blob: 336af363939ec1f91000402cacd462f8cf3cee80 [file] [log] [blame]
Glenn Kasten99e53b82012-01-19 08:59:58 -08001/*
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9** http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080018//#define LOG_NDEBUG 0
19#define LOG_TAG "AudioTrack"
20
Mark Salyzyn34fb2962014-06-18 16:30:56 -070021#include <inttypes.h>
Glenn Kastenc56f3422014-03-21 17:53:17 -070022#include <math.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080023#include <sys/resource.h>
Andy Hung959b5b82021-09-24 10:46:20 -070024#include <thread>
Mark Salyzyn34fb2962014-06-18 16:30:56 -070025
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -080026#include <android/media/IAudioPolicyService.h>
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -070027#include <android-base/macros.h>
Andy Hung3acde2c2021-11-11 09:18:08 -080028#include <android-base/stringprintf.h>
Andy Hung2b01f002017-07-05 12:01:36 -070029#include <audio_utils/clock.h>
Glenn Kasten9f80dd22012-12-18 15:57:32 -080030#include <audio_utils/primitives.h>
31#include <binder/IPCThreadState.h>
Vlad Popaad0fe922022-06-10 00:43:14 +020032#include <binder/IServiceManager.h>
Glenn Kasten9f80dd22012-12-18 15:57:32 -080033#include <media/AudioTrack.h>
34#include <utils/Log.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080035#include <private/media/AudioTrackShared.h>
Suren Baghdasaryan7435e7d2018-12-19 17:09:28 -080036#include <processgroup/sched_policy.h>
Glenn Kasten1ab85ec2013-05-31 09:18:43 -070037#include <media/IAudioFlinger.h>
Dean Wheatleya70eef72018-01-04 14:23:50 +110038#include <media/AudioParameter.h>
Andy Hungcd044842014-08-07 11:04:34 -070039#include <media/AudioResamplerPublic.h>
Michael Chana94fbb22018-04-24 14:31:19 +100040#include <media/AudioSystem.h>
Ray Essickf27e9872019-12-07 06:28:46 -080041#include <media/MediaMetricsItem.h>
Ray Essicked304702017-12-12 14:00:57 -080042#include <media/TypeConverter.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080043
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +010044#define WAIT_PERIOD_MS 10
45#define WAIT_STREAM_END_TIMEOUT_SEC 120
Vlad Popaad0fe922022-06-10 00:43:14 +020046
Andy Hung53c3b5f2014-12-15 16:42:05 -080047static const int kMaxLoopCountNotifications = 32;
Vlad Popaad0fe922022-06-10 00:43:14 +020048static constexpr char kAudioServiceName[] = "audio";
Glenn Kasten511754b2012-01-11 09:52:19 -080049
Kuowei Lid4adbdb2020-08-13 14:44:25 +080050using ::android::aidl_utils::statusTFromBinderStatus;
Andy Hung3acde2c2021-11-11 09:18:08 -080051using ::android::base::StringPrintf;
Kuowei Lid4adbdb2020-08-13 14:44:25 +080052
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080053namespace android {
Chia-chi Yeh33005a92010-06-16 06:33:13 +080054// ---------------------------------------------------------------------------
55
Ivan Lozano8cf3a072017-08-09 09:01:33 -070056using media::VolumeShaper;
Svet Ganov3e5f14f2021-05-13 22:51:08 +000057using android::content::AttributionSourceState;
Ivan Lozano8cf3a072017-08-09 09:01:33 -070058
Andy Hunga7f03352015-05-31 21:54:49 -070059// TODO: Move to a separate .h
60
Andy Hung4ede21d2014-12-12 15:37:34 -080061template <typename T>
Andy Hunga7f03352015-05-31 21:54:49 -070062static inline const T &min(const T &x, const T &y) {
Andy Hung4ede21d2014-12-12 15:37:34 -080063 return x < y ? x : y;
64}
65
Andy Hunga7f03352015-05-31 21:54:49 -070066template <typename T>
67static inline const T &max(const T &x, const T &y) {
68 return x > y ? x : y;
69}
70
71static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed)
72{
73 return ((double)frames * 1000000000) / ((double)sampleRate * speed);
74}
75
Andy Hung7f1bc8a2014-09-12 14:43:11 -070076static int64_t convertTimespecToUs(const struct timespec &tv)
77{
Chih-Hung Hsieh50fa06c2018-12-11 13:51:36 -080078 return tv.tv_sec * 1000000LL + tv.tv_nsec / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -070079}
80
Andy Hungffa36952017-08-17 10:41:51 -070081// TODO move to audio_utils.
82static inline struct timespec convertNsToTimespec(int64_t ns) {
83 struct timespec tv;
84 tv.tv_sec = static_cast<time_t>(ns / NANOS_PER_SECOND);
Andy Hung06a730b2020-04-09 13:28:31 -070085 tv.tv_nsec = static_cast<int64_t>(ns % NANOS_PER_SECOND);
Andy Hungffa36952017-08-17 10:41:51 -070086 return tv;
87}
88
Andy Hung7f1bc8a2014-09-12 14:43:11 -070089// current monotonic time in microseconds.
90static int64_t getNowUs()
91{
92 struct timespec tv;
93 (void) clock_gettime(CLOCK_MONOTONIC, &tv);
94 return convertTimespecToUs(tv);
95}
96
Andy Hung26145642015-04-15 21:56:53 -070097// FIXME: we don't use the pitch setting in the time stretcher (not working);
98// instead we emulate it using our sample rate converter.
99static const bool kFixPitch = true; // enable pitch fix
100static inline uint32_t adjustSampleRate(uint32_t sampleRate, float pitch)
101{
102 return kFixPitch ? (sampleRate * pitch + 0.5) : sampleRate;
103}
104
105static inline float adjustSpeed(float speed, float pitch)
106{
Ricardo Garcia6c7f0622015-04-30 18:39:16 -0700107 return kFixPitch ? speed / max(pitch, AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) : speed;
Andy Hung26145642015-04-15 21:56:53 -0700108}
109
110static inline float adjustPitch(float pitch)
111{
112 return kFixPitch ? AUDIO_TIMESTRETCH_PITCH_NORMAL : pitch;
113}
114
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800115// static
116status_t AudioTrack::getMinFrameCount(
Glenn Kastene33054e2012-11-14 12:54:39 -0800117 size_t* frameCount,
Glenn Kastenfff6d712012-01-12 16:38:12 -0800118 audio_stream_type_t streamType,
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800119 uint32_t sampleRate)
120{
Glenn Kastend65d73c2012-06-22 17:21:07 -0700121 if (frameCount == NULL) {
122 return BAD_VALUE;
123 }
Glenn Kasten04cd0182012-06-25 11:49:27 -0700124
Andy Hung0e48d252015-01-26 11:43:15 -0800125 // FIXME handle in server, like createTrack_l(), possible missing info:
Glenn Kastene0fa4672012-04-24 14:35:14 -0700126 // audio_io_handle_t output
127 // audio_format_t format
128 // audio_channel_mask_t channelMask
Andy Hung0e48d252015-01-26 11:43:15 -0800129 // audio_output_flags_t flags (FAST)
Glenn Kasten3b16c762012-11-14 08:44:39 -0800130 uint32_t afSampleRate;
Glenn Kasten66a04672014-01-08 08:53:44 -0800131 status_t status;
132 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
133 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700134 ALOGE("%s(): Unable to query output sample rate for stream type %d; status %d",
135 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800136 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800137 }
Glenn Kastene33054e2012-11-14 12:54:39 -0800138 size_t afFrameCount;
Glenn Kasten66a04672014-01-08 08:53:44 -0800139 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
140 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700141 ALOGE("%s(): Unable to query output frame count for stream type %d; status %d",
142 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800143 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800144 }
145 uint32_t afLatency;
Glenn Kasten66a04672014-01-08 08:53:44 -0800146 status = AudioSystem::getOutputLatency(&afLatency, streamType);
147 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700148 ALOGE("%s(): Unable to query output latency for stream type %d; status %d",
149 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800150 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800151 }
152
Andy Hung8edb8dc2015-03-26 19:13:55 -0700153 // When called from createTrack, speed is 1.0f (normal speed).
154 // This is rechecked again on setting playback rate (TODO: on setting sample rate, too).
Eric Laurent21da6472017-11-09 16:29:26 -0800155 *frameCount = AudioSystem::calculateMinFrameCount(afLatency, afFrameCount, afSampleRate,
156 sampleRate, 1.0f /*, 0 notificationsPerBufferReq*/);
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800157
Andy Hung0e48d252015-01-26 11:43:15 -0800158 // The formula above should always produce a non-zero value under normal circumstances:
159 // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX.
160 // Return error in the unlikely event that it does not, as that's part of the API contract.
Glenn Kasten66a04672014-01-08 08:53:44 -0800161 if (*frameCount == 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700162 ALOGE("%s(): failed for streamType %d, sampleRate %u",
163 __func__, streamType, sampleRate);
Glenn Kasten66a04672014-01-08 08:53:44 -0800164 return BAD_VALUE;
165 }
Andy Hungfb8ede22018-09-12 19:03:24 -0700166 ALOGV("%s(): getMinFrameCount=%zu: afFrameCount=%zu, afSampleRate=%u, afLatency=%u",
167 __func__, *frameCount, afFrameCount, afSampleRate, afLatency);
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800168 return NO_ERROR;
169}
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800170
Michael Chana94fbb22018-04-24 14:31:19 +1000171// static
172bool AudioTrack::isDirectOutputSupported(const audio_config_base_t& config,
173 const audio_attributes_t& attributes) {
174 ALOGV("%s()", __FUNCTION__);
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800175 const sp<media::IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
Michael Chana94fbb22018-04-24 14:31:19 +1000176 if (aps == 0) return false;
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800177
178 auto result = [&]() -> ConversionResult<bool> {
Mikhail Naganovdbf03642021-08-25 18:15:32 -0700179 media::audio::common::AudioConfigBase configAidl = VALUE_OR_RETURN(
Mikhail Naganov9dec7012021-07-21 10:30:57 -0700180 legacy2aidl_audio_config_base_t_AudioConfigBase(config, false /*isInput*/));
Mikhail Naganov1c400902023-05-17 11:48:43 -0700181 media::audio::common::AudioAttributes attributesAidl = VALUE_OR_RETURN(
182 legacy2aidl_audio_attributes_t_AudioAttributes(attributes));
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800183 bool retAidl;
184 RETURN_IF_ERROR(aidl_utils::statusTFromBinderStatus(
185 aps->isDirectOutputSupported(configAidl, attributesAidl, &retAidl)));
186 return retAidl;
187 }();
188 return result.value_or(false);
Michael Chana94fbb22018-04-24 14:31:19 +1000189}
190
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800191// ---------------------------------------------------------------------------
192
Ray Essicked304702017-12-12 14:00:57 -0800193void AudioTrack::MediaMetrics::gather(const AudioTrack *track)
194{
Ray Essick88394302018-01-24 14:52:05 -0800195 // only if we're in a good state...
196 // XXX: shall we gather alternative info if failing?
