blob: ca0db0da1552427ad5f27bc73356b90f30e18bd7 [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>
Robert Wu101ad252023-11-28 20:29:29 +000045#include <com_android_media_aaudio.h>
Phil Burke572f462017-04-20 13:03:19 -070046
Phil Burkc0c70e32017-02-09 13:18:38 -080047#include "AudioStreamInternal.h"
Phil Burk204a1632017-01-03 17:23:43 -080048
Phil Burka9876702020-04-20 18:16:15 -070049// We do this after the #includes because if a header uses ALOG.
50// it would fail on the reference to mInService.
51#undef LOG_TAG
52// This file is used in both client and server processes.
53// This is needed to make sense of the logs more easily.
54#define LOG_TAG (mInService ? "AudioStreamInternal_Service" : "AudioStreamInternal_Client")
55
Svet Ganov3e5f14f2021-05-13 22:51:08 +000056using android::content::AttributionSourceState;
Phil Burk204a1632017-01-03 17:23:43 -080057
Phil Burk5ed503c2017-02-01 09:38:15 -080058using namespace aaudio;
Phil Burk204a1632017-01-03 17:23:43 -080059
Phil Burke4d7bb42017-03-28 11:32:39 -070060#define MIN_TIMEOUT_NANOS (1000 * AAUDIO_NANOS_PER_MILLISECOND)
61
62// Wait at least this many times longer than the operation should take.
63#define MIN_TIMEOUT_OPERATIONS 4
64
Phil Burkbcc36742017-08-31 17:24:51 -070065#define LOG_TIMESTAMPS 0
Phil Burk87c9f642017-05-17 07:22:39 -070066
Phil Burkc0c70e32017-02-09 13:18:38 -080067AudioStreamInternal::AudioStreamInternal(AAudioServiceInterface &serviceInterface, bool inService)
Phil Burk204a1632017-01-03 17:23:43 -080068 : AudioStream()
69 , mClockModel()
Phil Burkec89b2e2017-06-20 15:05:06 -070070 , mInService(inService)
Phil Burkfd34a932017-07-19 07:03:52 -070071 , mServiceInterface(serviceInterface)
Phil Burka53ffa62018-10-10 16:21:37 -070072 , mAtomicInternalTimestamp()
Phil Burkfd34a932017-07-19 07:03:52 -070073 , mWakeupDelayNanos(AAudioProperty_getWakeupDelayMicros() * AAUDIO_NANOS_PER_MICROSECOND)
74 , mMinimumSleepNanos(AAudioProperty_getMinimumSleepMicros() * AAUDIO_NANOS_PER_MICROSECOND)
75 {
jiabin5f787812023-03-02 20:42:43 +000076
Phil Burk204a1632017-01-03 17:23:43 -080077}
78
79AudioStreamInternal::~AudioStreamInternal() {
Phil Burkdd582922020-10-15 20:29:51 +000080 ALOGD("%s() %p called", __func__, this);
Phil Burk204a1632017-01-03 17:23:43 -080081}
82
Phil Burk5ed503c2017-02-01 09:38:15 -080083aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) {
Phil Burk204a1632017-01-03 17:23:43 -080084
Phil Burk5ed503c2017-02-01 09:38:15 -080085 aaudio_result_t result = AAUDIO_OK;
86 AAudioStreamRequest request;
Phil Burk99306c82017-08-14 12:38:58 -070087 AAudioStreamConfiguration configurationOutput;
Phil Burk204a1632017-01-03 17:23:43 -080088
Phil Burk99306c82017-08-14 12:38:58 -070089 if (getState() != AAUDIO_STREAM_STATE_UNINITIALIZED) {
Phil Burkfbf031e2017-10-12 15:58:31 -070090 ALOGE("%s - already open! state = %d", __func__, getState());
Phil Burk99306c82017-08-14 12:38:58 -070091 return AAUDIO_ERROR_INVALID_STATE;
92 }
93
94 // Copy requested parameters to the stream.
Phil Burk204a1632017-01-03 17:23:43 -080095 result = AudioStream::open(builder);
96 if (result < 0) {
97 return result;
98 }
99
jiabinef348b82021-04-19 16:53:08 +0000100 const audio_format_t requestedFormat = getFormat();
Phil Burkc0c70e32017-02-09 13:18:38 -0800101 // We have to do volume scaling. So we prefer FLOAT format.
jiabinef348b82021-04-19 16:53:08 +0000102 if (requestedFormat == AUDIO_FORMAT_DEFAULT) {
Phil Burk0127c1b2018-03-29 13:48:06 -0700103 setFormat(AUDIO_FORMAT_PCM_FLOAT);
Phil Burkc0c70e32017-02-09 13:18:38 -0800104 }
Phil Burk04e805b2018-03-27 09:13:53 -0700105 // Request FLOAT for the shared mixer or the device.
Phil Burk0127c1b2018-03-29 13:48:06 -0700106 request.getConfiguration().setFormat(AUDIO_FORMAT_PCM_FLOAT);
Phil Burkc0c70e32017-02-09 13:18:38 -0800107
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000108 // TODO b/182392769: use attribution source util
109 AttributionSourceState attributionSource;
110 attributionSource.uid = VALUE_OR_FATAL(android::legacy2aidl_uid_t_int32_t(getuid()));
111 attributionSource.pid = VALUE_OR_FATAL(android::legacy2aidl_pid_t_int32_t(getpid()));
112 attributionSource.packageName = builder.getOpPackageName();
113 attributionSource.attributionTag = builder.getAttributionTag();
114 attributionSource.token = sp<android::BBinder>::make();
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700115
Phil Burkdec33ab2017-01-17 14:48:16 -0800116 // Build the request to send to the server.
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000117 request.setAttributionSource(attributionSource);
Phil Burk71f35bb2017-04-13 16:05:07 -0700118 request.setSharingModeMatchRequired(isSharingModeMatchRequired());
Phil Burk41f19d82018-02-13 14:59:10 -0800119 request.setInService(isInService());
Phil Burkc0c70e32017-02-09 13:18:38 -0800120
Phil Burk204a1632017-01-03 17:23:43 -0800121 request.getConfiguration().setDeviceId(getDeviceId());
122 request.getConfiguration().setSampleRate(getSampleRate());
Phil Burk39f02dd2017-08-04 09:13:31 -0700123 request.getConfiguration().setDirection(getDirection());
Phil Burk71f35bb2017-04-13 16:05:07 -0700124 request.getConfiguration().setSharingMode(getSharingMode());
jiabina9094092021-06-28 20:36:45 +0000125 request.getConfiguration().setChannelMask(getChannelMask());
Phil Burk71f35bb2017-04-13 16:05:07 -0700126
Phil Burka62fb952018-01-16 12:44:06 -0800127 request.getConfiguration().setUsage(getUsage());
128 request.getConfiguration().setContentType(getContentType());
Jean-Michel Trivi656bfdc2021-09-20 18:42:37 -0700129 request.getConfiguration().setSpatializationBehavior(getSpatializationBehavior());
130 request.getConfiguration().setIsContentSpatialized(isContentSpatialized());
Phil Burka62fb952018-01-16 12:44:06 -0800131 request.getConfiguration().setInputPreset(getInputPreset());
Eric Laurentd17c8502019-10-24 15:58:35 -0700132 request.getConfiguration().setPrivacySensitive(isPrivacySensitive());
Phil Burka62fb952018-01-16 12:44:06 -0800133
Phil Burk3df348f2017-02-08 11:41:55 -0800134 request.getConfiguration().setBufferCapacity(builder.getBufferCapacity());
Phil Burk204a1632017-01-03 17:23:43 -0800135
jiabin5f787812023-03-02 20:42:43 +0000136 mServiceStreamHandleInfo = mServiceInterface.openStream(request, configurationOutput);
137 if (getServiceHandle() < 0
jiabina9094092021-06-28 20:36:45 +0000138 && (request.getConfiguration().getSamplesPerFrame() == 1
139 || request.getConfiguration().getChannelMask() == AAUDIO_CHANNEL_MONO)
Phil Burk41f19d82018-02-13 14:59:10 -0800140 && getDirection() == AAUDIO_DIRECTION_OUTPUT
141 && !isInService()) {
142 // if that failed then try switching from mono to stereo if OUTPUT.
143 // Only do this in the client. Otherwise we end up with a mono mixer in the service
144 // that writes to a stereo MMAP stream.
