1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
|
/*
* Copyright 2015 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#include <sys/types.h>
#include <pthread.h>
#include <thread>
#include <oboe/AudioStream.h>
#include "OboeDebug.h"
#include "AudioClock.h"
#include <oboe/Utilities.h>
namespace oboe {
/*
* AudioStream
*/
AudioStream::AudioStream(const AudioStreamBuilder &builder)
: AudioStreamBase(builder) {
}
Result AudioStream::close() {
closePerformanceHint();
// Update local counters so they can be read after the close.
updateFramesWritten();
updateFramesRead();
return Result::OK;
}
// Call this from fireDataCallback() if you want to monitor CPU scheduler.
void AudioStream::checkScheduler() {
int scheduler = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK; // for current thread
if (scheduler != mPreviousScheduler) {
LOGD("AudioStream::%s() scheduler = %s", __func__,
((scheduler == SCHED_FIFO) ? "SCHED_FIFO" :
((scheduler == SCHED_OTHER) ? "SCHED_OTHER" :
((scheduler == SCHED_RR) ? "SCHED_RR" : "UNKNOWN")))
);
mPreviousScheduler = scheduler;
}
}
DataCallbackResult AudioStream::fireDataCallback(void *audioData, int32_t numFrames) {
if (!isDataCallbackEnabled()) {
LOGW("AudioStream::%s() called with data callback disabled!", __func__);
return DataCallbackResult::Stop; // Should not be getting called
}
beginPerformanceHintInCallback();
// Call the app to do the work.
DataCallbackResult result;
if (mDataCallback) {
result = mDataCallback->onAudioReady(this, audioData, numFrames);
} else {
result = onDefaultCallback(audioData, numFrames);
}
// On Oreo, we might get called after returning stop.
// So block that here.
setDataCallbackEnabled(result == DataCallbackResult::Continue);
endPerformanceHintInCallback(numFrames);
return result;
}
Result AudioStream::waitForStateTransition(StreamState startingState,
StreamState endingState,
int64_t timeoutNanoseconds)
{
StreamState state;
{
std::lock_guard<std::mutex> lock(mLock);
state = getState();
if (state == StreamState::Closed) {
return Result::ErrorClosed;
} else if (state == StreamState::Disconnected) {
return Result::ErrorDisconnected;
}
}
StreamState nextState = state;
// TODO Should this be a while()?!
if (state == startingState && state != endingState) {
Result result = waitForStateChange(state, &nextState, timeoutNanoseconds);
if (result != Result::OK) {
return result;
}
}
if (nextState != endingState) {
return Result::ErrorInvalidState;
} else {
return Result::OK;
}
}
Result AudioStream::start(int64_t timeoutNanoseconds)
{
Result result = requestStart();
if (result != Result::OK) return result;
if (timeoutNanoseconds <= 0) return result;
return waitForStateTransition(StreamState::Starting,
StreamState::Started, timeoutNanoseconds);
}
Result AudioStream::pause(int64_t timeoutNanoseconds)
{
Result result = requestPause();
if (result != Result::OK) return result;
if (timeoutNanoseconds <= 0) return result;
return waitForStateTransition(StreamState::Pausing,
StreamState::Paused, timeoutNanoseconds);
}
Result AudioStream::flush(int64_t timeoutNanoseconds)
{
Result result = requestFlush();
if (result != Result::OK) return result;
if (timeoutNanoseconds <= 0) return result;
return waitForStateTransition(StreamState::Flushing,
StreamState::Flushed, timeoutNanoseconds);
}
Result AudioStream::stop(int64_t timeoutNanoseconds)
{
Result result = requestStop();
if (result != Result::OK) return result;
if (timeoutNanoseconds <= 0) return result;
return waitForStateTransition(StreamState::Stopping,
StreamState::Stopped, timeoutNanoseconds);
}
int32_t AudioStream::getBytesPerSample() const {
return convertFormatToSizeInBytes(mFormat);
}
int64_t AudioStream::getFramesRead() {
updateFramesRead();
return mFramesRead;
}
int64_t AudioStream::getFramesWritten() {
updateFramesWritten();
return mFramesWritten;
}
ResultWithValue<int32_t> AudioStream::getAvailableFrames() {
int64_t readCounter = getFramesRead();
if (readCounter < 0) return ResultWithValue<int32_t>::createBasedOnSign(readCounter);
int64_t writeCounter = getFramesWritten();
if (writeCounter < 0) return ResultWithValue<int32_t>::createBasedOnSign(writeCounter);
int32_t framesAvailable = writeCounter - readCounter;
return ResultWithValue<int32_t>(framesAvailable);
}
ResultWithValue<int32_t> AudioStream::waitForAvailableFrames(int32_t numFrames,
int64_t timeoutNanoseconds) {
if (numFrames == 0) return Result::OK;
if (numFrames < 0) return Result::ErrorOutOfRange;
// Make sure we don't try to wait for more frames than the buffer can hold.
// Subtract framesPerBurst because this is often called from a callback
// and we don't want to be sleeping if the buffer is close to overflowing.
const int32_t maxAvailableFrames = getBufferCapacityInFrames() - getFramesPerBurst();
numFrames = std::min(numFrames, maxAvailableFrames);
// The capacity should never be less than one burst. But clip to zero just in case.
numFrames = std::max(0, numFrames);
int64_t framesAvailable = 0;
int64_t burstInNanos = getFramesPerBurst() * kNanosPerSecond / getSampleRate();
bool ready = false;
int64_t deadline = AudioClock::getNanoseconds() + timeoutNanoseconds;
do {
ResultWithValue<int32_t> result = getAvailableFrames();
if (!result) return result;
framesAvailable = result.value();
ready = (framesAvailable >= numFrames);
if (!ready) {
int64_t now = AudioClock::getNanoseconds();
if (now > deadline) break;
AudioClock::sleepForNanos(burstInNanos);
}
} while (!ready);
return (!ready)
? ResultWithValue<int32_t>(Result::ErrorTimeout)
: ResultWithValue<int32_t>(framesAvailable);
}
ResultWithValue<FrameTimestamp> AudioStream::getTimestamp(clockid_t clockId) {
FrameTimestamp frame;
Result result = getTimestamp(clockId, &frame.position, &frame.timestamp);
if (result == Result::OK){
return ResultWithValue<FrameTimestamp>(frame);
} else {
return ResultWithValue<FrameTimestamp>(static_cast<Result>(result));
}
}
void AudioStream::calculateDefaultDelayBeforeCloseMillis() {
// Calculate delay time before close based on burst duration.
// Start with a burst duration then add 1 msec as a safety margin.
mDelayBeforeCloseMillis = std::max(kMinDelayBeforeCloseMillis,
1 + ((mFramesPerBurst * 1000) / getSampleRate()));
LOGD("calculateDefaultDelayBeforeCloseMillis() default = %d",
static_cast<int>(mDelayBeforeCloseMillis));
}
} // namespace oboe
|