197 const status_t lstatus = track->initCheck();
198 if (lstatus != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700199 ALOGD("%s(): no metrics gathered, track status=%d", __func__, (int) lstatus);
Ray Essick88394302018-01-24 14:52:05 -0800200 return;
201 }
202
Andy Hungd0979812019-02-21 15:51:44 -0800203#define MM_PREFIX "android.media.audiotrack." // avoid cut-n-paste errors.
Ray Essicked304702017-12-12 14:00:57 -0800204
Andy Hungde602302021-12-07 21:35:49 -0800205 // Do not change this without changing the MediaMetricsService side.
Andy Hungd0979812019-02-21 15:51:44 -0800206 // Java API 28 entries, do not change.
Ray Essickf27e9872019-12-07 06:28:46 -0800207 mMetricsItem->setCString(MM_PREFIX "streamtype", toString(track->streamType()).c_str());
208 mMetricsItem->setCString(MM_PREFIX "type",
Andy Hungd0979812019-02-21 15:51:44 -0800209 toString(track->mAttributes.content_type).c_str());
Ray Essickf27e9872019-12-07 06:28:46 -0800210 mMetricsItem->setCString(MM_PREFIX "usage", toString(track->mAttributes.usage).c_str());
Ray Essicked304702017-12-12 14:00:57 -0800211
Andy Hungd0979812019-02-21 15:51:44 -0800212 // Non-API entries, these can change due to a Java string mistake.
Ray Essickf27e9872019-12-07 06:28:46 -0800213 mMetricsItem->setInt32(MM_PREFIX "sampleRate", (int32_t)track->mSampleRate);
214 mMetricsItem->setInt64(MM_PREFIX "channelMask", (int64_t)track->mChannelMask);
Andy Hungd0979812019-02-21 15:51:44 -0800215 // Non-API entries, these can change.
Ray Essickf27e9872019-12-07 06:28:46 -0800216 mMetricsItem->setInt32(MM_PREFIX "portId", (int32_t)track->mPortId);
217 mMetricsItem->setCString(MM_PREFIX "encoding", toString(track->mFormat).c_str());
218 mMetricsItem->setInt32(MM_PREFIX "frameCount", (int32_t)track->mFrameCount);
219 mMetricsItem->setCString(MM_PREFIX "attributes", toString(track->mAttributes).c_str());
Andy Hung6f451f02021-02-24 21:53:29 -0800220 mMetricsItem->setCString(MM_PREFIX "logSessionId", track->mLogSessionId.c_str());
Andy Hungde602302021-12-07 21:35:49 -0800221 mMetricsItem->setInt32(MM_PREFIX "underrunFrames", (int32_t)track->getUnderrunFrames());
Ray Essicked304702017-12-12 14:00:57 -0800222}
223
Ray Essick88394302018-01-24 14:52:05 -0800224// hand the user a snapshot of the metrics.
Ray Essickf27e9872019-12-07 06:28:46 -0800225status_t AudioTrack::getMetrics(mediametrics::Item * &item)
Ray Essick88394302018-01-24 14:52:05 -0800226{
227 mMediaMetrics.gather(this);
Ray Essickf27e9872019-12-07 06:28:46 -0800228 mediametrics::Item *tmp = mMediaMetrics.dup();
Ray Essick88394302018-01-24 14:52:05 -0800229 if (tmp == nullptr) {
230 return BAD_VALUE;
231 }
232 item = tmp;
233 return NO_ERROR;
234}
Ray Essicked304702017-12-12 14:00:57 -0800235
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000236AudioTrack::AudioTrack(const AttributionSourceState& attributionSource)
Andy Hung4521b9b2024-04-11 19:01:28 -0700237 : mClientAttributionSource(attributionSource)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800238{
239}
240
241AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800242 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800243 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800244 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700245 audio_channel_mask_t channelMask,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800246 size_t frameCount,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700247 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400248 const wp<IAudioTrackCallback> & callback,
Atneyaf86d2692021-10-14 14:02:36 -0400249 int32_t notificationFrames,
250 audio_session_t sessionId,
251 transfer_type transferType,
252 const audio_offload_info_t *offloadInfo,
253 const AttributionSourceState& attributionSource,
254 const audio_attributes_t* pAttributes,
255 bool doNotReconnect,
256 float maxRequiredSpeed,
257 audio_port_handle_t selectedDeviceId)
Atneyaf86d2692021-10-14 14:02:36 -0400258{
Andy Hung4521b9b2024-04-11 19:01:28 -0700259 mSetParams = std::make_unique<SetParams>(
260 streamType, sampleRate, format, channelMask, frameCount, flags, callback,
261 notificationFrames, nullptr /*sharedBuffer*/, false /*threadCanCallJava*/,
262 sessionId, transferType, offloadInfo, attributionSource, pAttributes,
263 doNotReconnect, maxRequiredSpeed, selectedDeviceId);
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400264}
265
266namespace {
267 class LegacyCallbackWrapper : public AudioTrack::IAudioTrackCallback {
268 const AudioTrack::legacy_callback_t mCallback;
269 void * const mData;
270 public:
271 LegacyCallbackWrapper(AudioTrack::legacy_callback_t callback, void* user)
272 : mCallback(callback), mData(user) {}
273 size_t onMoreData(const AudioTrack::Buffer & buffer) override {
274 AudioTrack::Buffer copy = buffer;
275 mCallback(AudioTrack::EVENT_MORE_DATA, mData, static_cast<void*>(&copy));
Atneya Nair03079272022-01-18 17:03:14 -0500276 return copy.size();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400277 }
278 void onUnderrun() override {
279 mCallback(AudioTrack::EVENT_UNDERRUN, mData, nullptr);
280 }
281 void onLoopEnd(int32_t loopsRemaining) override {
282 mCallback(AudioTrack::EVENT_LOOP_END, mData, &loopsRemaining);
283 }
284 void onMarker(uint32_t markerPosition) override {
285 mCallback(AudioTrack::EVENT_MARKER, mData, &markerPosition);
286 }
287 void onNewPos(uint32_t newPos) override {
288 mCallback(AudioTrack::EVENT_NEW_POS, mData, &newPos);
289 }
290 void onBufferEnd() override {
291 mCallback(AudioTrack::EVENT_BUFFER_END, mData, nullptr);
292 }
293 void onNewIAudioTrack() override {
294 mCallback(AudioTrack::EVENT_NEW_IAUDIOTRACK, mData, nullptr);
295 }
296 void onStreamEnd() override {
297 mCallback(AudioTrack::EVENT_STREAM_END, mData, nullptr);
298 }
299 size_t onCanWriteMoreData(const AudioTrack::Buffer & buffer) override {
300 AudioTrack::Buffer copy = buffer;
301 mCallback(AudioTrack::EVENT_CAN_WRITE_MORE_DATA, mData, static_cast<void*>(&copy));
Atneya Nair03079272022-01-18 17:03:14 -0500302 return copy.size();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400303 }
304 };
305}
Andreas Huberc8139852012-01-18 10:51:55 -0800306AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800307 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800308 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800309 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700310 audio_channel_mask_t channelMask,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800311 const sp<IMemory>& sharedBuffer,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700312 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400313 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700314 int32_t notificationFrames,
Glenn Kastend848eb42016-03-08 13:42:11 -0800315 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000316 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800317 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000318 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700319 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700320 bool doNotReconnect,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700321 float maxRequiredSpeed)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800322{
François Gaffie393f0e02019-04-10 09:09:08 +0200323 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Ippei Murata5fa32ed2018-10-01 17:37:50 +0900324
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500325 mSetParams = std::unique_ptr<SetParams>{
326 new SetParams{streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags,
327 callback, notificationFrames, sharedBuffer, false /*threadCanCallJava*/,
328 sessionId, transferType, offloadInfo, attributionSource, pAttributes,
329 doNotReconnect, maxRequiredSpeed, AUDIO_PORT_HANDLE_NONE}};
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800330}
331
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500332void AudioTrack::onFirstRef() {
333 if (mSetParams) {
334 set(*mSetParams);
335 mSetParams.reset();
336 }
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400337}
338
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800339AudioTrack::~AudioTrack()
340{
Ray Essicked304702017-12-12 14:00:57 -0800341 // pull together the numbers, before we clean up our structures
342 mMediaMetrics.gather(this);
343
Andy Hungb68f5eb2019-12-03 16:49:17 -0800344 mediametrics::LogItem(mMetricsId)
345 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DTOR)
Phil Burkd3813f32020-04-23 16:26:15 -0700346 .set(AMEDIAMETRICS_PROP_CALLERNAME,
347 mCallerName.empty()
Andy Hunga6b27032020-04-27 10:34:24 -0700348 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
Phil Burkd3813f32020-04-23 16:26:15 -0700349 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800350 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
351 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)mStatus)
352 .record();
353
Phil Burk7a9577c2021-03-12 20:12:11 +0000354 stopAndJoinCallbacks(); // checks mStatus
355
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800356 if (mStatus == NO_ERROR) {
Marco Nelissenf8880202014-11-14 07:58:25 -0800357 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten53cec222013-08-29 09:01:02 -0700358 mAudioTrack.clear();
Eric Laurent3bcffa12014-06-12 18:38:45 -0700359 mCblkMemory.clear();
360 mSharedBuffer.clear();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800361 IPCThreadState::self()->flushCommands();
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000362 pid_t clientPid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
Andy Hungfb8ede22018-09-12 19:03:24 -0700363 ALOGV("%s(%d), releasing session id %d from %d on behalf of %d",
Eric Laurent973db022018-11-20 14:54:31 -0800364 __func__, mPortId,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700365 mSessionId, IPCThreadState::self()->getCallingPid(), clientPid);
366 AudioSystem::releaseAudioSessionId(mSessionId, clientPid);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800367 }
368}
369
Phil Burk7a9577c2021-03-12 20:12:11 +0000370void AudioTrack::stopAndJoinCallbacks() {
Phil Burk7a9577c2021-03-12 20:12:11 +0000371 // Make sure that callback function exits in the case where
372 // it is looping on buffer full condition in obtainBuffer().
373 // Otherwise the callback thread will never exit.
374 stop();
375 if (mAudioTrackThread != 0) { // not thread safe
Phil Burk7a9577c2021-03-12 20:12:11 +0000376 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
Kuowei Lifdf8e0f2021-11-26 17:38:34 +0800377 mProxy->interrupt();
Phil Burk7a9577c2021-03-12 20:12:11 +0000378 mAudioTrackThread->requestExitAndWait();
379 mAudioTrackThread.clear();
380 }
zhenjun.zhang116df6a2021-12-08 09:17:13 +0800381
382 AutoMutex lock(mLock);
Phil Burk7a9577c2021-03-12 20:12:11 +0000383 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) {
384 // This may not stop all of these device callbacks!
385 // TODO: Add some sort of protection.
