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Phil Burk204a1632017-01-03 17:23:43 -08001/*
2 * Copyright (C) 2016 The Android Open Source Project
3 *
4 * Licensed under the Apache License, Version 2.0 (the "License");
5 * you may not use this file except in compliance with the License.
6 * You may obtain a copy of the License at
7 *
8 * http://www.apache.org/licenses/LICENSE-2.0
9 *
10 * Unless required by applicable law or agreed to in writing, software
11 * distributed under the License is distributed on an "AS IS" BASIS,
12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
13 * See the License for the specific language governing permissions and
14 * limitations under the License.
15 */
16
Phil Burka9876702020-04-20 18:16:15 -070017#define LOG_TAG "AudioStreamInternal"
Phil Burk204a1632017-01-03 17:23:43 -080018//#define LOG_NDEBUG 0
19#include <utils/Log.h>
20
Phil Burk4485d412017-05-09 15:55:02 -070021#define ATRACE_TAG ATRACE_TAG_AUDIO
22
Phil Burkc0c70e32017-02-09 13:18:38 -080023#include <stdint.h>
Phil Burk204a1632017-01-03 17:23:43 -080024
25#include <binder/IServiceManager.h>
26
Phil Burk5ed503c2017-02-01 09:38:15 -080027#include <aaudio/AAudio.h>
Phil Burkfd34a932017-07-19 07:03:52 -070028#include <cutils/properties.h>
Phil Burka9876702020-04-20 18:16:15 -070029
Robert Wud7400832021-12-04 01:11:19 +000030#include <media/AudioParameter.h>
jiabine504e7b2021-09-18 00:27:08 +000031#include <media/AudioSystem.h>
Phil Burka9876702020-04-20 18:16:15 -070032#include <media/MediaMetricsItem.h>
Phil Burk4485d412017-05-09 15:55:02 -070033#include <utils/Trace.h>
Phil Burk204a1632017-01-03 17:23:43 -080034
Phil Burkc0c70e32017-02-09 13:18:38 -080035#include "AudioEndpointParcelable.h"
36#include "binding/AAudioStreamRequest.h"
37#include "binding/AAudioStreamConfiguration.h"
Phil Burk5ed503c2017-02-01 09:38:15 -080038#include "binding/AAudioServiceMessage.h"
Phil Burka9876702020-04-20 18:16:15 -070039#include "core/AudioGlobal.h"
Phil Burk3df348f2017-02-08 11:41:55 -080040#include "core/AudioStreamBuilder.h"
Phil Burke572f462017-04-20 13:03:19 -070041#include "fifo/FifoBuffer.h"
Phil Burkfd34a932017-07-19 07:03:52 -070042#include "utility/AudioClock.h"
Philip P. Moltmannbda45752020-07-17 16:41:18 -070043#include <media/AidlConversion.h>
Phil Burke572f462017-04-20 13:03:19 -070044
Phil Burkc0c70e32017-02-09 13:18:38 -080045#include "AudioStreamInternal.h"
Phil Burk204a1632017-01-03 17:23:43 -080046
Phil Burka9876702020-04-20 18:16:15 -070047// We do this after the #includes because if a header uses ALOG.
48// it would fail on the reference to mInService.
49#undef LOG_TAG
50// This file is used in both client and server processes.
51// This is needed to make sense of the logs more easily.
52#define LOG_TAG (mInService ? "AudioStreamInternal_Service" : "AudioStreamInternal_Client")
53
Svet Ganov3e5f14f2021-05-13 22:51:08 +000054using android::content::AttributionSourceState;
Phil Burk204a1632017-01-03 17:23:43 -080055
Phil Burk5ed503c2017-02-01 09:38:15 -080056using namespace aaudio;
Phil Burk204a1632017-01-03 17:23:43 -080057
Phil Burke4d7bb42017-03-28 11:32:39 -070058#define MIN_TIMEOUT_NANOS (1000 * AAUDIO_NANOS_PER_MILLISECOND)
59
60// Wait at least this many times longer than the operation should take.
61#define MIN_TIMEOUT_OPERATIONS 4
62
Phil Burkbcc36742017-08-31 17:24:51 -070063#define LOG_TIMESTAMPS 0
Phil Burk87c9f642017-05-17 07:22:39 -070064
Phil Burkc0c70e32017-02-09 13:18:38 -080065AudioStreamInternal::AudioStreamInternal(AAudioServiceInterface &serviceInterface, bool inService)
Phil Burk204a1632017-01-03 17:23:43 -080066 : AudioStream()
67 , mClockModel()
Phil Burk5ed503c2017-02-01 09:38:15 -080068 , mServiceStreamHandle(AAUDIO_HANDLE_INVALID)
Phil Burkec89b2e2017-06-20 15:05:06 -070069 , mInService(inService)
Phil Burkfd34a932017-07-19 07:03:52 -070070 , mServiceInterface(serviceInterface)
Phil Burka53ffa62018-10-10 16:21:37 -070071 , mAtomicInternalTimestamp()
Phil Burkfd34a932017-07-19 07:03:52 -070072 , mWakeupDelayNanos(AAudioProperty_getWakeupDelayMicros() * AAUDIO_NANOS_PER_MICROSECOND)
73 , mMinimumSleepNanos(AAudioProperty_getMinimumSleepMicros() * AAUDIO_NANOS_PER_MICROSECOND)
74 {
Phil Burk204a1632017-01-03 17:23:43 -080075}
76
77AudioStreamInternal::~AudioStreamInternal() {
Phil Burkdd582922020-10-15 20:29:51 +000078 ALOGD("%s() %p called", __func__, this);
Phil Burk204a1632017-01-03 17:23:43 -080079}
80
Phil Burk5ed503c2017-02-01 09:38:15 -080081aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) {
Phil Burk204a1632017-01-03 17:23:43 -080082
Phil Burk5ed503c2017-02-01 09:38:15 -080083 aaudio_result_t result = AAUDIO_OK;
Phil Burk6479d502017-11-20 09:32:52 -080084 int32_t framesPerBurst;
Phil Burk3c4e6b52019-01-22 15:53:36 -080085 int32_t framesPerHardwareBurst;
Phil Burk5ed503c2017-02-01 09:38:15 -080086 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
jiabine504e7b2021-09-18 00:27:08 +0000100 const int32_t burstMinMicros = android::AudioSystem::getAAudioHardwareBurstMinUsec();
Phil Burk3c4e6b52019-01-22 15:53:36 -0800101 int32_t burstMicros = 0;
102
jiabinef348b82021-04-19 16:53:08 +0000103 const audio_format_t requestedFormat = getFormat();
Phil Burkc0c70e32017-02-09 13:18:38 -0800104 // We have to do volume scaling. So we prefer FLOAT format.
jiabinef348b82021-04-19 16:53:08 +0000105 if (requestedFormat == AUDIO_FORMAT_DEFAULT) {
Phil Burk0127c1b2018-03-29 13:48:06 -0700106 setFormat(AUDIO_FORMAT_PCM_FLOAT);
Phil Burkc0c70e32017-02-09 13:18:38 -0800107 }
Phil Burk04e805b2018-03-27 09:13:53 -0700108 // Request FLOAT for the shared mixer or the device.
