"
#include "webrtc/modules/audio_processing/common.h"
@@ -1709,6 +1712,7 @@
const std::string& out_filename,
Format format,
int max_size_bytes) {
+ rtc::TaskQueue worker_queue("ApmTest_worker_queue");
FILE* in_file = fopen(in_filename.c_str(), "rb");
ASSERT_TRUE(in_file != NULL);
audioproc::Event event_msg;
@@ -1734,10 +1738,12 @@
msg.num_reverse_channels(),
false);
if (first_init) {
- // StartDebugRecording() writes an additional init message. Don't start
+ // AttachAecDump() writes an additional init message. Don't start
// recording until after the first init to avoid the extra message.
- EXPECT_NOERR(
- apm_->StartDebugRecording(out_filename.c_str(), max_size_bytes));
+ auto aec_dump =
+ AecDumpFactory::Create(out_filename, max_size_bytes, &worker_queue);
+ EXPECT_TRUE(aec_dump);
+ apm_->AttachAecDump(std::move(aec_dump));
first_init = false;
}
@@ -1794,7 +1800,7 @@
ProcessStreamChooser(format);
}
}
- EXPECT_NOERR(apm_->StopDebugRecording());
+ apm_->DetachAecDump();
fclose(in_file);
}
@@ -1874,19 +1880,24 @@
// TODO(andrew): expand test to verify output.
TEST_F(ApmTest, DebugDump) {
+ rtc::TaskQueue worker_queue("ApmTest_worker_queue");
const std::string filename =
test::TempFilename(test::OutputPath(), "debug_aec");
- EXPECT_EQ(apm_->kNullPointerError,
- apm_->StartDebugRecording(static_cast<const char*>(NULL), -1));
+ {
+ auto aec_dump = AecDumpFactory::Create("", -1, &worker_queue);
+ EXPECT_FALSE(aec_dump);
+ }
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// Stopping without having started should be OK.
- EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
+ apm_->DetachAecDump();
- EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(filename.c_str(), -1));
+ auto aec_dump = AecDumpFactory::Create(filename, -1, &worker_queue);
+ EXPECT_TRUE(aec_dump);
+ apm_->AttachAecDump(std::move(aec_dump));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(revframe_));
- EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
+ apm_->DetachAecDump();
// Verify the file has been written.
FILE* fid = fopen(filename.c_str(), "r");
@@ -1896,10 +1907,6 @@
ASSERT_EQ(0, fclose(fid));
ASSERT_EQ(0, remove(filename.c_str()));
#else
- EXPECT_EQ(apm_->kUnsupportedFunctionError,
- apm_->StartDebugRecording(filename.c_str(), -1));
- EXPECT_EQ(apm_->kUnsupportedFunctionError, apm_->StopDebugRecording());
-
// Verify the file has NOT been written.
ASSERT_TRUE(fopen(filename.c_str(), "r") == NULL);
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
@@ -1907,21 +1914,23 @@
// TODO(andrew): expand test to verify output.
TEST_F(ApmTest, DebugDumpFromFileHandle) {
- FILE* fid = NULL;
- EXPECT_EQ(apm_->kNullPointerError, apm_->StartDebugRecording(fid, -1));
+ rtc::TaskQueue worker_queue("ApmTest_worker_queue");
+
const std::string filename =
test::TempFilename(test::OutputPath(), "debug_aec");
- fid = fopen(filename.c_str(), "w");
+ FILE* fid = fopen(filename.c_str(), "w");
ASSERT_TRUE(fid);
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// Stopping without having started should be OK.
- EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
+ apm_->DetachAecDump();
- EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(fid, -1));
+ auto aec_dump = AecDumpFactory::Create(fid, -1, &worker_queue);
+ EXPECT_TRUE(aec_dump);
+ apm_->AttachAecDump(std::move(aec_dump));
EXPECT_EQ(apm_->kNoError, apm_->ProcessReverseStream(revframe_));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
+ apm_->DetachAecDump();
// Verify the file has been written.
