1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file.
5 #include "base/basictypes.h"
6 #include "base/environment.h"
7 #include "base/file_util.h"
8 #include "base/memory/scoped_ptr.h"
9 #include "base/message_loop.h"
10 #include "base/path_service.h"
11 #include "base/synchronization/lock.h"
12 #include "base/test/test_timeouts.h"
13 #include "base/time.h"
14 #include "build/build_config.h"
15 #include "media/audio/audio_io.h"
16 #include "media/audio/audio_manager_base.h"
17 #include "media/base/seekable_buffer.h"
18 #include "testing/gmock/include/gmock/gmock.h"
19 #include "testing/gtest/include/gtest/gtest.h"
21 #if defined(OS_LINUX) || defined(OS_OPENBSD)
22 #include "media/audio/linux/audio_manager_linux.h"
23 #elif defined(OS_MACOSX)
24 #include "media/audio/mac/audio_manager_mac.h"
26 #include "media/audio/win/audio_manager_win.h"
27 #include "media/audio/win/core_audio_util_win.h"
28 #elif defined(OS_ANDROID)
29 #include "media/audio/android/audio_manager_android.h"
34 #if defined(OS_LINUX) || defined(OS_OPENBSD)
35 typedef AudioManagerLinux AudioManagerAnyPlatform
;
36 #elif defined(OS_MACOSX)
37 typedef AudioManagerMac AudioManagerAnyPlatform
;
39 typedef AudioManagerWin AudioManagerAnyPlatform
;
40 #elif defined(OS_ANDROID)
41 typedef AudioManagerAndroid AudioManagerAnyPlatform
;
44 // Limits the number of delay measurements we can store in an array and
45 // then write to file at end of the WASAPIAudioInputOutputFullDuplex test.
46 static const size_t kMaxDelayMeasurements
= 1000;
48 // Name of the output text file. The output file will be stored in the
49 // directory containing media_unittests.exe.
50 // Example: \src\build\Debug\audio_delay_values_ms.txt.
51 // See comments for the WASAPIAudioInputOutputFullDuplex test for more details
52 // about the file format.
53 static const char kDelayValuesFileName
[] = "audio_delay_values_ms.txt";
55 // Contains delay values which are reported during the full-duplex test.
56 // Total delay = |buffer_delay_ms| + |input_delay_ms| + |output_delay_ms|.
57 struct AudioDelayState
{
65 // Time in milliseconds since last delay report. Typical value is ~10 [ms].
68 // Size of internal sync buffer. Typical value is ~0 [ms].
71 // Reported capture/input delay. Typical value is ~10 [ms].
74 // Reported render/output delay. Typical value is ~40 [ms].
78 // This class mocks the platform specific audio manager and overrides
79 // the GetMessageLoop() method to ensure that we can run our tests on
80 // the main thread instead of the audio thread.
81 class MockAudioManager
: public AudioManagerAnyPlatform
{
84 virtual ~MockAudioManager() {}
86 virtual scoped_refptr
<base::MessageLoopProxy
> GetMessageLoop() OVERRIDE
{
87 return MessageLoop::current()->message_loop_proxy();
91 DISALLOW_COPY_AND_ASSIGN(MockAudioManager
);
94 // Test fixture class.
95 class AudioLowLatencyInputOutputTest
: public testing::Test
{
97 AudioLowLatencyInputOutputTest() {}
99 virtual ~AudioLowLatencyInputOutputTest() {}
101 AudioManager
* audio_manager() { return &mock_audio_manager_
; }
102 MessageLoopForUI
* message_loop() { return &message_loop_
; }
104 // Convenience method which ensures that we are not running on the build
105 // bots and that at least one valid input and output device can be found.
106 bool CanRunAudioTests() {
107 bool input
= audio_manager()->HasAudioInputDevices();
108 bool output
= audio_manager()->HasAudioOutputDevices();
109 LOG_IF(WARNING
, !input
) << "No input device detected.";
110 LOG_IF(WARNING
, !output
) << "No output device detected.";
111 return input
&& output
;
115 MessageLoopForUI message_loop_
;
116 MockAudioManager mock_audio_manager_
;
118 DISALLOW_COPY_AND_ASSIGN(AudioLowLatencyInputOutputTest
);
121 // This audio source/sink implementation should be used for manual tests
122 // only since delay measurements are stored on an output text file.
