blob: 84c715f440c86a8a22f0b285df8f499fcc67b02e [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"
jiabin5f787812023-03-02 20:42:43 +000036#include "binding/AAudioBinderClient.h"
Phil Burkc0c70e32017-02-09 13:18:38 -080037#include "binding/AAudioStreamRequest.h"
38#include "binding/AAudioStreamConfiguration.h"
Phil Burk5ed503c2017-02-01 09:38:15 -080039#include "binding/AAudioServiceMessage.h"
Phil Burka9876702020-04-20 18:16:15 -070040#include "core/AudioGlobal.h"
Phil Burk3df348f2017-02-08 11:41:55 -080041#include "core/AudioStreamBuilder.h"
Phil Burke572f462017-04-20 13:03:19 -070042#include "fifo/FifoBuffer.h"
Phil Burkfd34a932017-07-19 07:03:52 -070043#include "utility/AudioClock.h"
Philip P. Moltmannbda45752020-07-17 16:41:18 -070044#include <media/AidlConversion.h>
Phil Burke572f462017-04-20 13:03:19 -070045
Phil Burkc0c70e32017-02-09 13:18:38 -080046#include "AudioStreamInternal.h"
Phil Burk204a1632017-01-03 17:23:43 -080047
Phil Burka9876702020-04-20 18:16:15 -070048// We do this after the #includes because if a header uses ALOG.
49// it would fail on the reference to mInService.
50#undef LOG_TAG
51// This file is used in both client and server processes.
52// This is needed to make sense of the logs more easily.
53#define LOG_TAG (mInService ? "AudioStreamInternal_Service" : "AudioStreamInternal_Client")
54
Svet Ganov3e5f14f2021-05-13 22:51:08 +000055using android::content::AttributionSourceState;
Phil Burk204a1632017-01-03 17:23:43 -080056
Phil Burk5ed503c2017-02-01 09:38:15 -080057using namespace aaudio;
Phil Burk204a1632017-01-03 17:23:43 -080058
Phil Burke4d7bb42017-03-28 11:32:39 -070059#define MIN_TIMEOUT_NANOS (1000 * AAUDIO_NANOS_PER_MILLISECOND)
60
61// Wait at least this many times longer than the operation should take.
62#define MIN_TIMEOUT_OPERATIONS 4
63
Phil Burkbcc36742017-08-31 17:24:51 -070064#define LOG_TIMESTAMPS 0
Phil Burk87c9f642017-05-17 07:22:39 -070065
Phil Burkc0c70e32017-02-09 13:18:38 -080066AudioStreamInternal::AudioStreamInternal(AAudioServiceInterface &serviceInterface, bool inService)
Phil Burk204a1632017-01-03 17:23:43 -080067 : AudioStream()
68 , mClockModel()
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 {
jiabin5f787812023-03-02 20:42:43 +000075
Phil Burk204a1632017-01-03 17:23:43 -080076}
77
78AudioStreamInternal::~AudioStreamInternal() {
Phil Burkdd582922020-10-15 20:29:51 +000079 ALOGD("%s() %p called", __func__, this);
Phil Burk204a1632017-01-03 17:23:43 -080080}
81
Phil Burk5ed503c2017-02-01 09:38:15 -080082aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) {
Phil Burk204a1632017-01-03 17:23:43 -080083
Phil Burk5ed503c2017-02-01 09:38:15 -080084 aaudio_result_t result = AAUDIO_OK;
85 AAudioStreamRequest request;
Phil Burk99306c82017-08-14 12:38:58 -070086 AAudioStreamConfiguration configurationOutput;
Phil Burk204a1632017-01-03 17:23:43 -080087
Phil Burk99306c82017-08-14 12:38:58 -070088 if (getState() != AAUDIO_STREAM_STATE_UNINITIALIZED) {
Phil Burkfbf031e2017-10-12 15:58:31 -070089 ALOGE("%s - already open! state = %d", __func__, getState());
Phil Burk99306c82017-08-14 12:38:58 -070090 return AAUDIO_ERROR_INVALID_STATE;
91 }
92
93 // Copy requested parameters to the stream.
Phil Burk204a1632017-01-03 17:23:43 -080094 result = AudioStream::open(builder);
95 if (result < 0) {
96 return result;
97 }
98
jiabinef348b82021-04-19 16:53:08 +000099 const audio_format_t requestedFormat = getFormat();
Phil Burkc0c70e32017-02-09 13:18:38 -0800100 // We have to do volume scaling. So we prefer FLOAT format.
jiabinef348b82021-04-19 16:53:08 +0000101 if (requestedFormat == AUDIO_FORMAT_DEFAULT) {
Phil Burk0127c1b2018-03-29 13:48:06 -0700102 setFormat(AUDIO_FORMAT_PCM_FLOAT);
Phil Burkc0c70e32017-02-09 13:18:38 -0800103 }
Phil Burk04e805b2018-03-27 09:13:53 -0700104 // Request FLOAT for the shared mixer or the device.
Phil Burk0127c1b2018-03-29 13:48:06 -0700105 request.getConfiguration().setFormat(AUDIO_FORMAT_PCM_FLOAT);
Phil Burkc0c70e32017-02-09 13:18:38 -0800106
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000107 // TODO b/182392769: use attribution source util
108 AttributionSourceState attributionSource;
109 attributionSource.uid = VALUE_OR_FATAL(android::legacy2aidl_uid_t_int32_t(getuid()));
110 attributionSource.pid = VALUE_OR_FATAL(android::legacy2aidl_pid_t_int32_t(getpid()));
111 attributionSource.packageName = builder.getOpPackageName();
112 attributionSource.attributionTag = builder.getAttributionTag();
113 attributionSource.token = sp<android::BBinder>::make();
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700114
Phil Burkdec33ab2017-01-17 14:48:16 -0800115 // Build the request to send to the server.
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000116 request.setAttributionSource(attributionSource);
Phil Burk71f35bb2017-04-13 16:05:07 -0700117 request.setSharingModeMatchRequired(isSharingModeMatchRequired());
Phil Burk41f19d82018-02-13 14:59:10 -0800118 request.setInService(isInService());
Phil Burkc0c70e32017-02-09 13:18:38 -0800119
Phil Burk204a1632017-01-03 17:23:43 -0800120 request.getConfiguration().setDeviceId(getDeviceId());
121 request.getConfiguration().setSampleRate(getSampleRate());
Phil Burk39f02dd2017-08-04 09:13:31 -0700122 request.getConfiguration().setDirection(getDirection());
Phil Burk71f35bb2017-04-13 16:05:07 -0700123 request.getConfiguration().setSharingMode(getSharingMode());
jiabina9094092021-06-28 20:36:45 +0000124 request.getConfiguration().setChannelMask(getChannelMask());
Phil Burk71f35bb2017-04-13 16:05:07 -0700125
Phil Burka62fb952018-01-16 12:44:06 -0800126 request.getConfiguration().setUsage(getUsage());
127 request.getConfiguration().setContentType(getContentType());
Jean-Michel Trivi656bfdc2021-09-20 18:42:37 -0700128 request.getConfiguration().setSpatializationBehavior(getSpatializationBehavior());
129 request.getConfiguration().setIsContentSpatialized(isContentSpatialized());
Phil Burka62fb952018-01-16 12:44:06 -0800130 request.getConfiguration().setInputPreset(getInputPreset());
Eric Laurentd17c8502019-10-24 15:58:35 -0700131 request.getConfiguration().setPrivacySensitive(isPrivacySensitive());
Phil Burka62fb952018-01-16 12:44:06 -0800132
Phil Burk3df348f2017-02-08 11:41:55 -0800133 request.getConfiguration().setBufferCapacity(builder.getBufferCapacity());
Phil Burk204a1632017-01-03 17:23:43 -0800134
Robert Wu310037a2022-09-06 21:48:18 +0000135 request.getConfiguration().setHardwareSamplesPerFrame(builder.getHardwareSamplesPerFrame());
136 request.getConfiguration().setHardwareSampleRate(builder.getHardwareSampleRate());
137 request.getConfiguration().setHardwareFormat(builder.getHardwareFormat());
138
Phil Burk41f19d82018-02-13 14:59:10 -0800139 mDeviceChannelCount = getSamplesPerFrame(); // Assume it will be the same. Update if not.