Phil Burk0127c1b2018-03-29 13:48:06 -0700109 request.getConfiguration().setFormat(AUDIO_FORMAT_PCM_FLOAT);
Phil Burkc0c70e32017-02-09 13:18:38 -0800110
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000111 // TODO b/182392769: use attribution source util
112 AttributionSourceState attributionSource;
113 attributionSource.uid = VALUE_OR_FATAL(android::legacy2aidl_uid_t_int32_t(getuid()));
114 attributionSource.pid = VALUE_OR_FATAL(android::legacy2aidl_pid_t_int32_t(getpid()));
115 attributionSource.packageName = builder.getOpPackageName();
116 attributionSource.attributionTag = builder.getAttributionTag();
117 attributionSource.token = sp<android::BBinder>::make();
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700118
Phil Burkdec33ab2017-01-17 14:48:16 -0800119 // Build the request to send to the server.
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000120 request.setAttributionSource(attributionSource);
Phil Burk71f35bb2017-04-13 16:05:07 -0700121 request.setSharingModeMatchRequired(isSharingModeMatchRequired());
Phil Burk41f19d82018-02-13 14:59:10 -0800122 request.setInService(isInService());
Phil Burkc0c70e32017-02-09 13:18:38 -0800123
Phil Burk204a1632017-01-03 17:23:43 -0800124 request.getConfiguration().setDeviceId(getDeviceId());
125 request.getConfiguration().setSampleRate(getSampleRate());
Phil Burk39f02dd2017-08-04 09:13:31 -0700126 request.getConfiguration().setDirection(getDirection());
Phil Burk71f35bb2017-04-13 16:05:07 -0700127 request.getConfiguration().setSharingMode(getSharingMode());
jiabina9094092021-06-28 20:36:45 +0000128 request.getConfiguration().setChannelMask(getChannelMask());
Phil Burk71f35bb2017-04-13 16:05:07 -0700129
Phil Burka62fb952018-01-16 12:44:06 -0800130 request.getConfiguration().setUsage(getUsage());
131 request.getConfiguration().setContentType(getContentType());
Jean-Michel Trivi656bfdc2021-09-20 18:42:37 -0700132 request.getConfiguration().setSpatializationBehavior(getSpatializationBehavior());
133 request.getConfiguration().setIsContentSpatialized(isContentSpatialized());
Phil Burka62fb952018-01-16 12:44:06 -0800134 request.getConfiguration().setInputPreset(getInputPreset());
Eric Laurentd17c8502019-10-24 15:58:35 -0700135 request.getConfiguration().setPrivacySensitive(isPrivacySensitive());
Phil Burka62fb952018-01-16 12:44:06 -0800136
Phil Burk3df348f2017-02-08 11:41:55 -0800137 request.getConfiguration().setBufferCapacity(builder.getBufferCapacity());
Phil Burk204a1632017-01-03 17:23:43 -0800138
Phil Burk41f19d82018-02-13 14:59:10 -0800139 mDeviceChannelCount = getSamplesPerFrame(); // Assume it will be the same. Update if not.
140
Phil Burk99306c82017-08-14 12:38:58 -0700141 mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
Phil Burk41f19d82018-02-13 14:59:10 -0800142 if (mServiceStreamHandle < 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",
Phil Burk41f19d82018-02-13 14:59:10 -0800151 __func__, mServiceStreamHandle);
jiabina9094092021-06-28 20:36:45 +0000152 request.getConfiguration().setChannelMask(AAUDIO_CHANNEL_STEREO);
Phil Burk41f19d82018-02-13 14:59:10 -0800153 mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
154 }
Phil Burk204a1632017-01-03 17:23:43 -0800155 if (mServiceStreamHandle < 0) {
Phil Burk41f19d82018-02-13 14:59:10 -0800156 return mServiceStreamHandle;
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)
164 + std::to_string(mServiceStreamHandle);
165 }
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
200 result = mServiceInterface.getStreamDescription(mServiceStreamHandle, mEndPointParcelable);
201 if (result != AAUDIO_OK) {
202 goto error;
203 }
204
205 // Resolve parcelable into a descriptor.
206 result = mEndPointParcelable.resolve(&mEndpointDescriptor);
207 if (result != AAUDIO_OK) {
208 goto error;
209 }
210
211 // Configure endpoint based on descriptor.
Phil Burk5edc4ea2020-04-17 08:15:42 -0700212 mAudioEndpoint = std::make_unique<AudioEndpoint>();
213 result = mAudioEndpoint->configure(&mEndpointDescriptor, getDirection());
Phil Burk99306c82017-08-14 12:38:58 -0700214 if (result != AAUDIO_OK) {
215 goto error;
216 }
217
Phil Burk3c4e6b52019-01-22 15:53:36 -0800218 framesPerHardwareBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
219
220 // Scale up the burst size to meet the minimum equivalent in microseconds.
221 // This is to avoid waking the CPU too often when the HW burst is very small
222 // or at high sample rates.
223 framesPerBurst = framesPerHardwareBurst;
224 do {
225 if (burstMicros > 0) { // skip first loop
226 framesPerBurst *= 2;
227 }
228 burstMicros = framesPerBurst * static_cast<int64_t>(1000000) / getSampleRate();
229 } while (burstMicros < burstMinMicros);
230 ALOGD("%s() original HW burst = %d, minMicros = %d => SW burst = %d\n",
231 __func__, framesPerHardwareBurst, burstMinMicros, framesPerBurst);
232
233 // Validate final burst size.
Phil Burk6479d502017-11-20 09:32:52 -0800234 if (framesPerBurst < MIN_FRAMES_PER_BURST || framesPerBurst > MAX_FRAMES_PER_BURST) {
235 ALOGE("%s - framesPerBurst out of range = %d", __func__, framesPerBurst);
Phil Burk99306c82017-08-14 12:38:58 -0700236 result = AAUDIO_ERROR_OUT_OF_RANGE;
237 goto error;
238 }
Phil Burk8d97b8e2020-09-25 23:18:14 +0000239 setFramesPerBurst(framesPerBurst); // only save good value
Phil Burk6479d502017-11-20 09:32:52 -0800240
Phil Burk5edc4ea2020-04-17 08:15:42 -0700241 mBufferCapacityInFrames = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
Phil Burk8d97b8e2020-09-25 23:18:14 +0000242 if (mBufferCapacityInFrames < getFramesPerBurst()
Phil Burk5edc4ea2020-04-17 08:15:42 -0700243 || mBufferCapacityInFrames > MAX_BUFFER_CAPACITY_IN_FRAMES) {
244 ALOGE("%s - bufferCapacity out of range = %d", __func__, mBufferCapacityInFrames);
Phil Burk99306c82017-08-14 12:38:58 -0700245 result = AAUDIO_ERROR_OUT_OF_RANGE;
246 goto error;
247 }
248
249 mClockModel.setSampleRate(getSampleRate());
Phil Burk3c4e6b52019-01-22 15:53:36 -0800250 mClockModel.setFramesPerBurst(framesPerHardwareBurst);
Phil Burk99306c82017-08-14 12:38:58 -0700251
Phil Burk134f1972017-12-08 13:06:11 -0800252 if (isDataCallbackSet()) {
Phil Burk99306c82017-08-14 12:38:58 -0700253 mCallbackFrames = builder.getFramesPerDataCallback();
254 if (mCallbackFrames > getBufferCapacity() / 2) {
Phil Burk29ccc292019-04-15 08:58:08 -0700255 ALOGW("%s - framesPerCallback too big = %d, capacity = %d",
Phil Burkfbf031e2017-10-12 15:58:31 -0700256 __func__, mCallbackFrames, getBufferCapacity());
Phil Burk99306c82017-08-14 12:38:58 -0700257 result = AAUDIO_ERROR_OUT_OF_RANGE;
258 goto error;
259
260 } else if (mCallbackFrames < 0) {
Phil Burk29ccc292019-04-15 08:58:08 -0700261 ALOGW("%s - framesPerCallback negative", __func__);
Phil Burk99306c82017-08-14 12:38:58 -0700262 result = AAUDIO_ERROR_OUT_OF_RANGE;
263 goto error;
264
265 }
266 if (mCallbackFrames == AAUDIO_UNSPECIFIED) {
Phil Burk8d97b8e2020-09-25 23:18:14 +0000267 mCallbackFrames = getFramesPerBurst();
Phil Burk99306c82017-08-14 12:38:58 -0700268 }
269
Phil Burk0127c1b2018-03-29 13:48:06 -0700270 const int32_t callbackBufferSize = mCallbackFrames * getBytesPerFrame();
Phil Burkbf821e22020-04-17 11:51:43 -0700271 mCallbackBuffer = std::make_unique<uint8_t[]>(callbackBufferSize);
Phil Burk99306c82017-08-14 12:38:58 -0700272 }
273
Robert Wud7400832021-12-04 01:11:19 +0000274 // Exclusive output streams should combine channels when mono audio adjustment
275 // is enabled.
