blob: 8fe85692425766dc118035728ae40c0ccea4fac9 [file] [log] [blame]
Phil Burk204a1632017-01-03 17:23:43 -08001/*
2 * Copyright (C) 2016 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
Phil Burka9876702020-04-20 18:16:15 -070017#define LOG_TAG "AudioStreamInternal"
Phil Burk204a1632017-01-03 17:23:43 -080018//#define LOG_NDEBUG 0
19#include <utils/Log.h>
20
Phil Burk4485d412017-05-09 15:55:02 -070021#define ATRACE_TAG ATRACE_TAG_AUDIO
22
Phil Burkc0c70e32017-02-09 13:18:38 -080023#include <stdint.h>
Phil Burk204a1632017-01-03 17:23:43 -080024
25#include <binder/IServiceManager.h>
26
Phil Burk5ed503c2017-02-01 09:38:15 -080027#include <aaudio/AAudio.h>
Phil Burkfd34a932017-07-19 07:03:52 -070028#include <cutils/properties.h>
Phil Burka9876702020-04-20 18:16:15 -070029
Robert Wud7400832021-12-04 01:11:19 +000030#include <media/AudioParameter.h>
jiabine504e7b2021-09-18 00:27:08 +000031#include <media/AudioSystem.h>
Phil Burka9876702020-04-20 18:16:15 -070032#include <media/MediaMetricsItem.h>
Phil Burk4485d412017-05-09 15:55:02 -070033#include <utils/Trace.h>
Phil Burk204a1632017-01-03 17:23:43 -080034
Phil Burkc0c70e32017-02-09 13:18:38 -080035#include "AudioEndpointParcelable.h"
36#include "binding/AAudioStreamRequest.h"
37#include "binding/AAudioStreamConfiguration.h"
Phil Burk5ed503c2017-02-01 09:38:15 -080038#include "binding/AAudioServiceMessage.h"
Phil Burka9876702020-04-20 18:16:15 -070039#include "core/AudioGlobal.h"
Phil Burk3df348f2017-02-08 11:41:55 -080040#include "core/AudioStreamBuilder.h"
Phil Burke572f462017-04-20 13:03:19 -070041#include "fifo/FifoBuffer.h"
Phil Burkfd34a932017-07-19 07:03:52 -070042#include "utility/AudioClock.h"
Philip P. Moltmannbda45752020-07-17 16:41:18 -070043#include <media/AidlConversion.h>
Phil Burke572f462017-04-20 13:03:19 -070044
Phil Burkc0c70e32017-02-09 13:18:38 -080045#include "AudioStreamInternal.h"
Phil Burk204a1632017-01-03 17:23:43 -080046
Phil Burka9876702020-04-20 18:16:15 -070047// We do this after the #includes because if a header uses ALOG.
48// it would fail on the reference to mInService.
49#undef LOG_TAG
50// This file is used in both client and server processes.
51// This is needed to make sense of the logs more easily.
52#define LOG_TAG (mInService ? "AudioStreamInternal_Service" : "AudioStreamInternal_Client")
53
Svet Ganov3e5f14f2021-05-13 22:51:08 +000054using android::content::AttributionSourceState;
Phil Burk204a1632017-01-03 17:23:43 -080055
Phil Burk5ed503c2017-02-01 09:38:15 -080056using namespace aaudio;
Phil Burk204a1632017-01-03 17:23:43 -080057
Phil Burke4d7bb42017-03-28 11:32:39 -070058#define MIN_TIMEOUT_NANOS (1000 * AAUDIO_NANOS_PER_MILLISECOND)
59
60// Wait at least this many times longer than the operation should take.
61#define MIN_TIMEOUT_OPERATIONS 4
62
Phil Burkbcc36742017-08-31 17:24:51 -070063#define LOG_TIMESTAMPS 0
Phil Burk87c9f642017-05-17 07:22:39 -070064
Phil Burkc0c70e32017-02-09 13:18:38 -080065AudioStreamInternal::AudioStreamInternal(AAudioServiceInterface &serviceInterface, bool inService)
Phil Burk204a1632017-01-03 17:23:43 -080066 : AudioStream()
67 , mClockModel()
Phil Burk5ed503c2017-02-01 09:38:15 -080068 , mServiceStreamHandle(AAUDIO_HANDLE_INVALID)
Phil Burkec89b2e2017-06-20 15:05:06 -070069 , mInService(inService)
Phil Burkfd34a932017-07-19 07:03:52 -070070 , mServiceInterface(serviceInterface)
Phil Burka53ffa62018-10-10 16:21:37 -070071 , mAtomicInternalTimestamp()
Phil Burkfd34a932017-07-19 07:03:52 -070072 , mWakeupDelayNanos(AAudioProperty_getWakeupDelayMicros() * AAUDIO_NANOS_PER_MICROSECOND)
73 , mMinimumSleepNanos(AAudioProperty_getMinimumSleepMicros() * AAUDIO_NANOS_PER_MICROSECOND)
74 {
Phil Burk204a1632017-01-03 17:23:43 -080075}
76
77AudioStreamInternal::~AudioStreamInternal() {
Phil Burkdd582922020-10-15 20:29:51 +000078 ALOGD("%s() %p called", __func__, this);
Phil Burk204a1632017-01-03 17:23:43 -080079}
80
Phil Burk5ed503c2017-02-01 09:38:15 -080081aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) {
Phil Burk204a1632017-01-03 17:23:43 -080082
Phil Burk5ed503c2017-02-01 09:38:15 -080083 aaudio_result_t result = AAUDIO_OK;
84 AAudioStreamRequest request;
Phil Burk99306c82017-08-14 12:38:58 -070085 AAudioStreamConfiguration configurationOutput;
Phil Burk204a1632017-01-03 17:23:43 -080086
Phil Burk99306c82017-08-14 12:38:58 -070087 if (getState() != AAUDIO_STREAM_STATE_UNINITIALIZED) {
Phil Burkfbf031e2017-10-12 15:58:31 -070088 ALOGE("%s - already open! state = %d", __func__, getState());
Phil Burk99306c82017-08-14 12:38:58 -070089 return AAUDIO_ERROR_INVALID_STATE;
90 }
91
92 // Copy requested parameters to the stream.
Phil Burk204a1632017-01-03 17:23:43 -080093 result = AudioStream::open(builder);
94 if (result < 0) {
95 return result;
96 }
97
jiabinef348b82021-04-19 16:53:08 +000098 const audio_format_t requestedFormat = getFormat();
Phil Burkc0c70e32017-02-09 13:18:38 -080099 // We have to do volume scaling. So we prefer FLOAT format.
jiabinef348b82021-04-19 16:53:08 +0000100 if (requestedFormat == AUDIO_FORMAT_DEFAULT) {
Phil Burk0127c1b2018-03-29 13:48:06 -0700101 setFormat(AUDIO_FORMAT_PCM_FLOAT);
Phil Burkc0c70e32017-02-09 13:18:38 -0800102 }
Phil Burk04e805b2018-03-27 09:13:53 -0700103 // Request FLOAT for the shared mixer or the device.
Phil Burk0127c1b2018-03-29 13:48:06 -0700104 request.getConfiguration().setFormat(AUDIO_FORMAT_PCM_FLOAT);
Phil Burkc0c70e32017-02-09 13:18:38 -0800105
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000106 // TODO b/182392769: use attribution source util
107 AttributionSourceState attributionSource;
108 attributionSource.uid = VALUE_OR_FATAL(android::legacy2aidl_uid_t_int32_t(getuid()));
109 attributionSource.pid = VALUE_OR_FATAL(android::legacy2aidl_pid_t_int32_t(getpid()));
110 attributionSource.packageName = builder.getOpPackageName();
111 attributionSource.attributionTag = builder.getAttributionTag();
112 attributionSource.token = sp<android::BBinder>::make();
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700113
Phil Burkdec33ab2017-01-17 14:48:16 -0800114 // Build the request to send to the server.