Phil Burk0127c1b2018-03-29 13:48:06 -0700145 ALOGD("%s() - openStream() returned %d, try switching from MONO to STEREO",
jiabin5f787812023-03-02 20:42:43 +0000146 __func__, getServiceHandle());
jiabina9094092021-06-28 20:36:45 +0000147 request.getConfiguration().setChannelMask(AAUDIO_CHANNEL_STEREO);
jiabin5f787812023-03-02 20:42:43 +0000148 mServiceStreamHandleInfo = mServiceInterface.openStream(request, configurationOutput);
Phil Burk41f19d82018-02-13 14:59:10 -0800149 }
jiabin5f787812023-03-02 20:42:43 +0000150 if (getServiceHandle() < 0) {
151 return getServiceHandle();
Phil Burk204a1632017-01-03 17:23:43 -0800152 }
Phil Burk99306c82017-08-14 12:38:58 -0700153
Phil Burka9876702020-04-20 18:16:15 -0700154 // This must match the key generated in oboeservice/AAudioServiceStreamBase.cpp
155 // so the client can have permission to log.
jiabinfbf20302021-07-28 22:15:01 +0000156 if (!mInService) {
157 // No need to log if it is from service side.
158 mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_STREAM)
jiabin5f787812023-03-02 20:42:43 +0000159 + std::to_string(getServiceHandle());
jiabinfbf20302021-07-28 22:15:01 +0000160 }
Phil Burka9876702020-04-20 18:16:15 -0700161
jiabinef348b82021-04-19 16:53:08 +0000162 android::mediametrics::LogItem(mMetricsId)
163 .set(AMEDIAMETRICS_PROP_PERFORMANCEMODE,
jiabinc8da9032021-04-28 20:42:36 +0000164 AudioGlobal_convertPerformanceModeToText(builder.getPerformanceMode()))
165 .set(AMEDIAMETRICS_PROP_SHARINGMODE,
166 AudioGlobal_convertSharingModeToText(builder.getSharingMode()))
jiabinef348b82021-04-19 16:53:08 +0000167 .set(AMEDIAMETRICS_PROP_ENCODINGCLIENT,
168 android::toString(requestedFormat).c_str()).record();
169
Phil Burk99306c82017-08-14 12:38:58 -0700170 result = configurationOutput.validate();
171 if (result != AAUDIO_OK) {
172 goto error;
173 }
174 // Save results of the open.
jiabina9094092021-06-28 20:36:45 +0000175 if (getChannelMask() == AAUDIO_UNSPECIFIED) {
176 setChannelMask(configurationOutput.getChannelMask());
Phil Burk41f19d82018-02-13 14:59:10 -0800177 }
jiabina9094092021-06-28 20:36:45 +0000178
Phil Burk99306c82017-08-14 12:38:58 -0700179 setDeviceId(configurationOutput.getDeviceId());
Phil Burk4e1af9f2018-01-03 15:54:35 -0800180 setSessionId(configurationOutput.getSessionId());
Phil Burk99306c82017-08-14 12:38:58 -0700181 setSharingMode(configurationOutput.getSharingMode());
182
Phil Burka62fb952018-01-16 12:44:06 -0800183 setUsage(configurationOutput.getUsage());
184 setContentType(configurationOutput.getContentType());
Jean-Michel Trivi656bfdc2021-09-20 18:42:37 -0700185 setSpatializationBehavior(configurationOutput.getSpatializationBehavior());
186 setIsContentSpatialized(configurationOutput.isContentSpatialized());
Phil Burka62fb952018-01-16 12:44:06 -0800187 setInputPreset(configurationOutput.getInputPreset());
188
Robert Wud559ba52023-06-29 00:08:51 +0000189 setDeviceSampleRate(configurationOutput.getSampleRate());
190
191 if (getSampleRate() == AAUDIO_UNSPECIFIED) {
192 setSampleRate(configurationOutput.getSampleRate());
193 }
194
Robert Wu101ad252023-11-28 20:29:29 +0000195 if (!com::android::media::aaudio::sample_rate_conversion()) {
196 if (getSampleRate() != getDeviceSampleRate()) {
197 ALOGD("%s - skipping sample rate converter. SR = %d, Device SR = %d", __func__,
198 getSampleRate(), getDeviceSampleRate());
199 goto error;
200 }
Robert Wud559ba52023-06-29 00:08:51 +0000201 }
Robert Wud559ba52023-06-29 00:08:51 +0000202
Phil Burk99306c82017-08-14 12:38:58 -0700203 // Save device format so we can do format conversion and volume scaling together.
Phil Burk3d786cb2018-04-09 11:58:09 -0700204 setDeviceFormat(configurationOutput.getFormat());
Robert Wue8b58962023-07-21 19:48:56 +0000205 setDeviceSamplesPerFrame(configurationOutput.getSamplesPerFrame());
Phil Burk99306c82017-08-14 12:38:58 -0700206
Robert Wu310037a2022-09-06 21:48:18 +0000207 setHardwareSamplesPerFrame(configurationOutput.getHardwareSamplesPerFrame());
208 setHardwareSampleRate(configurationOutput.getHardwareSampleRate());
209 setHardwareFormat(configurationOutput.getHardwareFormat());
210
jiabin5f787812023-03-02 20:42:43 +0000211 result = mServiceInterface.getStreamDescription(mServiceStreamHandleInfo, mEndPointParcelable);
Phil Burk99306c82017-08-14 12:38:58 -0700212 if (result != AAUDIO_OK) {
213 goto error;
214 }
215
216 // Resolve parcelable into a descriptor.
217 result = mEndPointParcelable.resolve(&mEndpointDescriptor);
218 if (result != AAUDIO_OK) {
219 goto error;
220 }
221
222 // Configure endpoint based on descriptor.
Phil Burk5edc4ea2020-04-17 08:15:42 -0700223 mAudioEndpoint = std::make_unique<AudioEndpoint>();
224 result = mAudioEndpoint->configure(&mEndpointDescriptor, getDirection());
Phil Burk99306c82017-08-14 12:38:58 -0700225 if (result != AAUDIO_OK) {
226 goto error;
227 }
228
jiabinf7f06152021-11-22 18:10:14 +0000229 if ((result = configureDataInformation(builder.getFramesPerDataCallback())) != AAUDIO_OK) {
230 goto error;
231 }
232
233 setState(AAUDIO_STREAM_STATE_OPEN);
234
235 return result;
236
237error:
238 safeReleaseClose();
239 return result;
240}
241
242aaudio_result_t AudioStreamInternal::configureDataInformation(int32_t callbackFrames) {
Robert Wu32d319b2023-11-09 22:40:52 +0000243 int32_t originalFramesPerBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
244 int32_t deviceFramesPerBurst = originalFramesPerBurst;
Phil Burk3c4e6b52019-01-22 15:53:36 -0800245
246 // Scale up the burst size to meet the minimum equivalent in microseconds.
247 // This is to avoid waking the CPU too often when the HW burst is very small
Robert Wud559ba52023-06-29 00:08:51 +0000248 // or at high sample rates. The actual number of frames that we call back to
249 // the app with will be 0 < N <= framesPerBurst so round up the division.
jiabinf7f06152021-11-22 18:10:14 +0000250 int32_t burstMicros = 0;
251 const int32_t burstMinMicros = android::AudioSystem::getAAudioHardwareBurstMinUsec();
Phil Burk3c4e6b52019-01-22 15:53:36 -0800252 do {
253 if (burstMicros > 0) { // skip first loop
Robert Wud559ba52023-06-29 00:08:51 +0000254 deviceFramesPerBurst *= 2;
Phil Burk3c4e6b52019-01-22 15:53:36 -0800255 }
Robert Wu32d319b2023-11-09 22:40:52 +0000256 burstMicros = deviceFramesPerBurst * static_cast<int64_t>(1000000) / getDeviceSampleRate();
Phil Burk3c4e6b52019-01-22 15:53:36 -0800257 } while (burstMicros < burstMinMicros);
258 ALOGD("%s() original HW burst = %d, minMicros = %d => SW burst = %d\n",
Robert Wu32d319b2023-11-09 22:40:52 +0000259 __func__, originalFramesPerBurst, burstMinMicros, deviceFramesPerBurst);
Phil Burk3c4e6b52019-01-22 15:53:36 -0800260
261 // Validate final burst size.