276 if ((getDirection() == AAUDIO_DIRECTION_OUTPUT) &&
277 (getSharingMode() == AAUDIO_SHARING_MODE_EXCLUSIVE)) {
278 bool isMasterMono = false;
279 android::AudioSystem::getMasterMono(&isMasterMono);
280 setRequireMonoBlend(isMasterMono);
281 }
282
Phil Burkb31b66f2019-09-30 09:33:41 -0700283 // For debugging and analyzing the distribution of MMAP timestamps.
284 // For OUTPUT, use a NEGATIVE offset to move the CPU writes further BEFORE the HW reads.
285 // For INPUT, use a POSITIVE offset to move the CPU reads further AFTER the HW writes.
286 // You can use this offset to reduce glitching.
287 // You can also use this offset to force glitching. By iterating over multiple
288 // values you can reveal the distribution of the hardware timing jitter.
Phil Burk5edc4ea2020-04-17 08:15:42 -0700289 if (mAudioEndpoint->isFreeRunning()) { // MMAP?
Phil Burkb31b66f2019-09-30 09:33:41 -0700290 int32_t offsetMicros = (getDirection() == AAUDIO_DIRECTION_OUTPUT)
291 ? AAudioProperty_getOutputMMapOffsetMicros()
292 : AAudioProperty_getInputMMapOffsetMicros();
293 // This log is used to debug some tricky glitch issues. Please leave.
294 ALOGD_IF(offsetMicros, "%s() - %s mmap offset = %d micros",
295 __func__,
296 (getDirection() == AAUDIO_DIRECTION_OUTPUT) ? "output" : "input",
297 offsetMicros);
298 mTimeOffsetNanos = offsetMicros * AAUDIO_NANOS_PER_MICROSECOND;
299 }
300
Phil Burk5edc4ea2020-04-17 08:15:42 -0700301 setBufferSize(mBufferCapacityInFrames / 2); // Default buffer size to match Q
Phil Burk6c63ae32019-10-28 10:28:21 -0700302
Phil Burk99306c82017-08-14 12:38:58 -0700303 setState(AAUDIO_STREAM_STATE_OPEN);
Phil Burk99306c82017-08-14 12:38:58 -0700304
305 return result;
306
307error:
Phil Burkdd582922020-10-15 20:29:51 +0000308 safeReleaseClose();
Phil Burk204a1632017-01-03 17:23:43 -0800309 return result;
310}
311
Phil Burk13d3d832019-06-10 14:36:48 -0700312// This must be called under mStreamLock.
Phil Burk8b4e05e2019-12-17 12:12:09 -0800313aaudio_result_t AudioStreamInternal::release_l() {
Phil Burk965650e2017-09-07 21:00:09 -0700314 aaudio_result_t result = AAUDIO_OK;
Phil Burkdd582922020-10-15 20:29:51 +0000315 ALOGD("%s(): mServiceStreamHandle = 0x%08X", __func__, mServiceStreamHandle);
Phil Burk5ed503c2017-02-01 09:38:15 -0800316 if (mServiceStreamHandle != AAUDIO_HANDLE_INVALID) {
Phil Burk4485d412017-05-09 15:55:02 -0700317 aaudio_stream_state_t currentState = getState();
Phil Burk8b4e05e2019-12-17 12:12:09 -0800318 // Don't release a stream while it is running. Stop it first.
Phil Burk13d3d832019-06-10 14:36:48 -0700319 // If DISCONNECTED then we should still try to stop in case the
320 // error callback is still running.
321 if (isActive() || currentState == AAUDIO_STREAM_STATE_DISCONNECTED) {
Phil Burkdd582922020-10-15 20:29:51 +0000322 requestStop_l();
Phil Burk4485d412017-05-09 15:55:02 -0700323 }
Phil Burka9876702020-04-20 18:16:15 -0700324
Phil Burk64e16a72020-06-01 13:25:51 -0700325 logReleaseBufferState();
Phil Burka9876702020-04-20 18:16:15 -0700326
Phil Burkec89b2e2017-06-20 15:05:06 -0700327 setState(AAUDIO_STREAM_STATE_CLOSING);
Phil Burk5ed503c2017-02-01 09:38:15 -0800328 aaudio_handle_t serviceStreamHandle = mServiceStreamHandle;
329 mServiceStreamHandle = AAUDIO_HANDLE_INVALID;
Phil Burkc0c70e32017-02-09 13:18:38 -0800330
331 mServiceInterface.closeStream(serviceStreamHandle);
Phil Burkbf821e22020-04-17 11:51:43 -0700332 mCallbackBuffer.reset();
Phil Burk5edc4ea2020-04-17 08:15:42 -0700333
334 // Update local frame counters so we can query them after releasing the endpoint.
335 getFramesRead();
336 getFramesWritten();
337 mAudioEndpoint.reset();
Phil Burk965650e2017-09-07 21:00:09 -0700338 result = mEndPointParcelable.close();
Phil Burk8b4e05e2019-12-17 12:12:09 -0800339 aaudio_result_t result2 = AudioStream::release_l();
Phil Burk965650e2017-09-07 21:00:09 -0700340 return (result != AAUDIO_OK) ? result : result2;
Phil Burk204a1632017-01-03 17:23:43 -0800341 } else {
Phil Burk5ed503c2017-02-01 09:38:15 -0800342 return AAUDIO_ERROR_INVALID_HANDLE;
Phil Burk204a1632017-01-03 17:23:43 -0800343 }
344}
345
Phil Burke4d7bb42017-03-28 11:32:39 -0700346static void *aaudio_callback_thread_proc(void *context)
347{
348 AudioStreamInternal *stream = (AudioStreamInternal *)context;
Phil Burkfbf031e2017-10-12 15:58:31 -0700349 //LOGD("oboe_callback_thread, stream = %p", stream);
jiabind5bd06a2021-04-27 22:04:08 +0000350 if (stream != nullptr) {
Phil Burke4d7bb42017-03-28 11:32:39 -0700351 return stream->callbackLoop();
352 } else {
jiabind5bd06a2021-04-27 22:04:08 +0000353 return nullptr;
Phil Burke4d7bb42017-03-28 11:32:39 -0700354 }
355}
356
Phil Burkbcc36742017-08-31 17:24:51 -0700357/*
358 * It normally takes about 20-30 msec to start a stream on the server.