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000115 request.setAttributionSource(attributionSource);
Phil Burk71f35bb2017-04-13 16:05:07 -0700116 request.setSharingModeMatchRequired(isSharingModeMatchRequired());
Phil Burk41f19d82018-02-13 14:59:10 -0800117 request.setInService(isInService());
Phil Burkc0c70e32017-02-09 13:18:38 -0800118
Phil Burk204a1632017-01-03 17:23:43 -0800119 request.getConfiguration().setDeviceId(getDeviceId());
120 request.getConfiguration().setSampleRate(getSampleRate());
Phil Burk39f02dd2017-08-04 09:13:31 -0700121 request.getConfiguration().setDirection(getDirection());
Phil Burk71f35bb2017-04-13 16:05:07 -0700122 request.getConfiguration().setSharingMode(getSharingMode());
jiabina9094092021-06-28 20:36:45 +0000123 request.getConfiguration().setChannelMask(getChannelMask());
Phil Burk71f35bb2017-04-13 16:05:07 -0700124
Phil Burka62fb952018-01-16 12:44:06 -0800125 request.getConfiguration().setUsage(getUsage());
126 request.getConfiguration().setContentType(getContentType());
Jean-Michel Trivi656bfdc2021-09-20 18:42:37 -0700127 request.getConfiguration().setSpatializationBehavior(getSpatializationBehavior());
128 request.getConfiguration().setIsContentSpatialized(isContentSpatialized());
Phil Burka62fb952018-01-16 12:44:06 -0800129 request.getConfiguration().setInputPreset(getInputPreset());
Eric Laurentd17c8502019-10-24 15:58:35 -0700130 request.getConfiguration().setPrivacySensitive(isPrivacySensitive());
Phil Burka62fb952018-01-16 12:44:06 -0800131
Phil Burk3df348f2017-02-08 11:41:55 -0800132 request.getConfiguration().setBufferCapacity(builder.getBufferCapacity());
Phil Burk204a1632017-01-03 17:23:43 -0800133
Robert Wu310037a2022-09-06 21:48:18 +0000134 request.getConfiguration().setHardwareSamplesPerFrame(builder.getHardwareSamplesPerFrame());
135 request.getConfiguration().setHardwareSampleRate(builder.getHardwareSampleRate());
136 request.getConfiguration().setHardwareFormat(builder.getHardwareFormat());
137
Phil Burk41f19d82018-02-13 14:59:10 -0800138 mDeviceChannelCount = getSamplesPerFrame(); // Assume it will be the same. Update if not.
139
Phil Burk99306c82017-08-14 12:38:58 -0700140 mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
Phil Burk41f19d82018-02-13 14:59:10 -0800141 if (mServiceStreamHandle < 0
jiabina9094092021-06-28 20:36:45 +0000142 && (request.getConfiguration().getSamplesPerFrame() == 1
143 || request.getConfiguration().getChannelMask() == AAUDIO_CHANNEL_MONO)
Phil Burk41f19d82018-02-13 14:59:10 -0800144 && getDirection() == AAUDIO_DIRECTION_OUTPUT
145 && !isInService()) {
146 // if that failed then try switching from mono to stereo if OUTPUT.
147 // Only do this in the client. Otherwise we end up with a mono mixer in the service
148 // that writes to a stereo MMAP stream.
Phil Burk0127c1b2018-03-29 13:48:06 -0700149 ALOGD("%s() - openStream() returned %d, try switching from MONO to STEREO",
Phil Burk41f19d82018-02-13 14:59:10 -0800150 __func__, mServiceStreamHandle);
jiabina9094092021-06-28 20:36:45 +0000151 request.getConfiguration().setChannelMask(AAUDIO_CHANNEL_STEREO);
Phil Burk41f19d82018-02-13 14:59:10 -0800152 mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
153 }
Phil Burk204a1632017-01-03 17:23:43 -0800154 if (mServiceStreamHandle < 0) {
Phil Burk41f19d82018-02-13 14:59:10 -0800155 return mServiceStreamHandle;
Phil Burk204a1632017-01-03 17:23:43 -0800156 }
Phil Burk99306c82017-08-14 12:38:58 -0700157
Phil Burka9876702020-04-20 18:16:15 -0700158 // This must match the key generated in oboeservice/AAudioServiceStreamBase.cpp
159 // so the client can have permission to log.
jiabinfbf20302021-07-28 22:15:01 +0000160 if (!mInService) {
161 // No need to log if it is from service side.
162 mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_STREAM)
163 + std::to_string(mServiceStreamHandle);
164 }
Phil Burka9876702020-04-20 18:16:15 -0700165
jiabinef348b82021-04-19 16:53:08 +0000166 android::mediametrics::LogItem(mMetricsId)
167 .set(AMEDIAMETRICS_PROP_PERFORMANCEMODE,
jiabinc8da9032021-04-28 20:42:36 +0000168 AudioGlobal_convertPerformanceModeToText(builder.getPerformanceMode()))
169 .set(AMEDIAMETRICS_PROP_SHARINGMODE,
170 AudioGlobal_convertSharingModeToText(builder.getSharingMode()))
jiabinef348b82021-04-19 16:53:08 +0000171 .set(AMEDIAMETRICS_PROP_ENCODINGCLIENT,
172 android::toString(requestedFormat).c_str()).record();
173
Phil Burk99306c82017-08-14 12:38:58 -0700174 result = configurationOutput.validate();
175 if (result != AAUDIO_OK) {
176 goto error;
177 }
178 // Save results of the open.
jiabina9094092021-06-28 20:36:45 +0000179 if (getChannelMask() == AAUDIO_UNSPECIFIED) {
180 setChannelMask(configurationOutput.getChannelMask());
Phil Burk41f19d82018-02-13 14:59:10 -0800181 }
jiabina9094092021-06-28 20:36:45 +0000182
Phil Burk41f19d82018-02-13 14:59:10 -0800183 mDeviceChannelCount = configurationOutput.getSamplesPerFrame();
184
Phil Burk99306c82017-08-14 12:38:58 -0700185 setSampleRate(configurationOutput.getSampleRate());
Phil Burk99306c82017-08-14 12:38:58 -0700186 setDeviceId(configurationOutput.getDeviceId());
Phil Burk4e1af9f2018-01-03 15:54:35 -0800187 setSessionId(configurationOutput.getSessionId());
Phil Burk99306c82017-08-14 12:38:58 -0700188 setSharingMode(configurationOutput.getSharingMode());
189
Phil Burka62fb952018-01-16 12:44:06 -0800190 setUsage(configurationOutput.getUsage());
191 setContentType(configurationOutput.getContentType());
Jean-Michel Trivi656bfdc2021-09-20 18:42:37 -0700192 setSpatializationBehavior(configurationOutput.getSpatializationBehavior());
193 setIsContentSpatialized(configurationOutput.isContentSpatialized());
Phil Burka62fb952018-01-16 12:44:06 -0800194 setInputPreset(configurationOutput.getInputPreset());
195
Phil Burk99306c82017-08-14 12:38:58 -0700196 // Save device format so we can do format conversion and volume scaling together.
Phil Burk3d786cb2018-04-09 11:58:09 -0700197 setDeviceFormat(configurationOutput.getFormat());
Phil Burk99306c82017-08-14 12:38:58 -0700198
Robert Wu310037a2022-09-06 21:48:18 +0000199 setHardwareSamplesPerFrame(configurationOutput.getHardwareSamplesPerFrame());
200 setHardwareSampleRate(configurationOutput.getHardwareSampleRate());
201 setHardwareFormat(configurationOutput.getHardwareFormat());
202
Phil Burk99306c82017-08-14 12:38:58 -0700203 result = mServiceInterface.getStreamDescription(mServiceStreamHandle, mEndPointParcelable);
204 if (result != AAUDIO_OK) {
205 goto error;
206 }
207
208 // Resolve parcelable into a descriptor.
209 result = mEndPointParcelable.resolve(&mEndpointDescriptor);
210 if (result != AAUDIO_OK) {
211 goto error;
212 }
213
214 // Configure endpoint based on descriptor.