fid = fopen(filename.c_str(), "r");
@@ -1931,10 +1940,6 @@
ASSERT_EQ(0, fclose(fid));
ASSERT_EQ(0, remove(filename.c_str()));
#else
- EXPECT_EQ(apm_->kUnsupportedFunctionError,
- apm_->StartDebugRecording(fid, -1));
- EXPECT_EQ(apm_->kUnsupportedFunctionError, apm_->StopDebugRecording());
-
ASSERT_EQ(0, fclose(fid));
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
}
diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
index 58b47e2..35e2d2c 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
@@ -19,6 +19,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/common_audio/include/audio_util.h"
+#include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace webrtc {
@@ -79,7 +80,7 @@
AudioProcessingSimulator::AudioProcessingSimulator(
const SimulationSettings& settings)
- : settings_(settings) {
+ : settings_(settings), worker_queue_("file_writer_task_queue") {
if (settings_.ed_graph_output_filename &&
settings_.ed_graph_output_filename->size() > 0) {
residual_echo_likelihood_graph_writer_.open(
@@ -249,7 +250,7 @@
void AudioProcessingSimulator::DestroyAudioProcessor() {
if (settings_.aec_dump_output_filename) {
- RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->StopDebugRecording());
+ ap_->DetachAecDump();
}
}
@@ -389,11 +390,8 @@
}
if (settings_.aec_dump_output_filename) {
- size_t kMaxFilenameSize = AudioProcessing::kMaxFilenameSize;
- RTC_CHECK_LE(settings_.aec_dump_output_filename->size(), kMaxFilenameSize);
- RTC_CHECK_EQ(AudioProcessing::kNoError,
- ap_->StartDebugRecording(
- settings_.aec_dump_output_filename->c_str(), -1));
+ ap_->AttachAecDump(AecDumpFactory::Create(
+ *settings_.aec_dump_output_filename, -1, &worker_queue_));
}
}
diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.h b/webrtc/modules/audio_processing/test/audio_processing_simulator.h
index c9ac2e3..1b838d9 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.h
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.h
@@ -17,9 +17,10 @@
#include <memory>
#include <string>
-#include "webrtc/base/timeutils.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/optional.h"
+#include "webrtc/base/task_queue.h"
+#include "webrtc/base/timeutils.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
@@ -177,6 +178,8 @@
TickIntervalStats proc_time_;
std::ofstream residual_echo_likelihood_graph_writer_;
+ rtc::TaskQueue worker_queue_;
+
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
};
diff --git a/webrtc/modules/audio_processing/test/debug_dump_test.cc b/webrtc/modules/audio_processing/test/debug_dump_test.cc
index d67a73e..0e55453 100644
--- a/webrtc/modules/audio_processing/test/debug_dump_test.cc
+++ b/webrtc/modules/audio_processing/test/debug_dump_test.cc
@@ -14,7 +14,9 @@
#include <string>
#include <vector>
+#include "webrtc/base/task_queue.h"
#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "webrtc/modules/audio_processing/test/debug_dump_replayer.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
#include "webrtc/test/gtest.h"
@@ -104,6 +106,7 @@
std::unique_ptr<ChannelBuffer<float>> reverse_;
std::unique_ptr<ChannelBuffer<float>> output_;
+ rtc::TaskQueue worker_queue_;
std::unique_ptr<AudioProcessing> apm_;
const std::string dump_file_name_;
@@ -130,9 +133,9 @@
reverse_config_.num_channels())),
output_(new ChannelBuffer<float>(output_config_.num_frames(),
output_config_.num_channels())),
+ worker_queue_("debug_dump_generator_worker_queue"),
apm_(AudioProcessing::Create(config)),
- dump_file_name_(dump_file_name) {
-}
+ dump_file_name_(dump_file_name) {}
DebugDumpGenerator::DebugDumpGenerator(
const Config& config,
@@ -187,7 +190,8 @@
}
void DebugDumpGenerator::StartRecording() {
- apm_->StartDebugRecording(dump_file_name_.c_str(), -1);
+ apm_->AttachAecDump(
+ AecDumpFactory::Create(dump_file_name_.c_str(), -1, &worker_queue_));
}
void DebugDumpGenerator::Process(size_t num_blocks) {
@@ -211,7 +215,7 @@
}
void DebugDumpGenerator::StopRecording() {
- apm_->StopDebugRecording();
+ apm_->DetachAecDump();
}
void DebugDumpGenerator::ReadAndDeinterleave(ResampleInputAudioFile* audio,