359 * But the first time can take as much as 200-300 msec. The HW
360 * starts right away so by the time the client gets a chance to write into
361 * the buffer, it is already in a deep underflow state. That can cause the
362 * XRunCount to be non-zero, which could lead an app to tune its latency higher.
363 * To avoid this problem, we set a request for the processing code to start the
364 * client stream at the same position as the server stream.
365 * The processing code will then save the current offset
366 * between client and server and apply that to any position given to the app.
367 */
Phil Burkdd582922020-10-15 20:29:51 +0000368aaudio_result_t AudioStreamInternal::requestStart_l()
Phil Burk204a1632017-01-03 17:23:43 -0800369{
Phil Burk3316d5e2017-02-15 11:23:01 -0800370 int64_t startTime;
Phil Burk5ed503c2017-02-01 09:38:15 -0800371 if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700372 ALOGD("requestStart() mServiceStreamHandle invalid");
Phil Burk5ed503c2017-02-01 09:38:15 -0800373 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800374 }
Phil Burkec89b2e2017-06-20 15:05:06 -0700375 if (isActive()) {
Phil Burk29ccc292019-04-15 08:58:08 -0700376 ALOGD("requestStart() already active");
Phil Burkec89b2e2017-06-20 15:05:06 -0700377 return AAUDIO_ERROR_INVALID_STATE;
378 }
Phil Burkec89b2e2017-06-20 15:05:06 -0700379
Phil Burkbcc36742017-08-31 17:24:51 -0700380 aaudio_stream_state_t originalState = getState();
381 if (originalState == AAUDIO_STREAM_STATE_DISCONNECTED) {
Phil Burk29ccc292019-04-15 08:58:08 -0700382 ALOGD("requestStart() but DISCONNECTED");
Phil Burkbcc36742017-08-31 17:24:51 -0700383 return AAUDIO_ERROR_DISCONNECTED;
384 }
Phil Burkec89b2e2017-06-20 15:05:06 -0700385 setState(AAUDIO_STREAM_STATE_STARTING);
Phil Burkbcc36742017-08-31 17:24:51 -0700386
387 // Clear any stale timestamps from the previous run.
388 drainTimestampsFromService();
389
Phil Burkec8ca522020-05-19 10:05:58 -0700390 prepareBuffersForStart(); // tell subclasses to get ready
391
Phil Burk965650e2017-09-07 21:00:09 -0700392 aaudio_result_t result = mServiceInterface.startStream(mServiceStreamHandle);
Phil Burk6e463ce2020-04-13 10:20:20 -0700393 if (result == AAUDIO_ERROR_INVALID_HANDLE) {
394 ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
395 // Stealing was added in R. Coerce result to improve backward compatibility.
396 result = AAUDIO_ERROR_DISCONNECTED;
397 setState(AAUDIO_STREAM_STATE_DISCONNECTED);
398 }
Phil Burkc0c70e32017-02-09 13:18:38 -0800399
Phil Burk3316d5e2017-02-15 11:23:01 -0800400 startTime = AudioClock::getNanoseconds();
Phil Burk204a1632017-01-03 17:23:43 -0800401 mClockModel.start(startTime);
Phil Burkbcc36742017-08-31 17:24:51 -0700402 mNeedCatchUp.request(); // Ask data processing code to catch up when first timestamp received.
Phil Burke4d7bb42017-03-28 11:32:39 -0700403
Phil Burk965650e2017-09-07 21:00:09 -0700404 // Start data callback thread.
Phil Burk134f1972017-12-08 13:06:11 -0800405 if (result == AAUDIO_OK && isDataCallbackSet()) {
Phil Burke4d7bb42017-03-28 11:32:39 -0700406 // Launch the callback loop thread.
407 int64_t periodNanos = mCallbackFrames
408 * AAUDIO_NANOS_PER_SECOND
409 / getSampleRate();
410 mCallbackEnabled.store(true);
Phil Burkdd582922020-10-15 20:29:51 +0000411 result = createThread_l(periodNanos, aaudio_callback_thread_proc, this);
Phil Burke4d7bb42017-03-28 11:32:39 -0700412 }
Phil Burkec89b2e2017-06-20 15:05:06 -0700413 if (result != AAUDIO_OK) {
414 setState(originalState);
415 }
Phil Burke4d7bb42017-03-28 11:32:39 -0700416 return result;
Phil Burk204a1632017-01-03 17:23:43 -0800417}
418
Phil Burke4d7bb42017-03-28 11:32:39 -0700419int64_t AudioStreamInternal::calculateReasonableTimeout(int32_t framesPerOperation) {
420
421 // Wait for at least a second or some number of callbacks to join the thread.
Phil Burk71f35bb2017-04-13 16:05:07 -0700422 int64_t timeoutNanoseconds = (MIN_TIMEOUT_OPERATIONS
423 * framesPerOperation
424 * AAUDIO_NANOS_PER_SECOND)
425 / getSampleRate();
Phil Burke4d7bb42017-03-28 11:32:39 -0700426 if (timeoutNanoseconds < MIN_TIMEOUT_NANOS) { // arbitrary number of seconds
427 timeoutNanoseconds = MIN_TIMEOUT_NANOS;
428 }
429 return timeoutNanoseconds;
430}
431
Phil Burk87c9f642017-05-17 07:22:39 -0700432int64_t AudioStreamInternal::calculateReasonableTimeout() {
433 return calculateReasonableTimeout(getFramesPerBurst());
434}
435
Phil Burk13d3d832019-06-10 14:36:48 -0700436// This must be called under mStreamLock.
Phil Burkdd582922020-10-15 20:29:51 +0000437aaudio_result_t AudioStreamInternal::stopCallback_l()
Phil Burke4d7bb42017-03-28 11:32:39 -0700438{
Phil Burk13d3d832019-06-10 14:36:48 -0700439 if (isDataCallbackSet()
440 && (isActive() || getState() == AAUDIO_STREAM_STATE_DISCONNECTED)) {
Phil Burke4d7bb42017-03-28 11:32:39 -0700441 mCallbackEnabled.store(false);
jiabind5bd06a2021-04-27 22:04:08 +0000442 aaudio_result_t result = joinThread_l(nullptr); // may temporarily unlock mStreamLock
Phil Burk6e463ce2020-04-13 10:20:20 -0700443 if (result == AAUDIO_ERROR_INVALID_HANDLE) {
444 ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
445 result = AAUDIO_OK;
446 }
447 return result;
Phil Burke4d7bb42017-03-28 11:32:39 -0700448 } else {
Phil Burkdd582922020-10-15 20:29:51 +0000449 ALOGD("%s() skipped, isDataCallbackSet() = %d, isActive() = %d, getState() = %d", __func__,
450 isDataCallbackSet(), isActive(), getState());
Phil Burke4d7bb42017-03-28 11:32:39 -0700451 return AAUDIO_OK;
452 }
453}
454
Phil Burkdd582922020-10-15 20:29:51 +0000455aaudio_result_t AudioStreamInternal::requestStop_l() {
456 aaudio_result_t result = stopCallback_l();
Phil Burk5cc83c32017-11-28 15:43:18 -0800457 if (result != AAUDIO_OK) {
Phil Burkdd582922020-10-15 20:29:51 +0000458 ALOGW("%s() stop callback returned %d, returning early", __func__, result);
Phil Burk5cc83c32017-11-28 15:43:18 -0800459 return result;
460 }
Phil Burk13d3d832019-06-10 14:36:48 -0700461 // The stream may have been unlocked temporarily to let a callback finish
462 // and the callback may have stopped the stream.
