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Side by Side Diff: webrtc/modules/audio_processing/test/debug_dump_test.cc

Issue 1810463002: Adding DebugDumpReplayer. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: a nit Created 4 years, 9 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #include <stddef.h> // size_t 11 #include <stddef.h> // size_t
12 12
13 #include <memory>
14 #include <string> 13 #include <string>
15 #include <vector> 14 #include <vector>
16 15
17 #include "testing/gtest/include/gtest/gtest.h" 16 #include "testing/gtest/include/gtest/gtest.h"
18 #include "webrtc/base/checks.h"
19 #include "webrtc/common_audio/channel_buffer.h"
20 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h" 17 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
21 #include "webrtc/modules/audio_processing/debug.pb.h" 18 #include "webrtc/modules/audio_processing/test/debug_dump_replayer.h"
22 #include "webrtc/modules/audio_processing/include/audio_processing.h"
23 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
24 #include "webrtc/modules/audio_processing/test/test_utils.h" 19 #include "webrtc/modules/audio_processing/test/test_utils.h"
25 #include "webrtc/test/testsupport/fileutils.h" 20 #include "webrtc/test/testsupport/fileutils.h"
26 21
22
27 namespace webrtc { 23 namespace webrtc {
28 namespace test { 24 namespace test {
29 25
30 namespace { 26 namespace {
31 27
32 void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer, 28 void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer,
33 const StreamConfig& config) { 29 const StreamConfig& config) {
34 auto& buffer_ref = *buffer; 30 auto& buffer_ref = *buffer;
35 if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() || 31 if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
36 buffer_ref->num_channels() != config.num_channels()) { 32 buffer_ref->num_channels() != config.num_channels()) {
(...skipping 189 matching lines...) Expand 10 before | Expand all | Expand 10 after
226 for (size_t i = 0; i < num_frames; ++i) { 222 for (size_t i = 0; i < num_frames; ++i) {
227 buffer[channel][i] = S16ToFloat(signal[i * channels + channel]); 223 buffer[channel][i] = S16ToFloat(signal[i * channels + channel]);
228 } 224 }
229 } 225 }
230 } 226 }
231 227
232 } // namespace 228 } // namespace
233 229
234 class DebugDumpTest : public ::testing::Test { 230 class DebugDumpTest : public ::testing::Test {
235 public: 231 public:
236 DebugDumpTest();
237
238 // VerifyDebugDump replays a debug dump using APM and verifies that the result 232 // VerifyDebugDump replays a debug dump using APM and verifies that the result
239 // is bit-exact-identical to the output channel in the dump. This is only 233 // is bit-exact-identical to the output channel in the dump. This is only
240 // guaranteed if the debug dump is started on the first frame. 234 // guaranteed if the debug dump is started on the first frame.
241 void VerifyDebugDump(const std::string& dump_file_name); 235 void VerifyDebugDump(const std::string& dump_file_name);
242 236
243 private: 237 private:
244 // Following functions are facilities for replaying debug dumps. 238 DebugDumpReplayer debug_dump_replayer_;
245 void OnInitEvent(const audioproc::Init& msg);
246 void OnStreamEvent(const audioproc::Stream& msg);
247 void OnReverseStreamEvent(const audioproc::ReverseStream& msg);
248 void OnConfigEvent(const audioproc::Config& msg);
249
250 void MaybeRecreateApm(const audioproc::Config& msg);
251 void ConfigureApm(const audioproc::Config& msg);
252
253 // Buffer for APM input/output.