Robert Wu32d319b2023-11-09 22:40:52 +0000262 if (deviceFramesPerBurst < MIN_FRAMES_PER_BURST
263 || deviceFramesPerBurst > MAX_FRAMES_PER_BURST) {
264 ALOGE("%s - deviceFramesPerBurst out of range = %d", __func__, deviceFramesPerBurst);
jiabinf7f06152021-11-22 18:10:14 +0000265 return AAUDIO_ERROR_OUT_OF_RANGE;
Phil Burk99306c82017-08-14 12:38:58 -0700266 }
Robert Wu32d319b2023-11-09 22:40:52 +0000267
268 // Calculate the application framesPerBurst from the deviceFramesPerBurst
269 int32_t framesPerBurst = (static_cast<int64_t>(deviceFramesPerBurst) * getSampleRate() +
270 getDeviceSampleRate() - 1) / getDeviceSampleRate();
271
Robert Wud559ba52023-06-29 00:08:51 +0000272 setDeviceFramesPerBurst(deviceFramesPerBurst);
Phil Burk8d97b8e2020-09-25 23:18:14 +0000273 setFramesPerBurst(framesPerBurst); // only save good value
Phil Burk6479d502017-11-20 09:32:52 -0800274
Robert Wud559ba52023-06-29 00:08:51 +0000275 mDeviceBufferCapacityInFrames = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
276
277 mBufferCapacityInFrames = static_cast<int64_t>(mDeviceBufferCapacityInFrames)
278 * getSampleRate() / getDeviceSampleRate();
Phil Burk8d97b8e2020-09-25 23:18:14 +0000279 if (mBufferCapacityInFrames < getFramesPerBurst()
Phil Burk5edc4ea2020-04-17 08:15:42 -0700280 || mBufferCapacityInFrames > MAX_BUFFER_CAPACITY_IN_FRAMES) {
281 ALOGE("%s - bufferCapacity out of range = %d", __func__, mBufferCapacityInFrames);
jiabinf7f06152021-11-22 18:10:14 +0000282 return AAUDIO_ERROR_OUT_OF_RANGE;
Phil Burk99306c82017-08-14 12:38:58 -0700283 }
284
Robert Wud559ba52023-06-29 00:08:51 +0000285 mClockModel.setSampleRate(getDeviceSampleRate());
286 mClockModel.setFramesPerBurst(deviceFramesPerBurst);
Phil Burk99306c82017-08-14 12:38:58 -0700287
Phil Burk134f1972017-12-08 13:06:11 -0800288 if (isDataCallbackSet()) {
jiabinf7f06152021-11-22 18:10:14 +0000289 mCallbackFrames = callbackFrames;
Phil Burk99306c82017-08-14 12:38:58 -0700290 if (mCallbackFrames > getBufferCapacity() / 2) {
Phil Burk29ccc292019-04-15 08:58:08 -0700291 ALOGW("%s - framesPerCallback too big = %d, capacity = %d",
Phil Burkfbf031e2017-10-12 15:58:31 -0700292 __func__, mCallbackFrames, getBufferCapacity());
jiabinf7f06152021-11-22 18:10:14 +0000293 return AAUDIO_ERROR_OUT_OF_RANGE;
Phil Burk99306c82017-08-14 12:38:58 -0700294 } else if (mCallbackFrames < 0) {
Phil Burk29ccc292019-04-15 08:58:08 -0700295 ALOGW("%s - framesPerCallback negative", __func__);
jiabinf7f06152021-11-22 18:10:14 +0000296 return AAUDIO_ERROR_OUT_OF_RANGE;
Phil Burk99306c82017-08-14 12:38:58 -0700297 }
298 if (mCallbackFrames == AAUDIO_UNSPECIFIED) {
Phil Burk8d97b8e2020-09-25 23:18:14 +0000299 mCallbackFrames = getFramesPerBurst();
Phil Burk99306c82017-08-14 12:38:58 -0700300 }
301
Phil Burk0127c1b2018-03-29 13:48:06 -0700302 const int32_t callbackBufferSize = mCallbackFrames * getBytesPerFrame();
Phil Burkbf821e22020-04-17 11:51:43 -0700303 mCallbackBuffer = std::make_unique<uint8_t[]>(callbackBufferSize);
Phil Burk99306c82017-08-14 12:38:58 -0700304 }
305
Robert Wud7400832021-12-04 01:11:19 +0000306 // Exclusive output streams should combine channels when mono audio adjustment
Robert Wu8393bed2021-12-08 02:08:48 +0000307 // is enabled. They should also adjust for audio balance.
Robert Wud7400832021-12-04 01:11:19 +0000308 if ((getDirection() == AAUDIO_DIRECTION_OUTPUT) &&
309 (getSharingMode() == AAUDIO_SHARING_MODE_EXCLUSIVE)) {
310 bool isMasterMono = false;
311 android::AudioSystem::getMasterMono(&isMasterMono);
312 setRequireMonoBlend(isMasterMono);
Robert Wu8393bed2021-12-08 02:08:48 +0000313 float audioBalance = 0;
314 android::AudioSystem::getMasterBalance(&audioBalance);
315 setAudioBalance(audioBalance);
Robert Wud7400832021-12-04 01:11:19 +0000316 }
317
Phil Burkb31b66f2019-09-30 09:33:41 -0700318 // For debugging and analyzing the distribution of MMAP timestamps.
319 // For OUTPUT, use a NEGATIVE offset to move the CPU writes further BEFORE the HW reads.
320 // For INPUT, use a POSITIVE offset to move the CPU reads further AFTER the HW writes.
321 // You can use this offset to reduce glitching.
322 // You can also use this offset to force glitching. By iterating over multiple
323 // values you can reveal the distribution of the hardware timing jitter.
Phil Burk5edc4ea2020-04-17 08:15:42 -0700324 if (mAudioEndpoint->isFreeRunning()) { // MMAP?
Phil Burkb31b66f2019-09-30 09:33:41 -0700325 int32_t offsetMicros = (getDirection() == AAUDIO_DIRECTION_OUTPUT)
326 ? AAudioProperty_getOutputMMapOffsetMicros()
327 : AAudioProperty_getInputMMapOffsetMicros();
328 // This log is used to debug some tricky glitch issues. Please leave.
329 ALOGD_IF(offsetMicros, "%s() - %s mmap offset = %d micros",
330 __func__,
331 (getDirection() == AAUDIO_DIRECTION_OUTPUT) ? "output" : "input",
332 offsetMicros);
333 mTimeOffsetNanos = offsetMicros * AAUDIO_NANOS_PER_MICROSECOND;
334 }
335
Robert Wud559ba52023-06-29 00:08:51 +0000336 // Default buffer size to match Q
337 setBufferSize(mBufferCapacityInFrames / 2);
jiabinf7f06152021-11-22 18:10:14 +0000338 return AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800339}
340
Phil Burk13d3d832019-06-10 14:36:48 -0700341// This must be called under mStreamLock.
Phil Burk8b4e05e2019-12-17 12:12:09 -0800342aaudio_result_t AudioStreamInternal::release_l() {
Phil Burk965650e2017-09-07 21:00:09 -0700343 aaudio_result_t result = AAUDIO_OK;
jiabin5f787812023-03-02 20:42:43 +0000344 ALOGD("%s(): mServiceStreamHandle = 0x%08X", __func__, getServiceHandle());
345 if (getServiceHandle() != AAUDIO_HANDLE_INVALID) {
Phil Burk8b4e05e2019-12-17 12:12:09 -0800346 // Don't release a stream while it is running. Stop it first.
Phil Burk13d3d832019-06-10 14:36:48 -0700347 // If DISCONNECTED then we should still try to stop in case the
348 // error callback is still running.
jiabincb212cd2022-08-24 16:50:44 -0700349 if (isActive() || isDisconnected()) {
Phil Burkdd582922020-10-15 20:29:51 +0000350 requestStop_l();
Phil Burk4485d412017-05-09 15:55:02 -0700351 }
Phil Burka9876702020-04-20 18:16:15 -0700352
Phil Burk64e16a72020-06-01 13:25:51 -0700353 logReleaseBufferState();
Phil Burka9876702020-04-20 18:16:15 -0700354
Phil Burkec89b2e2017-06-20 15:05:06 -0700355 setState(AAUDIO_STREAM_STATE_CLOSING);
jiabin5f787812023-03-02 20:42:43 +0000356 auto serviceStreamHandleInfo = mServiceStreamHandleInfo;
357 mServiceStreamHandleInfo = AAudioHandleInfo();
Phil Burkc0c70e32017-02-09 13:18:38 -0800358
jiabin5f787812023-03-02 20:42:43 +0000359 mServiceInterface.closeStream(serviceStreamHandleInfo);
Phil Burkbf821e22020-04-17 11:51:43 -0700360 mCallbackBuffer.reset();
Phil Burk5edc4ea2020-04-17 08:15:42 -0700361
362 // Update local frame counters so we can query them after releasing the endpoint.