140
jiabin5f787812023-03-02 20:42:43 +0000141 mServiceStreamHandleInfo = mServiceInterface.openStream(request, configurationOutput);
142 if (getServiceHandle() < 0
jiabina9094092021-06-28 20:36:45 +0000143 && (request.getConfiguration().getSamplesPerFrame() == 1
144 || request.getConfiguration().getChannelMask() == AAUDIO_CHANNEL_MONO)
Phil Burk41f19d82018-02-13 14:59:10 -0800145 && getDirection() == AAUDIO_DIRECTION_OUTPUT
146 && !isInService()) {
147 // if that failed then try switching from mono to stereo if OUTPUT.
148 // Only do this in the client. Otherwise we end up with a mono mixer in the service
149 // that writes to a stereo MMAP stream.
Phil Burk0127c1b2018-03-29 13:48:06 -0700150 ALOGD("%s() - openStream() returned %d, try switching from MONO to STEREO",
jiabin5f787812023-03-02 20:42:43 +0000151 __func__, getServiceHandle());
jiabina9094092021-06-28 20:36:45 +0000152 request.getConfiguration().setChannelMask(AAUDIO_CHANNEL_STEREO);
jiabin5f787812023-03-02 20:42:43 +0000153 mServiceStreamHandleInfo = mServiceInterface.openStream(request, configurationOutput);
Phil Burk41f19d82018-02-13 14:59:10 -0800154 }
jiabin5f787812023-03-02 20:42:43 +0000155 if (getServiceHandle() < 0) {
156 return getServiceHandle();
Phil Burk204a1632017-01-03 17:23:43 -0800157 }
Phil Burk99306c82017-08-14 12:38:58 -0700158
Phil Burka9876702020-04-20 18:16:15 -0700159 // This must match the key generated in oboeservice/AAudioServiceStreamBase.cpp
160 // so the client can have permission to log.
jiabinfbf20302021-07-28 22:15:01 +0000161 if (!mInService) {
162 // No need to log if it is from service side.
163 mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_STREAM)
jiabin5f787812023-03-02 20:42:43 +0000164 + std::to_string(getServiceHandle());
jiabinfbf20302021-07-28 22:15:01 +0000165 }
Phil Burka9876702020-04-20 18:16:15 -0700166
jiabinef348b82021-04-19 16:53:08 +0000167 android::mediametrics::LogItem(mMetricsId)
168 .set(AMEDIAMETRICS_PROP_PERFORMANCEMODE,
jiabinc8da9032021-04-28 20:42:36 +0000169 AudioGlobal_convertPerformanceModeToText(builder.getPerformanceMode()))
170 .set(AMEDIAMETRICS_PROP_SHARINGMODE,
171 AudioGlobal_convertSharingModeToText(builder.getSharingMode()))
jiabinef348b82021-04-19 16:53:08 +0000172 .set(AMEDIAMETRICS_PROP_ENCODINGCLIENT,
173 android::toString(requestedFormat).c_str()).record();
174
Phil Burk99306c82017-08-14 12:38:58 -0700175 result = configurationOutput.validate();
176 if (result != AAUDIO_OK) {
177 goto error;
178 }
179 // Save results of the open.
jiabina9094092021-06-28 20:36:45 +0000180 if (getChannelMask() == AAUDIO_UNSPECIFIED) {
181 setChannelMask(configurationOutput.getChannelMask());
Phil Burk41f19d82018-02-13 14:59:10 -0800182 }
jiabina9094092021-06-28 20:36:45 +0000183
Phil Burk41f19d82018-02-13 14:59:10 -0800184 mDeviceChannelCount = configurationOutput.getSamplesPerFrame();
185
Phil Burk99306c82017-08-14 12:38:58 -0700186 setSampleRate(configurationOutput.getSampleRate());
Phil Burk99306c82017-08-14 12:38:58 -0700187 setDeviceId(configurationOutput.getDeviceId());
Phil Burk4e1af9f2018-01-03 15:54:35 -0800188 setSessionId(configurationOutput.getSessionId());
Phil Burk99306c82017-08-14 12:38:58 -0700189 setSharingMode(configurationOutput.getSharingMode());
190
Phil Burka62fb952018-01-16 12:44:06 -0800191 setUsage(configurationOutput.getUsage());
192 setContentType(configurationOutput.getContentType());
Jean-Michel Trivi656bfdc2021-09-20 18:42:37 -0700193 setSpatializationBehavior(configurationOutput.getSpatializationBehavior());
194 setIsContentSpatialized(configurationOutput.isContentSpatialized());
Phil Burka62fb952018-01-16 12:44:06 -0800195 setInputPreset(configurationOutput.getInputPreset());
196
Phil Burk99306c82017-08-14 12:38:58 -0700197 // Save device format so we can do format conversion and volume scaling together.
Phil Burk3d786cb2018-04-09 11:58:09 -0700198 setDeviceFormat(configurationOutput.getFormat());
Phil Burk99306c82017-08-14 12:38:58 -0700199
Robert Wu310037a2022-09-06 21:48:18 +0000200 setHardwareSamplesPerFrame(configurationOutput.getHardwareSamplesPerFrame());
201 setHardwareSampleRate(configurationOutput.getHardwareSampleRate());
202 setHardwareFormat(configurationOutput.getHardwareFormat());
203
jiabin5f787812023-03-02 20:42:43 +0000204 result = mServiceInterface.getStreamDescription(mServiceStreamHandleInfo, mEndPointParcelable);
Phil Burk99306c82017-08-14 12:38:58 -0700205 if (result != AAUDIO_OK) {
206 goto error;
207 }
208
209 // Resolve parcelable into a descriptor.
210 result = mEndPointParcelable.resolve(&mEndpointDescriptor);
211 if (result != AAUDIO_OK) {
212 goto error;
213 }
214
215 // Configure endpoint based on descriptor.
Phil Burk5edc4ea2020-04-17 08:15:42 -0700216 mAudioEndpoint = std::make_unique<AudioEndpoint>();
217 result = mAudioEndpoint->configure(&mEndpointDescriptor, getDirection());
Phil Burk99306c82017-08-14 12:38:58 -0700218 if (result != AAUDIO_OK) {
219 goto error;
220 }
221
jiabinf7f06152021-11-22 18:10:14 +0000222 if ((result = configureDataInformation(builder.getFramesPerDataCallback())) != AAUDIO_OK) {
223 goto error;
224 }
225
226 setState(AAUDIO_STREAM_STATE_OPEN);
227
228 return result;
229
230error:
231 safeReleaseClose();
232 return result;
233}
234
235aaudio_result_t AudioStreamInternal::configureDataInformation(int32_t callbackFrames) {
236 int32_t framesPerHardwareBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
Phil Burk3c4e6b52019-01-22 15:53:36 -0800237
238 // Scale up the burst size to meet the minimum equivalent in microseconds.
239 // This is to avoid waking the CPU too often when the HW burst is very small
240 // or at high sample rates.
jiabinf7f06152021-11-22 18:10:14 +0000241 int32_t framesPerBurst = framesPerHardwareBurst;
242 int32_t burstMicros = 0;
243 const int32_t burstMinMicros = android::AudioSystem::getAAudioHardwareBurstMinUsec();
Phil Burk3c4e6b52019-01-22 15:53:36 -0800244 do {
245 if (burstMicros > 0) { // skip first loop
246 framesPerBurst *= 2;
247 }
248 burstMicros = framesPerBurst * static_cast<int64_t>(1000000) / getSampleRate();
249 } while (burstMicros < burstMinMicros);
250 ALOGD("%s() original HW burst = %d, minMicros = %d => SW burst = %d\n",
251 __func__, framesPerHardwareBurst, burstMinMicros, framesPerBurst);
252
253 // Validate final burst size.