386 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
387 mDeviceCallback.clear();
388 }
389}
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400390status_t AudioTrack::set(
391 audio_stream_type_t streamType,
392 uint32_t sampleRate,
393 audio_format_t format,
394 audio_channel_mask_t channelMask,
395 size_t frameCount,
396 audio_output_flags_t flags,
397 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700398 int32_t notificationFrames,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800399 const sp<IMemory>& sharedBuffer,
Eric Laurentbe916aa2010-06-01 23:49:17 -0700400 bool threadCanCallJava,
Glenn Kastend848eb42016-03-08 13:42:11 -0800401 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000402 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800403 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000404 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700405 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700406 bool doNotReconnect,
jiabin156c6872017-10-06 09:47:15 -0700407 float maxRequiredSpeed,
408 audio_port_handle_t selectedDeviceId)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800409{
Atneya Nair14aabae2021-11-30 17:36:24 -0500410 LOG_ALWAYS_FATAL_IF(mInitialized, "%s: should not be called twice", __func__);
411 mInitialized = true;
Eric Laurentf32d7812017-11-30 14:44:07 -0800412 status_t status;
413 uint32_t channelCount;
414 pid_t callingPid;
415 pid_t myPid;
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000416 uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
417 pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
Andy Hung3acde2c2021-11-11 09:18:08 -0800418 std::string errorMessage;
Eric Laurent973db022018-11-20 14:54:31 -0800419 // Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
Andy Hungfb8ede22018-09-12 19:03:24 -0700420 ALOGV("%s(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
wanggang1471f644f2022-07-08 11:10:20 +0800421 "flags %#x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
Andy Hungfb8ede22018-09-12 19:03:24 -0700422 __func__,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800423 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000424 sessionId, transferType, attributionSource.uid, attributionSource.pid);
Glenn Kasten86f04662014-02-24 15:13:05 -0800425
Phil Burk33ff89b2015-11-30 11:16:01 -0800426 mThreadCanCallJava = threadCanCallJava;
Andy Hungde602302021-12-07 21:35:49 -0800427
428 // These variables are pulled in an error report, so we initialize them early.
jiabin156c6872017-10-06 09:47:15 -0700429 mSelectedDeviceId = selectedDeviceId;
Eric Laurent21da6472017-11-09 16:29:26 -0800430 mSessionId = sessionId;
Andy Hungde602302021-12-07 21:35:49 -0800431 mChannelMask = channelMask;
Andy Hungde602302021-12-07 21:35:49 -0800432 mReqFrameCount = mFrameCount = frameCount;
433 mSampleRate = sampleRate;
434 mOriginalSampleRate = sampleRate;
435 mAttributes = pAttributes != nullptr ? *pAttributes : AUDIO_ATTRIBUTES_INITIALIZER;
436 mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
Phil Burk33ff89b2015-11-30 11:16:01 -0800437
Eric Laurentd7f33c52022-01-06 13:54:56 +0100438 // update format and flags before storing them in mFormat, mOrigFlags and mFlags
439 if (pAttributes != NULL) {
440 // stream type shouldn't be looked at, this track has audio attributes
441 ALOGV("%s(): Building AudioTrack with attributes:"
442 " usage=%d content=%d flags=0x%x tags=[%s]",
443 __func__,
444 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
445 audio_flags_to_audio_output_flags(mAttributes.flags, &flags);
446 }
447
448 // these below should probably come from the audioFlinger too...
449 if (format == AUDIO_FORMAT_DEFAULT) {
450 format = AUDIO_FORMAT_PCM_16_BIT;
451 } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
452 flags = static_cast<audio_output_flags_t>(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
453 }
454
455 // force direct flag if format is not linear PCM
456 // or offload was requested
457 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
458 || !audio_is_linear_pcm(format)) {
459 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
460 ? "%s(): Offload request, forcing to Direct Output"
461 : "%s(): Not linear PCM, forcing to Direct Output",
462 __func__);
463 flags = (audio_output_flags_t)
464 // FIXME why can't we allow direct AND fast?
465 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
466 }
467
468 // force direct flag if HW A/V sync requested
469 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
470 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
471 }
472
473 mFormat = format;
474 mOrigFlags = mFlags = flags;
475
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800476 switch (transferType) {
477 case TRANSFER_DEFAULT:
478 if (sharedBuffer != 0) {
479 transferType = TRANSFER_SHARED;
Atneya Nairba809b82022-03-04 18:11:10 -0500480 } else if (callback == nullptr|| threadCanCallJava) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800481 transferType = TRANSFER_SYNC;
482 } else {
483 transferType = TRANSFER_CALLBACK;
484 }
485 break;
486 case TRANSFER_CALLBACK:
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700487 case TRANSFER_SYNC_NOTIF_CALLBACK:
Atneya Nairba809b82022-03-04 18:11:10 -0500488 if (callback == nullptr || sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800489 errorMessage = StringPrintf(
490 "%s: Transfer type %s but callback == nullptr || sharedBuffer != 0",
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700491 convertTransferToText(transferType), __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800492 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800493 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800494 }
495 break;
496 case TRANSFER_OBTAIN:
497 case TRANSFER_SYNC:
498 if (sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800499 errorMessage = StringPrintf(
500 "%s: Transfer type TRANSFER_OBTAIN but sharedBuffer != 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800501 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800502 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800503 }
504 break;
505 case TRANSFER_SHARED:
506 if (sharedBuffer == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800507 errorMessage = StringPrintf(
508 "%s: Transfer type TRANSFER_SHARED but sharedBuffer == 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800509 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800510 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800511 }
512 break;
513 default:
Andy Hung3acde2c2021-11-11 09:18:08 -0800514 errorMessage = StringPrintf("%s: Invalid transfer type %d", __func__, transferType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800515 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800516 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800517 }
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800518 mSharedBuffer = sharedBuffer;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800519 mTransfer = transferType;
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700520 mDoNotReconnect = doNotReconnect;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800521
Andy Hungfb8ede22018-09-12 19:03:24 -0700522 ALOGV_IF(sharedBuffer != 0, "%s(): sharedBuffer: %p, size: %zu",
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -0700523 __func__, sharedBuffer->unsecurePointer(), sharedBuffer->size());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800524
Glenn Kasten53cec222013-08-29 09:01:02 -0700525 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Eric Laurent1dd70b92009-04-21 07:56:33 -0700526 if (mAudioTrack != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800527 errorMessage = StringPrintf("%s: Track already in use", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800528 status = INVALID_OPERATION;
Andy Hung3acde2c2021-11-11 09:18:08 -0800529 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800530 }
531
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800532 // handle default values first.
Eric Laurente83b55d2014-11-14 10:06:21 -0800533 if (streamType == AUDIO_STREAM_DEFAULT) {
Dima Zavinfce7a472011-04-19 22:30:36 -0700534 streamType = AUDIO_STREAM_MUSIC;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800535 }
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700536 if (pAttributes == NULL) {
Eric Laurent223fd5c2014-11-11 13:43:36 -0800537 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800538 errorMessage = StringPrintf("%s: Invalid stream type %d", __func__, streamType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800539 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800540 goto error;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700541 }
Andy Hunga2159aa2021-07-20 13:01:52 -0700542 mOriginalStreamType = streamType;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700543 } else {
Andy Hunga2159aa2021-07-20 13:01:52 -0700544 mOriginalStreamType = AUDIO_STREAM_DEFAULT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800545 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800546
547 // validate parameters
Dima Zavinfce7a472011-04-19 22:30:36 -0700548 if (!audio_is_valid_format(format)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800549 errorMessage = StringPrintf("%s: Invalid format %#x", __func__, format);
Eric Laurentf32d7812017-11-30 14:44:07 -0800550 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800551 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800552 }
Eric Laurentc2f1f072009-07-17 12:17:14 -0700553
Glenn Kasten8ba90322013-10-30 11:29:27 -0700554 if (!audio_is_output_channel(channelMask)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800555 errorMessage = StringPrintf("%s: Invalid channel mask %#x", __func__, channelMask);
Eric Laurentf32d7812017-11-30 14:44:07 -0800556 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800557 goto error;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700558 }
Eric Laurentf32d7812017-11-30 14:44:07 -0800559 channelCount = audio_channel_count_from_out_mask(channelMask);
Glenn Kastene3247bf2014-02-24 15:19:07 -0800560 mChannelCount = channelCount;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700561
Dean Wheatleyd883e302023-10-20 06:11:43 +1100562 if (!(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
Glenn Kastenb7730382014-04-30 15:50:31 -0700563 // createTrack will return an error if PCM format is not supported by server,
564 // so no need to check for specific PCM formats here
Dean Wheatleyd883e302023-10-20 06:11:43 +1100565 ALOGW_IF(!audio_has_proportional_frames(format), "%s(): no direct flag for format 0x%x",
566 __func__, format);
Glenn Kastene3aa6592012-12-04 12:22:46 -0800567 }
Dean Wheatleyd883e302023-10-20 06:11:43 +1100568 mFrameSize = audio_bytes_per_frame(channelCount, format);
Glenn Kastene3aa6592012-12-04 12:22:46 -0800569
Eric Laurent0d6db582014-11-12 18:39:44 -0800570 // sampling rate must be specified for direct outputs
Eric Laurentd7f33c52022-01-06 13:54:56 +0100571 if (sampleRate == 0 && (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800572 errorMessage = StringPrintf(
573 "%s: sample rate must be specified for direct outputs", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800574 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800575 goto error;
Eric Laurent0d6db582014-11-12 18:39:44 -0800576 }
Andy Hungff874dc2016-04-11 16:49:09 -0700577 // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
578 mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
Eric Laurent0d6db582014-11-12 18:39:44 -0800579
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800580 // Make copy of input parameter offloadInfo so that in the future:
581 // (a) createTrack_l doesn't need it as an input parameter
582 // (b) we can support re-creation of offloaded tracks
583 if (offloadInfo != NULL) {
584 mOffloadInfoCopy = *offloadInfo;
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800585 } else {
Eric Laurent20b9ef02016-12-05 11:03:16 -0800586 memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
Ytai Ben-Tsviffa2fd92020-10-20 09:13:53 -0700587 mOffloadInfoCopy = AUDIO_INFO_INITIALIZER;
ziyang zhangb3ec8aa2022-05-10 13:28:38 +0800588 mOffloadInfoCopy.format = format;
589 mOffloadInfoCopy.sample_rate = sampleRate;
590 mOffloadInfoCopy.channel_mask = channelMask;
591 mOffloadInfoCopy.stream_type = streamType;
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800592 }
593
Glenn Kasten66e46352014-01-16 17:44:23 -0800594 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
595 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
Glenn Kasten05632a52012-01-03 14:22:33 -0800596 mSendLevel = 0.0f;
Glenn Kasten396fabd2014-01-08 08:54:23 -0800597 // mFrameCount is initialized in createTrack_l
Glenn Kastenea38ee72016-04-18 11:08:01 -0700598 if (notificationFrames >= 0) {
599 mNotificationFramesReq = notificationFrames;
600 mNotificationsPerBufferReq = 0;
601 } else {
Eric Laurentd7f33c52022-01-06 13:54:56 +0100602 if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800603 errorMessage = StringPrintf(
604 "%s: notificationFrames=%d not permitted for non-fast track",
Andy Hungfb8ede22018-09-12 19:03:24 -0700605 __func__, notificationFrames);
Eric Laurentf32d7812017-11-30 14:44:07 -0800606 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800607 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700608 }
609 if (frameCount > 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700610 ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
611 __func__, notificationFrames, frameCount);
Eric Laurentf32d7812017-11-30 14:44:07 -0800612 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800613 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700614 }
615 mNotificationFramesReq = 0;
616 const uint32_t minNotificationsPerBuffer = 1;
617 const uint32_t maxNotificationsPerBuffer = 8;
618 mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
619 max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
620 ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
Andy Hungfb8ede22018-09-12 19:03:24 -0700621 "%s(): notificationFrames=%d clamped to the range -%u to -%u",
622 __func__,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700623 notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
624 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800625 mNotificationFramesAct = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700626 // TODO b/182392553: refactor or remove
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000627 mClientAttributionSource = AttributionSourceState(attributionSource);
Eric Laurentf32d7812017-11-30 14:44:07 -0800628 callingPid = IPCThreadState::self()->getCallingPid();
629 myPid = getpid();
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700630 if (uid == -1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000631 mClientAttributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700632 IPCThreadState::self()->getCallingUid()));
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800633 }
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700634 if (pid == (pid_t)-1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000635 mClientAttributionSource.pid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(callingPid));
Marco Nelissend457c972014-02-11 08:47:07 -0800636 }
Eric Laurent2beeb502010-07-16 07:43:46 -0700637 mAuxEffectId = 0;
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400638 mCallback = callback;
Eric Laurentbe916aa2010-06-01 23:49:17 -0700639
Atneya Nairba809b82022-03-04 18:11:10 -0500640 if (callback != nullptr) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400641 mAudioTrackThread = sp<AudioTrackThread>::make(*this);
Glenn Kastena997e7a2012-08-07 09:44:19 -0700642 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
Glenn Kastenbfd31842015-03-20 09:01:44 -0700643 // thread begins in paused state, and will not reference us until start()
Glenn Kastena997e7a2012-08-07 09:44:19 -0700644 }
645
Eric Laurent34f1d8e2009-11-04 08:27:26 -0800646 // create the IAudioTrack
Francois Gaffie24a9fb02019-01-18 17:51:34 +0100647 {
648 AutoMutex lock(mLock);
649 status = createTrack_l();
650 }
Glenn Kastena997e7a2012-08-07 09:44:19 -0700651 if (status != NO_ERROR) {
652 if (mAudioTrackThread != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100653 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
654 mAudioTrackThread->requestExitAndWait();
Glenn Kastena997e7a2012-08-07 09:44:19 -0700655 mAudioTrackThread.clear();
656 }
Andy Hung3acde2c2021-11-11 09:18:08 -0800657 // We do not goto error to prevent double-logging errors.