Phil Burk5edc4ea2020-04-17 08:15:42 -0700215 mAudioEndpoint = std::make_unique<AudioEndpoint>();
216 result = mAudioEndpoint->configure(&mEndpointDescriptor, getDirection());
Phil Burk99306c82017-08-14 12:38:58 -0700217 if (result != AAUDIO_OK) {
218 goto error;
219 }
220
jiabinf7f06152021-11-22 18:10:14 +0000221 if ((result = configureDataInformation(builder.getFramesPerDataCallback())) != AAUDIO_OK) {
222 goto error;
223 }
224
225 setState(AAUDIO_STREAM_STATE_OPEN);
226
227 return result;
228
229error:
230 safeReleaseClose();
231 return result;
232}
233
234aaudio_result_t AudioStreamInternal::configureDataInformation(int32_t callbackFrames) {
235 int32_t framesPerHardwareBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
Phil Burk3c4e6b52019-01-22 15:53:36 -0800236
237 // Scale up the burst size to meet the minimum equivalent in microseconds.
238 // This is to avoid waking the CPU too often when the HW burst is very small
239 // or at high sample rates.
jiabinf7f06152021-11-22 18:10:14 +0000240 int32_t framesPerBurst = framesPerHardwareBurst;
241 int32_t burstMicros = 0;
242 const int32_t burstMinMicros = android::AudioSystem::getAAudioHardwareBurstMinUsec();
Phil Burk3c4e6b52019-01-22 15:53:36 -0800243 do {
244 if (burstMicros > 0) { // skip first loop
245 framesPerBurst *= 2;
246 }
247 burstMicros = framesPerBurst * static_cast<int64_t>(1000000) / getSampleRate();
248 } while (burstMicros < burstMinMicros);
249 ALOGD("%s() original HW burst = %d, minMicros = %d => SW burst = %d\n",
250 __func__, framesPerHardwareBurst, burstMinMicros, framesPerBurst);
251
252 // Validate final burst size.
Phil Burk6479d502017-11-20 09:32:52 -0800253 if (framesPerBurst < MIN_FRAMES_PER_BURST || framesPerBurst > MAX_FRAMES_PER_BURST) {
254 ALOGE("%s - framesPerBurst out of range = %d", __func__, framesPerBurst);
jiabinf7f06152021-11-22 18:10:14 +0000255 return AAUDIO_ERROR_OUT_OF_RANGE;
Phil Burk99306c82017-08-14 12:38:58 -0700256 }
Phil Burk8d97b8e2020-09-25 23:18:14 +0000257 setFramesPerBurst(framesPerBurst); // only save good value
Phil Burk6479d502017-11-20 09:32:52 -0800258
Phil Burk5edc4ea2020-04-17 08:15:42 -0700259 mBufferCapacityInFrames = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
Phil Burk8d97b8e2020-09-25 23:18:14 +0000260 if (mBufferCapacityInFrames < getFramesPerBurst()
Phil Burk5edc4ea2020-04-17 08:15:42 -0700261 || mBufferCapacityInFrames > MAX_BUFFER_CAPACITY_IN_FRAMES) {
262 ALOGE("%s - bufferCapacity out of range = %d", __func__, mBufferCapacityInFrames);
jiabinf7f06152021-11-22 18:10:14 +0000263 return AAUDIO_ERROR_OUT_OF_RANGE;
Phil Burk99306c82017-08-14 12:38:58 -0700264 }
265
266 mClockModel.setSampleRate(getSampleRate());
Phil Burk3c4e6b52019-01-22 15:53:36 -0800267 mClockModel.setFramesPerBurst(framesPerHardwareBurst);
Phil Burk99306c82017-08-14 12:38:58 -0700268
Phil Burk134f1972017-12-08 13:06:11 -0800269 if (isDataCallbackSet()) {
jiabinf7f06152021-11-22 18:10:14 +0000270 mCallbackFrames = callbackFrames;
Phil Burk99306c82017-08-14 12:38:58 -0700271 if (mCallbackFrames > getBufferCapacity() / 2) {
Phil Burk29ccc292019-04-15 08:58:08 -0700272 ALOGW("%s - framesPerCallback too big = %d, capacity = %d",
Phil Burkfbf031e2017-10-12 15:58:31 -0700273 __func__, mCallbackFrames, getBufferCapacity());
jiabinf7f06152021-11-22 18:10:14 +0000274 return AAUDIO_ERROR_OUT_OF_RANGE;
Phil Burk99306c82017-08-14 12:38:58 -0700275 } else if (mCallbackFrames < 0) {
Phil Burk29ccc292019-04-15 08:58:08 -0700276 ALOGW("%s - framesPerCallback negative", __func__);
jiabinf7f06152021-11-22 18:10:14 +0000277 return AAUDIO_ERROR_OUT_OF_RANGE;
Phil Burk99306c82017-08-14 12:38:58 -0700278 }
279 if (mCallbackFrames == AAUDIO_UNSPECIFIED) {
Phil Burk8d97b8e2020-09-25 23:18:14 +0000280 mCallbackFrames = getFramesPerBurst();
Phil Burk99306c82017-08-14 12:38:58 -0700281 }
282
Phil Burk0127c1b2018-03-29 13:48:06 -0700283 const int32_t callbackBufferSize = mCallbackFrames * getBytesPerFrame();
Phil Burkbf821e22020-04-17 11:51:43 -0700284 mCallbackBuffer = std::make_unique<uint8_t[]>(callbackBufferSize);
Phil Burk99306c82017-08-14 12:38:58 -0700285 }
286
Robert Wud7400832021-12-04 01:11:19 +0000287 // Exclusive output streams should combine channels when mono audio adjustment
Robert Wu8393bed2021-12-08 02:08:48 +0000288 // is enabled. They should also adjust for audio balance.
Robert Wud7400832021-12-04 01:11:19 +0000289 if ((getDirection() == AAUDIO_DIRECTION_OUTPUT) &&
290 (getSharingMode() == AAUDIO_SHARING_MODE_EXCLUSIVE)) {
291 bool isMasterMono = false;
292 android::AudioSystem::getMasterMono(&isMasterMono);
293 setRequireMonoBlend(isMasterMono);
Robert Wu8393bed2021-12-08 02:08:48 +0000294 float audioBalance = 0;
295 android::AudioSystem::getMasterBalance(&audioBalance);
296 setAudioBalance(audioBalance);
Robert Wud7400832021-12-04 01:11:19 +0000297 }
298
Phil Burkb31b66f2019-09-30 09:33:41 -0700299 // For debugging and analyzing the distribution of MMAP timestamps.
300 // For OUTPUT, use a NEGATIVE offset to move the CPU writes further BEFORE the HW reads.
301 // For INPUT, use a POSITIVE offset to move the CPU reads further AFTER the HW writes.
302 // You can use this offset to reduce glitching.
303 // You can also use this offset to force glitching. By iterating over multiple
304 // values you can reveal the distribution of the hardware timing jitter.
Phil Burk5edc4ea2020-04-17 08:15:42 -0700305 if (mAudioEndpoint->isFreeRunning()) { // MMAP?
Phil Burkb31b66f2019-09-30 09:33:41 -0700306 int32_t offsetMicros = (getDirection() == AAUDIO_DIRECTION_OUTPUT)
307 ? AAudioProperty_getOutputMMapOffsetMicros()
308 : AAudioProperty_getInputMMapOffsetMicros();
309 // This log is used to debug some tricky glitch issues. Please leave.
310 ALOGD_IF(offsetMicros, "%s() - %s mmap offset = %d micros",
311 __func__,
312 (getDirection() == AAUDIO_DIRECTION_OUTPUT) ? "output" : "input",
313 offsetMicros);
314 mTimeOffsetNanos = offsetMicros * AAUDIO_NANOS_PER_MICROSECOND;
315 }
316
Phil Burk5edc4ea2020-04-17 08:15:42 -0700317 setBufferSize(mBufferCapacityInFrames / 2); // Default buffer size to match Q
jiabinf7f06152021-11-22 18:10:14 +0000318 return AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800319}
320
Phil Burk13d3d832019-06-10 14:36:48 -0700321// This must be called under mStreamLock.
Phil Burk8b4e05e2019-12-17 12:12:09 -0800322aaudio_result_t AudioStreamInternal::release_l() {
Phil Burk965650e2017-09-07 21:00:09 -0700323 aaudio_result_t result = AAUDIO_OK;
Phil Burkdd582922020-10-15 20:29:51 +0000324 ALOGD("%s(): mServiceStreamHandle = 0x%08X", __func__, mServiceStreamHandle);
Phil Burk5ed503c2017-02-01 09:38:15 -0800325 if (mServiceStreamHandle != AAUDIO_HANDLE_INVALID) {
Phil Burk8b4e05e2019-12-17 12:12:09 -0800326 // Don't release a stream while it is running. Stop it first.
