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Change existing aec dump tests to use webrtc::AecDump.

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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}
This commit is contained in:
aleloi
2017-06-15 01:55:38 -07:00
committed by Commit Bot
parent af66f2ca8a
commit f4dd191b28
5 changed files with 51 additions and 35 deletions
Download Patch File Download Diff File Expand all files Collapse all files

6
webrtc/modules/audio_processing/BUILD.gn
View File

@ -595,6 +595,9 @@ if (rtc_include_tests) {
":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 @@ if (rtc_include_tests) {
":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",
]

51
webrtc/modules/audio_processing/audio_processing_unittest.cc
View File

@ -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 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
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 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
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 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
ProcessStreamChooser(format);
}
}
EXPECT_NOERR(apm_->StopDebugRecording());
apm_->DetachAecDump();
fclose(in_file);
}
@ -1874,19 +1880,24 @@ TEST_F(ApmTest, VerifyDebugDumpFloat) {
// 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 @@ TEST_F(ApmTest, DebugDump) {
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 @@ TEST_F(ApmTest, DebugDump) {
// 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 @@ TEST_F(ApmTest, DebugDumpFromFileHandle) {
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
}

12
webrtc/modules/audio_processing/test/audio_processing_simulator.cc
View File

@ -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 @@ void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
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::SetupOutput() {
void AudioProcessingSimulator::DestroyAudioProcessor() {
if (settings_.aec_dump_output_filename) {
RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->StopDebugRecording());
ap_->DetachAecDump();
}
}
@ -389,11 +390,8 @@ void AudioProcessingSimulator::CreateAudioProcessor() {
}
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_));
}
}

5
webrtc/modules/audio_processing/test/audio_processing_simulator.h
View File

@ -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 @@ class AudioProcessingSimulator {
TickIntervalStats proc_time_;
std::ofstream residual_echo_likelihood_graph_writer_;
rtc::TaskQueue worker_queue_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
};

12
webrtc/modules/audio_processing/test/debug_dump_test.cc
View File

@ -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 @@ class DebugDumpGenerator {
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 @@ DebugDumpGenerator::DebugDumpGenerator(const std::string& input_file_name,
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::SetOutputChannels(int channels) {
}
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::Process(size_t num_blocks) {
}
void DebugDumpGenerator::StopRecording() {
apm_->StopDebugRecording();
apm_->DetachAecDump();
}
void DebugDumpGenerator::ReadAndDeinterleave(ResampleInputAudioFile* audio,