Phil Burk6479d502017-11-20 09:32:52 -0800254 if (framesPerBurst < MIN_FRAMES_PER_BURST || framesPerBurst > MAX_FRAMES_PER_BURST) {
255 ALOGE("%s - framesPerBurst out of range = %d", __func__, framesPerBurst);
jiabinf7f06152021-11-22 18:10:14 +0000256 return AAUDIO_ERROR_OUT_OF_RANGE;
Phil Burk99306c82017-08-14 12:38:58 -0700257 }
Phil Burk8d97b8e2020-09-25 23:18:14 +0000258 setFramesPerBurst(framesPerBurst); // only save good value
Phil Burk6479d502017-11-20 09:32:52 -0800259
Phil Burk5edc4ea2020-04-17 08:15:42 -0700260 mBufferCapacityInFrames = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
Phil Burk8d97b8e2020-09-25 23:18:14 +0000261 if (mBufferCapacityInFrames < getFramesPerBurst()
Phil Burk5edc4ea2020-04-17 08:15:42 -0700262 || mBufferCapacityInFrames > MAX_BUFFER_CAPACITY_IN_FRAMES) {
263 ALOGE("%s - bufferCapacity out of range = %d", __func__, mBufferCapacityInFrames);
jiabinf7f06152021-11-22 18:10:14 +0000264 return AAUDIO_ERROR_OUT_OF_RANGE;
Phil Burk99306c82017-08-14 12:38:58 -0700265 }
266
267 mClockModel.setSampleRate(getSampleRate());
Phil Burk3c4e6b52019-01-22 15:53:36 -0800268 mClockModel.setFramesPerBurst(framesPerHardwareBurst);
Phil Burk99306c82017-08-14 12:38:58 -0700269
Phil Burk134f1972017-12-08 13:06:11 -0800270 if (isDataCallbackSet()) {
jiabinf7f06152021-11-22 18:10:14 +0000271 mCallbackFrames = callbackFrames;
Phil Burk99306c82017-08-14 12:38:58 -0700272 if (mCallbackFrames > getBufferCapacity() / 2) {
Phil Burk29ccc292019-04-15 08:58:08 -0700273 ALOGW("%s - framesPerCallback too big = %d, capacity = %d",
Phil Burkfbf031e2017-10-12 15:58:31 -0700274 __func__, mCallbackFrames, getBufferCapacity());
jiabinf7f06152021-11-22 18:10:14 +0000275 return AAUDIO_ERROR_OUT_OF_RANGE;
Phil Burk99306c82017-08-14 12:38:58 -0700276 } else if (mCallbackFrames < 0) {
Phil Burk29ccc292019-04-15 08:58:08 -0700277 ALOGW("%s - framesPerCallback negative", __func__);
jiabinf7f06152021-11-22 18:10:14 +0000278 return AAUDIO_ERROR_OUT_OF_RANGE;
Phil Burk99306c82017-08-14 12:38:58 -0700279 }
280 if (mCallbackFrames == AAUDIO_UNSPECIFIED) {
Phil Burk8d97b8e2020-09-25 23:18:14 +0000281 mCallbackFrames = getFramesPerBurst();
Phil Burk99306c82017-08-14 12:38:58 -0700282 }
283
Phil Burk0127c1b2018-03-29 13:48:06 -0700284 const int32_t callbackBufferSize = mCallbackFrames * getBytesPerFrame();
Phil Burkbf821e22020-04-17 11:51:43 -0700285 mCallbackBuffer = std::make_unique<uint8_t[]>(callbackBufferSize);
Phil Burk99306c82017-08-14 12:38:58 -0700286 }
287
Robert Wud7400832021-12-04 01:11:19 +0000288 // Exclusive output streams should combine channels when mono audio adjustment
Robert Wu8393bed2021-12-08 02:08:48 +0000289 // is enabled. They should also adjust for audio balance.
Robert Wud7400832021-12-04 01:11:19 +0000290 if ((getDirection() == AAUDIO_DIRECTION_OUTPUT) &&
291 (getSharingMode() == AAUDIO_SHARING_MODE_EXCLUSIVE)) {
292 bool isMasterMono = false;
293 android::AudioSystem::getMasterMono(&isMasterMono);
294 setRequireMonoBlend(isMasterMono);
Robert Wu8393bed2021-12-08 02:08:48 +0000295 float audioBalance = 0;
296 android::AudioSystem::getMasterBalance(&audioBalance);
297 setAudioBalance(audioBalance);
Robert Wud7400832021-12-04 01:11:19 +0000298 }
299
Phil Burkb31b66f2019-09-30 09:33:41 -0700300 // For debugging and analyzing the distribution of MMAP timestamps.
301 // For OUTPUT, use a NEGATIVE offset to move the CPU writes further BEFORE the HW reads.
302 // For INPUT, use a POSITIVE offset to move the CPU reads further AFTER the HW writes.
303 // You can use this offset to reduce glitching.
304 // You can also use this offset to force glitching. By iterating over multiple
305 // values you can reveal the distribution of the hardware timing jitter.
Phil Burk5edc4ea2020-04-17 08:15:42 -0700306 if (mAudioEndpoint->isFreeRunning()) { // MMAP?
Phil Burkb31b66f2019-09-30 09:33:41 -0700307 int32_t offsetMicros = (getDirection() == AAUDIO_DIRECTION_OUTPUT)
308 ? AAudioProperty_getOutputMMapOffsetMicros()
309 : AAudioProperty_getInputMMapOffsetMicros();
310 // This log is used to debug some tricky glitch issues. Please leave.
311 ALOGD_IF(offsetMicros, "%s() - %s mmap offset = %d micros",
312 __func__,
313 (getDirection() == AAUDIO_DIRECTION_OUTPUT) ? "output" : "input",
314 offsetMicros);
315 mTimeOffsetNanos = offsetMicros * AAUDIO_NANOS_PER_MICROSECOND;
316 }
317
Phil Burk5edc4ea2020-04-17 08:15:42 -0700318 setBufferSize(mBufferCapacityInFrames / 2); // Default buffer size to match Q
jiabinf7f06152021-11-22 18:10:14 +0000319 return AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800320}
321
Phil Burk13d3d832019-06-10 14:36:48 -0700322// This must be called under mStreamLock.
Phil Burk8b4e05e2019-12-17 12:12:09 -0800323aaudio_result_t AudioStreamInternal::release_l() {
Phil Burk965650e2017-09-07 21:00:09 -0700324 aaudio_result_t result = AAUDIO_OK;
jiabin5f787812023-03-02 20:42:43 +0000325 ALOGD("%s(): mServiceStreamHandle = 0x%08X", __func__, getServiceHandle());
326 if (getServiceHandle() != AAUDIO_HANDLE_INVALID) {
Phil Burk8b4e05e2019-12-17 12:12:09 -0800327 // Don't release a stream while it is running. Stop it first.
Phil Burk13d3d832019-06-10 14:36:48 -0700328 // If DISCONNECTED then we should still try to stop in case the
329 // error callback is still running.
jiabincb212cd2022-08-24 16:50:44 -0700330 if (isActive() || isDisconnected()) {
Phil Burkdd582922020-10-15 20:29:51 +0000331 requestStop_l();
Phil Burk4485d412017-05-09 15:55:02 -0700332 }
Phil Burka9876702020-04-20 18:16:15 -0700333
Phil Burk64e16a72020-06-01 13:25:51 -0700334 logReleaseBufferState();
Phil Burka9876702020-04-20 18:16:15 -0700335
Phil Burkec89b2e2017-06-20 15:05:06 -0700336 setState(AAUDIO_STREAM_STATE_CLOSING);
jiabin5f787812023-03-02 20:42:43 +0000337 auto serviceStreamHandleInfo = mServiceStreamHandleInfo;
338 mServiceStreamHandleInfo = AAudioHandleInfo();
Phil Burkc0c70e32017-02-09 13:18:38 -0800339
jiabin5f787812023-03-02 20:42:43 +0000340 mServiceInterface.closeStream(serviceStreamHandleInfo);
Phil Burkbf821e22020-04-17 11:51:43 -0700341 mCallbackBuffer.reset();
Phil Burk5edc4ea2020-04-17 08:15:42 -0700342
343 // Update local frame counters so we can query them after releasing the endpoint.