Phil Burk13d3d832019-06-10 14:36:48 -0700327 // If DISCONNECTED then we should still try to stop in case the
328 // error callback is still running.
jiabincb212cd2022-08-24 16:50:44 -0700329 if (isActive() || isDisconnected()) {
Phil Burkdd582922020-10-15 20:29:51 +0000330 requestStop_l();
Phil Burk4485d412017-05-09 15:55:02 -0700331 }
Phil Burka9876702020-04-20 18:16:15 -0700332
Phil Burk64e16a72020-06-01 13:25:51 -0700333 logReleaseBufferState();
Phil Burka9876702020-04-20 18:16:15 -0700334
Phil Burkec89b2e2017-06-20 15:05:06 -0700335 setState(AAUDIO_STREAM_STATE_CLOSING);
Phil Burk5ed503c2017-02-01 09:38:15 -0800336 aaudio_handle_t serviceStreamHandle = mServiceStreamHandle;
337 mServiceStreamHandle = AAUDIO_HANDLE_INVALID;
Phil Burkc0c70e32017-02-09 13:18:38 -0800338
339 mServiceInterface.closeStream(serviceStreamHandle);
Phil Burkbf821e22020-04-17 11:51:43 -0700340 mCallbackBuffer.reset();
Phil Burk5edc4ea2020-04-17 08:15:42 -0700341
342 // Update local frame counters so we can query them after releasing the endpoint.
343 getFramesRead();
344 getFramesWritten();
345 mAudioEndpoint.reset();
Phil Burk965650e2017-09-07 21:00:09 -0700346 result = mEndPointParcelable.close();
Phil Burk8b4e05e2019-12-17 12:12:09 -0800347 aaudio_result_t result2 = AudioStream::release_l();
Phil Burk965650e2017-09-07 21:00:09 -0700348 return (result != AAUDIO_OK) ? result : result2;
Phil Burk204a1632017-01-03 17:23:43 -0800349 } else {
Phil Burk5ed503c2017-02-01 09:38:15 -0800350 return AAUDIO_ERROR_INVALID_HANDLE;
Phil Burk204a1632017-01-03 17:23:43 -0800351 }
352}
353
Phil Burke4d7bb42017-03-28 11:32:39 -0700354static void *aaudio_callback_thread_proc(void *context)
355{
356 AudioStreamInternal *stream = (AudioStreamInternal *)context;
Phil Burkfbf031e2017-10-12 15:58:31 -0700357 //LOGD("oboe_callback_thread, stream = %p", stream);
jiabind5bd06a2021-04-27 22:04:08 +0000358 if (stream != nullptr) {
Phil Burke4d7bb42017-03-28 11:32:39 -0700359 return stream->callbackLoop();
360 } else {
jiabind5bd06a2021-04-27 22:04:08 +0000361 return nullptr;
Phil Burke4d7bb42017-03-28 11:32:39 -0700362 }
363}
364
jiabinf7f06152021-11-22 18:10:14 +0000365aaudio_result_t AudioStreamInternal::exitStandby_l() {
366 AudioEndpointParcelable endpointParcelable;
367 // The stream is in standby mode, copy all available data and then close the duplicated
368 // shared file descriptor so that it won't cause issue when the HAL try to reallocate new
369 // shared file descriptor when exiting from standby.
370 // Cache current read counter, which will be reset to new read and write counter
371 // when the new data queue and endpoint are reconfigured.
372 const android::fifo_counter_t readCounter = mAudioEndpoint->getDataReadCounter();
373 // Cache the buffer size which may be from client.
374 const int32_t previousBufferSize = mBufferSizeInFrames;
375 // Copy all available data from current data queue.
376 uint8_t buffer[getBufferCapacity() * getBytesPerFrame()];
377 android::fifo_frames_t fullFramesAvailable =
378 mAudioEndpoint->read(buffer, getBufferCapacity());
379 mEndPointParcelable.closeDataFileDescriptor();
380 aaudio_result_t result = mServiceInterface.exitStandby(
381 mServiceStreamHandle, endpointParcelable);
382 if (result != AAUDIO_OK) {
383 ALOGE("Failed to exit standby, error=%d", result);
384 goto exit;
385 }
386 // Reconstruct data queue descriptor using new shared file descriptor.
387 mEndPointParcelable.updateDataFileDescriptor(&endpointParcelable);
388 result = mEndPointParcelable.resolveDataQueue(&mEndpointDescriptor.dataQueueDescriptor);
389 if (result != AAUDIO_OK) {
390 ALOGE("Failed to resolve data queue after exiting standby, error=%d", result);
391 goto exit;
392 }
393 // Reconfigure audio endpoint with new data queue descriptor.
394 mAudioEndpoint->configureDataQueue(
395 mEndpointDescriptor.dataQueueDescriptor, getDirection());
396 // Set read and write counters with previous read counter, the later write action
397 // will make the counter at the correct place.
398 mAudioEndpoint->setDataReadCounter(readCounter);
399 mAudioEndpoint->setDataWriteCounter(readCounter);
400 result = configureDataInformation(mCallbackFrames);
401 if (result != AAUDIO_OK) {
402 ALOGE("Failed to configure data information after exiting standby, error=%d", result);
403 goto exit;
404 }
405 // Write data from previous data buffer to new endpoint.
406 if (android::fifo_frames_t framesWritten =
407 mAudioEndpoint->write(buffer, fullFramesAvailable);
408 framesWritten != fullFramesAvailable) {
409 ALOGW("Some data lost after exiting standby, frames written: %d, "
410 "frames to write: %d", framesWritten, fullFramesAvailable);
411 }
412 // Reset previous buffer size as it may be requested by the client.
413 setBufferSize(previousBufferSize);
414
415exit:
416 return result;
417}
418
Phil Burkbcc36742017-08-31 17:24:51 -0700419/*
420 * It normally takes about 20-30 msec to start a stream on the server.
421 * But the first time can take as much as 200-300 msec. The HW
422 * starts right away so by the time the client gets a chance to write into
423 * the buffer, it is already in a deep underflow state. That can cause the
424 * XRunCount to be non-zero, which could lead an app to tune its latency higher.
425 * To avoid this problem, we set a request for the processing code to start the
426 * client stream at the same position as the server stream.
427 * The processing code will then save the current offset
428 * between client and server and apply that to any position given to the app.
429 */
Phil Burkdd582922020-10-15 20:29:51 +0000430aaudio_result_t AudioStreamInternal::requestStart_l()
Phil Burk204a1632017-01-03 17:23:43 -0800431{
Phil Burk3316d5e2017-02-15 11:23:01 -0800432 int64_t startTime;
Phil Burk5ed503c2017-02-01 09:38:15 -0800433 if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700434 ALOGD("requestStart() mServiceStreamHandle invalid");
Phil Burk5ed503c2017-02-01 09:38:15 -0800435 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800436 }
Phil Burkec89b2e2017-06-20 15:05:06 -0700437 if (isActive()) {
Phil Burk29ccc292019-04-15 08:58:08 -0700438 ALOGD("requestStart() already active");
Phil Burkec89b2e2017-06-20 15:05:06 -0700439 return AAUDIO_ERROR_INVALID_STATE;
440 }
Phil Burkec89b2e2017-06-20 15:05:06 -0700441
jiabincb212cd2022-08-24 16:50:44 -0700442 if (isDisconnected()) {
Phil Burk29ccc292019-04-15 08:58:08 -0700443 ALOGD("requestStart() but DISCONNECTED");
Phil Burkbcc36742017-08-31 17:24:51 -0700444 return AAUDIO_ERROR_DISCONNECTED;
445 }
jiabincb212cd2022-08-24 16:50:44 -0700446 aaudio_stream_state_t originalState = getState();
Phil Burkec89b2e2017-06-20 15:05:06 -0700447 setState(AAUDIO_STREAM_STATE_STARTING);
Phil Burkbcc36742017-08-31 17:24:51 -0700448
449 // Clear any stale timestamps from the previous run.