463 // Check to make sure the stream still needs to be stopped.
Phil Burk0bd745e2020-10-17 18:20:01 +0000464 // See also AudioStream::safeStop_l().
Phil Burk13d3d832019-06-10 14:36:48 -0700465 if (!(isActive() || getState() == AAUDIO_STREAM_STATE_DISCONNECTED)) {
Phil Burkdd582922020-10-15 20:29:51 +0000466 ALOGD("%s() returning early, not active or disconnected", __func__);
Phil Burk13d3d832019-06-10 14:36:48 -0700467 return AAUDIO_OK;
468 }
Phil Burk5cc83c32017-11-28 15:43:18 -0800469
Phil Burk71f35bb2017-04-13 16:05:07 -0700470 if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700471 ALOGW("%s() mServiceStreamHandle invalid = 0x%08X",
472 __func__, mServiceStreamHandle);
Phil Burk71f35bb2017-04-13 16:05:07 -0700473 return AAUDIO_ERROR_INVALID_STATE;
474 }
475
476 mClockModel.stop(AudioClock::getNanoseconds());
477 setState(AAUDIO_STREAM_STATE_STOPPING);
Phil Burka53ffa62018-10-10 16:21:37 -0700478 mAtomicInternalTimestamp.clear();
Phil Burk965650e2017-09-07 21:00:09 -0700479
Phil Burk6e463ce2020-04-13 10:20:20 -0700480 result = mServiceInterface.stopStream(mServiceStreamHandle);
481 if (result == AAUDIO_ERROR_INVALID_HANDLE) {
482 ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
483 result = AAUDIO_OK;
484 }
485 return result;
Phil Burk71f35bb2017-04-13 16:05:07 -0700486}
487
Phil Burk5ed503c2017-02-01 09:38:15 -0800488aaudio_result_t AudioStreamInternal::registerThread() {
Phil Burk5ed503c2017-02-01 09:38:15 -0800489 if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700490 ALOGW("%s() mServiceStreamHandle invalid", __func__);
Phil Burk5ed503c2017-02-01 09:38:15 -0800491 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800492 }
Phil Burkc0c70e32017-02-09 13:18:38 -0800493 return mServiceInterface.registerAudioThread(mServiceStreamHandle,
Phil Burkc0c70e32017-02-09 13:18:38 -0800494 gettid(),
495 getPeriodNanoseconds());
Phil Burk204a1632017-01-03 17:23:43 -0800496}
497
Phil Burk5ed503c2017-02-01 09:38:15 -0800498aaudio_result_t AudioStreamInternal::unregisterThread() {
Phil Burk5ed503c2017-02-01 09:38:15 -0800499 if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
Phil Burk29ccc292019-04-15 08:58:08 -0700500 ALOGW("%s() mServiceStreamHandle invalid", __func__);
Phil Burk5ed503c2017-02-01 09:38:15 -0800501 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800502 }
Phil Burk2ac035f2017-06-23 14:51:14 -0700503 return mServiceInterface.unregisterAudioThread(mServiceStreamHandle, gettid());
Phil Burk204a1632017-01-03 17:23:43 -0800504}
505
Eric Laurentcb4dae22017-07-01 19:39:32 -0700506aaudio_result_t AudioStreamInternal::startClient(const android::AudioClient& client,
jiabind1f1cb62020-03-24 11:57:57 -0700507 const audio_attributes_t *attr,
Phil Burkbbd52862018-04-13 11:37:42 -0700508 audio_port_handle_t *portHandle) {
509 ALOGV("%s() called", __func__);
Eric Laurentcb4dae22017-07-01 19:39:32 -0700510 if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
511 return AAUDIO_ERROR_INVALID_STATE;
512 }
Phil Burkbbd52862018-04-13 11:37:42 -0700513 aaudio_result_t result = mServiceInterface.startClient(mServiceStreamHandle,
jiabind1f1cb62020-03-24 11:57:57 -0700514 client, attr, portHandle);
Phil Burkbbd52862018-04-13 11:37:42 -0700515 ALOGV("%s(%d) returning %d", __func__, *portHandle, result);
516 return result;
Eric Laurentcb4dae22017-07-01 19:39:32 -0700517}
518
Phil Burkbbd52862018-04-13 11:37:42 -0700519aaudio_result_t AudioStreamInternal::stopClient(audio_port_handle_t portHandle) {
520 ALOGV("%s(%d) called", __func__, portHandle);
Eric Laurentcb4dae22017-07-01 19:39:32 -0700521 if (mServiceStreamHandle == AAUDIO_HANDLE_INVALID) {
522 return AAUDIO_ERROR_INVALID_STATE;
523 }
Phil Burkbbd52862018-04-13 11:37:42 -0700524 aaudio_result_t result = mServiceInterface.stopClient(mServiceStreamHandle, portHandle);
525 ALOGV("%s(%d) returning %d", __func__, portHandle, result);
526 return result;
Eric Laurentcb4dae22017-07-01 19:39:32 -0700527}
528
jiabind5bd06a2021-04-27 22:04:08 +0000529aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t /*clockId*/,
Phil Burk3316d5e2017-02-15 11:23:01 -0800530 int64_t *framePosition,
531 int64_t *timeNanoseconds) {
Phil Burk97350f92017-07-21 15:59:44 -0700532 // Generated in server and passed to client. Return latest.