344 getFramesRead();
345 getFramesWritten();
346 mAudioEndpoint.reset();
Phil Burk965650e2017-09-07 21:00:09 -0700347 result = mEndPointParcelable.close();
Phil Burk8b4e05e2019-12-17 12:12:09 -0800348 aaudio_result_t result2 = AudioStream::release_l();
Phil Burk965650e2017-09-07 21:00:09 -0700349 return (result != AAUDIO_OK) ? result : result2;
Phil Burk204a1632017-01-03 17:23:43 -0800350 } else {
Phil Burk5ed503c2017-02-01 09:38:15 -0800351 return AAUDIO_ERROR_INVALID_HANDLE;
Phil Burk204a1632017-01-03 17:23:43 -0800352 }
353}
354
Phil Burke4d7bb42017-03-28 11:32:39 -0700355static void *aaudio_callback_thread_proc(void *context)
356{
357 AudioStreamInternal *stream = (AudioStreamInternal *)context;
Phil Burkfbf031e2017-10-12 15:58:31 -0700358 //LOGD("oboe_callback_thread, stream = %p", stream);
jiabind5bd06a2021-04-27 22:04:08 +0000359 if (stream != nullptr) {
Phil Burke4d7bb42017-03-28 11:32:39 -0700360 return stream->callbackLoop();
361 } else {
jiabind5bd06a2021-04-27 22:04:08 +0000362 return nullptr;
Phil Burke4d7bb42017-03-28 11:32:39 -0700363 }
364}
365
jiabinf7f06152021-11-22 18:10:14 +0000366aaudio_result_t AudioStreamInternal::exitStandby_l() {
367 AudioEndpointParcelable endpointParcelable;
368 // The stream is in standby mode, copy all available data and then close the duplicated
369 // shared file descriptor so that it won't cause issue when the HAL try to reallocate new
370 // shared file descriptor when exiting from standby.
371 // Cache current read counter, which will be reset to new read and write counter
372 // when the new data queue and endpoint are reconfigured.
373 const android::fifo_counter_t readCounter = mAudioEndpoint->getDataReadCounter();
374 // Cache the buffer size which may be from client.
375 const int32_t previousBufferSize = mBufferSizeInFrames;
376 // Copy all available data from current data queue.
377 uint8_t buffer[getBufferCapacity() * getBytesPerFrame()];
378 android::fifo_frames_t fullFramesAvailable =
379 mAudioEndpoint->read(buffer, getBufferCapacity());
380 mEndPointParcelable.closeDataFileDescriptor();
381 aaudio_result_t result = mServiceInterface.exitStandby(
jiabin5f787812023-03-02 20:42:43 +0000382 mServiceStreamHandleInfo, endpointParcelable);
jiabinf7f06152021-11-22 18:10:14 +0000383 if (result != AAUDIO_OK) {
384 ALOGE("Failed to exit standby, error=%d", result);
385 goto exit;
386 }
387 // Reconstruct data queue descriptor using new shared file descriptor.
jiabinfc791ee2023-02-15 19:43:40 +0000388 result = mEndPointParcelable.updateDataFileDescriptor(&endpointParcelable);
389 if (result != AAUDIO_OK) {
390 ALOGE("%s failed to update data file descriptor, error=%d", __func__, result);
391 goto exit;
392 }
jiabinf7f06152021-11-22 18:10:14 +0000393 result = mEndPointParcelable.resolveDataQueue(&mEndpointDescriptor.dataQueueDescriptor);
394 if (result != AAUDIO_OK) {
395 ALOGE("Failed to resolve data queue after exiting standby, error=%d", result);
396 goto exit;
397 }
398 // Reconfigure audio endpoint with new data queue descriptor.
399 mAudioEndpoint->configureDataQueue(
400 mEndpointDescriptor.dataQueueDescriptor, getDirection());
401 // Set read and write counters with previous read counter, the later write action
402 // will make the counter at the correct place.
403 mAudioEndpoint->setDataReadCounter(readCounter);
404 mAudioEndpoint->setDataWriteCounter(readCounter);
405 result = configureDataInformation(mCallbackFrames);
406 if (result != AAUDIO_OK) {
407 ALOGE("Failed to configure data information after exiting standby, error=%d", result);
408 goto exit;
409 }
410 // Write data from previous data buffer to new endpoint.
411 if (android::fifo_frames_t framesWritten =
412 mAudioEndpoint->write(buffer, fullFramesAvailable);
413 framesWritten != fullFramesAvailable) {
414 ALOGW("Some data lost after exiting standby, frames written: %d, "
415 "frames to write: %d", framesWritten, fullFramesAvailable);
416 }
417 // Reset previous buffer size as it may be requested by the client.
418 setBufferSize(previousBufferSize);
419
420exit:
421 return result;
422}
423
Phil Burkbcc36742017-08-31 17:24:51 -0700424/*
425 * It normally takes about 20-30 msec to start a stream on the server.
426 * But the first time can take as much as 200-300 msec. The HW
427 * starts right away so by the time the client gets a chance to write into
428 * the buffer, it is already in a deep underflow state. That can cause the
429 * XRunCount to be non-zero, which could lead an app to tune its latency higher.
430 * To avoid this problem, we set a request for the processing code to start the
431 * client stream at the same position as the server stream.
432 * The processing code will then save the current offset
433 * between client and server and apply that to any position given to the app.
434 */
Phil Burkdd582922020-10-15 20:29:51 +0000435aaudio_result_t AudioStreamInternal::requestStart_l()
Phil Burk204a1632017-01-03 17:23:43 -0800436{
Phil Burk3316d5e2017-02-15 11:23:01 -0800437 int64_t startTime;
jiabin5f787812023-03-02 20:42:43 +0000438 if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700439 ALOGD("requestStart() mServiceStreamHandle invalid");
Phil Burk5ed503c2017-02-01 09:38:15 -0800440 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800441 }
Phil Burkec89b2e2017-06-20 15:05:06 -0700442 if (isActive()) {
Phil Burk29ccc292019-04-15 08:58:08 -0700443 ALOGD("requestStart() already active");
Phil Burkec89b2e2017-06-20 15:05:06 -0700444 return AAUDIO_ERROR_INVALID_STATE;
445 }
Phil Burkec89b2e2017-06-20 15:05:06 -0700446
jiabincb212cd2022-08-24 16:50:44 -0700447 if (isDisconnected()) {
Phil Burk29ccc292019-04-15 08:58:08 -0700448 ALOGD("requestStart() but DISCONNECTED");
Phil Burkbcc36742017-08-31 17:24:51 -0700449 return AAUDIO_ERROR_DISCONNECTED;
450 }
jiabincb212cd2022-08-24 16:50:44 -0700451 aaudio_stream_state_t originalState = getState();
Phil Burkec89b2e2017-06-20 15:05:06 -0700452 setState(AAUDIO_STREAM_STATE_STARTING);
Phil Burkbcc36742017-08-31 17:24:51 -0700453
454 // Clear any stale timestamps from the previous run.
455 drainTimestampsFromService();
456
Phil Burkec8ca522020-05-19 10:05:58 -0700457 prepareBuffersForStart(); // tell subclasses to get ready
458
jiabin5f787812023-03-02 20:42:43 +0000459 aaudio_result_t result = mServiceInterface.startStream(mServiceStreamHandleInfo);
jiabinf7f06152021-11-22 18:10:14 +0000460 if (result == AAUDIO_ERROR_STANDBY) {
461 // The stream is at standby mode. Need to exit standby before starting the stream.
462 result = exitStandby_l();
463 if (result == AAUDIO_OK) {
jiabin5f787812023-03-02 20:42:43 +0000464 result = mServiceInterface.startStream(mServiceStreamHandleInfo);
jiabinf7f06152021-11-22 18:10:14 +0000465 }
466 }
467 if (result != AAUDIO_OK) {
468 ALOGD("%s() error = %d, stream was probably stolen", __func__, result);
Phil Burk6e463ce2020-04-13 10:20:20 -0700469 // Stealing was added in R. Coerce result to improve backward compatibility.
470 result = AAUDIO_ERROR_DISCONNECTED;
jiabincb212cd2022-08-24 16:50:44 -0700471 setDisconnected();
Phil Burk6e463ce2020-04-13 10:20:20 -0700472 }
Phil Burkc0c70e32017-02-09 13:18:38 -0800473
Phil Burk3316d5e2017-02-15 11:23:01 -0800474 startTime = AudioClock::getNanoseconds();
Phil Burk204a1632017-01-03 17:23:43 -0800475 mClockModel.start(startTime);
Phil Burkbcc36742017-08-31 17:24:51 -0700476 mNeedCatchUp.request(); // Ask data processing code to catch up when first timestamp received.