Eric Laurentf32d7812017-11-30 14:44:07 -0800658 goto exit;
Glenn Kasten5d464eb2012-06-22 17:19:53 -0700659 }
660
Andy Hung4ede21d2014-12-12 15:37:34 -0800661 mLoopCount = 0;
662 mLoopStart = 0;
663 mLoopEnd = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -0800664 mLoopCountNotified = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800665 mMarkerPosition = 0;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -0700666 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800667 mNewPosition = 0;
668 mUpdatePeriod = 0;
Glenn Kasten200092b2014-08-15 15:13:30 -0700669 mPosition = 0;
670 mReleased = 0;
Andy Hungffa36952017-08-17 10:41:51 -0700671 mStartNs = 0;
672 mStartFromZeroUs = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700673 AudioSystem::acquireAudioSessionId(mSessionId, pid, uid);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800674 mSequence = 1;
675 mObservedSequence = mSequence;
676 mInUnderrun = false;
Phil Burk1b420972015-04-22 10:52:21 -0700677 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700678 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700679 mTimestampRetrogradePositionReported = false;
680 mTimestampRetrogradeTimeReported = false;
681 mTimestampStallReported = false;
682 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700683 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Andy Hung65ffdfc2016-10-10 15:52:11 -0700684 mStartTs.mPosition = 0;
Phil Burk2812d9e2016-01-04 10:34:30 -0800685 mUnderrunCountOffset = 0;
Andy Hungea2b9c02016-02-12 17:06:53 -0800686 mFramesWritten = 0;
687 mFramesWrittenServerOffset = 0;
Andy Hungf20a4e92016-08-15 19:10:34 -0700688 mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
Ivan Lozano8cf3a072017-08-09 09:01:33 -0700689 mVolumeHandler = new media::VolumeHandler();
Eric Laurentf32d7812017-11-30 14:44:07 -0800690
Andy Hung3acde2c2021-11-11 09:18:08 -0800691error:
692 if (status != NO_ERROR) {
693 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
694 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
695 }
696 // fall through
Eric Laurentf32d7812017-11-30 14:44:07 -0800697exit:
698 mStatus = status;
699 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800700}
701
Mikhail Naganov55773032020-10-01 15:08:13 -0700702
703status_t AudioTrack::set(
704 audio_stream_type_t streamType,
705 uint32_t sampleRate,
706 audio_format_t format,
707 uint32_t channelMask,
708 size_t frameCount,
709 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400710 legacy_callback_t callback,
Mikhail Naganov55773032020-10-01 15:08:13 -0700711 void* user,
712 int32_t notificationFrames,
713 const sp<IMemory>& sharedBuffer,
714 bool threadCanCallJava,
715 audio_session_t sessionId,
716 transfer_type transferType,
717 const audio_offload_info_t *offloadInfo,
718 uid_t uid,
719 pid_t pid,
720 const audio_attributes_t* pAttributes,
721 bool doNotReconnect,
722 float maxRequiredSpeed,
723 audio_port_handle_t selectedDeviceId)
724{
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000725 AttributionSourceState attributionSource;
726 attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(uid));
727 attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(pid));
728 attributionSource.token = sp<BBinder>::make();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400729 if (callback) {
730 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
731 } else if (user) {
732 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
733 }
734 return set(streamType, sampleRate, format, static_cast<audio_channel_mask_t>(channelMask),
735 frameCount, flags, mLegacyCallbackWrapper, notificationFrames, sharedBuffer,
736 threadCanCallJava, sessionId, transferType, offloadInfo, attributionSource,
737 pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
Mikhail Naganov55773032020-10-01 15:08:13 -0700738}
739
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800740// -------------------------------------------------------------------------
741
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100742status_t AudioTrack::start()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800743{
Eric Laurentf5aafb22010-11-18 08:40:16 -0800744 AutoMutex lock(mLock);
Andy Hungb68f5eb2019-12-03 16:49:17 -0800745
Andy Hung10fb4be2020-05-27 22:22:22 -0700746 if (mState == STATE_ACTIVE) {
747 return INVALID_OPERATION;
748 }
749
750 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
751
752 // Defer logging here due to OpenSL ES repeated start calls.
753 // TODO(b/154868033) after fix, restore this logging back to the beginning of start().
754 const int64_t beginNs = systemTime();
Andy Hungb68f5eb2019-12-03 16:49:17 -0800755 status_t status = NO_ERROR; // logged: make sure to set this before returning.
Andy Hung06a730b2020-04-09 13:28:31 -0700756 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800757 mediametrics::LogItem(mMetricsId)
Andy Hunga6b27032020-04-27 10:34:24 -0700758 .set(AMEDIAMETRICS_PROP_CALLERNAME,
759 mCallerName.empty()
760 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
761 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800762 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_START)
Andy Hungea840382020-05-05 21:50:17 -0700763 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800764 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
765 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
766 .record(); });
767
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800768
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800769 mInUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800770
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800771 State previousState = mState;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100772 if (previousState == STATE_PAUSED_STOPPING) {
773 mState = STATE_STOPPING;
774 } else {
775 mState = STATE_ACTIVE;
776 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700777 (void) updateAndGetPosition_l();
Andy Hung65ffdfc2016-10-10 15:52:11 -0700778
779 // save start timestamp
jiabin94ed47c2023-07-27 23:34:20 +0000780 if (isAfTrackOffloadedOrDirect_l()) {
Andy Hung65ffdfc2016-10-10 15:52:11 -0700781 if (getTimestamp_l(mStartTs) != OK) {
782 mStartTs.mPosition = 0;
783 }
784 } else {
785 if (getTimestamp_l(&mStartEts) != OK) {
786 mStartEts.clear();
787 }
788 }
Andy Hungffa36952017-08-17 10:41:51 -0700789 mStartNs = systemTime(); // save this for timestamp adjustment after starting.
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800790 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
791 // reset current position as seen by client to 0
Glenn Kasten200092b2014-08-15 15:13:30 -0700792 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -0700793 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700794 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700795 mTimestampRetrogradePositionReported = false;
796 mTimestampRetrogradeTimeReported = false;
797 mTimestampStallReported = false;
798 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700799 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Phil Burk1b420972015-04-22 10:52:21 -0700800
jiabin94ed47c2023-07-27 23:34:20 +0000801 if (!isAfTrackOffloadedOrDirect_l()
Andy Hung65ffdfc2016-10-10 15:52:11 -0700802 && mStartEts.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) {
Andy Hunge1e98462016-04-12 10:18:51 -0700803 // Server side has consumed something, but is it finished consuming?
804 // It is possible since flush and stop are asynchronous that the server
805 // is still active at this point.
Andy Hungfb8ede22018-09-12 19:03:24 -0700806 ALOGV("%s(%d): server read:%lld cumulative flushed:%lld client written:%lld",
Eric Laurent973db022018-11-20 14:54:31 -0800807 __func__, mPortId,
Andy Hunge1e98462016-04-12 10:18:51 -0700808 (long long)(mFramesWrittenServerOffset
Andy Hung65ffdfc2016-10-10 15:52:11 -0700809 + mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]),
810 (long long)mStartEts.mFlushed,
Andy Hunge1e98462016-04-12 10:18:51 -0700811 (long long)mFramesWritten);
Andy Hungc4e60eb2017-09-06 11:14:57 -0700812 // mStartEts is already adjusted by mFramesWrittenServerOffset, so we delta adjust.
813 mFramesWrittenServerOffset -= mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER];
Andy Hung61be8412015-10-06 10:51:09 -0700814 }
Andy Hunge1e98462016-04-12 10:18:51 -0700815 mFramesWritten = 0;
816 mProxy->clearTimestamp(); // need new server push for valid timestamp
817 mMarkerReached = false;
Andy Hung61be8412015-10-06 10:51:09 -0700818
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700819 // For offloaded tracks, we don't know if the hardware counters are really zero here,
820 // since the flush is asynchronous and stop may not fully drain.
821 // We save the time when the track is started to later verify whether
822 // the counters are realistic (i.e. start from zero after this time).
Andy Hungffa36952017-08-17 10:41:51 -0700823 mStartFromZeroUs = mStartNs / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700824
Eric Laurentec9a0322013-08-28 10:23:01 -0700825 // force refresh of remaining frames by processAudioBuffer() as last
826 // write before stop could be partial.
827 mRefreshRemaining = true;
Tomoharu Kasahara6e8bf982017-06-09 16:17:33 +0900828
829 // for static track, clear the old flags when starting from stopped state
830 if (mSharedBuffer != 0) {
831 android_atomic_and(
832 ~(CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
833 &mCblk->mFlags);
834 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800835 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700836 mNewPosition = mPosition + mUpdatePeriod;
Andy Hung4be3b832016-10-13 17:51:43 -0700837 int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800838
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800839 if (!(flags & CBLK_INVALID)) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -0800840 mAudioTrack->start(&status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800841 if (status == DEAD_OBJECT) {
842 flags |= CBLK_INVALID;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800843 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800844 }
845 if (flags & CBLK_INVALID) {
846 status = restoreTrack_l("start");
847 }
848
Andy Hung79629f02016-03-24 13:57:40 -0700849 // resume or pause the callback thread as needed.