450 drainTimestampsFromService();
451
Phil Burkec8ca522020-05-19 10:05:58 -0700452 prepareBuffersForStart(); // tell subclasses to get ready
453
Phil Burk965650e2017-09-07 21:00:09 -0700454 aaudio_result_t result = mServiceInterface.startStream(mServiceStreamHandle);
jiabinf7f06152021-11-22 18:10:14 +0000455 if (result == AAUDIO_ERROR_STANDBY) {
456 // The stream is at standby mode. Need to exit standby before starting the stream.
457 result = exitStandby_l();
458 if (result == AAUDIO_OK) {
459 result = mServiceInterface.startStream(mServiceStreamHandle);
460 }
461 }
462 if (result != AAUDIO_OK) {
463 ALOGD("%s() error = %d, stream was probably stolen", __func__, result);
Phil Burk6e463ce2020-04-13 10:20:20 -0700464 // Stealing was added in R. Coerce result to improve backward compatibility.
465 result = AAUDIO_ERROR_DISCONNECTED;
jiabincb212cd2022-08-24 16:50:44 -0700466 setDisconnected();
Phil Burk6e463ce2020-04-13 10:20:20 -0700467 }
Phil Burkc0c70e32017-02-09 13:18:38 -0800468
Phil Burk3316d5e2017-02-15 11:23:01 -0800469 startTime = AudioClock::getNanoseconds();
Phil Burk204a1632017-01-03 17:23:43 -0800470 mClockModel.start(startTime);
Phil Burkbcc36742017-08-31 17:24:51 -0700471 mNeedCatchUp.request(); // Ask data processing code to catch up when first timestamp received.
Phil Burke4d7bb42017-03-28 11:32:39 -0700472
Phil Burk965650e2017-09-07 21:00:09 -0700473 // Start data callback thread.
Phil Burk134f1972017-12-08 13:06:11 -0800474 if (result == AAUDIO_OK && isDataCallbackSet()) {
Phil Burke4d7bb42017-03-28 11:32:39 -0700475 // Launch the callback loop thread.
476 int64_t periodNanos = mCallbackFrames
477 * AAUDIO_NANOS_PER_SECOND
478 / getSampleRate();
479 mCallbackEnabled.store(true);
Phil Burkdd582922020-10-15 20:29:51 +0000480 result = createThread_l(periodNanos, aaudio_callback_thread_proc, this);
Phil Burke4d7bb42017-03-28 11:32:39 -0700481 }
Phil Burkec89b2e2017-06-20 15:05:06 -0700482 if (result != AAUDIO_OK) {
483 setState(originalState);
484 }
Phil Burke4d7bb42017-03-28 11:32:39 -0700485 return result;
Phil Burk204a1632017-01-03 17:23:43 -0800486}
487
Phil Burke4d7bb42017-03-28 11:32:39 -0700488int64_t AudioStreamInternal::calculateReasonableTimeout(int32_t framesPerOperation) {
489
490 // Wait for at least a second or some number of callbacks to join the thread.
Phil Burk71f35bb2017-04-13 16:05:07 -0700491 int64_t timeoutNanoseconds = (MIN_TIMEOUT_OPERATIONS
492 * framesPerOperation
493 * AAUDIO_NANOS_PER_SECOND)
494 / getSampleRate();
Phil Burke4d7bb42017-03-28 11:32:39 -0700495 if (timeoutNanoseconds < MIN_TIMEOUT_NANOS) { // arbitrary number of seconds
496 timeoutNanoseconds = MIN_TIMEOUT_NANOS;
497 }
498 return timeoutNanoseconds;
499}
500
Phil Burk87c9f642017-05-17 07:22:39 -0700501int64_t AudioStreamInternal::calculateReasonableTimeout() {
502 return calculateReasonableTimeout(getFramesPerBurst());
503}
504
Phil Burk13d3d832019-06-10 14:36:48 -0700505// This must be called under mStreamLock.
Phil Burkdd582922020-10-15 20:29:51 +0000506aaudio_result_t AudioStreamInternal::stopCallback_l()
Phil Burke4d7bb42017-03-28 11:32:39 -0700507{
jiabincb212cd2022-08-24 16:50:44 -0700508 if (isDataCallbackSet() && (isActive() || isDisconnected())) {
Phil Burke4d7bb42017-03-28 11:32:39 -0700509 mCallbackEnabled.store(false);
jiabind5bd06a2021-04-27 22:04:08 +0000510 aaudio_result_t result = joinThread_l(nullptr); // may temporarily unlock mStreamLock
Phil Burk6e463ce2020-04-13 10:20:20 -0700511 if (result == AAUDIO_ERROR_INVALID_HANDLE) {
512 ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
513 result = AAUDIO_OK;
514 }
515 return result;
Phil Burke4d7bb42017-03-28 11:32:39 -0700516 } else {
Phil Burkdd582922020-10-15 20:29:51 +0000517 ALOGD("%s() skipped, isDataCallbackSet() = %d, isActive() = %d, getState() = %d", __func__,
518 isDataCallbackSet(), isActive(), getState());
Phil Burke4d7bb42017-03-28 11:32:39 -0700519 return AAUDIO_OK;
520 }
521}
522
Phil Burkdd582922020-10-15 20:29:51 +0000523aaudio_result_t AudioStreamInternal::requestStop_l() {
524 aaudio_result_t result = stopCallback_l();
Phil Burk5cc83c32017-11-28 15:43:18 -0800525 if (result != AAUDIO_OK) {
Phil Burkdd582922020-10-15 20:29:51 +0000526 ALOGW("%s() stop callback returned %d, returning early", __func__, result);
Phil Burk5cc83c32017-11-28 15:43:18 -0800527 return result;
528 }
Phil Burk13d3d832019-06-10 14:36:48 -0700529 // The stream may have been unlocked temporarily to let a callback finish
530 // and the callback may have stopped the stream.
531 // Check to make sure the stream still needs to be stopped.
Phil Burk0bd745e2020-10-17 18:20:01 +0000532 // See also AudioStream::safeStop_l().