254 std::unique_ptr<ChannelBuffer<float>> input_;
255 std::unique_ptr<ChannelBuffer<float>> reverse_;
256 std::unique_ptr<ChannelBuffer<float>> output_;
257
258 std::unique_ptr<AudioProcessing> apm_;
259
260 StreamConfig input_config_;
261 StreamConfig reverse_config_;
262 StreamConfig output_config_;
263 }; 239 };
264 240
265 DebugDumpTest::DebugDumpTest() 241 void DebugDumpTest::VerifyDebugDump(const std::string& in_filename) {
266 : input_(nullptr), // will be created upon usage. 242 ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(in_filename));
267 reverse_(nullptr),
268 output_(nullptr),
269 apm_(nullptr) {
270 }
271 243
272 void DebugDumpTest::VerifyDebugDump(const std::string& in_filename) { 244 if (const rtc::Optional<audioproc::Event> event =
273 FILE* in_file = fopen(in_filename.c_str(), "rb"); 245 debug_dump_replayer_.GetNextEvent()) {
274 ASSERT_TRUE(in_file); 246 debug_dump_replayer_.RunNextEvent();
275 audioproc::Event event_msg; 247 if (event->type() == audioproc::Event::STREAM) {
276 248 const audioproc::Stream* msg = &event->stream();
277 while (ReadMessageFromFile(in_file, &event_msg)) { 249 const StreamConfig output_config = debug_dump_replayer_.GetOutputConfig();
278 switch (event_msg.type()) { 250 const ChannelBuffer<float>* output = debug_dump_replayer_.GetOutput();
279 case audioproc::Event::INIT: 251 // Check that output of APM is bit-exact to the output in the dump.
280 OnInitEvent(event_msg.init()); 252 ASSERT_EQ(output_config.num_channels(),
281 break; 253 static_cast<size_t>(msg->output_channel_size()));
282 case audioproc::Event::STREAM: 254 ASSERT_EQ(output_config.num_frames() * sizeof(float),
283 OnStreamEvent(event_msg.stream()); 255 msg->output_channel(0).size());
284 break; 256 for (int i = 0; i < msg->output_channel_size(); ++i) {
285 case audioproc::Event::REVERSE_STREAM: 257 ASSERT_EQ(0, memcmp(output->channels()[i],
286 OnReverseStreamEvent(event_msg.reverse_stream()); 258 msg->output_channel(i).data(),
287 break; 259 msg->output_channel(i).size()));
288 case audioproc::Event::CONFIG: 260 }
289 OnConfigEvent(event_msg.config());
290 break;
291 case audioproc::Event::UNKNOWN_EVENT:
292 // We do not expect receive UNKNOWN event currently.
293 FAIL();
294 } 261 }
295 } 262 }
296 fclose(in_file);
297 }
298
299 // OnInitEvent reset the input/output/reserve channel format.
300 void DebugDumpTest::OnInitEvent(const audioproc::Init& msg) {
301 ASSERT_TRUE(msg.has_num_input_channels());
302 ASSERT_TRUE(msg.has_output_sample_rate());
303 ASSERT_TRUE(msg.has_num_output_channels());
304 ASSERT_TRUE(msg.has_reverse_sample_rate());
305 ASSERT_TRUE(msg.has_num_reverse_channels());
306
307 input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels());
308 output_config_ =
309 StreamConfig(msg.output_sample_rate(), msg.num_output_channels());
310 reverse_config_ =
311 StreamConfig(msg.reverse_sample_rate(), msg.num_reverse_channels());
312
313 MaybeResetBuffer(&input_, input_config_);
314 MaybeResetBuffer(&output_, output_config_);
315 MaybeResetBuffer(&reverse_, reverse_config_);
316 }
317
318 // OnStreamEvent replays an input signal and verifies the output.
319 void DebugDumpTest::OnStreamEvent(const audioproc::Stream& msg) {
320 // APM should have been created.
321 ASSERT_TRUE(apm_.get());
322
323 EXPECT_NOERR(apm_->gain_control()->set_stream_analog_level(msg.level()));
324 EXPECT_NOERR(apm_->set_stream_delay_ms(msg.delay()));
325 apm_->echo_cancellation()->set_stream_drift_samples(msg.drift());
326 if (msg.has_keypress())
327 apm_->set_stream_key_pressed(msg.keypress());
328 else
329 apm_->set_stream_key_pressed(true);
330
331 ASSERT_EQ(input_config_.num_channels(),
332 static_cast<size_t>(msg.input_channel_size()));
333 ASSERT_EQ(input_config_.num_frames() * sizeof(float),
334 msg.input_channel(0).size());
335
336 for (int i = 0; i < msg.input_channel_size(); ++i) {
337 memcpy(input_->channels()[i], msg.input_channel(i).data(),
338 msg.input_channel(i).size());
339 }
340
341 ASSERT_EQ(AudioProcessing::kNoError,
342 apm_->ProcessStream(input_->channels(), input_config_,
343 output_config_, output_->channels()));
344
345 // Check that output of APM is bit-exact to the output in the dump.