363 getFramesRead();
364 getFramesWritten();
365 mAudioEndpoint.reset();
Phil Burk965650e2017-09-07 21:00:09 -0700366 result = mEndPointParcelable.close();
Phil Burk8b4e05e2019-12-17 12:12:09 -0800367 aaudio_result_t result2 = AudioStream::release_l();
Phil Burk965650e2017-09-07 21:00:09 -0700368 return (result != AAUDIO_OK) ? result : result2;
Phil Burk204a1632017-01-03 17:23:43 -0800369 } else {
Phil Burk5ed503c2017-02-01 09:38:15 -0800370 return AAUDIO_ERROR_INVALID_HANDLE;
Phil Burk204a1632017-01-03 17:23:43 -0800371 }
372}
373
Phil Burke4d7bb42017-03-28 11:32:39 -0700374static void *aaudio_callback_thread_proc(void *context)
375{
376 AudioStreamInternal *stream = (AudioStreamInternal *)context;
Phil Burkfbf031e2017-10-12 15:58:31 -0700377 //LOGD("oboe_callback_thread, stream = %p", stream);
jiabind5bd06a2021-04-27 22:04:08 +0000378 if (stream != nullptr) {
Phil Burke4d7bb42017-03-28 11:32:39 -0700379 return stream->callbackLoop();
380 } else {
jiabind5bd06a2021-04-27 22:04:08 +0000381 return nullptr;
Phil Burke4d7bb42017-03-28 11:32:39 -0700382 }
383}
384
jiabinf7f06152021-11-22 18:10:14 +0000385aaudio_result_t AudioStreamInternal::exitStandby_l() {
386 AudioEndpointParcelable endpointParcelable;
387 // The stream is in standby mode, copy all available data and then close the duplicated
388 // shared file descriptor so that it won't cause issue when the HAL try to reallocate new
389 // shared file descriptor when exiting from standby.
390 // Cache current read counter, which will be reset to new read and write counter
391 // when the new data queue and endpoint are reconfigured.
392 const android::fifo_counter_t readCounter = mAudioEndpoint->getDataReadCounter();
393 // Cache the buffer size which may be from client.
394 const int32_t previousBufferSize = mBufferSizeInFrames;
395 // Copy all available data from current data queue.
Robert Wud559ba52023-06-29 00:08:51 +0000396 uint8_t buffer[getDeviceBufferCapacity() * getBytesPerFrame()];
397 android::fifo_frames_t fullFramesAvailable = mAudioEndpoint->read(buffer,
398 getDeviceBufferCapacity());
jiabinf7f06152021-11-22 18:10:14 +0000399 mEndPointParcelable.closeDataFileDescriptor();
400 aaudio_result_t result = mServiceInterface.exitStandby(
jiabin5f787812023-03-02 20:42:43 +0000401 mServiceStreamHandleInfo, endpointParcelable);
jiabinf7f06152021-11-22 18:10:14 +0000402 if (result != AAUDIO_OK) {
403 ALOGE("Failed to exit standby, error=%d", result);
404 goto exit;
405 }
406 // Reconstruct data queue descriptor using new shared file descriptor.
jiabinfc791ee2023-02-15 19:43:40 +0000407 result = mEndPointParcelable.updateDataFileDescriptor(&endpointParcelable);
408 if (result != AAUDIO_OK) {
409 ALOGE("%s failed to update data file descriptor, error=%d", __func__, result);
410 goto exit;
411 }
jiabinf7f06152021-11-22 18:10:14 +0000412 result = mEndPointParcelable.resolveDataQueue(&mEndpointDescriptor.dataQueueDescriptor);
413 if (result != AAUDIO_OK) {
414 ALOGE("Failed to resolve data queue after exiting standby, error=%d", result);
415 goto exit;
416 }
417 // Reconfigure audio endpoint with new data queue descriptor.
418 mAudioEndpoint->configureDataQueue(
419 mEndpointDescriptor.dataQueueDescriptor, getDirection());
420 // Set read and write counters with previous read counter, the later write action
421 // will make the counter at the correct place.
422 mAudioEndpoint->setDataReadCounter(readCounter);
423 mAudioEndpoint->setDataWriteCounter(readCounter);
424 result = configureDataInformation(mCallbackFrames);
425 if (result != AAUDIO_OK) {
426 ALOGE("Failed to configure data information after exiting standby, error=%d", result);
427 goto exit;
428 }
429 // Write data from previous data buffer to new endpoint.
Robert Wud559ba52023-06-29 00:08:51 +0000430 if (const android::fifo_frames_t framesWritten =
jiabinf7f06152021-11-22 18:10:14 +0000431 mAudioEndpoint->write(buffer, fullFramesAvailable);
432 framesWritten != fullFramesAvailable) {
433 ALOGW("Some data lost after exiting standby, frames written: %d, "
434 "frames to write: %d", framesWritten, fullFramesAvailable);
435 }
436 // Reset previous buffer size as it may be requested by the client.
437 setBufferSize(previousBufferSize);
438
439exit:
440 return result;
441}
442
Phil Burkbcc36742017-08-31 17:24:51 -0700443/*
444 * It normally takes about 20-30 msec to start a stream on the server.
445 * But the first time can take as much as 200-300 msec. The HW
446 * starts right away so by the time the client gets a chance to write into
447 * the buffer, it is already in a deep underflow state. That can cause the
448 * XRunCount to be non-zero, which could lead an app to tune its latency higher.
449 * To avoid this problem, we set a request for the processing code to start the
450 * client stream at the same position as the server stream.
451 * The processing code will then save the current offset
452 * between client and server and apply that to any position given to the app.
453 */
Phil Burkdd582922020-10-15 20:29:51 +0000454aaudio_result_t AudioStreamInternal::requestStart_l()
Phil Burk204a1632017-01-03 17:23:43 -0800455{
Phil Burk3316d5e2017-02-15 11:23:01 -0800456 int64_t startTime;
jiabin5f787812023-03-02 20:42:43 +0000457 if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700458 ALOGD("requestStart() mServiceStreamHandle invalid");
Phil Burk5ed503c2017-02-01 09:38:15 -0800459 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800460 }
Phil Burkec89b2e2017-06-20 15:05:06 -0700461 if (isActive()) {
Phil Burk29ccc292019-04-15 08:58:08 -0700462 ALOGD("requestStart() already active");
Phil Burkec89b2e2017-06-20 15:05:06 -0700463 return AAUDIO_ERROR_INVALID_STATE;
464 }
Phil Burkec89b2e2017-06-20 15:05:06 -0700465
jiabincb212cd2022-08-24 16:50:44 -0700466 if (isDisconnected()) {
Phil Burk29ccc292019-04-15 08:58:08 -0700467 ALOGD("requestStart() but DISCONNECTED");
Phil Burkbcc36742017-08-31 17:24:51 -0700468 return AAUDIO_ERROR_DISCONNECTED;
469 }
Robert Wud559ba52023-06-29 00:08:51 +0000470 const aaudio_stream_state_t originalState = getState();
Phil Burkec89b2e2017-06-20 15:05:06 -0700471 setState(AAUDIO_STREAM_STATE_STARTING);
Phil Burkbcc36742017-08-31 17:24:51 -0700472
473 // Clear any stale timestamps from the previous run.
474 drainTimestampsFromService();
475
Phil Burkec8ca522020-05-19 10:05:58 -0700476 prepareBuffersForStart(); // tell subclasses to get ready
477
jiabin5f787812023-03-02 20:42:43 +0000478 aaudio_result_t result = mServiceInterface.startStream(mServiceStreamHandleInfo);
jiabinf7f06152021-11-22 18:10:14 +0000479 if (result == AAUDIO_ERROR_STANDBY) {
480 // The stream is at standby mode. Need to exit standby before starting the stream.
481 result = exitStandby_l();
482 if (result == AAUDIO_OK) {
jiabin5f787812023-03-02 20:42:43 +0000483 result = mServiceInterface.startStream(mServiceStreamHandleInfo);
jiabinf7f06152021-11-22 18:10:14 +0000484 }
485 }
486 if (result != AAUDIO_OK) {
487 ALOGD("%s() error = %d, stream was probably stolen", __func__, result);
Phil Burk6e463ce2020-04-13 10:20:20 -0700488 // Stealing was added in R. Coerce result to improve backward compatibility.