jiabincb212cd2022-08-24 16:50:44 -0700533 if (!(isActive() || isDisconnected())) {
Phil Burkdd582922020-10-15 20:29:51 +0000534 ALOGD("%s() returning early, not active or disconnected", __func__);
Phil Burk13d3d832019-06-10 14:36:48 -0700535 return AAUDIO_OK;
536 }
Phil Burk5cc83c32017-11-28 15:43:18 -0800537
Phil Burk71f35bb2017-04-13 16:05:07 -0700538 if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700539 ALOGW("%s() mServiceStreamHandle invalid = 0x%08X",
540 __func__, mServiceStreamHandle);
Phil Burk71f35bb2017-04-13 16:05:07 -0700541 return AAUDIO_ERROR_INVALID_STATE;
542 }
543
544 mClockModel.stop(AudioClock::getNanoseconds());
545 setState(AAUDIO_STREAM_STATE_STOPPING);
Phil Burka53ffa62018-10-10 16:21:37 -0700546 mAtomicInternalTimestamp.clear();
Phil Burk965650e2017-09-07 21:00:09 -0700547
Phil Burk6e463ce2020-04-13 10:20:20 -0700548 result = mServiceInterface.stopStream(mServiceStreamHandle);
549 if (result == AAUDIO_ERROR_INVALID_HANDLE) {
550 ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
551 result = AAUDIO_OK;
552 }
553 return result;
Phil Burk71f35bb2017-04-13 16:05:07 -0700554}
555
Phil Burk5ed503c2017-02-01 09:38:15 -0800556aaudio_result_t AudioStreamInternal::registerThread() {
Phil Burk5ed503c2017-02-01 09:38:15 -0800557 if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700558 ALOGW("%s() mServiceStreamHandle invalid", __func__);
Phil Burk5ed503c2017-02-01 09:38:15 -0800559 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800560 }
Phil Burkc0c70e32017-02-09 13:18:38 -0800561 return mServiceInterface.registerAudioThread(mServiceStreamHandle,
Phil Burkc0c70e32017-02-09 13:18:38 -0800562 gettid(),
563 getPeriodNanoseconds());
Phil Burk204a1632017-01-03 17:23:43 -0800564}
565
Phil Burk5ed503c2017-02-01 09:38:15 -0800566aaudio_result_t AudioStreamInternal::unregisterThread() {
Phil Burk5ed503c2017-02-01 09:38:15 -0800567 if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700568 ALOGW("%s() mServiceStreamHandle invalid", __func__);
Phil Burk5ed503c2017-02-01 09:38:15 -0800569 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800570 }
Phil Burk2ac035f2017-06-23 14:51:14 -0700571 return mServiceInterface.unregisterAudioThread(mServiceStreamHandle, gettid());
Phil Burk204a1632017-01-03 17:23:43 -0800572}
573
Eric Laurentcb4dae22017-07-01 19:39:32 -0700574aaudio_result_t AudioStreamInternal::startClient(const android::AudioClient& client,
jiabind1f1cb62020-03-24 11:57:57 -0700575 const audio_attributes_t *attr,
Phil Burkbbd52862018-04-13 11:37:42 -0700576 audio_port_handle_t *portHandle) {
577 ALOGV("%s() called", __func__);
Eric Laurentcb4dae22017-07-01 19:39:32 -0700578 if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
579 return AAUDIO_ERROR_INVALID_STATE;
580 }
Phil Burkbbd52862018-04-13 11:37:42 -0700581 aaudio_result_t result = mServiceInterface.startClient(mServiceStreamHandle,
jiabind1f1cb62020-03-24 11:57:57 -0700582 client, attr, portHandle);
Phil Burkbbd52862018-04-13 11:37:42 -0700583 ALOGV("%s(%d) returning %d", __func__, *portHandle, result);
584 return result;
Eric Laurentcb4dae22017-07-01 19:39:32 -0700585}
586
Phil Burkbbd52862018-04-13 11:37:42 -0700587aaudio_result_t AudioStreamInternal::stopClient(audio_port_handle_t portHandle) {
588 ALOGV("%s(%d) called", __func__, portHandle);
Eric Laurentcb4dae22017-07-01 19:39:32 -0700589 if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
590 return AAUDIO_ERROR_INVALID_STATE;
591 }
Phil Burkbbd52862018-04-13 11:37:42 -0700592 aaudio_result_t result = mServiceInterface.stopClient(mServiceStreamHandle, portHandle);
593 ALOGV("%s(%d) returning %d", __func__, portHandle, result);
594 return result;
Eric Laurentcb4dae22017-07-01 19:39:32 -0700595}
596
jiabind5bd06a2021-04-27 22:04:08 +0000597aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t /*clockId*/,
Phil Burk3316d5e2017-02-15 11:23:01 -0800598 int64_t *framePosition,
599 int64_t *timeNanoseconds) {
Phil Burk97350f92017-07-21 15:59:44 -0700600 // Generated in server and passed to client. Return latest.
Phil Burka53ffa62018-10-10 16:21:37 -0700601 if (mAtomicInternalTimestamp.isValid()) {
602 Timestamp timestamp = mAtomicInternalTimestamp.read();
Phil Burkbcc36742017-08-31 17:24:51 -0700603 int64_t position = timestamp.getPosition() + mFramesOffsetFromService;
604 if (position >= 0) {
605 *framePosition = position;
606 *timeNanoseconds = timestamp.getNanoseconds();
607 return AAUDIO_OK;
608 }
Phil Burk97350f92017-07-21 15:59:44 -0700609 }
Phil Burkc75d97f2017-09-08 15:48:36 -0700610 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800611}
612
Phil Burkec89b2e2017-06-20 15:05:06 -0700613void AudioStreamInternal::logTimestamp(AAudioServiceMessage &command) {
Phil Burk204a1632017-01-03 17:23:43 -0800614 static int64_t oldPosition = 0;
Phil Burk3316d5e2017-02-15 11:23:01 -0800615 static int64_t oldTime = 0;
Phil Burk204a1632017-01-03 17:23:43 -0800616 int64_t framePosition = command.timestamp.position;
Phil Burk3316d5e2017-02-15 11:23:01 -0800617 int64_t nanoTime = command.timestamp.timestamp;
Phil Burkbcc36742017-08-31 17:24:51 -0700618 ALOGD("logTimestamp: timestamp says framePosition = %8lld at nanoTime %lld",
Phil Burk204a1632017-01-03 17:23:43 -0800619 (long long) framePosition,
620 (long long) nanoTime);
621 int64_t nanosDelta = nanoTime - oldTime;
622 if (nanosDelta > 0 && oldTime > 0) {
623 int64_t framesDelta = framePosition - oldPosition;
Phil Burk5ed503c2017-02-01 09:38:15 -0800624 int64_t rate = (framesDelta * AAUDIO_NANOS_PER_SECOND) / nanosDelta;
Phil Burkbcc36742017-08-31 17:24:51 -0700625 ALOGD("logTimestamp: framesDelta = %8lld, nanosDelta = %8lld, rate = %lld",
Phil Burkec89b2e2017-06-20 15:05:06 -0700626 (long long) framesDelta, (long long) nanosDelta, (long long) rate);
Phil Burk204a1632017-01-03 17:23:43 -0800627 }
628 oldPosition = framePosition;
629 oldTime = nanoTime;
630}
Phil Burk204a1632017-01-03 17:23:43 -0800631
Phil Burk97350f92017-07-21 15:59:44 -0700632aaudio_result_t AudioStreamInternal::onTimestampService(AAudioServiceMessage *message) {
Phil Burk204a1632017-01-03 17:23:43 -0800633#if LOG_TIMESTAMPS
Phil Burkec89b2e2017-06-20 15:05:06 -0700634 logTimestamp(*message);
Phil Burk204a1632017-01-03 17:23:43 -0800635#endif
Phil Burkb31b66f2019-09-30 09:33:41 -0700636 processTimestamp(message->timestamp.position,
637 message->timestamp.timestamp + mTimeOffsetNanos);
Phil Burk5ed503c2017-02-01 09:38:15 -0800638 return AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800639}
640
Phil Burk97350f92017-07-21 15:59:44 -0700641aaudio_result_t AudioStreamInternal::onTimestampHardware(AAudioServiceMessage *message) {
642 Timestamp timestamp(message->timestamp.position, message->timestamp.timestamp);
Phil Burka53ffa62018-10-10 16:21:37 -0700643 mAtomicInternalTimestamp.write(timestamp);
Phil Burk97350f92017-07-21 15:59:44 -0700644 return AAUDIO_OK;
645}
646
Phil Burk5ed503c2017-02-01 09:38:15 -0800647aaudio_result_t AudioStreamInternal::onEventFromServer(AAudioServiceMessage *message) {
648 aaudio_result_t result = AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800649 switch (message->event.event) {
Phil Burk5ed503c2017-02-01 09:38:15 -0800650 case AAUDIO_SERVICE_EVENT_STARTED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700651 ALOGD("%s - got AAUDIO_SERVICE_EVENT_STARTED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700652 if (getState() == AAUDIO_STREAM_STATE_STARTING) {
653 setState(AAUDIO_STREAM_STATE_STARTED);
654 }
Vlad Popaec1788e2022-08-04 11:23:30 +0200655 mPlayerBase->triggerPortIdUpdate(static_cast<audio_port_handle_t>(
656 message->event.dataLong));
Phil Burk204a1632017-01-03 17:23:43 -0800657 break;
Phil Burk5ed503c2017-02-01 09:38:15 -0800658 case AAUDIO_SERVICE_EVENT_PAUSED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700659 ALOGD("%s - got AAUDIO_SERVICE_EVENT_PAUSED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700660 if (getState() == AAUDIO_STREAM_STATE_PAUSING) {
661 setState(AAUDIO_STREAM_STATE_PAUSED);
662 }
Phil Burk204a1632017-01-03 17:23:43 -0800663 break;
Phil Burk71f35bb2017-04-13 16:05:07 -0700664 case AAUDIO_SERVICE_EVENT_STOPPED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700665 ALOGD("%s - got AAUDIO_SERVICE_EVENT_STOPPED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700666 if (getState() == AAUDIO_STREAM_STATE_STOPPING) {
667 setState(AAUDIO_STREAM_STATE_STOPPED);
668 }
Phil Burk71f35bb2017-04-13 16:05:07 -0700669 break;
Phil Burk5ed503c2017-02-01 09:38:15 -0800670 case AAUDIO_SERVICE_EVENT_FLUSHED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700671 ALOGD("%s - got AAUDIO_SERVICE_EVENT_FLUSHED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700672 if (getState() == AAUDIO_STREAM_STATE_FLUSHING) {
673 setState(AAUDIO_STREAM_STATE_FLUSHED);
674 onFlushFromServer();
675 }
Phil Burk204a1632017-01-03 17:23:43 -0800676 break;
Phil Burk5ed503c2017-02-01 09:38:15 -0800677 case AAUDIO_SERVICE_EVENT_DISCONNECTED:
Phil Burkea04d972017-08-07 12:30:44 -0700678 // Prevent hardware from looping on old data and making buzzing sounds.