123 // All incoming/recorded audio packets are stored in an intermediate media
124 // buffer which the renderer reads from when it needs audio for playout.
125 // The total effect is that recorded audio is played out in loop back using
126 // a sync buffer as temporary storage.
127 class FullDuplexAudioSinkSource
128 : public AudioInputStream::AudioInputCallback
,
129 public AudioOutputStream::AudioSourceCallback
{
131 FullDuplexAudioSinkSource(int sample_rate
,
132 int samples_per_packet
,
134 : sample_rate_(sample_rate
),
135 samples_per_packet_(samples_per_packet
),
137 input_elements_to_write_(0),
138 output_elements_to_write_(0),
139 previous_write_time_(base::Time::Now()) {
140 // Size in bytes of each audio frame (4 bytes for 16-bit stereo PCM).
141 frame_size_
= (16 / 8) * channels_
;
143 // Start with the smallest possible buffer size. It will be increased
144 // dynamically during the test if required.
146 new media::SeekableBuffer(0, samples_per_packet_
* frame_size_
));
148 frames_to_ms_
= static_cast<double>(1000.0 / sample_rate_
);
149 delay_states_
.reset(new AudioDelayState
[kMaxDelayMeasurements
]);
152 virtual ~FullDuplexAudioSinkSource() {
153 // Get complete file path to output file in the directory containing
154 // media_unittests.exe. Example: src/build/Debug/audio_delay_values_ms.txt.
155 base::FilePath file_name
;
156 EXPECT_TRUE(PathService::Get(base::DIR_EXE
, &file_name
));
157 file_name
= file_name
.AppendASCII(kDelayValuesFileName
);
159 FILE* text_file
= file_util::OpenFile(file_name
, "wt");
160 DLOG_IF(ERROR
, !text_file
) << "Failed to open log file.";
161 LOG(INFO
) << ">> Output file " << file_name
.value() << " has been created.";
163 // Write the array which contains time-stamps, buffer size and
164 // audio delays values to a text file.
165 size_t elements_written
= 0;
166 while (elements_written
<
167 std::min(input_elements_to_write_
, output_elements_to_write_
)) {
168 const AudioDelayState state
= delay_states_
[elements_written
];
169 fprintf(text_file
, "%d %d %d %d\n",
171 state
.buffer_delay_ms
,
172 state
.input_delay_ms
,
173 state
.output_delay_ms
);
177 file_util::CloseFile(text_file
);
180 // AudioInputStream::AudioInputCallback.
181 virtual void OnData(AudioInputStream
* stream
,
182 const uint8
* src
, uint32 size
,
183 uint32 hardware_delay_bytes
,
184 double volume
) OVERRIDE
{
185 base::AutoLock
lock(lock_
);
187 // Update three components in the AudioDelayState for this recorded
189 base::Time now_time
= base::Time::Now();
190 int diff
= (now_time
- previous_write_time_
).InMilliseconds();
191 previous_write_time_
= now_time
;
192 if (input_elements_to_write_
< kMaxDelayMeasurements
) {
193 delay_states_
[input_elements_to_write_
].delta_time_ms
= diff
;
194 delay_states_
[input_elements_to_write_
].buffer_delay_ms
=
195 BytesToMilliseconds(buffer_
->forward_bytes());
196 delay_states_
[input_elements_to_write_
].input_delay_ms
=
197 BytesToMilliseconds(hardware_delay_bytes
);
198 ++input_elements_to_write_
;
201 // Store the captured audio packet in a seekable media buffer.
202 if (!buffer_
->Append(src
, size
)) {
203 // An attempt to write outside the buffer limits has been made.
204 // Double the buffer capacity to ensure that we have a buffer large
205 // enough to handle the current sample test scenario.
206 buffer_
->set_forward_capacity(2 * buffer_
->forward_capacity());
211 virtual void OnClose(AudioInputStream
* stream
) OVERRIDE
{}
212 virtual void OnError(AudioInputStream
* stream
) OVERRIDE
{}
214 // AudioOutputStream::AudioSourceCallback.