346 ASSERT_EQ(output_config_.num_channels(),
347 static_cast<size_t>(msg.output_channel_size()));
348 ASSERT_EQ(output_config_.num_frames() * sizeof(float),
349 msg.output_channel(0).size());
350 for (int i = 0; i < msg.output_channel_size(); ++i) {
351 ASSERT_EQ(0, memcmp(output_->channels()[i], msg.output_channel(i).data(),
352 msg.output_channel(i).size()));
353 }
354 } 263 }
355 264
356 void DebugDumpTest::OnReverseStreamEvent(const audioproc::ReverseStream& msg) {
357 // APM should have been created.
358 ASSERT_TRUE(apm_.get());
359
360 ASSERT_GT(msg.channel_size(), 0);
361 ASSERT_EQ(reverse_config_.num_channels(),
362 static_cast<size_t>(msg.channel_size()));
363 ASSERT_EQ(reverse_config_.num_frames() * sizeof(float),
364 msg.channel(0).size());
365
366 for (int i = 0; i < msg.channel_size(); ++i) {
367 memcpy(reverse_->channels()[i], msg.channel(i).data(),
368 msg.channel(i).size());
369 }
370
371 ASSERT_EQ(AudioProcessing::kNoError,
372 apm_->ProcessReverseStream(reverse_->channels(),
373 reverse_config_,
374 reverse_config_,
375 reverse_->channels()));
376 }
377
378 void DebugDumpTest::OnConfigEvent(const audioproc::Config& msg) {
379 MaybeRecreateApm(msg);
380 ConfigureApm(msg);
381 }
382
383 void DebugDumpTest::MaybeRecreateApm(const audioproc::Config& msg) {
384 // These configurations cannot be changed on the fly.
385 Config config;
386 ASSERT_TRUE(msg.has_aec_delay_agnostic_enabled());
387 config.Set<DelayAgnostic>(
388 new DelayAgnostic(msg.aec_delay_agnostic_enabled()));
389
390 ASSERT_TRUE(msg.has_noise_robust_agc_enabled());
391 config.Set<ExperimentalAgc>(
392 new ExperimentalAgc(msg.noise_robust_agc_enabled()));
393
394 ASSERT_TRUE(msg.has_transient_suppression_enabled());
395 config.Set<ExperimentalNs>(
396 new ExperimentalNs(msg.transient_suppression_enabled()));
397
398 ASSERT_TRUE(msg.has_aec_extended_filter_enabled());
399 config.Set<ExtendedFilter>(new ExtendedFilter(
400 msg.aec_extended_filter_enabled()));
401
402 // We only create APM once, since changes on these fields should not
403 // happen in current implementation.
404 if (!apm_.get()) {
405 apm_.reset(AudioProcessing::Create(config));
406 }
407 }
408
409 void DebugDumpTest::ConfigureApm(const audioproc::Config& msg) {
410 // AEC configs.
411 ASSERT_TRUE(msg.has_aec_enabled());
412 EXPECT_EQ(AudioProcessing::kNoError,
413 apm_->echo_cancellation()->Enable(msg.aec_enabled()));
414
415 ASSERT_TRUE(msg.has_aec_drift_compensation_enabled());
416 EXPECT_EQ(AudioProcessing::kNoError,
417 apm_->echo_cancellation()->enable_drift_compensation(
418 msg.aec_drift_compensation_enabled()));
419
420 ASSERT_TRUE(msg.has_aec_suppression_level());
421 EXPECT_EQ(AudioProcessing::kNoError,
422 apm_->echo_cancellation()->set_suppression_level(
423 static_cast<EchoCancellation::SuppressionLevel>(
424 msg.aec_suppression_level())));
425
426 // AECM configs.