Phil Burka53ffa62018-10-10 16:21:37 -0700533 if (mAtomicInternalTimestamp.isValid()) {
534 Timestamp timestamp = mAtomicInternalTimestamp.read();
Phil Burkbcc36742017-08-31 17:24:51 -0700535 int64_t position = timestamp.getPosition() + mFramesOffsetFromService;
536 if (position >= 0) {
537 *framePosition = position;
538 *timeNanoseconds = timestamp.getNanoseconds();
539 return AAUDIO_OK;
540 }
Phil Burk97350f92017-07-21 15:59:44 -0700541 }
Phil Burkc75d97f2017-09-08 15:48:36 -0700542 return AAUDIO_ERROR_INVALID_STATE;
Phil Burk204a1632017-01-03 17:23:43 -0800543}
544
Phil Burk0befec62017-07-28 15:12:13 -0700545aaudio_result_t AudioStreamInternal::updateStateMachine() {
Phil Burke4d7bb42017-03-28 11:32:39 -0700546 if (isDataCallbackActive()) {
547 return AAUDIO_OK; // state is getting updated by the callback thread read/write call
548 }
Phil Burk204a1632017-01-03 17:23:43 -0800549 return processCommands();
550}
551
Phil Burkec89b2e2017-06-20 15:05:06 -0700552void AudioStreamInternal::logTimestamp(AAudioServiceMessage &command) {
Phil Burk204a1632017-01-03 17:23:43 -0800553 static int64_t oldPosition = 0;
Phil Burk3316d5e2017-02-15 11:23:01 -0800554 static int64_t oldTime = 0;
Phil Burk204a1632017-01-03 17:23:43 -0800555 int64_t framePosition = command.timestamp.position;
Phil Burk3316d5e2017-02-15 11:23:01 -0800556 int64_t nanoTime = command.timestamp.timestamp;
Phil Burkbcc36742017-08-31 17:24:51 -0700557 ALOGD("logTimestamp: timestamp says framePosition = %8lld at nanoTime %lld",
Phil Burk204a1632017-01-03 17:23:43 -0800558 (long long) framePosition,
559 (long long) nanoTime);
560 int64_t nanosDelta = nanoTime - oldTime;
561 if (nanosDelta > 0 && oldTime > 0) {
562 int64_t framesDelta = framePosition - oldPosition;
Phil Burk5ed503c2017-02-01 09:38:15 -0800563 int64_t rate = (framesDelta * AAUDIO_NANOS_PER_SECOND) / nanosDelta;
Phil Burkbcc36742017-08-31 17:24:51 -0700564 ALOGD("logTimestamp: framesDelta = %8lld, nanosDelta = %8lld, rate = %lld",
Phil Burkec89b2e2017-06-20 15:05:06 -0700565 (long long) framesDelta, (long long) nanosDelta, (long long) rate);
Phil Burk204a1632017-01-03 17:23:43 -0800566 }
567 oldPosition = framePosition;
568 oldTime = nanoTime;
569}
Phil Burk204a1632017-01-03 17:23:43 -0800570
Phil Burk97350f92017-07-21 15:59:44 -0700571aaudio_result_t AudioStreamInternal::onTimestampService(AAudioServiceMessage *message) {
Phil Burk204a1632017-01-03 17:23:43 -0800572#if LOG_TIMESTAMPS
Phil Burkec89b2e2017-06-20 15:05:06 -0700573 logTimestamp(*message);
Phil Burk204a1632017-01-03 17:23:43 -0800574#endif
Phil Burkb31b66f2019-09-30 09:33:41 -0700575 processTimestamp(message->timestamp.position,
576 message->timestamp.timestamp + mTimeOffsetNanos);
Phil Burk5ed503c2017-02-01 09:38:15 -0800577 return AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800578}
579
Phil Burk97350f92017-07-21 15:59:44 -0700580aaudio_result_t AudioStreamInternal::onTimestampHardware(AAudioServiceMessage *message) {
581 Timestamp timestamp(message->timestamp.position, message->timestamp.timestamp);
Phil Burka53ffa62018-10-10 16:21:37 -0700582 mAtomicInternalTimestamp.write(timestamp);
Phil Burk97350f92017-07-21 15:59:44 -0700583 return AAUDIO_OK;
584}
585
Phil Burk5ed503c2017-02-01 09:38:15 -0800586aaudio_result_t AudioStreamInternal::onEventFromServer(AAudioServiceMessage *message) {
587 aaudio_result_t result = AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800588 switch (message->event.event) {
Phil Burk5ed503c2017-02-01 09:38:15 -0800589 case AAUDIO_SERVICE_EVENT_STARTED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700590 ALOGD("%s - got AAUDIO_SERVICE_EVENT_STARTED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700591 if (getState() == AAUDIO_STREAM_STATE_STARTING) {
592 setState(AAUDIO_STREAM_STATE_STARTED);
593 }
Phil Burk204a1632017-01-03 17:23:43 -0800594 break;
Phil Burk5ed503c2017-02-01 09:38:15 -0800595 case AAUDIO_SERVICE_EVENT_PAUSED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700596 ALOGD("%s - got AAUDIO_SERVICE_EVENT_PAUSED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700597 if (getState() == AAUDIO_STREAM_STATE_PAUSING) {
598 setState(AAUDIO_STREAM_STATE_PAUSED);
599 }
Phil Burk204a1632017-01-03 17:23:43 -0800600 break;
Phil Burk71f35bb2017-04-13 16:05:07 -0700601 case AAUDIO_SERVICE_EVENT_STOPPED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700602 ALOGD("%s - got AAUDIO_SERVICE_EVENT_STOPPED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700603 if (getState() == AAUDIO_STREAM_STATE_STOPPING) {
604 setState(AAUDIO_STREAM_STATE_STOPPED);
605 }
Phil Burk71f35bb2017-04-13 16:05:07 -0700606 break;
Phil Burk5ed503c2017-02-01 09:38:15 -0800607 case AAUDIO_SERVICE_EVENT_FLUSHED:
Phil Burkfbf031e2017-10-12 15:58:31 -0700608 ALOGD("%s - got AAUDIO_SERVICE_EVENT_FLUSHED", __func__);
Phil Burk87c9f642017-05-17 07:22:39 -0700609 if (getState() == AAUDIO_STREAM_STATE_FLUSHING) {
610 setState(AAUDIO_STREAM_STATE_FLUSHED);
611 onFlushFromServer();
612 }
Phil Burk204a1632017-01-03 17:23:43 -0800613 break;
Phil Burk5ed503c2017-02-01 09:38:15 -0800614 case AAUDIO_SERVICE_EVENT_DISCONNECTED:
Phil Burkea04d972017-08-07 12:30:44 -0700615 // Prevent hardware from looping on old data and making buzzing sounds.
616 if (getDirection() == AAUDIO_DIRECTION_OUTPUT) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700617 mAudioEndpoint->eraseDataMemory();
Phil Burkea04d972017-08-07 12:30:44 -0700618 }
Phil Burk5ed503c2017-02-01 09:38:15 -0800619 result = AAUDIO_ERROR_DISCONNECTED;
Phil Burkc0c70e32017-02-09 13:18:38 -0800620 setState(AAUDIO_STREAM_STATE_DISCONNECTED);
Phil Burkfbf031e2017-10-12 15:58:31 -0700621 ALOGW("%s - AAUDIO_SERVICE_EVENT_DISCONNECTED - FIFO cleared", __func__);
Phil Burk204a1632017-01-03 17:23:43 -0800622 break;
Phil Burkc0c70e32017-02-09 13:18:38 -0800623 case AAUDIO_SERVICE_EVENT_VOLUME:
Phil Burk55e5eab2018-04-10 15:16:38 -0700624 ALOGD("%s - AAUDIO_SERVICE_EVENT_VOLUME %lf", __func__, message->event.dataDouble);
Eric Laurenta2f296e2017-06-21 18:51:47 -0700625 mStreamVolume = (float)message->event.dataDouble;
626 doSetVolume();
Phil Burkc0c70e32017-02-09 13:18:38 -0800627 break;
Phil Burk23296382017-11-20 15:45:11 -0800628 case AAUDIO_SERVICE_EVENT_XRUN:
629 mXRunCount = static_cast<int32_t>(message->event.dataLong);
630 break;
Phil Burk204a1632017-01-03 17:23:43 -0800631 default:
Phil Burkfbf031e2017-10-12 15:58:31 -0700632 ALOGE("%s - Unrecognized event = %d", __func__, (int) message->event.event);
Phil Burk204a1632017-01-03 17:23:43 -0800633 break;
634 }
635 return result;
636}
637
Phil Burkbcc36742017-08-31 17:24:51 -0700638aaudio_result_t AudioStreamInternal::drainTimestampsFromService() {
639 aaudio_result_t result = AAUDIO_OK;
640
641 while (result == AAUDIO_OK) {
642 AAudioServiceMessage message;
Phil Burk5edc4ea2020-04-17 08:15:42 -0700643 if (!mAudioEndpoint) {
644 break;
645 }
646 if (mAudioEndpoint->readUpCommand(&message) != 1) {
Phil Burkbcc36742017-08-31 17:24:51 -0700647 break; // no command this time, no problem
648 }
649 switch (message.what) {
650 // ignore most messages
651 case AAudioServiceMessage::code::TIMESTAMP_SERVICE:
652 case AAudioServiceMessage::code::TIMESTAMP_HARDWARE:
653 break;
654
655 case AAudioServiceMessage::code::EVENT:
656 result = onEventFromServer(&message);
657 break;
658
659 default:
Phil Burkfbf031e2017-10-12 15:58:31 -0700660 ALOGE("%s - unrecognized message.what = %d", __func__, (int) message.what);
Phil Burkbcc36742017-08-31 17:24:51 -0700661 result = AAUDIO_ERROR_INTERNAL;
662 break;
663 }
664 }
665 return result;
666}
667
Phil Burk204a1632017-01-03 17:23:43 -0800668// Process all the commands coming from the server.