215 virtual int OnMoreData(AudioBus
* audio_bus
,
216 AudioBuffersState buffers_state
) OVERRIDE
{
217 base::AutoLock
lock(lock_
);
219 // Update one component in the AudioDelayState for the packet
220 // which is about to be played out.
221 if (output_elements_to_write_
< kMaxDelayMeasurements
) {
222 int output_delay_bytes
= buffers_state
.hardware_delay_bytes
;
224 // Special fix for Windows in combination with Wave where the
225 // pending bytes field of the audio buffer state is used to
227 if (!CoreAudioUtil::IsSupported()) {
228 output_delay_bytes
= buffers_state
.pending_bytes
;
231 delay_states_
[output_elements_to_write_
].output_delay_ms
=
232 BytesToMilliseconds(output_delay_bytes
);
233 ++output_elements_to_write_
;
238 // Read the data from the seekable media buffer which contains
239 // captured data at the same size and sample rate as the output side.
240 if (buffer_
->GetCurrentChunk(&source
, &size
) && size
> 0) {
241 EXPECT_EQ(channels_
, audio_bus
->channels());
242 size
= std::min(audio_bus
->frames() * frame_size_
, size
);
243 EXPECT_EQ(static_cast<size_t>(size
) % sizeof(*audio_bus
->channel(0)), 0U);
244 audio_bus
->FromInterleaved(
245 source
, size
/ frame_size_
, frame_size_
/ channels_
);
247 return size
/ frame_size_
;
253 virtual int OnMoreIOData(AudioBus
* source
,
255 AudioBuffersState buffers_state
) OVERRIDE
{
260 virtual void OnError(AudioOutputStream
* stream
) OVERRIDE
{}
261 virtual void WaitTillDataReady() OVERRIDE
{}
264 // Converts from bytes to milliseconds taking the sample rate and size
265 // of an audio frame into account.
266 int BytesToMilliseconds(uint32 delay_bytes
) const {
267 return static_cast<int>((delay_bytes
/ frame_size_
) * frames_to_ms_
+ 0.5);
272 scoped_ptr
<media::SeekableBuffer
> buffer_
;
274 int samples_per_packet_
;
277 double frames_to_ms_
;
278 scoped_ptr
<AudioDelayState
[]> delay_states_
;
279 size_t input_elements_to_write_
;
280 size_t output_elements_to_write_
;
281 base::Time previous_write_time_
;
284 class AudioInputStreamTraits
{
286 typedef AudioInputStream StreamType
;
288 static AudioParameters
GetDefaultAudioStreamParameters(
289 AudioManager
* audio_manager
) {
290 return audio_manager
->GetInputStreamParameters(
291 AudioManagerBase::kDefaultDeviceId
);
294 static StreamType
* CreateStream(AudioManager
* audio_manager
,
295 const AudioParameters
& params
) {
296 return audio_manager
->MakeAudioInputStream(params
,
297 AudioManagerBase::kDefaultDeviceId
);
301 class AudioOutputStreamTraits
{
303 typedef AudioOutputStream StreamType
;
305 static AudioParameters
GetDefaultAudioStreamParameters(
306 AudioManager
* audio_manager
) {
307 return audio_manager
->GetDefaultOutputStreamParameters();
310 static StreamType
* CreateStream(AudioManager
* audio_manager
,
311 const AudioParameters
& params
) {
312 return audio_manager
->MakeAudioOutputStream(params
);
316 // Traits template holding a trait of StreamType. It encapsulates
317 // AudioInputStream and AudioOutputStream stream types.
318 template <typename StreamTraits
>
319 class StreamWrapper
{
321 typedef typename
StreamTraits::StreamType StreamType
;
323 explicit StreamWrapper(AudioManager
* audio_manager
)
325 audio_manager_(audio_manager
),
326 format_(AudioParameters::AUDIO_PCM_LOW_LATENCY
),
327 #if defined(OS_ANDROID)
328 channel_layout_(CHANNEL_LAYOUT_MONO
),
330 channel_layout_(CHANNEL_LAYOUT_STEREO
),
332 bits_per_sample_(16) {
333 // Use the preferred sample rate.
334 const AudioParameters
& params
=
335 StreamTraits::GetDefaultAudioStreamParameters(audio_manager_
);
336 sample_rate_
= params
.sample_rate();
338 // Use the preferred buffer size. Note that the input side uses the same
339 // size as the output side in this implementation.
