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1 /*
2  * Copyright (C) 2017 The Android Open Source Project
3  *
4  * Licensed under the Apache License, Version 2.0 (the "License");
5  * you may not use this file except in compliance with the License.
6  * You may obtain a copy of the License at
7  *
8  *      http://www.apache.org/licenses/LICENSE-2.0
9  *
10  * Unless required by applicable law or agreed to in writing, software
11  * distributed under the License is distributed on an "AS IS" BASIS,
12  * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
13  * See the License for the specific language governing permissions and
14  * limitations under the License.
15  */
16 
17 #define LOG_TAG "AAudioServiceEndpointPlay"
18 //#define LOG_NDEBUG 0
19 #include <utils/Log.h>
20 
21 #include <assert.h>
22 #include <map>
23 #include <mutex>
24 #include <utils/Singleton.h>
25 
26 #include "AAudioEndpointManager.h"
27 #include "AAudioServiceEndpoint.h"
28 #include <algorithm>
29 #include <mutex>
30 #include <vector>
31 
32 #include "core/AudioStreamBuilder.h"
33 #include "AAudioServiceEndpoint.h"
34 #include "AAudioServiceStreamShared.h"
35 #include "AAudioServiceEndpointPlay.h"
36 #include "AAudioServiceEndpointShared.h"
37 #include "AAudioServiceStreamBase.h"
38 
39 using namespace android;  // TODO just import names needed
40 using namespace aaudio;   // TODO just import names needed
41 
42 #define BURSTS_PER_BUFFER_DEFAULT   2
43 
AAudioServiceEndpointPlay(AAudioService & audioService)44 AAudioServiceEndpointPlay::AAudioServiceEndpointPlay(AAudioService& audioService)
45         : AAudioServiceEndpointShared(
46                 new AudioStreamInternalPlay(audioService.asAAudioServiceInterface(), true)) {}
47 
open(const aaudio::AAudioStreamRequest & request)48 aaudio_result_t AAudioServiceEndpointPlay::open(const aaudio::AAudioStreamRequest &request) {
49     aaudio_result_t result = AAudioServiceEndpointShared::open(request);
50     if (result == AAUDIO_OK) {
51         mMixer.allocate(getStreamInternal()->getSamplesPerFrame(),
52                         getStreamInternal()->getFramesPerBurst());
53 
54         int32_t burstsPerBuffer = AAudioProperty_getMixerBursts();
55         if (burstsPerBuffer == 0) {
56             mLatencyTuningEnabled = true;
57             burstsPerBuffer = BURSTS_PER_BUFFER_DEFAULT;
58         }
59         int32_t desiredBufferSize = burstsPerBuffer * getStreamInternal()->getFramesPerBurst();
60         getStreamInternal()->setBufferSize(desiredBufferSize);
61     }
62     return result;
63 }
64 
65 // Mix data from each application stream and write result to the shared MMAP stream.
callbackLoop()66 void *AAudioServiceEndpointPlay::callbackLoop() {
67     ALOGD("%s() entering >>>>>>>>>>>>>>> MIXER", __func__);
68     aaudio_result_t result = AAUDIO_OK;
69     int64_t timeoutNanos = getStreamInternal()->calculateReasonableTimeout();
70 
71     // result might be a frame count
72     while (mCallbackEnabled.load() && getStreamInternal()->isActive() && (result >= 0)) {
73         // Mix data from each active stream.
74         mMixer.clear();
75 
76         { // brackets are for lock_guard
77             int index = 0;
78             int64_t mmapFramesWritten = getStreamInternal()->getFramesWritten();
79 
80             std::lock_guard <std::mutex> lock(mLockStreams);
81             for (const auto& clientStream : mRegisteredStreams) {
82                 int64_t clientFramesRead = 0;
83                 bool allowUnderflow = true;
84 
85                 if (clientStream->isSuspended()) {
86                     continue; // dead stream
87                 }
88 
89                 aaudio_stream_state_t state = clientStream->getState();
90                 if (state == AAUDIO_STREAM_STATE_STOPPING) {
91                     allowUnderflow = false; // just read what is already in the FIFO
92                 } else if (state != AAUDIO_STREAM_STATE_STARTED) {
93                     continue; // this stream is not running so skip it.
94                 }
95 
96                 sp<AAudioServiceStreamShared> streamShared =
97                         static_cast<AAudioServiceStreamShared *>(clientStream.get());
98 
99                 {
100                     // Lock the AudioFifo to protect against close.
101                     std::lock_guard <std::mutex> lock(streamShared->audioDataQueueLock);
102                     std::shared_ptr<SharedRingBuffer> audioDataQueue
103                             = streamShared->getAudioDataQueue_l();
104                     std::shared_ptr<FifoBuffer> fifo;
105                     if (audioDataQueue && (fifo = audioDataQueue->getFifoBuffer())) {
106 
107                         // Determine offset between framePosition in client's stream
108                         // vs the underlying MMAP stream.
109                         clientFramesRead = fifo->getReadCounter();
110                         // These two indices refer to the same frame.
111                         int64_t positionOffset = mmapFramesWritten - clientFramesRead;
112                         streamShared->setTimestampPositionOffset(positionOffset);
113 
114                         int32_t framesMixed = mMixer.mix(index, fifo, allowUnderflow);
115 
116                         if (streamShared->isFlowing()) {
117                             // Consider it an underflow if we got less than a burst
118                             // after the data started flowing.
119                             bool underflowed = allowUnderflow
120                                                && framesMixed < mMixer.getFramesPerBurst();
121                             if (underflowed) {
122                                 streamShared->incrementXRunCount();
123                             }
124                         } else if (framesMixed > 0) {
125                             // Mark beginning of data flow after a start.
126                             streamShared->setFlowing(true);
127                         }
128                         clientFramesRead = fifo->getReadCounter();
129                     }
130                 }
131 
132                 if (clientFramesRead > 0) {
133                     // This timestamp represents the completion of data being read out of the
134                     // client buffer. It is sent to the client and used in the timing model
135                     // to decide when the client has room to write more data.
136                     Timestamp timestamp(clientFramesRead, AudioClock::getNanoseconds());
137                     streamShared->markTransferTime(timestamp);
138                 }
139 
140                 index++; // just used for labelling tracks in systrace
141             }
142         }
143 
144         // Write mixer output to stream using a blocking write.
145         result = getStreamInternal()->write(mMixer.getOutputBuffer(),
146                                             getFramesPerBurst(), timeoutNanos);
147         if (result == AAUDIO_ERROR_DISCONNECTED) {
148             ALOGD("%s() write() returned AAUDIO_ERROR_DISCONNECTED", __func__);
149             // We do not need the returned vector.
150             (void) AAudioServiceEndpointShared::disconnectRegisteredStreams();
151             break;
152         } else if (result != getFramesPerBurst()) {
153             ALOGW("callbackLoop() wrote %d / %d",
154                   result, getFramesPerBurst());
155             break;
156         }
157     }
158 
159     ALOGD("%s() exiting, enabled = %d, state = %d, result = %d <<<<<<<<<<<<< MIXER",
160           __func__, mCallbackEnabled.load(), getStreamInternal()->getState(), result);
161     return NULL; // TODO review
162 }
163