489 result = AAUDIO_ERROR_DISCONNECTED;
jiabincb212cd2022-08-24 16:50:44 -0700490 setDisconnected();
Phil Burk6e463ce2020-04-13 10:20:20 -0700491 }
Phil Burkc0c70e32017-02-09 13:18:38 -0800492
Phil Burk3316d5e2017-02-15 11:23:01 -0800493 startTime = AudioClock::getNanoseconds();
Phil Burk204a1632017-01-03 17:23:43 -0800494 mClockModel.start(startTime);
Phil Burkbcc36742017-08-31 17:24:51 -0700495 mNeedCatchUp.request(); // Ask data processing code to catch up when first timestamp received.
Phil Burke4d7bb42017-03-28 11:32:39 -0700496
Phil Burk965650e2017-09-07 21:00:09 -0700497 // Start data callback thread.
Phil Burk134f1972017-12-08 13:06:11 -0800498 if (result == AAUDIO_OK && isDataCallbackSet()) {
Phil Burke4d7bb42017-03-28 11:32:39 -0700499 // Launch the callback loop thread.
500 int64_t periodNanos = mCallbackFrames
501 * AAUDIO_NANOS_PER_SECOND
502 / getSampleRate();
503 mCallbackEnabled.store(true);
Phil Burkdd582922020-10-15 20:29:51 +0000504 result = createThread_l(periodNanos, aaudio_callback_thread_proc, this);
Phil Burke4d7bb42017-03-28 11:32:39 -0700505 }
Phil Burkec89b2e2017-06-20 15:05:06 -0700506 if (result != AAUDIO_OK) {
507 setState(originalState);
508 }
Phil Burke4d7bb42017-03-28 11:32:39 -0700509 return result;
Phil Burk204a1632017-01-03 17:23:43 -0800510}
511
Phil Burke4d7bb42017-03-28 11:32:39 -0700512int64_t AudioStreamInternal::calculateReasonableTimeout(int32_t framesPerOperation) {
513
514 // Wait for at least a second or some number of callbacks to join the thread.
Phil Burk71f35bb2017-04-13 16:05:07 -0700515 int64_t timeoutNanoseconds = (MIN_TIMEOUT_OPERATIONS
516 * framesPerOperation
517 * AAUDIO_NANOS_PER_SECOND)
518 / getSampleRate();
Phil Burke4d7bb42017-03-28 11:32:39 -0700519 if (timeoutNanoseconds < MIN_TIMEOUT_NANOS) { // arbitrary number of seconds
520 timeoutNanoseconds = MIN_TIMEOUT_NANOS;
521 }
522 return timeoutNanoseconds;
523}
524
Phil Burk87c9f642017-05-17 07:22:39 -0700525int64_t AudioStreamInternal::calculateReasonableTimeout() {
526 return calculateReasonableTimeout(getFramesPerBurst());
527}
528
Phil Burk13d3d832019-06-10 14:36:48 -0700529// This must be called under mStreamLock.
Phil Burkdd582922020-10-15 20:29:51 +0000530aaudio_result_t AudioStreamInternal::stopCallback_l()
Phil Burke4d7bb42017-03-28 11:32:39 -0700531{
jiabincb212cd2022-08-24 16:50:44 -0700532 if (isDataCallbackSet() && (isActive() || isDisconnected())) {
Phil Burke4d7bb42017-03-28 11:32:39 -0700533 mCallbackEnabled.store(false);
jiabind5bd06a2021-04-27 22:04:08 +0000534 aaudio_result_t result = joinThread_l(nullptr); // may temporarily unlock mStreamLock
Phil Burk6e463ce2020-04-13 10:20:20 -0700535 if (result == AAUDIO_ERROR_INVALID_HANDLE) {
536 ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
537 result = AAUDIO_OK;
538 }
539 return result;
Phil Burke4d7bb42017-03-28 11:32:39 -0700540 } else {
Phil Burkdd582922020-10-15 20:29:51 +0000541 ALOGD("%s() skipped, isDataCallbackSet() = %d, isActive() = %d, getState() = %d", __func__,
542 isDataCallbackSet(), isActive(), getState());
Phil Burke4d7bb42017-03-28 11:32:39 -0700543 return AAUDIO_OK;
544 }
545}
546
Phil Burkdd582922020-10-15 20:29:51 +0000547aaudio_result_t AudioStreamInternal::requestStop_l() {
548 aaudio_result_t result = stopCallback_l();
Phil Burk5cc83c32017-11-28 15:43:18 -0800549 if (result != AAUDIO_OK) {
Phil Burkdd582922020-10-15 20:29:51 +0000550 ALOGW("%s() stop callback returned %d, returning early", __func__, result);
Phil Burk5cc83c32017-11-28 15:43:18 -0800551 return result;
552 }
Phil Burk13d3d832019-06-10 14:36:48 -0700553 // The stream may have been unlocked temporarily to let a callback finish
554 // and the callback may have stopped the stream.
555 // Check to make sure the stream still needs to be stopped.
Phil Burk0bd745e2020-10-17 18:20:01 +0000556 // See also AudioStream::safeStop_l().
jiabincb212cd2022-08-24 16:50:44 -0700557 if (!(isActive() || isDisconnected())) {
Phil Burkdd582922020-10-15 20:29:51 +0000558 ALOGD("%s() returning early, not active or disconnected", __func__);
Phil Burk13d3d832019-06-10 14:36:48 -0700559 return AAUDIO_OK;
560 }
Phil Burk5cc83c32017-11-28 15:43:18 -0800561
jiabin5f787812023-03-02 20:42:43 +0000562 if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700563 ALOGW("%s() mServiceStreamHandle invalid = 0x%08X",
jiabin5f787812023-03-02 20:42:43 +0000564 __func__, getServiceHandle());
Phil Burk71f35bb2017-04-13 16:05:07 -0700565 return AAUDIO_ERROR_INVALID_STATE;
566 }
567
568 mClockModel.stop(AudioClock::getNanoseconds());
569 setState(AAUDIO_STREAM_STATE_STOPPING);
Phil Burka53ffa62018-10-10 16:21:37 -0700570 mAtomicInternalTimestamp.clear();
Phil Burk965650e2017-09-07 21:00:09 -0700571
jiabin5f787812023-03-02 20:42:43 +0000572 result = mServiceInterface.stopStream(mServiceStreamHandleInfo);
Phil Burk6e463ce2020-04-13 10:20:20 -0700573 if (result == AAUDIO_ERROR_INVALID_HANDLE) {
574 ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
575 result = AAUDIO_OK;
576 }
577 return result;
Phil Burk71f35bb2017-04-13 16:05:07 -0700578}
579
Phil Burk5ed503c2017-02-01 09:38:15 -0800580aaudio_result_t AudioStreamInternal::registerThread() {
jiabin5f787812023-03-02 20:42:43 +0000581 if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700582 ALOGW("%s() mServiceStreamHandle invalid", __func__);
Phil Burk5ed503c2017-02-01 09:38:15 -0800583 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800584 }
jiabin5f787812023-03-02 20:42:43 +0000585 return mServiceInterface.registerAudioThread(mServiceStreamHandleInfo,
586 gettid(),
587 getPeriodNanoseconds());
Phil Burk204a1632017-01-03 17:23:43 -0800588}
589
Phil Burk5ed503c2017-02-01 09:38:15 -0800590aaudio_result_t AudioStreamInternal::unregisterThread() {
jiabin5f787812023-03-02 20:42:43 +0000591 if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700592 ALOGW("%s() mServiceStreamHandle invalid", __func__);
Phil Burk5ed503c2017-02-01 09:38:15 -0800593 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800594 }
jiabin5f787812023-03-02 20:42:43 +0000595 return mServiceInterface.unregisterAudioThread(mServiceStreamHandleInfo, gettid());
Phil Burk204a1632017-01-03 17:23:43 -0800596}
597
Eric Laurentcb4dae22017-07-01 19:39:32 -0700598aaudio_result_t AudioStreamInternal::startClient(const android::AudioClient& client,
jiabind1f1cb62020-03-24 11:57:57 -0700599 const audio_attributes_t *attr,
Phil Burkbbd52862018-04-13 11:37:42 -0700600 audio_port_handle_t *portHandle) {
601 ALOGV("%s() called", __func__);
jiabin5f787812023-03-02 20:42:43 +0000602 if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
Eric Laurentcb4dae22017-07-01 19:39:32 -0700603 return AAUDIO_ERROR_INVALID_STATE;
604 }
jiabin5f787812023-03-02 20:42:43 +0000605 aaudio_result_t result = mServiceInterface.startClient(mServiceStreamHandleInfo,
jiabind1f1cb62020-03-24 11:57:57 -0700606 client, attr, portHandle);
Phil Burkbbd52862018-04-13 11:37:42 -0700607 ALOGV("%s(%d) returning %d", __func__, *portHandle, result);
608 return result;
Eric Laurentcb4dae22017-07-01 19:39:32 -0700609}
610
Phil Burkbbd52862018-04-13 11:37:42 -0700611aaudio_result_t AudioStreamInternal::stopClient(audio_port_handle_t portHandle) {
612 ALOGV("%s(%d) called", __func__, portHandle);
jiabin5f787812023-03-02 20:42:43 +0000613 if (getServiceHandle() == AAUDIO_HANDLE_INVALID) {
Eric Laurentcb4dae22017-07-01 19:39:32 -0700614 return AAUDIO_ERROR_INVALID_STATE;
615 }
jiabin5f787812023-03-02 20:42:43 +0000616 aaudio_result_t result = mServiceInterface.stopClient(mServiceStreamHandleInfo, portHandle);
Phil Burkbbd52862018-04-13 11:37:42 -0700617 ALOGV("%s(%d) returning %d", __func__, portHandle, result);
618 return result;
Eric Laurentcb4dae22017-07-01 19:39:32 -0700619}
620
jiabind5bd06a2021-04-27 22:04:08 +0000621aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t /*clockId*/,
Phil Burk3316d5e2017-02-15 11:23:01 -0800622 int64_t *framePosition,
623 int64_t *timeNanoseconds) {
Phil Burk97350f92017-07-21 15:59:44 -0700624 // Generated in server and passed to client. Return latest.