679 if (getDirection() == AAUDIO_DIRECTION_OUTPUT) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700680 mAudioEndpoint->eraseDataMemory();
Phil Burkea04d972017-08-07 12:30:44 -0700681 }
Phil Burk5ed503c2017-02-01 09:38:15 -0800682 result = AAUDIO_ERROR_DISCONNECTED;
jiabincb212cd2022-08-24 16:50:44 -0700683 setDisconnected();
Phil Burkfbf031e2017-10-12 15:58:31 -0700684 ALOGW("%s - AAUDIO_SERVICE_EVENT_DISCONNECTED - FIFO cleared", __func__);
Phil Burk204a1632017-01-03 17:23:43 -0800685 break;
Phil Burkc0c70e32017-02-09 13:18:38 -0800686 case AAUDIO_SERVICE_EVENT_VOLUME:
Phil Burk55e5eab2018-04-10 15:16:38 -0700687 ALOGD("%s - AAUDIO_SERVICE_EVENT_VOLUME %lf", __func__, message->event.dataDouble);
Eric Laurenta2f296e2017-06-21 18:51:47 -0700688 mStreamVolume = (float)message->event.dataDouble;
689 doSetVolume();
Phil Burkc0c70e32017-02-09 13:18:38 -0800690 break;
Phil Burk23296382017-11-20 15:45:11 -0800691 case AAUDIO_SERVICE_EVENT_XRUN:
692 mXRunCount = static_cast<int32_t>(message->event.dataLong);
693 break;
Phil Burk204a1632017-01-03 17:23:43 -0800694 default:
Phil Burkfbf031e2017-10-12 15:58:31 -0700695 ALOGE("%s - Unrecognized event = %d", __func__, (int) message->event.event);
Phil Burk204a1632017-01-03 17:23:43 -0800696 break;
697 }
698 return result;
699}
700
Phil Burkbcc36742017-08-31 17:24:51 -0700701aaudio_result_t AudioStreamInternal::drainTimestampsFromService() {
702 aaudio_result_t result = AAUDIO_OK;
703
704 while (result == AAUDIO_OK) {
705 AAudioServiceMessage message;
Phil Burk5edc4ea2020-04-17 08:15:42 -0700706 if (!mAudioEndpoint) {
707 break;
708 }
709 if (mAudioEndpoint->readUpCommand(&message) != 1) {
Phil Burkbcc36742017-08-31 17:24:51 -0700710 break; // no command this time, no problem
711 }
712 switch (message.what) {
713 // ignore most messages
714 case AAudioServiceMessage::code::TIMESTAMP_SERVICE:
715 case AAudioServiceMessage::code::TIMESTAMP_HARDWARE:
716 break;
717
718 case AAudioServiceMessage::code::EVENT:
719 result = onEventFromServer(&message);
720 break;
721
722 default:
Phil Burkfbf031e2017-10-12 15:58:31 -0700723 ALOGE("%s - unrecognized message.what = %d", __func__, (int) message.what);
Phil Burkbcc36742017-08-31 17:24:51 -0700724 result = AAUDIO_ERROR_INTERNAL;
725 break;
726 }
727 }
728 return result;
729}
730
Phil Burk204a1632017-01-03 17:23:43 -0800731// Process all the commands coming from the server.
Phil Burk5ed503c2017-02-01 09:38:15 -0800732aaudio_result_t AudioStreamInternal::processCommands() {
733 aaudio_result_t result = AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800734
Phil Burk5ed503c2017-02-01 09:38:15 -0800735 while (result == AAUDIO_OK) {
736 AAudioServiceMessage message;
Phil Burk5edc4ea2020-04-17 08:15:42 -0700737 if (!mAudioEndpoint) {
738 break;
739 }
740 if (mAudioEndpoint->readUpCommand(&message) != 1) {
Phil Burk204a1632017-01-03 17:23:43 -0800741 break; // no command this time, no problem
742 }
743 switch (message.what) {
Phil Burk97350f92017-07-21 15:59:44 -0700744 case AAudioServiceMessage::code::TIMESTAMP_SERVICE:
745 result = onTimestampService(&message);
746 break;
747
748 case AAudioServiceMessage::code::TIMESTAMP_HARDWARE:
749 result = onTimestampHardware(&message);
Phil Burk204a1632017-01-03 17:23:43 -0800750 break;
751
Phil Burk5ed503c2017-02-01 09:38:15 -0800752 case AAudioServiceMessage::code::EVENT:
Phil Burk204a1632017-01-03 17:23:43 -0800753 result = onEventFromServer(&message);
754 break;
755
756 default:
Phil Burkfbf031e2017-10-12 15:58:31 -0700757 ALOGE("%s - unrecognized message.what = %d", __func__, (int) message.what);
Phil Burk17fff382017-05-16 14:06:45 -0700758 result = AAUDIO_ERROR_INTERNAL;
Phil Burk204a1632017-01-03 17:23:43 -0800759 break;
760 }
761 }
762 return result;
763}
764
Phil Burk87c9f642017-05-17 07:22:39 -0700765// Read or write the data, block if needed and timeoutMillis > 0
766aaudio_result_t AudioStreamInternal::processData(void *buffer, int32_t numFrames,
767 int64_t timeoutNanoseconds)
Phil Burk204a1632017-01-03 17:23:43 -0800768{
Phil Burkfd34a932017-07-19 07:03:52 -0700769 const char * traceName = "aaProc";
770 const char * fifoName = "aaRdy";
Phil Burk4485d412017-05-09 15:55:02 -0700771 ATRACE_BEGIN(traceName);
Phil Burk4485d412017-05-09 15:55:02 -0700772 if (ATRACE_ENABLED()) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700773 int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
Phil Burkfd34a932017-07-19 07:03:52 -0700774 ATRACE_INT(fifoName, fullFrames);
Phil Burk4485d412017-05-09 15:55:02 -0700775 }
776
Phil Burkec89b2e2017-06-20 15:05:06 -0700777 aaudio_result_t result = AAUDIO_OK;
778 int32_t loopCount = 0;
779 uint8_t* audioData = (uint8_t*)buffer;
780 int64_t currentTimeNanos = AudioClock::getNanoseconds();
781 const int64_t entryTimeNanos = currentTimeNanos;
782 const int64_t deadlineNanos = currentTimeNanos + timeoutNanoseconds;
783 int32_t framesLeft = numFrames;
784
Phil Burk87c9f642017-05-17 07:22:39 -0700785 // Loop until all the data has been processed or until a timeout occurs.
Phil Burk204a1632017-01-03 17:23:43 -0800786 while (framesLeft > 0) {
Phil Burkec89b2e2017-06-20 15:05:06 -0700787 // The call to processDataNow() will not block. It will just process as much as it can.
Phil Burk3316d5e2017-02-15 11:23:01 -0800788 int64_t wakeTimeNanos = 0;
Phil Burk87c9f642017-05-17 07:22:39 -0700789 aaudio_result_t framesProcessed = processDataNow(audioData, framesLeft,
790 currentTimeNanos, &wakeTimeNanos);
791 if (framesProcessed < 0) {
Phil Burk87c9f642017-05-17 07:22:39 -0700792 result = framesProcessed;
Phil Burk204a1632017-01-03 17:23:43 -0800793 break;
794 }
Phil Burk87c9f642017-05-17 07:22:39 -0700795 framesLeft -= (int32_t) framesProcessed;
796 audioData += framesProcessed * getBytesPerFrame();
Phil Burk204a1632017-01-03 17:23:43 -0800797
798 // Should we block?
799 if (timeoutNanoseconds == 0) {
800 break; // don't block
Phil Burk8d4f0062019-10-03 15:55:41 -0700801 } else if (wakeTimeNanos != 0) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700802 if (!mAudioEndpoint->isFreeRunning()) {
Phil Burkfd34a932017-07-19 07:03:52 -0700803 // If there is software on the other end of the FIFO then it may get delayed.