Phil Burke4d7bb42017-03-28 11:32:39 -0700477
Phil Burk965650e2017-09-07 21:00:09 -0700478 // Start data callback thread.
Phil Burk134f1972017-12-08 13:06:11 -0800479 if (result == AAUDIO_OK && isDataCallbackSet()) {
Phil Burke4d7bb42017-03-28 11:32:39 -0700480 // Launch the callback loop thread.
481 int64_t periodNanos = mCallbackFrames
482 * AAUDIO_NANOS_PER_SECOND
483 / getSampleRate();
484 mCallbackEnabled.store(true);
Phil Burkdd582922020-10-15 20:29:51 +0000485 result = createThread_l(periodNanos, aaudio_callback_thread_proc, this);
Phil Burke4d7bb42017-03-28 11:32:39 -0700486 }
Phil Burkec89b2e2017-06-20 15:05:06 -0700487 if (result != AAUDIO_OK) {
488 setState(originalState);
489 }
Phil Burke4d7bb42017-03-28 11:32:39 -0700490 return result;
Phil Burk204a1632017-01-03 17:23:43 -0800491}
492
Phil Burke4d7bb42017-03-28 11:32:39 -0700493int64_t AudioStreamInternal::calculateReasonableTimeout(int32_t framesPerOperation) {
494
495 // Wait for at least a second or some number of callbacks to join the thread.
Phil Burk71f35bb2017-04-13 16:05:07 -0700496 int64_t timeoutNanoseconds = (MIN_TIMEOUT_OPERATIONS
497 * framesPerOperation
498 * AAUDIO_NANOS_PER_SECOND)
499 / getSampleRate();
Phil Burke4d7bb42017-03-28 11:32:39 -0700500 if (timeoutNanoseconds < MIN_TIMEOUT_NANOS) { // arbitrary number of seconds
501 timeoutNanoseconds = MIN_TIMEOUT_NANOS;
502 }
503 return timeoutNanoseconds;
504}
505
Phil Burk87c9f642017-05-17 07:22:39 -0700506int64_t AudioStreamInternal::calculateReasonableTimeout() {
507 return calculateReasonableTimeout(getFramesPerBurst());
508}
509
Phil Burk13d3d832019-06-10 14:36:48 -0700510// This must be called under mStreamLock.
Phil Burkdd582922020-10-15 20:29:51 +0000511aaudio_result_t AudioStreamInternal::stopCallback_l()
Phil Burke4d7bb42017-03-28 11:32:39 -0700512{
jiabincb212cd2022-08-24 16:50:44 -0700513 if (isDataCallbackSet() && (isActive() || isDisconnected())) {
Phil Burke4d7bb42017-03-28 11:32:39 -0700514 mCallbackEnabled.store(false);
jiabind5bd06a2021-04-27 22:04:08 +0000515 aaudio_result_t result = joinThread_l(nullptr); // may temporarily unlock mStreamLock
Phil Burk6e463ce2020-04-13 10:20:20 -0700516 if (result == AAUDIO_ERROR_INVALID_HANDLE) {
517 ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
518 result = AAUDIO_OK;
519 }
520 return result;
Phil Burke4d7bb42017-03-28 11:32:39 -0700521 } else {
Phil Burkdd582922020-10-15 20:29:51 +0000522 ALOGD("%s() skipped, isDataCallbackSet() = %d, isActive() = %d, getState() = %d", __func__,
523 isDataCallbackSet(), isActive(), getState());
Phil Burke4d7bb42017-03-28 11:32:39 -0700524 return AAUDIO_OK;
525 }
526}
527
Phil Burkdd582922020-10-15 20:29:51 +0000528aaudio_result_t AudioStreamInternal::requestStop_l() {
529 aaudio_result_t result = stopCallback_l();
Phil Burk5cc83c32017-11-28 15:43:18 -0800530 if (result != AAUDIO_OK) {
Phil Burkdd582922020-10-15 20:29:51 +0000531 ALOGW("%s() stop callback returned %d, returning early", __func__, result);
Phil Burk5cc83c32017-11-28 15:43:18 -0800532 return result;
533 }
Phil Burk13d3d832019-06-10 14:36:48 -0700534 // The stream may have been unlocked temporarily to let a callback finish
535 // and the callback may have stopped the stream.
536 // Check to make sure the stream still needs to be stopped.
Phil Burk0bd745e2020-10-17 18:20:01 +0000537 // See also AudioStream::safeStop_l().
jiabincb212cd2022-08-24 16:50:44 -0700538 if (!(isActive() || isDisconnected())) {
Phil Burkdd582922020-10-15 20:29:51 +0000539 ALOGD("%s() returning early, not active or disconnected", __func__);
Phil Burk13d3d832019-06-10 14:36:48 -0700540 return AAUDIO_OK;
541 }
Phil Burk5cc83c32017-11-28 15:43:18 -0800542
jiabin5f787812023-03-02 20:42:43 +0000543 if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700544 ALOGW("%s() mServiceStreamHandle invalid = 0x%08X",
jiabin5f787812023-03-02 20:42:43 +0000545 __func__, getServiceHandle());
Phil Burk71f35bb2017-04-13 16:05:07 -0700546 return AAUDIO_ERROR_INVALID_STATE;
547 }
548
549 mClockModel.stop(AudioClock::getNanoseconds());
550 setState(AAUDIO_STREAM_STATE_STOPPING);
Phil Burka53ffa62018-10-10 16:21:37 -0700551 mAtomicInternalTimestamp.clear();
Phil Burk965650e2017-09-07 21:00:09 -0700552
jiabin5f787812023-03-02 20:42:43 +0000553 result = mServiceInterface.stopStream(mServiceStreamHandleInfo);
Phil Burk6e463ce2020-04-13 10:20:20 -0700554 if (result == AAUDIO_ERROR_INVALID_HANDLE) {
555 ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
556 result = AAUDIO_OK;
557 }
558 return result;
Phil Burk71f35bb2017-04-13 16:05:07 -0700559}
560
Phil Burk5ed503c2017-02-01 09:38:15 -0800561aaudio_result_t AudioStreamInternal::registerThread() {
jiabin5f787812023-03-02 20:42:43 +0000562 if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700563 ALOGW("%s() mServiceStreamHandle invalid", __func__);
Phil Burk5ed503c2017-02-01 09:38:15 -0800564 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800565 }
jiabin5f787812023-03-02 20:42:43 +0000566 return mServiceInterface.registerAudioThread(mServiceStreamHandleInfo,
567 gettid(),
568 getPeriodNanoseconds());
Phil Burk204a1632017-01-03 17:23:43 -0800569}
570
Phil Burk5ed503c2017-02-01 09:38:15 -0800571aaudio_result_t AudioStreamInternal::unregisterThread() {
jiabin5f787812023-03-02 20:42:43 +0000572 if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700573 ALOGW("%s() mServiceStreamHandle invalid", __func__);
Phil Burk5ed503c2017-02-01 09:38:15 -0800574 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800575 }
jiabin5f787812023-03-02 20:42:43 +0000576 return mServiceInterface.unregisterAudioThread(mServiceStreamHandleInfo, gettid());
Phil Burk204a1632017-01-03 17:23:43 -0800577}
578
Eric Laurentcb4dae22017-07-01 19:39:32 -0700579aaudio_result_t AudioStreamInternal::startClient(const android::AudioClient& client,
jiabind1f1cb62020-03-24 11:57:57 -0700580 const audio_attributes_t *attr,
Phil Burkbbd52862018-04-13 11:37:42 -0700581 audio_port_handle_t *portHandle) {
582 ALOGV("%s() called", __func__);
jiabin5f787812023-03-02 20:42:43 +0000583 if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
Eric Laurentcb4dae22017-07-01 19:39:32 -0700584 return AAUDIO_ERROR_INVALID_STATE;
585 }
jiabin5f787812023-03-02 20:42:43 +0000586 aaudio_result_t result = mServiceInterface.startClient(mServiceStreamHandleInfo,
jiabind1f1cb62020-03-24 11:57:57 -0700587 client, attr, portHandle);
Phil Burkbbd52862018-04-13 11:37:42 -0700588 ALOGV("%s(%d) returning %d", __func__, *portHandle, result);
589 return result;
Eric Laurentcb4dae22017-07-01 19:39:32 -0700590}
591
Phil Burkbbd52862018-04-13 11:37:42 -0700592aaudio_result_t AudioStreamInternal::stopClient(audio_port_handle_t portHandle) {
593 ALOGV("%s(%d) called", __func__, portHandle);
jiabin5f787812023-03-02 20:42:43 +0000594 if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
Eric Laurentcb4dae22017-07-01 19:39:32 -0700595 return AAUDIO_ERROR_INVALID_STATE;
596 }
jiabin5f787812023-03-02 20:42:43 +0000597 aaudio_result_t result = mServiceInterface.stopClient(mServiceStreamHandleInfo, portHandle);
Phil Burkbbd52862018-04-13 11:37:42 -0700598 ALOGV("%s(%d) returning %d", __func__, portHandle, result);
599 return result;
Eric Laurentcb4dae22017-07-01 19:39:32 -0700600}
601
jiabind5bd06a2021-04-27 22:04:08 +0000602aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t /*clockId*/,
Phil Burk3316d5e2017-02-15 11:23:01 -0800603 int64_t *framePosition,
604 int64_t *timeNanoseconds) {
Phil Burk97350f92017-07-21 15:59:44 -0700605 // Generated in server and passed to client. Return latest.