Phil Burk5ed503c2017-02-01 09:38:15 -0800669aaudio_result_t AudioStreamInternal::processCommands() {
670 aaudio_result_t result = AAUDIO_OK;
Phil Burk204a1632017-01-03 17:23:43 -0800671
Phil Burk5ed503c2017-02-01 09:38:15 -0800672 while (result == AAUDIO_OK) {
673 AAudioServiceMessage message;
Phil Burk5edc4ea2020-04-17 08:15:42 -0700674 if (!mAudioEndpoint) {
675 break;
676 }
677 if (mAudioEndpoint->readUpCommand(&message) != 1) {
Phil Burk204a1632017-01-03 17:23:43 -0800678 break; // no command this time, no problem
679 }
680 switch (message.what) {
Phil Burk97350f92017-07-21 15:59:44 -0700681 case AAudioServiceMessage::code::TIMESTAMP_SERVICE:
682 result = onTimestampService(&message);
683 break;
684
685 case AAudioServiceMessage::code::TIMESTAMP_HARDWARE:
686 result = onTimestampHardware(&message);
Phil Burk204a1632017-01-03 17:23:43 -0800687 break;
688
Phil Burk5ed503c2017-02-01 09:38:15 -0800689 case AAudioServiceMessage::code::EVENT:
Phil Burk204a1632017-01-03 17:23:43 -0800690 result = onEventFromServer(&message);
691 break;
692
693 default:
Phil Burkfbf031e2017-10-12 15:58:31 -0700694 ALOGE("%s - unrecognized message.what = %d", __func__, (int) message.what);
Phil Burk17fff382017-05-16 14:06:45 -0700695 result = AAUDIO_ERROR_INTERNAL;
Phil Burk204a1632017-01-03 17:23:43 -0800696 break;
697 }
698 }
699 return result;
700}
701
Phil Burk87c9f642017-05-17 07:22:39 -0700702// Read or write the data, block if needed and timeoutMillis > 0
703aaudio_result_t AudioStreamInternal::processData(void *buffer, int32_t numFrames,
704 int64_t timeoutNanoseconds)
Phil Burk204a1632017-01-03 17:23:43 -0800705{
Phil Burkfd34a932017-07-19 07:03:52 -0700706 const char * traceName = "aaProc";
707 const char * fifoName = "aaRdy";
Phil Burk4485d412017-05-09 15:55:02 -0700708 ATRACE_BEGIN(traceName);
Phil Burk4485d412017-05-09 15:55:02 -0700709 if (ATRACE_ENABLED()) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700710 int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
Phil Burkfd34a932017-07-19 07:03:52 -0700711 ATRACE_INT(fifoName, fullFrames);
Phil Burk4485d412017-05-09 15:55:02 -0700712 }
713
Phil Burkec89b2e2017-06-20 15:05:06 -0700714 aaudio_result_t result = AAUDIO_OK;
715 int32_t loopCount = 0;
716 uint8_t* audioData = (uint8_t*)buffer;
717 int64_t currentTimeNanos = AudioClock::getNanoseconds();
718 const int64_t entryTimeNanos = currentTimeNanos;
719 const int64_t deadlineNanos = currentTimeNanos + timeoutNanoseconds;
720 int32_t framesLeft = numFrames;
721
Phil Burk87c9f642017-05-17 07:22:39 -0700722 // Loop until all the data has been processed or until a timeout occurs.
Phil Burk204a1632017-01-03 17:23:43 -0800723 while (framesLeft > 0) {
Phil Burkec89b2e2017-06-20 15:05:06 -0700724 // The call to processDataNow() will not block. It will just process as much as it can.
Phil Burk3316d5e2017-02-15 11:23:01 -0800725 int64_t wakeTimeNanos = 0;
Phil Burk87c9f642017-05-17 07:22:39 -0700726 aaudio_result_t framesProcessed = processDataNow(audioData, framesLeft,
727 currentTimeNanos, &wakeTimeNanos);
728 if (framesProcessed < 0) {
Phil Burk87c9f642017-05-17 07:22:39 -0700729 result = framesProcessed;
Phil Burk204a1632017-01-03 17:23:43 -0800730 break;
731 }
Phil Burk87c9f642017-05-17 07:22:39 -0700732 framesLeft -= (int32_t) framesProcessed;
733 audioData += framesProcessed * getBytesPerFrame();
Phil Burk204a1632017-01-03 17:23:43 -0800734
735 // Should we block?
736 if (timeoutNanoseconds == 0) {
737 break; // don't block
Phil Burk8d4f0062019-10-03 15:55:41 -0700738 } else if (wakeTimeNanos != 0) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700739 if (!mAudioEndpoint->isFreeRunning()) {
Phil Burkfd34a932017-07-19 07:03:52 -0700740 // If there is software on the other end of the FIFO then it may get delayed.
741 // So wake up just a little after we expect it to be ready.
742 wakeTimeNanos += mWakeupDelayNanos;
Phil Burk204a1632017-01-03 17:23:43 -0800743 }
Phil Burkfd34a932017-07-19 07:03:52 -0700744
Phil Burk2bc7c182017-08-28 11:45:01 -0700745 currentTimeNanos = AudioClock::getNanoseconds();
746 int64_t earliestWakeTime = currentTimeNanos + mMinimumSleepNanos;
747 // Guarantee a minimum sleep time.
748 if (wakeTimeNanos < earliestWakeTime) {
749 wakeTimeNanos = earliestWakeTime;
750 }
751
Phil Burk204a1632017-01-03 17:23:43 -0800752 if (wakeTimeNanos > deadlineNanos) {
753 // If we time out, just return the framesWritten so far.