Phil Burka53ffa62018-10-10 16:21:37 -0700625 if (mAtomicInternalTimestamp.isValid()) {
626 Timestamp timestamp = mAtomicInternalTimestamp.read();
Robert Wud559ba52023-06-29 00:08:51 +0000627 // This should not overflow as timestamp.getPosition() should be a position in a buffer and
628 // not the actual timestamp. timestamp.getNanoseconds() below uses the actual timestamp.
629 // At 48000 Hz we can run for over 100 years before overflowing the int64_t.
630 int64_t position = (timestamp.getPosition() + mFramesOffsetFromService) * getSampleRate() /
631 getDeviceSampleRate();
Phil Burkbcc36742017-08-31 17:24:51 -0700632 if (position >= 0) {
633 *framePosition = position;
634 *timeNanoseconds = timestamp.getNanoseconds();
635 return AAUDIO_OK;
636 }
Phil Burk97350f92017-07-21 15:59:44 -0700637 }
Phil Burkc75d97f2017-09-08 15:48:36 -0700638 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800639}
640
Phil Burkec89b2e2017-06-20 15:05:06 -0700641void AudioStreamInternal::logTimestamp(AAudioServiceMessage &command) {
Phil Burk204a1632017-01-03 17:23:43 -0800642 static int64_t oldPosition = 0;
Phil Burk3316d5e2017-02-15 11:23:01 -0800643 static int64_t oldTime = 0;
Phil Burk204a1632017-01-03 17:23:43 -0800644 int64_t framePosition = command.timestamp.position;
Phil Burk3316d5e2017-02-15 11:23:01 -0800645 int64_t nanoTime = command.timestamp.timestamp;
Phil Burkbcc36742017-08-31 17:24:51 -0700646 ALOGD("logTimestamp: timestamp says framePosition = %8lld at nanoTime %lld",
Phil Burk204a1632017-01-03 17:23:43 -0800647 (long long) framePosition,
648 (long long) nanoTime);
649 int64_t nanosDelta = nanoTime - oldTime;
650 if (nanosDelta > 0 && oldTime > 0) {
651 int64_t framesDelta = framePosition - oldPosition;
Phil Burk5ed503c2017-02-01 09:38:15 -0800652 int64_t rate = (framesDelta * AAUDIO_NANOS_PER_SECOND) / nanosDelta;
Phil Burkbcc36742017-08-31 17:24:51 -0700653 ALOGD("logTimestamp: framesDelta = %8lld, nanosDelta = %8lld, rate = %lld",
Phil Burkec89b2e2017-06-20 15:05:06 -0700654 (long long) framesDelta, (long long) nanosDelta, (long long) rate);
Phil Burk204a1632017-01-03 17:23:43 -0800655 }
656 oldPosition = framePosition;
657 oldTime = nanoTime;
658}
Phil Burk204a1632017-01-03 17:23:43 -0800659
Phil Burk97350f92017-07-21 15:59:44 -0700660aaudio_result_t AudioStreamInternal::onTimestampService(AAudioServiceMessage *message) {
Phil Burk204a1632017-01-03 17:23:43 -0800661#if LOG_TIMESTAMPS
Phil Burkec89b2e2017-06-20 15:05:06 -0700662 logTimestamp(*message);
Phil Burk204a1632017-01-03 17:23:43 -0800663#endif
Phil Burkb31b66f2019-09-30 09:33:41 -0700664 processTimestamp(message->timestamp.position,
665 message->timestamp.timestamp + mTimeOffsetNanos);
Phil Burk5ed503c2017-02-01 09:38:15 -0800666 return AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800667}
668
Phil Burk97350f92017-07-21 15:59:44 -0700669aaudio_result_t AudioStreamInternal::onTimestampHardware(AAudioServiceMessage *message) {
670 Timestamp timestamp(message->timestamp.position, message->timestamp.timestamp);
Phil Burka53ffa62018-10-10 16:21:37 -0700671 mAtomicInternalTimestamp.write(timestamp);
Phil Burk97350f92017-07-21 15:59:44 -0700672 return AAUDIO_OK;
673}
674
Phil Burk5ed503c2017-02-01 09:38:15 -0800675aaudio_result_t AudioStreamInternal::onEventFromServer(AAudioServiceMessage *message) {
676 aaudio_result_t result = AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800677 switch (message->event.event) {
Phil Burk5ed503c2017-02-01 09:38:15 -0800678 case AAUDIO_SERVICE_EVENT_STARTED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700679 ALOGD("%s - got AAUDIO_SERVICE_EVENT_STARTED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700680 if (getState() == AAUDIO_STREAM_STATE_STARTING) {
681 setState(AAUDIO_STREAM_STATE_STARTED);
682 }
Vlad Popaec1788e2022-08-04 11:23:30 +0200683 mPlayerBase->triggerPortIdUpdate(static_cast<audio_port_handle_t>(
684 message->event.dataLong));
Phil Burk204a1632017-01-03 17:23:43 -0800685 break;
Phil Burk5ed503c2017-02-01 09:38:15 -0800686 case AAUDIO_SERVICE_EVENT_PAUSED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700687 ALOGD("%s - got AAUDIO_SERVICE_EVENT_PAUSED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700688 if (getState() == AAUDIO_STREAM_STATE_PAUSING) {
689 setState(AAUDIO_STREAM_STATE_PAUSED);
690 }
Phil Burk204a1632017-01-03 17:23:43 -0800691 break;
Phil Burk71f35bb2017-04-13 16:05:07 -0700692 case AAUDIO_SERVICE_EVENT_STOPPED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700693 ALOGD("%s - got AAUDIO_SERVICE_EVENT_STOPPED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700694 if (getState() == AAUDIO_STREAM_STATE_STOPPING) {
695 setState(AAUDIO_STREAM_STATE_STOPPED);
696 }
Phil Burk71f35bb2017-04-13 16:05:07 -0700697 break;
Phil Burk5ed503c2017-02-01 09:38:15 -0800698 case AAUDIO_SERVICE_EVENT_FLUSHED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700699 ALOGD("%s - got AAUDIO_SERVICE_EVENT_FLUSHED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700700 if (getState() == AAUDIO_STREAM_STATE_FLUSHING) {
701 setState(AAUDIO_STREAM_STATE_FLUSHED);
702 onFlushFromServer();
703 }
Phil Burk204a1632017-01-03 17:23:43 -0800704 break;
Phil Burk5ed503c2017-02-01 09:38:15 -0800705 case AAUDIO_SERVICE_EVENT_DISCONNECTED:
Phil Burkea04d972017-08-07 12:30:44 -0700706 // Prevent hardware from looping on old data and making buzzing sounds.