Phil Burka53ffa62018-10-10 16:21:37 -0700606 if (mAtomicInternalTimestamp.isValid()) {
607 Timestamp timestamp = mAtomicInternalTimestamp.read();
Phil Burkbcc36742017-08-31 17:24:51 -0700608 int64_t position = timestamp.getPosition() + mFramesOffsetFromService;
609 if (position >= 0) {
610 *framePosition = position;
611 *timeNanoseconds = timestamp.getNanoseconds();
612 return AAUDIO_OK;
613 }
Phil Burk97350f92017-07-21 15:59:44 -0700614 }
Phil Burkc75d97f2017-09-08 15:48:36 -0700615 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800616}
617
Phil Burkec89b2e2017-06-20 15:05:06 -0700618void AudioStreamInternal::logTimestamp(AAudioServiceMessage &command) {
Phil Burk204a1632017-01-03 17:23:43 -0800619 static int64_t oldPosition = 0;
Phil Burk3316d5e2017-02-15 11:23:01 -0800620 static int64_t oldTime = 0;
Phil Burk204a1632017-01-03 17:23:43 -0800621 int64_t framePosition = command.timestamp.position;
Phil Burk3316d5e2017-02-15 11:23:01 -0800622 int64_t nanoTime = command.timestamp.timestamp;
Phil Burkbcc36742017-08-31 17:24:51 -0700623 ALOGD("logTimestamp: timestamp says framePosition = %8lld at nanoTime %lld",
Phil Burk204a1632017-01-03 17:23:43 -0800624 (long long) framePosition,
625 (long long) nanoTime);
626 int64_t nanosDelta = nanoTime - oldTime;
627 if (nanosDelta > 0 && oldTime > 0) {
628 int64_t framesDelta = framePosition - oldPosition;
Phil Burk5ed503c2017-02-01 09:38:15 -0800629 int64_t rate = (framesDelta * AAUDIO_NANOS_PER_SECOND) / nanosDelta;
Phil Burkbcc36742017-08-31 17:24:51 -0700630 ALOGD("logTimestamp: framesDelta = %8lld, nanosDelta = %8lld, rate = %lld",
Phil Burkec89b2e2017-06-20 15:05:06 -0700631 (long long) framesDelta, (long long) nanosDelta, (long long) rate);
Phil Burk204a1632017-01-03 17:23:43 -0800632 }
633 oldPosition = framePosition;
634 oldTime = nanoTime;
635}
Phil Burk204a1632017-01-03 17:23:43 -0800636
Phil Burk97350f92017-07-21 15:59:44 -0700637aaudio_result_t AudioStreamInternal::onTimestampService(AAudioServiceMessage *message) {
Phil Burk204a1632017-01-03 17:23:43 -0800638#if LOG_TIMESTAMPS
Phil Burkec89b2e2017-06-20 15:05:06 -0700639 logTimestamp(*message);
Phil Burk204a1632017-01-03 17:23:43 -0800640#endif
Phil Burkb31b66f2019-09-30 09:33:41 -0700641 processTimestamp(message->timestamp.position,
642 message->timestamp.timestamp + mTimeOffsetNanos);
Phil Burk5ed503c2017-02-01 09:38:15 -0800643 return AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800644}
645
Phil Burk97350f92017-07-21 15:59:44 -0700646aaudio_result_t AudioStreamInternal::onTimestampHardware(AAudioServiceMessage *message) {
647 Timestamp timestamp(message->timestamp.position, message->timestamp.timestamp);
Phil Burka53ffa62018-10-10 16:21:37 -0700648 mAtomicInternalTimestamp.write(timestamp);
Phil Burk97350f92017-07-21 15:59:44 -0700649 return AAUDIO_OK;
650}
651
Phil Burk5ed503c2017-02-01 09:38:15 -0800652aaudio_result_t AudioStreamInternal::onEventFromServer(AAudioServiceMessage *message) {
653 aaudio_result_t result = AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800654 switch (message->event.event) {
Phil Burk5ed503c2017-02-01 09:38:15 -0800655 case AAUDIO_SERVICE_EVENT_STARTED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700656 ALOGD("%s - got AAUDIO_SERVICE_EVENT_STARTED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700657 if (getState() == AAUDIO_STREAM_STATE_STARTING) {
658 setState(AAUDIO_STREAM_STATE_STARTED);
659 }
Vlad Popaec1788e2022-08-04 11:23:30 +0200660 mPlayerBase->triggerPortIdUpdate(static_cast<audio_port_handle_t>(
661 message->event.dataLong));
Phil Burk204a1632017-01-03 17:23:43 -0800662 break;
Phil Burk5ed503c2017-02-01 09:38:15 -0800663 case AAUDIO_SERVICE_EVENT_PAUSED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700664 ALOGD("%s - got AAUDIO_SERVICE_EVENT_PAUSED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700665 if (getState() == AAUDIO_STREAM_STATE_PAUSING) {
666 setState(AAUDIO_STREAM_STATE_PAUSED);
667 }
Phil Burk204a1632017-01-03 17:23:43 -0800668 break;
Phil Burk71f35bb2017-04-13 16:05:07 -0700669 case AAUDIO_SERVICE_EVENT_STOPPED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700670 ALOGD("%s - got AAUDIO_SERVICE_EVENT_STOPPED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700671 if (getState() == AAUDIO_STREAM_STATE_STOPPING) {
672 setState(AAUDIO_STREAM_STATE_STOPPED);
673 }
Phil Burk71f35bb2017-04-13 16:05:07 -0700674 break;
Phil Burk5ed503c2017-02-01 09:38:15 -0800675 case AAUDIO_SERVICE_EVENT_FLUSHED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700676 ALOGD("%s - got AAUDIO_SERVICE_EVENT_FLUSHED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700677 if (getState() == AAUDIO_STREAM_STATE_FLUSHING) {
678 setState(AAUDIO_STREAM_STATE_FLUSHED);
679 onFlushFromServer();
680 }
Phil Burk204a1632017-01-03 17:23:43 -0800681 break;
Phil Burk5ed503c2017-02-01 09:38:15 -0800682 case AAUDIO_SERVICE_EVENT_DISCONNECTED:
Phil Burkea04d972017-08-07 12:30:44 -0700683 // Prevent hardware from looping on old data and making buzzing sounds.