804 // So wake up just a little after we expect it to be ready.
805 wakeTimeNanos += mWakeupDelayNanos;
Phil Burk204a1632017-01-03 17:23:43 -0800806 }
Phil Burkfd34a932017-07-19 07:03:52 -0700807
Phil Burk2bc7c182017-08-28 11:45:01 -0700808 currentTimeNanos = AudioClock::getNanoseconds();
809 int64_t earliestWakeTime = currentTimeNanos + mMinimumSleepNanos;
810 // Guarantee a minimum sleep time.
811 if (wakeTimeNanos < earliestWakeTime) {
812 wakeTimeNanos = earliestWakeTime;
813 }
814
Phil Burk204a1632017-01-03 17:23:43 -0800815 if (wakeTimeNanos > deadlineNanos) {
816 // If we time out, just return the framesWritten so far.
Phil Burkfbf031e2017-10-12 15:58:31 -0700817 ALOGW("processData(): entered at %lld nanos, currently %lld",
Phil Burkec89b2e2017-06-20 15:05:06 -0700818 (long long) entryTimeNanos, (long long) currentTimeNanos);
Phil Burkfbf031e2017-10-12 15:58:31 -0700819 ALOGW("processData(): TIMEOUT after %lld nanos",
Phil Burkc0c70e32017-02-09 13:18:38 -0800820 (long long) timeoutNanoseconds);
Phil Burkfbf031e2017-10-12 15:58:31 -0700821 ALOGW("processData(): wakeTime = %lld, deadline = %lld nanos",
Phil Burk87c9f642017-05-17 07:22:39 -0700822 (long long) wakeTimeNanos, (long long) deadlineNanos);
Phil Burkfbf031e2017-10-12 15:58:31 -0700823 ALOGW("processData(): past deadline by %d micros",
Phil Burk87c9f642017-05-17 07:22:39 -0700824 (int)((wakeTimeNanos - deadlineNanos) / AAUDIO_NANOS_PER_MICROSECOND));
Phil Burkec89b2e2017-06-20 15:05:06 -0700825 mClockModel.dump();
Phil Burk5edc4ea2020-04-17 08:15:42 -0700826 mAudioEndpoint->dump();
Phil Burk204a1632017-01-03 17:23:43 -0800827 break;
828 }
829
Phil Burkfd34a932017-07-19 07:03:52 -0700830 if (ATRACE_ENABLED()) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700831 int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
Phil Burkfd34a932017-07-19 07:03:52 -0700832 ATRACE_INT(fifoName, fullFrames);
833 int64_t sleepForNanos = wakeTimeNanos - currentTimeNanos;
834 ATRACE_INT("aaSlpNs", (int32_t)sleepForNanos);
835 }
836
837 AudioClock::sleepUntilNanoTime(wakeTimeNanos);
Phil Burk204a1632017-01-03 17:23:43 -0800838 currentTimeNanos = AudioClock::getNanoseconds();
839 }
840 }
841
Phil Burkfd34a932017-07-19 07:03:52 -0700842 if (ATRACE_ENABLED()) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700843 int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
Phil Burkfd34a932017-07-19 07:03:52 -0700844 ATRACE_INT(fifoName, fullFrames);
845 }
846
Phil Burk87c9f642017-05-17 07:22:39 -0700847 // return error or framesProcessed
Phil Burkc0c70e32017-02-09 13:18:38 -0800848 (void) loopCount;
Phil Burk4485d412017-05-09 15:55:02 -0700849 ATRACE_END();
Phil Burk204a1632017-01-03 17:23:43 -0800850 return (result < 0) ? result : numFrames - framesLeft;
851}
852
Phil Burk3316d5e2017-02-15 11:23:01 -0800853void AudioStreamInternal::processTimestamp(uint64_t position, int64_t time) {
Phil Burk87c9f642017-05-17 07:22:39 -0700854 mClockModel.processTimestamp(position, time);
Phil Burk204a1632017-01-03 17:23:43 -0800855}
856
Phil Burk3316d5e2017-02-15 11:23:01 -0800857aaudio_result_t AudioStreamInternal::setBufferSize(int32_t requestedFrames) {
Phil Burk6479d502017-11-20 09:32:52 -0800858 int32_t adjustedFrames = requestedFrames;
Phil Burk8d97b8e2020-09-25 23:18:14 +0000859 const int32_t maximumSize = getBufferCapacity() - getFramesPerBurst();
Phil Burk5347dca2020-04-08 16:31:07 -0700860 // Minimum size should be a multiple number of bursts.
Phil Burk8d97b8e2020-09-25 23:18:14 +0000861 const int32_t minimumSize = 1 * getFramesPerBurst();
Phil Burk6479d502017-11-20 09:32:52 -0800862
863 // Clip to minimum size so that rounding up will work better.
Phil Burk8d4f0062019-10-03 15:55:41 -0700864 adjustedFrames = std::max(minimumSize, adjustedFrames);
Phil Burk71f35bb2017-04-13 16:05:07 -0700865
Phil Burk8d4f0062019-10-03 15:55:41 -0700866 // Prevent arithmetic overflow by clipping before we round.
867 if (adjustedFrames >= maximumSize) {
Phil Burk6479d502017-11-20 09:32:52 -0800868 adjustedFrames = maximumSize;
869 } else {
870 // Round to the next highest burst size.
Phil Burk8d97b8e2020-09-25 23:18:14 +0000871 int32_t numBursts = (adjustedFrames + getFramesPerBurst() - 1) / getFramesPerBurst();
872 adjustedFrames = numBursts * getFramesPerBurst();
873 // Clip just in case maximumSize is not a multiple of getFramesPerBurst().
Phil Burk5347dca2020-04-08 16:31:07 -0700874 adjustedFrames = std::min(maximumSize, adjustedFrames);
Phil Burk6479d502017-11-20 09:32:52 -0800875 }
876
Phil Burk5edc4ea2020-04-17 08:15:42 -0700877 if (mAudioEndpoint) {
878 // Clip against the actual size from the endpoint.
879 int32_t actualFrames = 0;
880 // Set to maximum size so we can write extra data when ready in order to reduce glitches.
881 // The amount we keep in the buffer is controlled by mBufferSizeInFrames.
882 mAudioEndpoint->setBufferSizeInFrames(maximumSize, &actualFrames);
883 // actualFrames should be <= actual maximum size of endpoint
884 adjustedFrames = std::min(actualFrames, adjustedFrames);
885 }
Phil Burk8d4f0062019-10-03 15:55:41 -0700886
Phil Burk64e16a72020-06-01 13:25:51 -0700887 if (adjustedFrames != mBufferSizeInFrames) {
888 android::mediametrics::LogItem(mMetricsId)
889 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
890 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, adjustedFrames)
891 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getXRunCount())
892 .record();
893 }
894
Phil Burk8d4f0062019-10-03 15:55:41 -0700895 mBufferSizeInFrames = adjustedFrames;
Phil Burk6c63ae32019-10-28 10:28:21 -0700896 ALOGV("%s(%d) returns %d", __func__, requestedFrames, adjustedFrames);
Phil Burk8d4f0062019-10-03 15:55:41 -0700897 return (aaudio_result_t) adjustedFrames;
Phil Burk204a1632017-01-03 17:23:43 -0800898}
899
Phil Burk87c9f642017-05-17 07:22:39 -0700900int32_t AudioStreamInternal::getBufferSize() const {
Phil Burk8d4f0062019-10-03 15:55:41 -0700901 return mBufferSizeInFrames;
Phil Burk204a1632017-01-03 17:23:43 -0800902}
903
Phil Burk87c9f642017-05-17 07:22:39 -0700904int32_t AudioStreamInternal::getBufferCapacity() const {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700905 return mBufferCapacityInFrames;
Phil Burk204a1632017-01-03 17:23:43 -0800906}
907
Phil Burk377c1c22018-12-12 16:06:54 -0800908bool AudioStreamInternal::isClockModelInControl() const {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700909 return isActive() && mAudioEndpoint->isFreeRunning() && mClockModel.isRunning();
Phil Burk377c1c22018-12-12 16:06:54 -0800910}