850 sp<AudioTrackThread> t = mAudioTrackThread;
851 if (status == NO_ERROR) {
852 if (t != 0) {
853 if (previousState == STATE_STOPPING) {
854 mProxy->interrupt();
855 } else {
856 t->resume();
857 }
858 } else {
859 mPreviousPriority = getpriority(PRIO_PROCESS, 0);
860 get_sched_policy(0, &mPreviousSchedulingGroup);
861 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
862 }
Andy Hung39399b62017-04-21 15:07:45 -0700863
864 // Start our local VolumeHandler for restoration purposes.
865 mVolumeHandler->setStarted();
Andy Hung79629f02016-03-24 13:57:40 -0700866 } else {
Eric Laurent973db022018-11-20 14:54:31 -0800867 ALOGE("%s(%d): status %d", __func__, mPortId, status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800868 mState = previousState;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800869 if (t != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100870 if (previousState != STATE_STOPPING) {
871 t->pause();
872 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800873 } else {
Glenn Kasten87913512011-06-22 16:15:25 -0700874 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
Glenn Kastena6364332012-04-19 09:35:04 -0700875 set_sched_policy(0, mPreviousSchedulingGroup);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800876 }
877 }
878
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100879 return status;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800880}
881
882void AudioTrack::stop()
883{
Andy Hungb68f5eb2019-12-03 16:49:17 -0800884 const int64_t beginNs = systemTime();
885
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800886 AutoMutex lock(mLock);
Andy Hung1f950512024-04-11 19:03:35 -0700887 if (mProxy == nullptr) return; // not successfully initialized.
Andy Hung06a730b2020-04-09 13:28:31 -0700888 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800889 mediametrics::LogItem(mMetricsId)
890 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_STOP)
Andy Hungea840382020-05-05 21:50:17 -0700891 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800892 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Phil Burk64e16a72020-06-01 13:25:51 -0700893 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
894 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getUnderrunCount_l())
Phil Burka9876702020-04-20 18:16:15 -0700895 .record();
Phil Burka9876702020-04-20 18:16:15 -0700896 });
Andy Hungb68f5eb2019-12-03 16:49:17 -0800897
Eric Laurent973db022018-11-20 14:54:31 -0800898 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -0700899
Glenn Kasten397edb32013-08-30 15:10:13 -0700900 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800901 return;
902 }
903
Glenn Kasten23a75452014-01-13 10:37:17 -0800904 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100905 mState = STATE_STOPPING;
906 } else {
907 mState = STATE_STOPPED;
Andy Hung2148bf02016-11-28 19:01:02 -0800908 ALOGD_IF(mSharedBuffer == nullptr,
Eric Laurent973db022018-11-20 14:54:31 -0800909 "%s(%d): called with %u frames delivered", __func__, mPortId, mReleased.value());
Andy Hungc2813e52014-10-16 17:54:34 -0700910 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100911 }
912
Andy Hung1d3556d2018-03-29 16:30:14 -0700913 mProxy->stop(); // notify server not to read beyond current client position until start().
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800914 mProxy->interrupt();
915 mAudioTrack->stop();
Andy Hung61be8412015-10-06 10:51:09 -0700916
917 // Note: legacy handling - stop does not clear playback marker
918 // and periodic update counter, but flush does for streaming tracks.
Andy Hung9b461582014-12-01 17:56:29 -0800919
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800920 if (mSharedBuffer != 0) {
Andy Hung9b461582014-12-01 17:56:29 -0800921 // clear buffer position and loop count.
Andy Hung9b461582014-12-01 17:56:29 -0800922 mStaticProxy->setBufferPositionAndLoop(0 /* position */,
923 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800924 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100925
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800926 sp<AudioTrackThread> t = mAudioTrackThread;
927 if (t != 0) {
Glenn Kasten23a75452014-01-13 10:37:17 -0800928 if (!isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100929 t->pause();
Jean-Michel Trivia60e9a22018-11-15 16:13:36 -0800930 } else if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
Jean-Michel Trivi0248a2a2018-11-21 10:05:57 -0800931 // causes wake up of the playback thread, that will callback the client for
932 // EVENT_STREAM_END in processAudioBuffer()
933 t->wake();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100934 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800935 } else {
936 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
937 set_sched_policy(0, mPreviousSchedulingGroup);
938 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800939}
940
941bool AudioTrack::stopped() const
942{
Glenn Kasten9a2aaf92012-01-03 09:42:47 -0800943 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800944 return mState != STATE_ACTIVE;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800945}
946
947void AudioTrack::flush()
948{
Andy Hungb68f5eb2019-12-03 16:49:17 -0800949 const int64_t beginNs = systemTime();
Andy Hungfb8ede22018-09-12 19:03:24 -0700950 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -0700951 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800952 mediametrics::LogItem(mMetricsId)
953 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_FLUSH)
Andy Hungea840382020-05-05 21:50:17 -0700954 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800955 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
956 .record(); });
957
Eric Laurent973db022018-11-20 14:54:31 -0800958 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -0700959
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800960 if (mSharedBuffer != 0) {
961 return;
Glenn Kasten4bae3642012-11-30 13:41:12 -0800962 }
Andy Hung4c5ed302018-05-09 11:16:21 -0700963 if (mState == STATE_ACTIVE) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800964 return;
965 }
966 flush_l();
Eric Laurent1703cdf2011-03-07 14:52:59 -0800967}
968
Eric Laurent1703cdf2011-03-07 14:52:59 -0800969void AudioTrack::flush_l()
970{
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800971 ALOG_ASSERT(mState != STATE_ACTIVE);
Eric Laurentc2f1f072009-07-17 12:17:14 -0700972
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -0700973 // clear playback marker and periodic update counter
974 mMarkerPosition = 0;
975 mMarkerReached = false;
976 mUpdatePeriod = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100977 mRefreshRemaining = true;
Eric Laurentc2f1f072009-07-17 12:17:14 -0700978
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800979 mState = STATE_FLUSHED;
Andy Hungc2813e52014-10-16 17:54:34 -0700980 mReleased = 0;
Glenn Kasten23a75452014-01-13 10:37:17 -0800981 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100982 mProxy->interrupt();
983 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800984 mProxy->flush();
Glenn Kasten4bae3642012-11-30 13:41:12 -0800985 mAudioTrack->flush();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800986}
987
Andy Hung959b5b82021-09-24 10:46:20 -0700988bool AudioTrack::pauseAndWait(const std::chrono::milliseconds& timeout)
989{
990 using namespace std::chrono_literals;
991
Andy Hungd87a53a2022-01-19 16:56:17 -0800992 // We use atomic access here for state variables - these are used as hints
993 // to ensure we have ramped down audio.
994 const int priorState = mProxy->getState();
995 const uint32_t priorPosition = mProxy->getPosition().unsignedValue();
996
Andy Hung959b5b82021-09-24 10:46:20 -0700997 pause();
998
Andy Hungd87a53a2022-01-19 16:56:17 -0800999 // Only if we were previously active, do we wait to ramp down the audio.
1000 if (priorState != CBLK_STATE_ACTIVE) return true;
1001
Andy Hung959b5b82021-09-24 10:46:20 -07001002 AutoMutex lock(mLock);
1003 // offload and direct tracks do not wait because pause volume ramp is handled by hardware.
1004 if (isOffloadedOrDirect_l()) return true;
1005
1006 // Wait for the track state to be anything besides pausing.
1007 // This ensures that the volume has ramped down.
1008 constexpr auto SLEEP_INTERVAL_MS = 10ms;
Andy Hungd87a53a2022-01-19 16:56:17 -08001009 constexpr auto POSITION_TIMEOUT_MS = 40ms; // don't wait longer than this for position change.
Andy Hung959b5b82021-09-24 10:46:20 -07001010 auto begin = std::chrono::steady_clock::now();
1011 while (true) {
Andy Hungd87a53a2022-01-19 16:56:17 -08001012 // Wait for state and position to change.
1013 // After pause() the server state should be PAUSING, but that may immediately
1014 // convert to PAUSED by prepareTracks before data is read into the mixer.
1015 // Hence we check that the state is not PAUSING and that the server position
1016 // has advanced to be a more reliable estimate that the volume ramp has completed.
Andy Hung959b5b82021-09-24 10:46:20 -07001017 const int state = mProxy->getState();
Andy Hungd87a53a2022-01-19 16:56:17 -08001018 const uint32_t position = mProxy->getPosition().unsignedValue();
Andy Hung959b5b82021-09-24 10:46:20 -07001019
1020 mLock.unlock(); // only local variables accessed until lock.
1021 auto elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(
1022 std::chrono::steady_clock::now() - begin);
Andy Hungd87a53a2022-01-19 16:56:17 -08001023 if (state != CBLK_STATE_PAUSING &&
1024 (elapsed >= POSITION_TIMEOUT_MS || position != priorPosition)) {
1025 ALOGV("%s: success state:%d, position:%u after %lld ms"
1026 " (prior state:%d prior position:%u)",
1027 __func__, state, position, elapsed.count(), priorState, priorPosition);
Andy Hung959b5b82021-09-24 10:46:20 -07001028 return true;
1029 }
1030 std::chrono::milliseconds remaining = timeout - elapsed;
1031 if (remaining.count() <= 0) {
1032 ALOGW("%s: timeout expired state:%d still pausing:%d after %lld ms",
1033 __func__, state, CBLK_STATE_PAUSING, elapsed.count());
1034 return false;
1035 }
1036 // It is conceivable that the track is restored while sleeping;
1037 // as this logic is advisory, we allow that.
1038 std::this_thread::sleep_for(std::min(remaining, SLEEP_INTERVAL_MS));
1039 mLock.lock();
1040 }
1041}
1042
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001043void AudioTrack::pause()
1044{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001045 const int64_t beginNs = systemTime();
Eric Laurentf5aafb22010-11-18 08:40:16 -08001046 AutoMutex lock(mLock);
Andy Hunga6b27032020-04-27 10:34:24 -07001047 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001048 mediametrics::LogItem(mMetricsId)
1049 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_PAUSE)
Andy Hungea840382020-05-05 21:50:17 -07001050 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001051 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
1052 .record(); });
1053
Eric Laurent973db022018-11-20 14:54:31 -08001054 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001055
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001056 if (mState == STATE_ACTIVE) {
1057 mState = STATE_PAUSED;
1058 } else if (mState == STATE_STOPPING) {
1059 mState = STATE_PAUSED_STOPPING;
1060 } else {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001061 return;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001062 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001063 mProxy->interrupt();
1064 mAudioTrack->pause();
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001065
Marco Nelissen3a90f282014-03-10 11:21:43 -07001066 if (isOffloaded_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001067 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001068 // An offload output can be re-used between two audio tracks having
1069 // the same configuration. A timestamp query for a paused track
1070 // while the other is running would return an incorrect time.
1071 // To fix this, cache the playback position on a pause() and return
1072 // this time when requested until the track is resumed.