684 if (getDirection() == AAUDIO_DIRECTION_OUTPUT) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700685 mAudioEndpoint->eraseDataMemory();
Phil Burkea04d972017-08-07 12:30:44 -0700686 }
Phil Burk5ed503c2017-02-01 09:38:15 -0800687 result = AAUDIO_ERROR_DISCONNECTED;
jiabincb212cd2022-08-24 16:50:44 -0700688 setDisconnected();
Phil Burkfbf031e2017-10-12 15:58:31 -0700689 ALOGW("%s - AAUDIO_SERVICE_EVENT_DISCONNECTED - FIFO cleared", __func__);
Phil Burk204a1632017-01-03 17:23:43 -0800690 break;
Phil Burkc0c70e32017-02-09 13:18:38 -0800691 case AAUDIO_SERVICE_EVENT_VOLUME:
Phil Burk55e5eab2018-04-10 15:16:38 -0700692 ALOGD("%s - AAUDIO_SERVICE_EVENT_VOLUME %lf", __func__, message->event.dataDouble);
Eric Laurenta2f296e2017-06-21 18:51:47 -0700693 mStreamVolume = (float)message->event.dataDouble;
694 doSetVolume();
Phil Burkc0c70e32017-02-09 13:18:38 -0800695 break;
Phil Burk23296382017-11-20 15:45:11 -0800696 case AAUDIO_SERVICE_EVENT_XRUN:
697 mXRunCount = static_cast<int32_t>(message->event.dataLong);
698 break;
Phil Burk204a1632017-01-03 17:23:43 -0800699 default:
Phil Burkfbf031e2017-10-12 15:58:31 -0700700 ALOGE("%s - Unrecognized event = %d", __func__, (int) message->event.event);
Phil Burk204a1632017-01-03 17:23:43 -0800701 break;
702 }
703 return result;
704}
705
Phil Burkbcc36742017-08-31 17:24:51 -0700706aaudio_result_t AudioStreamInternal::drainTimestampsFromService() {
707 aaudio_result_t result = AAUDIO_OK;
708
709 while (result == AAUDIO_OK) {
710 AAudioServiceMessage message;
Phil Burk5edc4ea2020-04-17 08:15:42 -0700711 if (!mAudioEndpoint) {
712 break;
713 }
714 if (mAudioEndpoint->readUpCommand(&message) != 1) {
Phil Burkbcc36742017-08-31 17:24:51 -0700715 break; // no command this time, no problem
716 }
717 switch (message.what) {
718 // ignore most messages
719 case AAudioServiceMessage::code::TIMESTAMP_SERVICE:
720 case AAudioServiceMessage::code::TIMESTAMP_HARDWARE:
721 break;
722
723 case AAudioServiceMessage::code::EVENT:
724 result = onEventFromServer(&message);
725 break;
726
727 default:
Phil Burkfbf031e2017-10-12 15:58:31 -0700728 ALOGE("%s - unrecognized message.what = %d", __func__, (int) message.what);
Phil Burkbcc36742017-08-31 17:24:51 -0700729 result = AAUDIO_ERROR_INTERNAL;
730 break;
731 }
732 }
733 return result;
734}
735
Phil Burk204a1632017-01-03 17:23:43 -0800736// Process all the commands coming from the server.
Phil Burk5ed503c2017-02-01 09:38:15 -0800737aaudio_result_t AudioStreamInternal::processCommands() {
738 aaudio_result_t result = AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800739
Phil Burk5ed503c2017-02-01 09:38:15 -0800740 while (result == AAUDIO_OK) {
741 AAudioServiceMessage message;
Phil Burk5edc4ea2020-04-17 08:15:42 -0700742 if (!mAudioEndpoint) {
743 break;
744 }
745 if (mAudioEndpoint->readUpCommand(&message) != 1) {
Phil Burk204a1632017-01-03 17:23:43 -0800746 break; // no command this time, no problem
747 }
748 switch (message.what) {
Phil Burk97350f92017-07-21 15:59:44 -0700749 case AAudioServiceMessage::code::TIMESTAMP_SERVICE:
750 result = onTimestampService(&message);
751 break;
752
753 case AAudioServiceMessage::code::TIMESTAMP_HARDWARE:
754 result = onTimestampHardware(&message);
Phil Burk204a1632017-01-03 17:23:43 -0800755 break;
756
Phil Burk5ed503c2017-02-01 09:38:15 -0800757 case AAudioServiceMessage::code::EVENT:
Phil Burk204a1632017-01-03 17:23:43 -0800758 result = onEventFromServer(&message);
759 break;
760
761 default:
Phil Burkfbf031e2017-10-12 15:58:31 -0700762 ALOGE("%s - unrecognized message.what = %d", __func__, (int) message.what);
Phil Burk17fff382017-05-16 14:06:45 -0700763 result = AAUDIO_ERROR_INTERNAL;
Phil Burk204a1632017-01-03 17:23:43 -0800764 break;
765 }
766 }
767 return result;
768}
769
Phil Burk87c9f642017-05-17 07:22:39 -0700770// Read or write the data, block if needed and timeoutMillis > 0
771aaudio_result_t AudioStreamInternal::processData(void *buffer, int32_t numFrames,
772 int64_t timeoutNanoseconds)
Phil Burk204a1632017-01-03 17:23:43 -0800773{
jiabin5f787812023-03-02 20:42:43 +0000774 if (isDisconnected()) {
775 return AAUDIO_ERROR_DISCONNECTED;
776 }
777 if (!mInService &&
778 AAudioBinderClient::getInstance().getServiceLifetimeId() != getServiceLifetimeId()) {
779 // The service lifetime id will be changed whenever the binder died. In that case, if
780 // the service lifetime id from AAudioBinderClient is different from the cached one,
781 // returns AAUDIO_ERROR_DISCONNECTED.
782 // Note that only compare the service lifetime id if it is not in service as the streams
783 // in service will all be gone when aaudio service dies.
784 mClockModel.stop(AudioClock::getNanoseconds());
785 // Set the stream as disconnected as the service lifetime id will only change when
786 // the binder dies.
787 setDisconnected();
788 return AAUDIO_ERROR_DISCONNECTED;
789 }
Phil Burkfd34a932017-07-19 07:03:52 -0700790 const char * traceName = "aaProc";
791 const char * fifoName = "aaRdy";
Phil Burk4485d412017-05-09 15:55:02 -0700792 ATRACE_BEGIN(traceName);
Phil Burk4485d412017-05-09 15:55:02 -0700793 if (ATRACE_ENABLED()) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700794 int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
Phil Burkfd34a932017-07-19 07:03:52 -0700795 ATRACE_INT(fifoName, fullFrames);
Phil Burk4485d412017-05-09 15:55:02 -0700796 }
797
Phil Burkec89b2e2017-06-20 15:05:06 -0700798 aaudio_result_t result = AAUDIO_OK;
799 int32_t loopCount = 0;
800 uint8_t* audioData = (uint8_t*)buffer;
801 int64_t currentTimeNanos = AudioClock::getNanoseconds();
802 const int64_t entryTimeNanos = currentTimeNanos;
803 const int64_t deadlineNanos = currentTimeNanos + timeoutNanoseconds;
804 int32_t framesLeft = numFrames;
805
Phil Burk87c9f642017-05-17 07:22:39 -0700806 // Loop until all the data has been processed or until a timeout occurs.
Phil Burk204a1632017-01-03 17:23:43 -0800807 while (framesLeft > 0) {
Phil Burkec89b2e2017-06-20 15:05:06 -0700808 // The call to processDataNow() will not block. It will just process as much as it can.
Phil Burk3316d5e2017-02-15 11:23:01 -0800809 int64_t wakeTimeNanos = 0;
Phil Burk87c9f642017-05-17 07:22:39 -0700810 aaudio_result_t framesProcessed = processDataNow(audioData, framesLeft,
811 currentTimeNanos, &wakeTimeNanos);
812 if (framesProcessed < 0) {
Phil Burk87c9f642017-05-17 07:22:39 -0700813 result = framesProcessed;
Phil Burk204a1632017-01-03 17:23:43 -0800814 break;
815 }
Phil Burk87c9f642017-05-17 07:22:39 -0700816 framesLeft -= (int32_t) framesProcessed;
817 audioData += framesProcessed * getBytesPerFrame();
Phil Burk204a1632017-01-03 17:23:43 -0800818
819 // Should we block?
820 if (timeoutNanoseconds == 0) {
821 break; // don't block
Phil Burk8d4f0062019-10-03 15:55:41 -0700822 } else if (wakeTimeNanos != 0) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700823 if (!mAudioEndpoint->isFreeRunning()) {
Phil Burkfd34a932017-07-19 07:03:52 -0700824 // If there is software on the other end of the FIFO then it may get delayed.