340 samples_per_packet_
= params
.frames_per_buffer();
343 virtual ~StreamWrapper() {}
345 // Creates an Audio[Input|Output]Stream stream object using default
347 StreamType
* Create() {
348 return CreateStream();
351 int channels() const {
352 return ChannelLayoutToChannelCount(channel_layout_
);
354 int bits_per_sample() const { return bits_per_sample_
; }
355 int sample_rate() const { return sample_rate_
; }
356 int samples_per_packet() const { return samples_per_packet_
; }
359 StreamType
* CreateStream() {
360 StreamType
* stream
= StreamTraits::CreateStream(audio_manager_
,
361 AudioParameters(format_
, channel_layout_
, sample_rate_
,
362 bits_per_sample_
, samples_per_packet_
));
367 AudioManager
* audio_manager_
;
368 AudioParameters::Format format_
;
369 ChannelLayout channel_layout_
;
370 int bits_per_sample_
;
372 int samples_per_packet_
;
375 typedef StreamWrapper
<AudioInputStreamTraits
> AudioInputStreamWrapper
;
376 typedef StreamWrapper
<AudioOutputStreamTraits
> AudioOutputStreamWrapper
;
378 // This test is intended for manual tests and should only be enabled
379 // when it is required to make a real-time test of audio in full duplex and
380 // at the same time create a text file which contains measured delay values.
381 // The file can later be analyzed off line using e.g. MATLAB.
383 // D=load('audio_delay_values_ms.txt');
385 // plot(x, D(:,2), x, D(:,3), x, D(:,4), x, D(:,2)+D(:,3)+D(:,4));
386 // axis([0, max(x), 0, max(D(:,2)+D(:,3)+D(:,4))+10]);
387 // legend('buffer delay','input delay','output delay','total delay');
388 // xlabel('time [msec]')
389 // ylabel('delay [msec]')
390 // title('Full-duplex audio delay measurement');
391 TEST_F(AudioLowLatencyInputOutputTest
, DISABLED_FullDuplexDelayMeasurement
) {
392 if (!CanRunAudioTests())
395 AudioInputStreamWrapper
aisw(audio_manager());
396 AudioInputStream
* ais
= aisw
.Create();
399 AudioOutputStreamWrapper
aosw(audio_manager());
400 AudioOutputStream
* aos
= aosw
.Create();
403 // This test only supports identical parameters in both directions.
404 // TODO(henrika): it is possible to cut delay here by using different
405 // buffer sizes for input and output.
406 if (aisw
.sample_rate() != aosw
.sample_rate() ||
407 aisw
.samples_per_packet() != aosw
.samples_per_packet() ||
408 aisw
.channels()!= aosw
.channels() ||
409 aisw
.bits_per_sample() != aosw
.bits_per_sample()) {
410 LOG(ERROR
) << "This test requires symmetric input and output parameters. "
411 "Ensure that sample rate and number of channels are identical in "
418 EXPECT_TRUE(ais
->Open());
419 EXPECT_TRUE(aos
->Open());
421 FullDuplexAudioSinkSource
full_duplex(
422 aisw
.sample_rate(), aisw
.samples_per_packet(), aisw
.channels());
424 LOG(INFO
) << ">> You should now be able to hear yourself in loopback...";
425 DLOG(INFO
) << " sample_rate : " << aisw
.sample_rate();
426 DLOG(INFO
) << " samples_per_packet: " << aisw
.samples_per_packet();
427 DLOG(INFO
) << " channels : " << aisw
.channels();
429 ais
->Start(&full_duplex
);
430 aos
->Start(&full_duplex
);
432 // Wait for approximately 10 seconds. The user shall hear his own voice
433 // in loop back during this time. At the same time, delay recordings are
434 // performed and stored in the output text file.
435 message_loop()->PostDelayedTask(FROM_HERE
,
436 MessageLoop::QuitClosure(), TestTimeouts::action_timeout());
437 message_loop()->Run();
442 // All Close() operations that run on the mocked audio thread,
443 // should be synchronous and not post additional close tasks to
444 // mocked the audio thread. Hence, there is no need to call
445 // message_loop()->RunUntilIdle() after the Close() methods.