1073
1074 // OffloadThread sends HAL pause in its threadLoop. Time saved
1075 // here can be slightly off.
1076
1077 // TODO: check return code for getRenderPosition.
1078
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001079 uint32_t halFrames;
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001080 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition);
Andy Hungfb8ede22018-09-12 19:03:24 -07001081 ALOGV("%s(%d): for offload, cache current position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001082 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001083 }
1084 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001085}
1086
Eric Laurentbe916aa2010-06-01 23:49:17 -07001087status_t AudioTrack::setVolume(float left, float right)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001088{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001089 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1090 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY ||
1091 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001092 return BAD_VALUE;
1093 }
1094
Andy Hungb68f5eb2019-12-03 16:49:17 -08001095 mediametrics::LogItem(mMetricsId)
1096 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOLUME)
1097 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)left)
1098 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)right)
1099 .record();
1100
Eric Laurent1703cdf2011-03-07 14:52:59 -08001101 AutoMutex lock(mLock);
Glenn Kasten66e46352014-01-16 17:44:23 -08001102 mVolume[AUDIO_INTERLEAVE_LEFT] = left;
1103 mVolume[AUDIO_INTERLEAVE_RIGHT] = right;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001104
Glenn Kastenc56f3422014-03-21 17:53:17 -07001105 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right)));
Eric Laurentbe916aa2010-06-01 23:49:17 -07001106
Glenn Kasten23a75452014-01-13 10:37:17 -08001107 if (isOffloaded_l()) {
Eric Laurent59fe0102013-09-27 18:48:26 -07001108 mAudioTrack->signal();
1109 }
Eric Laurentbe916aa2010-06-01 23:49:17 -07001110 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001111}
1112
Glenn Kastenb1c09932012-02-27 16:21:04 -08001113status_t AudioTrack::setVolume(float volume)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001114{
Glenn Kastenb1c09932012-02-27 16:21:04 -08001115 return setVolume(volume, volume);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001116}
1117
Eric Laurent2beeb502010-07-16 07:43:46 -07001118status_t AudioTrack::setAuxEffectSendLevel(float level)
Eric Laurentbe916aa2010-06-01 23:49:17 -07001119{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001120 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1121 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001122 return BAD_VALUE;
1123 }
1124
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001125 AutoMutex lock(mLock);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001126 mSendLevel = level;
Glenn Kastene3aa6592012-12-04 12:22:46 -08001127 mProxy->setSendLevel(level);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001128
1129 return NO_ERROR;
1130}
1131
Glenn Kastena5224f32012-01-04 12:41:44 -08001132void AudioTrack::getAuxEffectSendLevel(float* level) const
Eric Laurentbe916aa2010-06-01 23:49:17 -07001133{
1134 if (level != NULL) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001135 *level = mSendLevel;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001136 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001137}
1138
Glenn Kasten3b16c762012-11-14 08:44:39 -08001139status_t AudioTrack::setSampleRate(uint32_t rate)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001140{
Andy Hung5cbb5782015-03-27 18:39:59 -07001141 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08001142 ALOGV("%s(%d): prior state:%s rate:%u", __func__, mPortId, stateToString(mState), rate);
Andy Hungfb8ede22018-09-12 19:03:24 -07001143
Andy Hung5cbb5782015-03-27 18:39:59 -07001144 if (rate == mSampleRate) {
1145 return NO_ERROR;
1146 }
jiabinf4de6112018-12-19 12:40:08 -08001147 if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)
1148 || (mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL)) {
John Grossman4ff14ba2012-02-08 16:37:41 -08001149 return INVALID_OPERATION;
1150 }
Eric Laurent0d6db582014-11-12 18:39:44 -08001151 if (mOutput == AUDIO_IO_HANDLE_NONE) {
1152 return NO_INIT;
1153 }
Andy Hung5cbb5782015-03-27 18:39:59 -07001154 // NOTE: it is theoretically possible, but highly unlikely, that a device change
1155 // could mean a previously allowed sampling rate is no longer allowed.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001156 uint32_t afSamplingRate;
Eric Laurent0d6db582014-11-12 18:39:44 -08001157 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
Eric Laurent57326622009-07-07 07:10:45 -07001158 return NO_INIT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001159 }
Andy Hung26145642015-04-15 21:56:53 -07001160 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001161 const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001162 if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001163 return BAD_VALUE;
1164 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001165 // TODO: Should we also check if the buffer size is compatible?
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001166
Glenn Kastene3aa6592012-12-04 12:22:46 -08001167 mSampleRate = rate;
Andy Hung26145642015-04-15 21:56:53 -07001168 mProxy->setSampleRate(effectiveSampleRate);
Glenn Kastene3aa6592012-12-04 12:22:46 -08001169
Andy Hunge02df772024-06-10 17:27:28 -07001170 mediametrics::LogItem(mMetricsId)
1171 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSAMPLERATE)
1172 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE AMEDIAMETRICS_PROP_SAMPLERATE,
1173 static_cast<int32_t>(effectiveSampleRate))
1174 .set(AMEDIAMETRICS_PROP_SAMPLERATE, static_cast<int32_t>(rate))
1175 .record();
1176
Eric Laurent57326622009-07-07 07:10:45 -07001177 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001178}
1179
Glenn Kastena5224f32012-01-04 12:41:44 -08001180uint32_t AudioTrack::getSampleRate() const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001181{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001182 AutoMutex lock(mLock);
Eric Laurent6f59db12013-07-26 17:16:50 -07001183
1184 // sample rate can be updated during playback by the offloaded decoder so we need to
1185 // query the HAL and update if needed.
1186// FIXME use Proxy return channel to update the rate from server and avoid polling here
Eric Laurentab5cdba2014-06-09 17:22:27 -07001187 if (isOffloadedOrDirect_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001188 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent6f59db12013-07-26 17:16:50 -07001189 uint32_t sampleRate = 0;
Jean-Michel Trivib7f24b12014-06-11 10:05:30 -07001190 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
Eric Laurent6f59db12013-07-26 17:16:50 -07001191 if (status == NO_ERROR) {
1192 mSampleRate = sampleRate;
1193 }
1194 }
1195 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001196 return mSampleRate;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001197}
1198
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001199uint32_t AudioTrack::getOriginalSampleRate() const
1200{
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001201 return mOriginalSampleRate;
1202}
1203
Robert Wu310037a2022-09-06 21:48:18 +00001204uint32_t AudioTrack::getHalSampleRate() const
1205{
1206 return mAfSampleRate;
1207}
1208
1209uint32_t AudioTrack::getHalChannelCount() const
1210{
1211 return mAfChannelCount;
1212}
1213
1214audio_format_t AudioTrack::getHalFormat() const
1215{
1216 return mAfFormat;
1217}
1218
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001219status_t AudioTrack::setDualMonoMode(audio_dual_mono_mode_t mode)
1220{
1221 AutoMutex lock(mLock);
1222 return setDualMonoMode_l(mode);
1223}
1224
1225status_t AudioTrack::setDualMonoMode_l(audio_dual_mono_mode_t mode)
1226{
1227 const status_t status = statusTFromBinderStatus(
1228 mAudioTrack->setDualMonoMode(VALUE_OR_RETURN_STATUS(
1229 legacy2aidl_audio_dual_mono_mode_t_AudioDualMonoMode(mode))));
1230 if (status == NO_ERROR) mDualMonoMode = mode;
1231 return status;
1232}
1233
1234status_t AudioTrack::getDualMonoMode(audio_dual_mono_mode_t* mode) const
1235{
1236 AutoMutex lock(mLock);
Mikhail Naganovf53e1822022-12-18 02:48:14 +00001237 media::audio::common::AudioDualMonoMode mediaMode;
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001238 const status_t status = statusTFromBinderStatus(mAudioTrack->getDualMonoMode(&mediaMode));
1239 if (status == NO_ERROR) {
1240 *mode = VALUE_OR_RETURN_STATUS(
1241 aidl2legacy_AudioDualMonoMode_audio_dual_mono_mode_t(mediaMode));
1242 }
1243 return status;
1244}
1245
1246status_t AudioTrack::setAudioDescriptionMixLevel(float leveldB)
1247{
1248 AutoMutex lock(mLock);
1249 return setAudioDescriptionMixLevel_l(leveldB);
1250}
1251
1252status_t AudioTrack::setAudioDescriptionMixLevel_l(float leveldB)
1253{
1254 const status_t status = statusTFromBinderStatus(
1255 mAudioTrack->setAudioDescriptionMixLevel(leveldB));
1256 if (status == NO_ERROR) mAudioDescriptionMixLeveldB = leveldB;
1257 return status;
1258}
1259
1260status_t AudioTrack::getAudioDescriptionMixLevel(float* leveldB) const
1261{
1262 AutoMutex lock(mLock);
1263 return statusTFromBinderStatus(mAudioTrack->getAudioDescriptionMixLevel(leveldB));
1264}
1265
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001266status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
Andy Hung8edb8dc2015-03-26 19:13:55 -07001267{
Andy Hung8edb8dc2015-03-26 19:13:55 -07001268 AutoMutex lock(mLock);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001269 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
Andy Hung8edb8dc2015-03-26 19:13:55 -07001270 return NO_ERROR;
1271 }
Glenn Kastend79072e2016-01-06 08:41:20 -08001272 if (isOffloadedOrDirect_l()) {
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001273 const status_t status = statusTFromBinderStatus(mAudioTrack->setPlaybackRateParameters(
1274 VALUE_OR_RETURN_STATUS(
1275 legacy2aidl_audio_playback_rate_t_AudioPlaybackRate(playbackRate))));
1276 if (status == NO_ERROR) {
1277 mPlaybackRate = playbackRate;
Bertil Ã…kesson081fb742022-07-14 16:39:36 +02001278 } else if (status == INVALID_OPERATION
1279 && playbackRate.mSpeed == 1.0f && mPlaybackRate.mPitch == 1.0f) {
1280 mPlaybackRate = playbackRate;
1281 return NO_ERROR;
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001282 }
1283 return status;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001284 }
1285 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
1286 return INVALID_OPERATION;
1287 }
Andy Hungff874dc2016-04-11 16:49:09 -07001288
Andy Hungfb8ede22018-09-12 19:03:24 -07001289 ALOGV("%s(%d): mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001290 __func__, mPortId, mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001291 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001292 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
1293 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
1294 const float effectivePitch = adjustPitch(playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001295 AudioPlaybackRate playbackRateTemp = playbackRate;
1296 playbackRateTemp.mSpeed = effectiveSpeed;
1297 playbackRateTemp.mPitch = effectivePitch;
1298
Andy Hungfb8ede22018-09-12 19:03:24 -07001299 ALOGV("%s(%d) (effective) mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001300 __func__, mPortId, effectiveRate, effectiveSpeed, effectivePitch);
Andy Hungff874dc2016-04-11 16:49:09 -07001301
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001302 if (!isAudioPlaybackRateValid(playbackRateTemp)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001303 ALOGW("%s(%d) (%f, %f) failed (effective rate out of bounds)",
Eric Laurent973db022018-11-20 14:54:31 -08001304 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001305 return BAD_VALUE;
1306 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001307 // Check if the buffer size is compatible.