825 // So wake up just a little after we expect it to be ready.
826 wakeTimeNanos += mWakeupDelayNanos;
Phil Burk204a1632017-01-03 17:23:43 -0800827 }
Phil Burkfd34a932017-07-19 07:03:52 -0700828
Phil Burk2bc7c182017-08-28 11:45:01 -0700829 currentTimeNanos = AudioClock::getNanoseconds();
830 int64_t earliestWakeTime = currentTimeNanos + mMinimumSleepNanos;
831 // Guarantee a minimum sleep time.
832 if (wakeTimeNanos < earliestWakeTime) {
833 wakeTimeNanos = earliestWakeTime;
834 }
835
Phil Burk204a1632017-01-03 17:23:43 -0800836 if (wakeTimeNanos > deadlineNanos) {
837 // If we time out, just return the framesWritten so far.
Phil Burkfbf031e2017-10-12 15:58:31 -0700838 ALOGW("processData(): entered at %lld nanos, currently %lld",
Phil Burkec89b2e2017-06-20 15:05:06 -0700839 (long long) entryTimeNanos, (long long) currentTimeNanos);
Phil Burkfbf031e2017-10-12 15:58:31 -0700840 ALOGW("processData(): TIMEOUT after %lld nanos",
Phil Burkc0c70e32017-02-09 13:18:38 -0800841 (long long) timeoutNanoseconds);
Phil Burkfbf031e2017-10-12 15:58:31 -0700842 ALOGW("processData(): wakeTime = %lld, deadline = %lld nanos",
Phil Burk87c9f642017-05-17 07:22:39 -0700843 (long long) wakeTimeNanos, (long long) deadlineNanos);
Phil Burkfbf031e2017-10-12 15:58:31 -0700844 ALOGW("processData(): past deadline by %d micros",
Phil Burk87c9f642017-05-17 07:22:39 -0700845 (int)((wakeTimeNanos - deadlineNanos) / AAUDIO_NANOS_PER_MICROSECOND));
Phil Burkec89b2e2017-06-20 15:05:06 -0700846 mClockModel.dump();
Phil Burk5edc4ea2020-04-17 08:15:42 -0700847 mAudioEndpoint->dump();
Phil Burk204a1632017-01-03 17:23:43 -0800848 break;
849 }
850
Phil Burkfd34a932017-07-19 07:03:52 -0700851 if (ATRACE_ENABLED()) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700852 int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
Phil Burkfd34a932017-07-19 07:03:52 -0700853 ATRACE_INT(fifoName, fullFrames);
854 int64_t sleepForNanos = wakeTimeNanos - currentTimeNanos;
855 ATRACE_INT("aaSlpNs", (int32_t)sleepForNanos);
856 }
857
858 AudioClock::sleepUntilNanoTime(wakeTimeNanos);
Phil Burk204a1632017-01-03 17:23:43 -0800859 currentTimeNanos = AudioClock::getNanoseconds();
860 }
861 }
862
Phil Burkfd34a932017-07-19 07:03:52 -0700863 if (ATRACE_ENABLED()) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700864 int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
Phil Burkfd34a932017-07-19 07:03:52 -0700865 ATRACE_INT(fifoName, fullFrames);
866 }
867
Phil Burk87c9f642017-05-17 07:22:39 -0700868 // return error or framesProcessed
Phil Burkc0c70e32017-02-09 13:18:38 -0800869 (void) loopCount;
Phil Burk4485d412017-05-09 15:55:02 -0700870 ATRACE_END();
Phil Burk204a1632017-01-03 17:23:43 -0800871 return (result < 0) ? result : numFrames - framesLeft;
872}
873
Phil Burk3316d5e2017-02-15 11:23:01 -0800874void AudioStreamInternal::processTimestamp(uint64_t position, int64_t time) {
Phil Burk87c9f642017-05-17 07:22:39 -0700875 mClockModel.processTimestamp(position, time);
Phil Burk204a1632017-01-03 17:23:43 -0800876}
877
Phil Burk3316d5e2017-02-15 11:23:01 -0800878aaudio_result_t AudioStreamInternal::setBufferSize(int32_t requestedFrames) {
Phil Burk6479d502017-11-20 09:32:52 -0800879 int32_t adjustedFrames = requestedFrames;
Phil Burk8d97b8e2020-09-25 23:18:14 +0000880 const int32_t maximumSize = getBufferCapacity() - getFramesPerBurst();
Phil Burk5347dca2020-04-08 16:31:07 -0700881 // Minimum size should be a multiple number of bursts.
Phil Burk8d97b8e2020-09-25 23:18:14 +0000882 const int32_t minimumSize = 1 * getFramesPerBurst();
Phil Burk6479d502017-11-20 09:32:52 -0800883
884 // Clip to minimum size so that rounding up will work better.
Phil Burk8d4f0062019-10-03 15:55:41 -0700885 adjustedFrames = std::max(minimumSize, adjustedFrames);
Phil Burk71f35bb2017-04-13 16:05:07 -0700886
Phil Burk8d4f0062019-10-03 15:55:41 -0700887 // Prevent arithmetic overflow by clipping before we round.
888 if (adjustedFrames >= maximumSize) {
Phil Burk6479d502017-11-20 09:32:52 -0800889 adjustedFrames = maximumSize;
890 } else {
891 // Round to the next highest burst size.
Phil Burk8d97b8e2020-09-25 23:18:14 +0000892 int32_t numBursts = (adjustedFrames + getFramesPerBurst() - 1) / getFramesPerBurst();
893 adjustedFrames = numBursts * getFramesPerBurst();
894 // Clip just in case maximumSize is not a multiple of getFramesPerBurst().
Phil Burk5347dca2020-04-08 16:31:07 -0700895 adjustedFrames = std::min(maximumSize, adjustedFrames);
Phil Burk6479d502017-11-20 09:32:52 -0800896 }
897
Phil Burk5edc4ea2020-04-17 08:15:42 -0700898 if (mAudioEndpoint) {
899 // Clip against the actual size from the endpoint.
900 int32_t actualFrames = 0;
901 // Set to maximum size so we can write extra data when ready in order to reduce glitches.
902 // The amount we keep in the buffer is controlled by mBufferSizeInFrames.
903 mAudioEndpoint->setBufferSizeInFrames(maximumSize, &actualFrames);
904 // actualFrames should be <= actual maximum size of endpoint
905 adjustedFrames = std::min(actualFrames, adjustedFrames);
906 }
Phil Burk8d4f0062019-10-03 15:55:41 -0700907
Phil Burk64e16a72020-06-01 13:25:51 -0700908 if (adjustedFrames != mBufferSizeInFrames) {
909 android::mediametrics::LogItem(mMetricsId)
910 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
911 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, adjustedFrames)
912 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getXRunCount())
913 .record();
914 }
915
Phil Burk8d4f0062019-10-03 15:55:41 -0700916 mBufferSizeInFrames = adjustedFrames;
Phil Burk6c63ae32019-10-28 10:28:21 -0700917 ALOGV("%s(%d) returns %d", __func__, requestedFrames, adjustedFrames);
Phil Burk8d4f0062019-10-03 15:55:41 -0700918 return (aaudio_result_t) adjustedFrames;
Phil Burk204a1632017-01-03 17:23:43 -0800919}
920
Phil Burk87c9f642017-05-17 07:22:39 -0700921int32_t AudioStreamInternal::getBufferSize() const {
Phil Burk8d4f0062019-10-03 15:55:41 -0700922 return mBufferSizeInFrames;
Phil Burk204a1632017-01-03 17:23:43 -0800923}
924
Phil Burk87c9f642017-05-17 07:22:39 -0700925int32_t AudioStreamInternal::getBufferCapacity() const {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700926 return mBufferCapacityInFrames;
Phil Burk204a1632017-01-03 17:23:43 -0800927}
928
Phil Burk377c1c22018-12-12 16:06:54 -0800929bool AudioStreamInternal::isClockModelInControl() const {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700930 return isActive() && mAudioEndpoint->isFreeRunning() && mClockModel.isRunning();
Phil Burk377c1c22018-12-12 16:06:54 -0800931}