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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

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
}