707 if (getDirection() == AAUDIO_DIRECTION_OUTPUT) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700708 mAudioEndpoint->eraseDataMemory();
Phil Burkea04d972017-08-07 12:30:44 -0700709 }
Phil Burk5ed503c2017-02-01 09:38:15 -0800710 result = AAUDIO_ERROR_DISCONNECTED;
jiabincb212cd2022-08-24 16:50:44 -0700711 setDisconnected();
Phil Burkfbf031e2017-10-12 15:58:31 -0700712 ALOGW("%s - AAUDIO_SERVICE_EVENT_DISCONNECTED - FIFO cleared", __func__);
Phil Burk204a1632017-01-03 17:23:43 -0800713 break;
Phil Burkc0c70e32017-02-09 13:18:38 -0800714 case AAUDIO_SERVICE_EVENT_VOLUME:
Phil Burk55e5eab2018-04-10 15:16:38 -0700715 ALOGD("%s - AAUDIO_SERVICE_EVENT_VOLUME %lf", __func__, message->event.dataDouble);
Eric Laurenta2f296e2017-06-21 18:51:47 -0700716 mStreamVolume = (float)message->event.dataDouble;
717 doSetVolume();
Phil Burkc0c70e32017-02-09 13:18:38 -0800718 break;
Phil Burk23296382017-11-20 15:45:11 -0800719 case AAUDIO_SERVICE_EVENT_XRUN:
720 mXRunCount = static_cast<int32_t>(message->event.dataLong);
721 break;
Phil Burk204a1632017-01-03 17:23:43 -0800722 default:
Phil Burkfbf031e2017-10-12 15:58:31 -0700723 ALOGE("%s - Unrecognized event = %d", __func__, (int) message->event.event);
Phil Burk204a1632017-01-03 17:23:43 -0800724 break;
725 }
726 return result;
727}
728
Phil Burkbcc36742017-08-31 17:24:51 -0700729aaudio_result_t AudioStreamInternal::drainTimestampsFromService() {
730 aaudio_result_t result = AAUDIO_OK;
731
732 while (result == AAUDIO_OK) {
733 AAudioServiceMessage message;
Phil Burk5edc4ea2020-04-17 08:15:42 -0700734 if (!mAudioEndpoint) {
735 break;
736 }
737 if (mAudioEndpoint->readUpCommand(&message) != 1) {
Phil Burkbcc36742017-08-31 17:24:51 -0700738 break; // no command this time, no problem
739 }
740 switch (message.what) {
741 // ignore most messages
742 case AAudioServiceMessage::code::TIMESTAMP_SERVICE:
743 case AAudioServiceMessage::code::TIMESTAMP_HARDWARE:
744 break;
745
746 case AAudioServiceMessage::code::EVENT:
747 result = onEventFromServer(&message);
748 break;
749
750 default:
Phil Burkfbf031e2017-10-12 15:58:31 -0700751 ALOGE("%s - unrecognized message.what = %d", __func__, (int) message.what);
Phil Burkbcc36742017-08-31 17:24:51 -0700752 result = AAUDIO_ERROR_INTERNAL;
753 break;
754 }
755 }
756 return result;
757}
758
Phil Burk204a1632017-01-03 17:23:43 -0800759// Process all the commands coming from the server.
Phil Burk5ed503c2017-02-01 09:38:15 -0800760aaudio_result_t AudioStreamInternal::processCommands() {
761 aaudio_result_t result = AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800762
Phil Burk5ed503c2017-02-01 09:38:15 -0800763 while (result == AAUDIO_OK) {
764 AAudioServiceMessage message;
Phil Burk5edc4ea2020-04-17 08:15:42 -0700765 if (!mAudioEndpoint) {
766 break;
767 }
768 if (mAudioEndpoint->readUpCommand(&message) != 1) {
Phil Burk204a1632017-01-03 17:23:43 -0800769 break; // no command this time, no problem
770 }
771 switch (message.what) {
Phil Burk97350f92017-07-21 15:59:44 -0700772 case AAudioServiceMessage::code::TIMESTAMP_SERVICE:
773 result = onTimestampService(&message);
774 break;
775
776 case AAudioServiceMessage::code::TIMESTAMP_HARDWARE:
777 result = onTimestampHardware(&message);
Phil Burk204a1632017-01-03 17:23:43 -0800778 break;
779
Phil Burk5ed503c2017-02-01 09:38:15 -0800780 case AAudioServiceMessage::code::EVENT:
Phil Burk204a1632017-01-03 17:23:43 -0800781 result = onEventFromServer(&message);
782 break;
783
784 default:
Phil Burkfbf031e2017-10-12 15:58:31 -0700785 ALOGE("%s - unrecognized message.what = %d", __func__, (int) message.what);
Phil Burk17fff382017-05-16 14:06:45 -0700786 result = AAUDIO_ERROR_INTERNAL;
Phil Burk204a1632017-01-03 17:23:43 -0800787 break;
788 }
789 }
790 return result;
791}
792
Phil Burk87c9f642017-05-17 07:22:39 -0700793// Read or write the data, block if needed and timeoutMillis > 0
794aaudio_result_t AudioStreamInternal::processData(void *buffer, int32_t numFrames,
795 int64_t timeoutNanoseconds)
Phil Burk204a1632017-01-03 17:23:43 -0800796{
jiabin5f787812023-03-02 20:42:43 +0000797 if (isDisconnected()) {
798 return AAUDIO_ERROR_DISCONNECTED;
799 }
800 if (!mInService &&
801 AAudioBinderClient::getInstance().getServiceLifetimeId() != getServiceLifetimeId()) {
802 // The service lifetime id will be changed whenever the binder died. In that case, if
803 // the service lifetime id from AAudioBinderClient is different from the cached one,
804 // returns AAUDIO_ERROR_DISCONNECTED.
805 // Note that only compare the service lifetime id if it is not in service as the streams
806 // in service will all be gone when aaudio service dies.
807 mClockModel.stop(AudioClock::getNanoseconds());
808 // Set the stream as disconnected as the service lifetime id will only change when
809 // the binder dies.
810 setDisconnected();
811 return AAUDIO_ERROR_DISCONNECTED;
812 }
Phil Burkfd34a932017-07-19 07:03:52 -0700813 const char * traceName = "aaProc";
814 const char * fifoName = "aaRdy";
Phil Burk4485d412017-05-09 15:55:02 -0700815 ATRACE_BEGIN(traceName);
Phil Burk4485d412017-05-09 15:55:02 -0700816 if (ATRACE_ENABLED()) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700817 int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
Phil Burkfd34a932017-07-19 07:03:52 -0700818 ATRACE_INT(fifoName, fullFrames);
Phil Burk4485d412017-05-09 15:55:02 -0700819 }
820
Phil Burkec89b2e2017-06-20 15:05:06 -0700821 aaudio_result_t result = AAUDIO_OK;
822 int32_t loopCount = 0;
823 uint8_t* audioData = (uint8_t*)buffer;
824 int64_t currentTimeNanos = AudioClock::getNanoseconds();
825 const int64_t entryTimeNanos = currentTimeNanos;
826 const int64_t deadlineNanos = currentTimeNanos + timeoutNanoseconds;
827 int32_t framesLeft = numFrames;
828
Phil Burk87c9f642017-05-17 07:22:39 -0700829 // Loop until all the data has been processed or until a timeout occurs.
Phil Burk204a1632017-01-03 17:23:43 -0800830 while (framesLeft > 0) {
Phil Burkec89b2e2017-06-20 15:05:06 -0700831 // The call to processDataNow() will not block. It will just process as much as it can.
Phil Burk3316d5e2017-02-15 11:23:01 -0800832 int64_t wakeTimeNanos = 0;
Phil Burk87c9f642017-05-17 07:22:39 -0700833 aaudio_result_t framesProcessed = processDataNow(audioData, framesLeft,
834 currentTimeNanos, &wakeTimeNanos);
835 if (framesProcessed < 0) {
Phil Burk87c9f642017-05-17 07:22:39 -0700836 result = framesProcessed;
Phil Burk204a1632017-01-03 17:23:43 -0800837 break;
838 }
Phil Burk87c9f642017-05-17 07:22:39 -0700839 framesLeft -= (int32_t) framesProcessed;
840 audioData += framesProcessed * getBytesPerFrame();
Phil Burk204a1632017-01-03 17:23:43 -0800841
842 // Should we block?
843 if (timeoutNanoseconds == 0) {
844 break; // don't block
Phil Burk8d4f0062019-10-03 15:55:41 -0700845 } else if (wakeTimeNanos != 0) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700846 if (!mAudioEndpoint->isFreeRunning()) {
Phil Burkfd34a932017-07-19 07:03:52 -0700847 // If there is software on the other end of the FIFO then it may get delayed.
848 // So wake up just a little after we expect it to be ready.