Phil Burkcf5f6d22017-05-26 12:35:07 -0700754 // TODO remove after we fix the deadline bug
Phil Burkfbf031e2017-10-12 15:58:31 -0700755 ALOGW("processData(): entered at %lld nanos, currently %lld",
Phil Burkec89b2e2017-06-20 15:05:06 -0700756 (long long) entryTimeNanos, (long long) currentTimeNanos);
Phil Burkfbf031e2017-10-12 15:58:31 -0700757 ALOGW("processData(): TIMEOUT after %lld nanos",
Phil Burkc0c70e32017-02-09 13:18:38 -0800758 (long long) timeoutNanoseconds);
Phil Burkfbf031e2017-10-12 15:58:31 -0700759 ALOGW("processData(): wakeTime = %lld, deadline = %lld nanos",
Phil Burk87c9f642017-05-17 07:22:39 -0700760 (long long) wakeTimeNanos, (long long) deadlineNanos);
Phil Burkfbf031e2017-10-12 15:58:31 -0700761 ALOGW("processData(): past deadline by %d micros",
Phil Burk87c9f642017-05-17 07:22:39 -0700762 (int)((wakeTimeNanos - deadlineNanos) / AAUDIO_NANOS_PER_MICROSECOND));
Phil Burkec89b2e2017-06-20 15:05:06 -0700763 mClockModel.dump();
Phil Burk5edc4ea2020-04-17 08:15:42 -0700764 mAudioEndpoint->dump();
Phil Burk204a1632017-01-03 17:23:43 -0800765 break;
766 }
767
Phil Burkfd34a932017-07-19 07:03:52 -0700768 if (ATRACE_ENABLED()) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700769 int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
Phil Burkfd34a932017-07-19 07:03:52 -0700770 ATRACE_INT(fifoName, fullFrames);
771 int64_t sleepForNanos = wakeTimeNanos - currentTimeNanos;
772 ATRACE_INT("aaSlpNs", (int32_t)sleepForNanos);
773 }
774
775 AudioClock::sleepUntilNanoTime(wakeTimeNanos);
Phil Burk204a1632017-01-03 17:23:43 -0800776 currentTimeNanos = AudioClock::getNanoseconds();
777 }
778 }
779
Phil Burkfd34a932017-07-19 07:03:52 -0700780 if (ATRACE_ENABLED()) {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700781 int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
Phil Burkfd34a932017-07-19 07:03:52 -0700782 ATRACE_INT(fifoName, fullFrames);
783 }
784
Phil Burk87c9f642017-05-17 07:22:39 -0700785 // return error or framesProcessed
Phil Burkc0c70e32017-02-09 13:18:38 -0800786 (void) loopCount;
Phil Burk4485d412017-05-09 15:55:02 -0700787 ATRACE_END();
Phil Burk204a1632017-01-03 17:23:43 -0800788 return (result < 0) ? result : numFrames - framesLeft;
789}
790
Phil Burk3316d5e2017-02-15 11:23:01 -0800791void AudioStreamInternal::processTimestamp(uint64_t position, int64_t time) {
Phil Burk87c9f642017-05-17 07:22:39 -0700792 mClockModel.processTimestamp(position, time);
Phil Burk204a1632017-01-03 17:23:43 -0800793}
794
Phil Burk3316d5e2017-02-15 11:23:01 -0800795aaudio_result_t AudioStreamInternal::setBufferSize(int32_t requestedFrames) {
Phil Burk6479d502017-11-20 09:32:52 -0800796 int32_t adjustedFrames = requestedFrames;
Phil Burk8d97b8e2020-09-25 23:18:14 +0000797 const int32_t maximumSize = getBufferCapacity() - getFramesPerBurst();
Phil Burk5347dca2020-04-08 16:31:07 -0700798 // Minimum size should be a multiple number of bursts.
Phil Burk8d97b8e2020-09-25 23:18:14 +0000799 const int32_t minimumSize = 1 * getFramesPerBurst();
Phil Burk6479d502017-11-20 09:32:52 -0800800
801 // Clip to minimum size so that rounding up will work better.
Phil Burk8d4f0062019-10-03 15:55:41 -0700802 adjustedFrames = std::max(minimumSize, adjustedFrames);
Phil Burk71f35bb2017-04-13 16:05:07 -0700803
Phil Burk8d4f0062019-10-03 15:55:41 -0700804 // Prevent arithmetic overflow by clipping before we round.
805 if (adjustedFrames >= maximumSize) {
Phil Burk6479d502017-11-20 09:32:52 -0800806 adjustedFrames = maximumSize;
807 } else {
808 // Round to the next highest burst size.
Phil Burk8d97b8e2020-09-25 23:18:14 +0000809 int32_t numBursts = (adjustedFrames + getFramesPerBurst() - 1) / getFramesPerBurst();
810 adjustedFrames = numBursts * getFramesPerBurst();
811 // Clip just in case maximumSize is not a multiple of getFramesPerBurst().
Phil Burk5347dca2020-04-08 16:31:07 -0700812 adjustedFrames = std::min(maximumSize, adjustedFrames);
Phil Burk6479d502017-11-20 09:32:52 -0800813 }
814
Phil Burk5edc4ea2020-04-17 08:15:42 -0700815 if (mAudioEndpoint) {
816 // Clip against the actual size from the endpoint.
817 int32_t actualFrames = 0;
818 // Set to maximum size so we can write extra data when ready in order to reduce glitches.
819 // The amount we keep in the buffer is controlled by mBufferSizeInFrames.
820 mAudioEndpoint->setBufferSizeInFrames(maximumSize, &actualFrames);
821 // actualFrames should be <= actual maximum size of endpoint
822 adjustedFrames = std::min(actualFrames, adjustedFrames);
823 }
Phil Burk8d4f0062019-10-03 15:55:41 -0700824
Phil Burk64e16a72020-06-01 13:25:51 -0700825 if (adjustedFrames != mBufferSizeInFrames) {
826 android::mediametrics::LogItem(mMetricsId)
827 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
828 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, adjustedFrames)
829 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getXRunCount())
830 .record();
831 }
832
Phil Burk8d4f0062019-10-03 15:55:41 -0700833 mBufferSizeInFrames = adjustedFrames;
Phil Burk6c63ae32019-10-28 10:28:21 -0700834 ALOGV("%s(%d) returns %d", __func__, requestedFrames, adjustedFrames);
Phil Burk8d4f0062019-10-03 15:55:41 -0700835 return (aaudio_result_t) adjustedFrames;
Phil Burk204a1632017-01-03 17:23:43 -0800836}
837
Phil Burk87c9f642017-05-17 07:22:39 -0700838int32_t AudioStreamInternal::getBufferSize() const {
Phil Burk8d4f0062019-10-03 15:55:41 -0700839 return mBufferSizeInFrames;
Phil Burk204a1632017-01-03 17:23:43 -0800840}
841
Phil Burk87c9f642017-05-17 07:22:39 -0700842int32_t AudioStreamInternal::getBufferCapacity() const {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700843 return mBufferCapacityInFrames;
Phil Burk204a1632017-01-03 17:23:43 -0800844}
845
Phil Burk377c1c22018-12-12 16:06:54 -0800846bool AudioStreamInternal::isClockModelInControl() const {
Phil Burk5edc4ea2020-04-17 08:15:42 -0700847 return isActive() && mAudioEndpoint->isFreeRunning() && mClockModel.isRunning();
Phil Burk377c1c22018-12-12 16:06:54 -0800848}