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         : mStreamInternalPlay(audioService, true) {
46     ALOGD("%s(%p) created", __func__, this);
47     mStreamInternal = &mStreamInternalPlay;
48 }
49 
~AAudioServiceEndpointPlay()50 AAudioServiceEndpointPlay::~AAudioServiceEndpointPlay() {
51     ALOGD("%s(%p) destroyed", __func__, this);
52 }
53 
open(const aaudio::AAudioStreamRequest & request)54 aaudio_result_t AAudioServiceEndpointPlay::open(const aaudio::AAudioStreamRequest &request) {
55     aaudio_result_t result = AAudioServiceEndpointShared::open(request);
56     if (result == AAUDIO_OK) {
57         mMixer.allocate(getStreamInternal()->getSamplesPerFrame(),
58                         getStreamInternal()->getFramesPerBurst());
59 
60         int32_t burstsPerBuffer = AAudioProperty_getMixerBursts();
61         if (burstsPerBuffer == 0) {
62             mLatencyTuningEnabled = true;
63             burstsPerBuffer = BURSTS_PER_BUFFER_DEFAULT;
64         }
65         int32_t desiredBufferSize = burstsPerBuffer * getStreamInternal()->getFramesPerBurst();
66         getStreamInternal()->setBufferSize(desiredBufferSize);
67     }
68     return result;
69 }
70 
71 // Mix data from each application stream and write result to the shared MMAP stream.
callbackLoop()72 void *AAudioServiceEndpointPlay::callbackLoop() {
73     ALOGD("%s() entering >>>>>>>>>>>>>>> MIXER", __func__);
74     aaudio_result_t result = AAUDIO_OK;
75     int64_t timeoutNanos = getStreamInternal()->calculateReasonableTimeout();
76 
77     // result might be a frame count
78     while (mCallbackEnabled.load() && getStreamInternal()->isActive() && (result >= 0)) {
79         // Mix data from each active stream.
80         mMixer.clear();
81 
82         { // brackets are for lock_guard
83             int index = 0;
84             int64_t mmapFramesWritten = getStreamInternal()->getFramesWritten();
85 
86             std::lock_guard <std::mutex> lock(mLockStreams);
87             for (const auto clientStream : mRegisteredStreams) {
88                 int64_t clientFramesRead = 0;
89                 bool allowUnderflow = true;
90 
91                 aaudio_stream_state_t state = clientStream->getState();
92                 if (state == AAUDIO_STREAM_STATE_STOPPING) {
93                     allowUnderflow = false; // just read what is already in the FIFO
94                 } else if (state != AAUDIO_STREAM_STATE_STARTED) {
95                     continue; // this stream is not running so skip it.
96                 }
97 
98                 sp<AAudioServiceStreamShared> streamShared =
99                         static_cast<AAudioServiceStreamShared *>(clientStream.get());
100 
101                 {
102                     // Lock the AudioFifo to protect against close.
103                     std::lock_guard <std::mutex> lock(streamShared->getAudioDataQueueLock());
104 
105                     FifoBuffer *fifo = streamShared->getAudioDataFifoBuffer_l();
106                     if (fifo != nullptr) {
107 
108                         // Determine offset between framePosition in client's stream
109                         // vs the underlying MMAP stream.
110                         clientFramesRead = fifo->getReadCounter();
111                         // These two indices refer to the same frame.
112                         int64_t positionOffset = mmapFramesWritten - clientFramesRead;
113                         streamShared->setTimestampPositionOffset(positionOffset);
114 
115                         int32_t framesMixed = mMixer.mix(index, fifo, allowUnderflow);
116 
117                         if (streamShared->isFlowing()) {
118                             // Consider it an underflow if we got less than a burst
119                             // after the data started flowing.
120                             bool underflowed = allowUnderflow
121                                                && framesMixed < mMixer.getFramesPerBurst();
122                             if (underflowed) {
123                                 streamShared->incrementXRunCount();
124                             }
125                         } else if (framesMixed > 0) {
126                             // Mark beginning of data flow after a start.
127                             streamShared->setFlowing(true);
128                         }
129                         clientFramesRead = fifo->getReadCounter();
130                     }
131                 }
132 
133                 if (clientFramesRead > 0) {
134                     // This timestamp represents the completion of data being read out of the
135                     // client buffer. It is sent to the client and used in the timing model
136                     // to decide when the client has room to write more data.
137                     Timestamp timestamp(clientFramesRead, AudioClock::getNanoseconds());
138                     streamShared->markTransferTime(timestamp);
139                 }
140 
141                 index++; // just used for labelling tracks in systrace
142             }
143         }
144 
145         // Write mixer output to stream using a blocking write.
146         result = getStreamInternal()->write(mMixer.getOutputBuffer(),
147                                             getFramesPerBurst(), timeoutNanos);
148         if (result == AAUDIO_ERROR_DISCONNECTED) {
149             AAudioServiceEndpointShared::disconnectRegisteredStreams();
150             break;
151         } else if (result != getFramesPerBurst()) {
152             ALOGW("callbackLoop() wrote %d / %d",
153                   result, getFramesPerBurst());
154             break;
155         }
156     }
157 
158     ALOGD("%s() exiting, enabled = %d, state = %d, result = %d <<<<<<<<<<<<< MIXER",
159           __func__, mCallbackEnabled.load(), getStreamInternal()->getState(), result);
160     return NULL; // TODO review
161 }
162