849 wakeTimeNanos += mWakeupDelayNanos;
Phil Burk204a1632017-01-03 17:23:43 -0800850 }
Phil Burkfd34a932017-07-19 07:03:52 -0700851
Phil Burk2bc7c182017-08-28 11:45:01 -0700852 currentTimeNanos = AudioClock::getNanoseconds();
853 int64_t earliestWakeTime = currentTimeNanos + mMinimumSleepNanos;
854 // Guarantee a minimum sleep time.
855 if (wakeTimeNanos < earliestWakeTime) {
856 wakeTimeNanos = earliestWakeTime;
857 }
858
Phil Burk204a1632017-01-03 17:23:43 -0800859 if (wakeTimeNanos > deadlineNanos) {
860 // If we time out, just return the framesWritten so far.
Phil Burkfbf031e2017-10-12 15:58:31 -0700861 ALOGW("processData(): entered at %lld nanos, currently %lld",
Phil Burkec89b2e2017-06-20 15:05:06 -0700862 (long long) entryTimeNanos, (long long) currentTimeNanos);
Phil Burkfbf031e2017-10-12 15:58:31 -0700863 ALOGW("processData(): TIMEOUT after %lld nanos",
Phil Burkc0c70e32017-02-09 13:18:38 -0800864 (long long) timeoutNanoseconds);
Phil Burkfbf031e2017-10-12 15:58:31 -0700865 ALOGW("processData(): wakeTime = %lld, deadline = %lld nanos",
Phil Burk87c9f642017-05-17 07:22:39 -0700866 (long long) wakeTimeNanos, (long long) deadlineNanos);
Phil Burkfbf031e2017-10-12 15:58:31 -0700867 ALOGW("processData(): past deadline by %d micros",
Phil Burk87c9f642017-05-17 07:22:39 -0700868 (int)((wakeTimeNanos - deadlineNanos) / AAUDIO_NANOS_PER_MICROSECOND));
Phil Burkec89b2e2017-06-20 15:05:06 -0700869 mClockModel.dump();
Phil Burk5edc4ea2020-04-17 08:15:42 -0700870 mAudioEndpoint->dump();
Phil Burk204a1632017-01-03 17:23:43 -0800871 break;
872 }
873
Phil Burkfd34a932017-07-19 07:03:52 -0700874 if (ATRACE_ENABLED()) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700875 int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
Phil Burkfd34a932017-07-19 07:03:52 -0700876 ATRACE_INT(fifoName, fullFrames);
877 int64_t sleepForNanos = wakeTimeNanos - currentTimeNanos;
878 ATRACE_INT("aaSlpNs", (int32_t)sleepForNanos);
879 }
880
881 AudioClock::sleepUntilNanoTime(wakeTimeNanos);
Phil Burk204a1632017-01-03 17:23:43 -0800882 currentTimeNanos = AudioClock::getNanoseconds();
883 }
884 }
885
Phil Burkfd34a932017-07-19 07:03:52 -0700886 if (ATRACE_ENABLED()) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700887 int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
Phil Burkfd34a932017-07-19 07:03:52 -0700888 ATRACE_INT(fifoName, fullFrames);
889 }
890
Phil Burk87c9f642017-05-17 07:22:39 -0700891 // return error or framesProcessed
Phil Burkc0c70e32017-02-09 13:18:38 -0800892 (void) loopCount;
Phil Burk4485d412017-05-09 15:55:02 -0700893 ATRACE_END();
Phil Burk204a1632017-01-03 17:23:43 -0800894 return (result < 0) ? result : numFrames - framesLeft;
895}
896
Phil Burk3316d5e2017-02-15 11:23:01 -0800897void AudioStreamInternal::processTimestamp(uint64_t position, int64_t time) {
Phil Burk87c9f642017-05-17 07:22:39 -0700898 mClockModel.processTimestamp(position, time);
Phil Burk204a1632017-01-03 17:23:43 -0800899}
900
Phil Burk3316d5e2017-02-15 11:23:01 -0800901aaudio_result_t AudioStreamInternal::setBufferSize(int32_t requestedFrames) {
Phil Burk8d97b8e2020-09-25 23:18:14 +0000902 const int32_t maximumSize = getBufferCapacity() - getFramesPerBurst();
Robert Wu32d319b2023-11-09 22:40:52 +0000903 int32_t adjustedFrames = std::min(requestedFrames, maximumSize);
904 // Buffer sizes should always be a multiple of framesPerBurst.
905 int32_t numBursts = (static_cast<int64_t>(adjustedFrames) + getFramesPerBurst() - 1) /
906 getFramesPerBurst();
Phil Burk6479d502017-11-20 09:32:52 -0800907
Robert Wu32d319b2023-11-09 22:40:52 +0000908 // Use at least one burst
909 if (numBursts == 0) {
910 numBursts = 1;
Phil Burk6479d502017-11-20 09:32:52 -0800911 }
912
Phil Burk5edc4ea2020-04-17 08:15:42 -0700913 if (mAudioEndpoint) {
914 // Clip against the actual size from the endpoint.
Robert Wud559ba52023-06-29 00:08:51 +0000915 int32_t actualFramesDevice = 0;
Robert Wu32d319b2023-11-09 22:40:52 +0000916 int32_t maximumFramesDevice = getDeviceBufferCapacity() - getDeviceFramesPerBurst();
Phil Burk5edc4ea2020-04-17 08:15:42 -0700917 // Set to maximum size so we can write extra data when ready in order to reduce glitches.
918 // The amount we keep in the buffer is controlled by mBufferSizeInFrames.
Robert Wud559ba52023-06-29 00:08:51 +0000919 mAudioEndpoint->setBufferSizeInFrames(maximumFramesDevice, &actualFramesDevice);
Robert Wu32d319b2023-11-09 22:40:52 +0000920 int32_t actualNumBursts = actualFramesDevice / getDeviceFramesPerBurst();
921 numBursts = std::min(numBursts, actualNumBursts);
Phil Burk5edc4ea2020-04-17 08:15:42 -0700922 }
Phil Burk8d4f0062019-10-03 15:55:41 -0700923
Robert Wu32d319b2023-11-09 22:40:52 +0000924 const int32_t bufferSizeInFrames = numBursts * getFramesPerBurst();
925 const int32_t deviceBufferSizeInFrames = numBursts * getDeviceFramesPerBurst();
Robert Wud559ba52023-06-29 00:08:51 +0000926
927 if (deviceBufferSizeInFrames != mDeviceBufferSizeInFrames) {
Phil Burk64e16a72020-06-01 13:25:51 -0700928 android::mediametrics::LogItem(mMetricsId)
929 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
Robert Wud559ba52023-06-29 00:08:51 +0000930 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, deviceBufferSizeInFrames)
Phil Burk64e16a72020-06-01 13:25:51 -0700931 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getXRunCount())
932 .record();
933 }
934
Robert Wud559ba52023-06-29 00:08:51 +0000935 mBufferSizeInFrames = bufferSizeInFrames;
936 mDeviceBufferSizeInFrames = deviceBufferSizeInFrames;
Phil Burk6c63ae32019-10-28 10:28:21 -0700937 ALOGV("%s(%d) returns %d", __func__, requestedFrames, adjustedFrames);
Phil Burk8d4f0062019-10-03 15:55:41 -0700938 return (aaudio_result_t) adjustedFrames;
Phil Burk204a1632017-01-03 17:23:43 -0800939}
940
Phil Burk87c9f642017-05-17 07:22:39 -0700941int32_t AudioStreamInternal::getBufferSize() const {
Phil Burk8d4f0062019-10-03 15:55:41 -0700942 return mBufferSizeInFrames;
Phil Burk204a1632017-01-03 17:23:43 -0800943}
944
Robert Wud559ba52023-06-29 00:08:51 +0000945int32_t AudioStreamInternal::getDeviceBufferSize() const {
946 return mDeviceBufferSizeInFrames;
947}
948
Phil Burk87c9f642017-05-17 07:22:39 -0700949int32_t AudioStreamInternal::getBufferCapacity() const {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700950 return mBufferCapacityInFrames;
Phil Burk204a1632017-01-03 17:23:43 -0800951}
952
Robert Wud559ba52023-06-29 00:08:51 +0000953int32_t AudioStreamInternal::getDeviceBufferCapacity() const {
954 return mDeviceBufferCapacityInFrames;
955}
956
Phil Burk377c1c22018-12-12 16:06:54 -0800957bool AudioStreamInternal::isClockModelInControl() const {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700958 return isActive() && mAudioEndpoint->isFreeRunning() && mClockModel.isRunning();
Phil Burk377c1c22018-12-12 16:06:54 -0800959}