Andy Hung26145642015-04-15 21:56:53 -07001308 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001309 ALOGW("%s(%d) (%f, %f) failed (buffer size)",
Eric Laurent973db022018-11-20 14:54:31 -08001310 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung8edb8dc2015-03-26 19:13:55 -07001311 return BAD_VALUE;
1312 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001313
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001314 // Check resampler ratios are within bounds
Glenn Kastend3bb6452016-12-05 18:14:37 -08001315 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
1316 (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001317 ALOGW("%s(%d) (%f, %f) failed. Resample rate exceeds max accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001318 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001319 return BAD_VALUE;
1320 }
1321
Dan Austine34eae22015-10-27 16:14:52 -07001322 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001323 ALOGW("%s(%d) (%f, %f) failed. Resample rate below min accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001324 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001325 return BAD_VALUE;
1326 }
1327 mPlaybackRate = playbackRate;
1328 //set effective rates
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001329 mProxy->setPlaybackRate(playbackRateTemp);
Andy Hung26145642015-04-15 21:56:53 -07001330 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
Andy Hungb68f5eb2019-12-03 16:49:17 -08001331
1332 mediametrics::LogItem(mMetricsId)
1333 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYBACKPARAM)
1334 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
1335 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
1336 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
1337 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1338 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveRate)
1339 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1340 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)playbackRateTemp.mSpeed)
1341 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1342 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)playbackRateTemp.mPitch)
1343 .record();
1344
Andy Hung8edb8dc2015-03-26 19:13:55 -07001345 return NO_ERROR;
1346}
1347
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001348const AudioPlaybackRate& AudioTrack::getPlaybackRate()
Andy Hung8edb8dc2015-03-26 19:13:55 -07001349{
1350 AutoMutex lock(mLock);
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001351 if (isOffloadedOrDirect_l()) {
Mikhail Naganovf53e1822022-12-18 02:48:14 +00001352 media::audio::common::AudioPlaybackRate playbackRateTemp;
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001353 const status_t status = statusTFromBinderStatus(
1354 mAudioTrack->getPlaybackRateParameters(&playbackRateTemp));
1355 if (status == NO_ERROR) { // update local version if changed.
1356 mPlaybackRate =
1357 aidl2legacy_AudioPlaybackRate_audio_playback_rate_t(playbackRateTemp).value();
1358 }
1359 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001360 return mPlaybackRate;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001361}
1362
Phil Burkc0adecb2016-01-08 12:44:11 -08001363ssize_t AudioTrack::getBufferSizeInFrames()
1364{
1365 AutoMutex lock(mLock);
1366 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1367 return NO_INIT;
1368 }
Phil Burka9876702020-04-20 18:16:15 -07001369
Phil Burke8972b02016-03-04 11:29:57 -08001370 return (ssize_t) mProxy->getBufferSizeInFrames();
Phil Burkc0adecb2016-01-08 12:44:11 -08001371}
1372
Andy Hungf2c87b32016-04-07 19:49:29 -07001373status_t AudioTrack::getBufferDurationInUs(int64_t *duration)
1374{
1375 if (duration == nullptr) {
1376 return BAD_VALUE;
1377 }
1378 AutoMutex lock(mLock);
1379 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1380 return NO_INIT;
1381 }
1382 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames();
1383 if (bufferSizeInFrames < 0) {
1384 return (status_t)bufferSizeInFrames;
1385 }
1386 *duration = (int64_t)((double)bufferSizeInFrames * 1000000
1387 / ((double)mSampleRate * mPlaybackRate.mSpeed));
1388 return NO_ERROR;
1389}
1390
Phil Burkc0adecb2016-01-08 12:44:11 -08001391ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames)
1392{
1393 AutoMutex lock(mLock);
1394 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1395 return NO_INIT;
1396 }
Phil Burka9876702020-04-20 18:16:15 -07001397
1398 ssize_t originalBufferSize = mProxy->getBufferSizeInFrames();
1399 ssize_t finalBufferSize = mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
1400 if (originalBufferSize != finalBufferSize) {
Phil Burk64e16a72020-06-01 13:25:51 -07001401 android::mediametrics::LogItem(mMetricsId)
1402 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
1403 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1404 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t)getUnderrunCount_l())
1405 .record();
Phil Burka9876702020-04-20 18:16:15 -07001406 }
1407 return finalBufferSize;
Phil Burkc0adecb2016-01-08 12:44:11 -08001408}
1409
Andy Hung3c7f47a2021-03-16 17:30:09 -07001410ssize_t AudioTrack::getStartThresholdInFrames() const
1411{
1412 AutoMutex lock(mLock);
1413 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1414 return NO_INIT;
1415 }
1416 return (ssize_t) mProxy->getStartThresholdInFrames();
1417}
1418
1419ssize_t AudioTrack::setStartThresholdInFrames(size_t startThresholdInFrames)
1420{
1421 if (startThresholdInFrames > INT32_MAX || startThresholdInFrames == 0) {
1422 // contractually we could simply return the current threshold in frames
1423 // to indicate the request was ignored, but we return an error here.
1424 return BAD_VALUE;
1425 }
1426 AutoMutex lock(mLock);
1427 // We do not permit calling setStartThresholdInFrames() between the AudioTrack
1428 // default ctor AudioTrack() and set(...) but rather fail such an attempt.
1429 // (To do so would require a cached mOrigStartThresholdInFrames and we may
1430 // not have proper validation for the actual set value).
1431 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1432 return NO_INIT;
1433 }
1434 const uint32_t original = mProxy->getStartThresholdInFrames();
1435 const uint32_t final = mProxy->setStartThresholdInFrames(startThresholdInFrames);
1436 if (original != final) {
1437 android::mediametrics::LogItem(mMetricsId)
1438 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD)
1439 .set(AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES, (int32_t)final)
1440 .record();
1441 if (original > final) {
1442 // restart track if it was disabled by audioflinger due to previous underrun
1443 // and we reduced the number of frames for the threshold.
1444 restartIfDisabled();
1445 }
1446 }
1447 return final;
1448}
1449
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001450status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1451{
Glenn Kastend79072e2016-01-06 08:41:20 -08001452 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001453 return INVALID_OPERATION;
1454 }
1455
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001456 if (loopCount == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001457 ;
1458 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount &&
1459 loopEnd - loopStart >= MIN_LOOP) {
1460 ;
1461 } else {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001462 return BAD_VALUE;
1463 }
1464
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001465 AutoMutex lock(mLock);
1466 // See setPosition() regarding setting parameters such as loop points or position while active
1467 if (mState == STATE_ACTIVE) {
1468 return INVALID_OPERATION;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001469 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001470 setLoop_l(loopStart, loopEnd, loopCount);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001471 return NO_ERROR;
1472}
1473
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001474void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1475{
Andy Hung4ede21d2014-12-12 15:37:34 -08001476 // We do not update the periodic notification point.
1477 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
1478 mLoopCount = loopCount;
1479 mLoopEnd = loopEnd;
1480 mLoopStart = loopStart;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001481 mLoopCountNotified = loopCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001482 mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
Andy Hung3c09c782014-12-29 18:39:32 -08001483
1484 // Waking the AudioTrackThread is not needed as this cannot be called when active.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001485}
1486
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001487status_t AudioTrack::setMarkerPosition(uint32_t marker)
1488{
Atneya Nair14aabae2021-11-30 17:36:24 -05001489 AutoMutex lock(mLock);
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001490 // The only purpose of setting marker position is to get a callback
Atneya Nair14aabae2021-11-30 17:36:24 -05001491 if (!mCallback.promote() || isOffloadedOrDirect_l()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001492 return INVALID_OPERATION;
1493 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001494
1495 mMarkerPosition = marker;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001496 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001497
Andy Hung3c09c782014-12-29 18:39:32 -08001498 sp<AudioTrackThread> t = mAudioTrackThread;
1499 if (t != 0) {
1500 t->wake();
1501 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001502 return NO_ERROR;
1503}
1504
Glenn Kastena5224f32012-01-04 12:41:44 -08001505status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001506{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001507 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001508 return INVALID_OPERATION;
1509 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001510 if (marker == NULL) {
1511 return BAD_VALUE;
1512 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001513
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001514 AutoMutex lock(mLock);
Andy Hung90e8a972015-11-09 16:42:40 -08001515 mMarkerPosition.getValue(marker);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001516
1517 return NO_ERROR;
1518}
1519
1520status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
1521{
Atneya Nair14aabae2021-11-30 17:36:24 -05001522 AutoMutex lock(mLock);
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001523 // The only purpose of setting position update period is to get a callback
Atneya Nair14aabae2021-11-30 17:36:24 -05001524 if (!mCallback.promote() || isOffloadedOrDirect_l()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001525 return INVALID_OPERATION;
1526 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001527
Glenn Kasten200092b2014-08-15 15:13:30 -07001528 mNewPosition = updateAndGetPosition_l() + updatePeriod;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001529 mUpdatePeriod = updatePeriod;
Glenn Kasten2b2165c2014-01-13 08:53:36 -08001530
Andy Hung3c09c782014-12-29 18:39:32 -08001531 sp<AudioTrackThread> t = mAudioTrackThread;
1532 if (t != 0) {
1533 t->wake();
1534 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001535 return NO_ERROR;
1536}
1537
Glenn Kastena5224f32012-01-04 12:41:44 -08001538status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001539{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001540 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001541 return INVALID_OPERATION;
1542 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001543 if (updatePeriod == NULL) {
1544 return BAD_VALUE;
1545 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001546
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001547 AutoMutex lock(mLock);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001548 *updatePeriod = mUpdatePeriod;
1549
1550 return NO_ERROR;
1551}
1552
1553status_t AudioTrack::setPosition(uint32_t position)
1554{
Glenn Kastend79072e2016-01-06 08:41:20 -08001555 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001556 return INVALID_OPERATION;
1557 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001558 if (position > mFrameCount) {
1559 return BAD_VALUE;
1560 }
John Grossman4ff14ba2012-02-08 16:37:41 -08001561
Eric Laurent1703cdf2011-03-07 14:52:59 -08001562 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001563 // Currently we require that the player is inactive before setting parameters such as position
1564 // or loop points. Otherwise, there could be a race condition: the application could read the
1565 // current position, compute a new position or loop parameters, and then set that position or
1566 // loop parameters but it would do the "wrong" thing since the position has continued to advance
1567 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app
1568 // to specify how it wants to handle such scenarios.
1569 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001570 return INVALID_OPERATION;
1571 }
Andy Hung9b461582014-12-01 17:56:29 -08001572 // After setting the position, use full update period before notification.
Glenn Kasten200092b2014-08-15 15:13:30 -07001573 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001574 mStaticProxy->setBufferPosition(position);
Andy Hung3c09c782014-12-29 18:39:32 -08001575
1576 // Waking the AudioTrackThread is not needed as this cannot be called when active.