427 ASSERT_TRUE(msg.has_aecm_enabled());
428 EXPECT_EQ(AudioProcessing::kNoError,
429 apm_->echo_control_mobile()->Enable(msg.aecm_enabled()));
430
431 ASSERT_TRUE(msg.has_aecm_comfort_noise_enabled());
432 EXPECT_EQ(AudioProcessing::kNoError,
433 apm_->echo_control_mobile()->enable_comfort_noise(
434 msg.aecm_comfort_noise_enabled()));
435
436 ASSERT_TRUE(msg.has_aecm_routing_mode());
437 EXPECT_EQ(AudioProcessing::kNoError,
438 apm_->echo_control_mobile()->set_routing_mode(
439 static_cast<EchoControlMobile::RoutingMode>(
440 msg.aecm_routing_mode())));
441
442 // AGC configs.
443 ASSERT_TRUE(msg.has_agc_enabled());
444 EXPECT_EQ(AudioProcessing::kNoError,
445 apm_->gain_control()->Enable(msg.agc_enabled()));
446
447 ASSERT_TRUE(msg.has_agc_mode());
448 EXPECT_EQ(AudioProcessing::kNoError,
449 apm_->gain_control()->set_mode(
450 static_cast<GainControl::Mode>(msg.agc_mode())));
451
452 ASSERT_TRUE(msg.has_agc_limiter_enabled());
453 EXPECT_EQ(AudioProcessing::kNoError,
454 apm_->gain_control()->enable_limiter(msg.agc_limiter_enabled()));
455
456 // HPF configs.
457 ASSERT_TRUE(msg.has_hpf_enabled());
458 EXPECT_EQ(AudioProcessing::kNoError,
459 apm_->high_pass_filter()->Enable(msg.hpf_enabled()));
460
461 // NS configs.
462 ASSERT_TRUE(msg.has_ns_enabled());
463 EXPECT_EQ(AudioProcessing::kNoError,
464 apm_->noise_suppression()->Enable(msg.ns_enabled()));
465
466 ASSERT_TRUE(msg.has_ns_level());
467 EXPECT_EQ(AudioProcessing::kNoError,
468 apm_->noise_suppression()->set_level(
469 static_cast<NoiseSuppression::Level>(msg.ns_level())));
470 }
471
472 TEST_F(DebugDumpTest, SimpleCase) { 265 TEST_F(DebugDumpTest, SimpleCase) {
473 Config config; 266 Config config;
474 DebugDumpGenerator generator(config); 267 DebugDumpGenerator generator(config);
475 generator.StartRecording(); 268 generator.StartRecording();
476 generator.Process(100); 269 generator.Process(100);
477 generator.StopRecording(); 270 generator.StopRecording();
478 VerifyDebugDump(generator.dump_file_name()); 271 VerifyDebugDump(generator.dump_file_name());
479 } 272 }
480 273
481 TEST_F(DebugDumpTest, ChangeInputFormat) { 274 TEST_F(DebugDumpTest, ChangeInputFormat) {
(...skipping 122 matching lines...) Expand 10 before | Expand all | Expand 10 after
604 config.Set<ExperimentalNs>(new ExperimentalNs(true)); 397 config.Set<ExperimentalNs>(new ExperimentalNs(true));
605 DebugDumpGenerator generator(config); 398 DebugDumpGenerator generator(config);
606 generator.StartRecording(); 399 generator.StartRecording();
607 generator.Process(100); 400 generator.Process(100);
608 generator.StopRecording(); 401 generator.StopRecording();
609 VerifyDebugDump(generator.dump_file_name()); 402 VerifyDebugDump(generator.dump_file_name());
610 } 403 }
611 404
612 } // namespace test 405 } // namespace test
613 } // namespace webrtc 406 } // namespace webrtc
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