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001577 return NO_ERROR;
1578}
1579
Glenn Kasten200092b2014-08-15 15:13:30 -07001580status_t AudioTrack::getPosition(uint32_t *position)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001581{
Glenn Kastend65d73c2012-06-22 17:21:07 -07001582 if (position == NULL) {
1583 return BAD_VALUE;
1584 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001585
Eric Laurent1703cdf2011-03-07 14:52:59 -08001586 AutoMutex lock(mLock);
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001587 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
1588 if (mState == STATE_STOPPED || mState == STATE_FLUSHED) {
1589 *position = 0;
1590 return NO_ERROR;
1591 }
Andy Hung7a490e72016-03-23 15:58:10 -07001592 // FIXME: offloaded and direct tracks call into the HAL for render positions
1593 // for compressed/synced data; however, we use proxy position for pure linear pcm data
1594 // as we do not know the capability of the HAL for pcm position support and standby.
1595 // There may be some latency differences between the HAL position and the proxy position.
1596 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) {
Eric Laurentab5cdba2014-06-09 17:22:27 -07001597 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001598 ALOGV("%s(%d): called in paused state, return cached position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001599 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001600 *position = mPausedPosition;
1601 return NO_ERROR;
1602 }
1603
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001604 uint32_t dspFrames = 0;
Glenn Kasten142f5192014-03-25 17:44:59 -07001605 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung1f1db832015-06-08 13:26:10 -07001606 uint32_t halFrames; // actually unused
Andy Hung1f1db832015-06-08 13:26:10 -07001607 // FIXME: on getRenderPosition() error, we return OK with frame position 0.
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001608 if (AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames) != NO_ERROR) {
1609 *position = 0;
1610 return NO_ERROR;
1611 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001612 }
1613 *position = dspFrames;
1614 } else {
Eric Laurent275e8e92014-11-30 15:14:47 -08001615 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung1f1db832015-06-08 13:26:10 -07001616 (void) restoreTrack_l("getPosition");
1617 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l()
1618 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position.
Eric Laurent275e8e92014-11-30 15:14:47 -08001619 }
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001620 *position = updateAndGetPosition_l().value();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001621 }
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001622
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001623 return NO_ERROR;
1624}
1625
Kévin PETIT377b2ec2014-02-03 12:35:36 +00001626status_t AudioTrack::getBufferPosition(uint32_t *position)
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001627{
Glenn Kastend79072e2016-01-06 08:41:20 -08001628 if (mSharedBuffer == 0) {
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001629 return INVALID_OPERATION;
1630 }
1631 if (position == NULL) {
1632 return BAD_VALUE;
1633 }
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001634
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001635 AutoMutex lock(mLock);
1636 *position = mStaticProxy->getBufferPosition();
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001637 return NO_ERROR;
1638}
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001639
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001640status_t AudioTrack::reload()
1641{
Glenn Kastend79072e2016-01-06 08:41:20 -08001642 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001643 return INVALID_OPERATION;
1644 }
1645
Eric Laurent1703cdf2011-03-07 14:52:59 -08001646 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001647 // See setPosition() regarding setting parameters such as loop points or position while active
1648 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001649 return INVALID_OPERATION;
1650 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001651 mNewPosition = mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001652 (void) updateAndGetPosition_l();
1653 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07001654 mPreviousTimestampValid = false;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001655#if 0
Andy Hung9b461582014-12-01 17:56:29 -08001656 // The documentation is not clear on the behavior of reload() and the restoration
Andy Hung53c3b5f2014-12-15 16:42:05 -08001657 // of loop count. Historically we have not restored loop count, start, end,
1658 // but it makes sense if one desires to repeat playing a particular sound.
1659 if (mLoopCount != 0) {
1660 mLoopCountNotified = mLoopCount;
1661 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
1662 }
1663#endif
Andy Hung9b461582014-12-01 17:56:29 -08001664 mStaticProxy->setBufferPosition(0);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001665 return NO_ERROR;
1666}
1667
Glenn Kasten38e905b2014-01-13 10:21:48 -08001668audio_io_handle_t AudioTrack::getOutput() const
Eric Laurentc2f1f072009-07-17 12:17:14 -07001669{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001670 AutoMutex lock(mLock);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001671 return mOutput;
Eric Laurent1703cdf2011-03-07 14:52:59 -08001672}
1673
Paul McLeanaa981192015-03-21 09:55:15 -07001674status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) {
Kuowei Li72c8b062023-08-31 13:38:32 +08001675 status_t result = NO_ERROR;
Paul McLeanaa981192015-03-21 09:55:15 -07001676 AutoMutex lock(mLock);
Kuowei Li72c8b062023-08-31 13:38:32 +08001677 ALOGV("%s(%d): deviceId=%d mSelectedDeviceId=%d",
1678 __func__, mPortId, deviceId, mSelectedDeviceId);
Andy Hungd0239ae2024-08-22 16:22:16 -07001679 const int64_t beginNs = systemTime();
1680 mediametrics::Defer defer([&] {
1681 mediametrics::LogItem(mMetricsId)
1682 .set(AMEDIAMETRICS_PROP_CALLERNAME,
1683 mCallerName.empty()
1684 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
1685 : mCallerName.c_str())
1686 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPREFERREDDEVICE)
1687 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
1688 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)deviceId)
1689 .record(); });
1690
Paul McLeanaa981192015-03-21 09:55:15 -07001691 if (mSelectedDeviceId != deviceId) {
1692 mSelectedDeviceId = deviceId;
Dorin Drimusefc130c2024-01-12 16:51:56 +00001693 if (mStatus == NO_ERROR) {
Kuowei Li72c8b062023-08-31 13:38:32 +08001694 if (isOffloadedOrDirect_l()) {
gmanam7b69bd42024-04-26 14:46:10 +05301695 if (isPlaying_l()) {
Kuowei Li72c8b062023-08-31 13:38:32 +08001696 ALOGW("%s(%d). Offloaded or Direct track is not STOPPED or FLUSHED. "
1697 "State: %s.",
1698 __func__, mPortId, stateToString(mState));
1699 result = INVALID_OPERATION;
gmanam7b69bd42024-04-26 14:46:10 +05301700 } else {
1701 ALOGD("%s(%d): creating a new AudioTrack", __func__, mPortId);
1702 result = restoreTrack_l("setOutputDevice", true /* forceRestore */);
Dorin Drimusefc130c2024-01-12 16:51:56 +00001703 }
Eric Laurent72af8012023-03-15 17:36:22 +01001704 } else {
Kuowei Li72c8b062023-08-31 13:38:32 +08001705 // allow track invalidation when track is not playing to propagate
1706 // the updated mSelectedDeviceId
1707 if (isPlaying_l()) {
1708 if (mSelectedDeviceId != mRoutedDeviceId) {
1709 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
1710 mProxy->interrupt();
1711 }
1712 } else {
1713 // if the track is idle, try to restore now and
1714 // defer to next start if not possible
1715 if (restoreTrack_l("setOutputDevice") != OK) {
1716 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
1717 }
Eric Laurent72af8012023-03-15 17:36:22 +01001718 }
1719 }
Eric Laurentfb00fc72017-05-25 18:17:12 -07001720 }
Paul McLeanaa981192015-03-21 09:55:15 -07001721 }
Kuowei Li72c8b062023-08-31 13:38:32 +08001722 return result;
Paul McLeanaa981192015-03-21 09:55:15 -07001723}
1724
1725audio_port_handle_t AudioTrack::getOutputDevice() {
1726 AutoMutex lock(mLock);
1727 return mSelectedDeviceId;
1728}
1729
Eric Laurentad2e7b92017-09-14 20:06:42 -07001730// must be called with mLock held
1731void AudioTrack::updateRoutedDeviceId_l()
1732{
1733 // if the track is inactive, do not update actual device as the output stream maybe routed
1734 // to a device not relevant to this client because of other active use cases.
1735 if (mState != STATE_ACTIVE) {
1736 return;
1737 }
1738 if (mOutput != AUDIO_IO_HANDLE_NONE) {
1739 audio_port_handle_t deviceId = AudioSystem::getDeviceIdForIo(mOutput);
1740 if (deviceId != AUDIO_PORT_HANDLE_NONE) {
1741 mRoutedDeviceId = deviceId;
1742 }
1743 }
1744}
1745
Eric Laurent296fb132015-05-01 11:38:42 -07001746audio_port_handle_t AudioTrack::getRoutedDeviceId() {
1747 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001748 updateRoutedDeviceId_l();
1749 return mRoutedDeviceId;
Eric Laurent296fb132015-05-01 11:38:42 -07001750}
1751
Eric Laurentbe916aa2010-06-01 23:49:17 -07001752status_t AudioTrack::attachAuxEffect(int effectId)
1753{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001754 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001755 status_t status;
1756 mAudioTrack->attachAuxEffect(effectId, &status);
Eric Laurent2beeb502010-07-16 07:43:46 -07001757 if (status == NO_ERROR) {
1758 mAuxEffectId = effectId;
1759 }
1760 return status;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001761}
1762
Eric Laurente83b55d2014-11-14 10:06:21 -08001763audio_stream_type_t AudioTrack::streamType() const
1764{
Eric Laurente83b55d2014-11-14 10:06:21 -08001765 return mStreamType;
1766}
1767
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001768uint32_t AudioTrack::latency()
1769{
1770 AutoMutex lock(mLock);
1771 updateLatency_l();
1772 return mLatency;
1773}
1774
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001775// -------------------------------------------------------------------------
1776
Eric Laurent1703cdf2011-03-07 14:52:59 -08001777// must be called with mLock held
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001778void AudioTrack::updateLatency_l()
1779{
1780 status_t status = AudioSystem::getLatency(mOutput, &mAfLatency);
1781 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08001782 ALOGW("%s(%d): getLatency(%d) failed status %d", __func__, mPortId, mOutput, status);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001783 } else {
1784 // FIXME don't believe this lie
Andy Hung13969262017-09-11 17:24:21 -07001785 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001786 }
1787}
1788
Phil Burkadbb75a2017-06-16 12:19:42 -07001789// TODO Move this macro to a common header file for enum to string conversion in audio framework.
1790#define MEDIA_CASE_ENUM(name) case name: return #name
1791const char * AudioTrack::convertTransferToText(transfer_type transferType) {
1792 switch (transferType) {
1793 MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
1794 MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
1795 MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
1796 MEDIA_CASE_ENUM(TRANSFER_SYNC);
1797 MEDIA_CASE_ENUM(TRANSFER_SHARED);
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001798 MEDIA_CASE_ENUM(TRANSFER_SYNC_NOTIF_CALLBACK);
Phil Burkadbb75a2017-06-16 12:19:42 -07001799 default:
1800 return "UNRECOGNIZED";
1801 }
1802}
1803
Glenn Kasten200092b2014-08-15 15:13:30 -07001804status_t AudioTrack::createTrack_l()
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001805{
Eric Laurentf32d7812017-11-30 14:44:07 -08001806 status_t status;
1807 bool callbackAdded = false;
Andy Hung3acde2c2021-11-11 09:18:08 -08001808 std::string errorMessage;
Eric Laurentf32d7812017-11-30 14:44:07 -08001809
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001810 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
1811 if (audioFlinger == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001812 errorMessage = StringPrintf("%s(%d): Could not get audioflinger",
Eric Laurent973db022018-11-20 14:54:31 -08001813 __func__, mPortId);
Andy Hung3acde2c2021-11-11 09:18:08 -08001814 status = DEAD_OBJECT;
Eric Laurentf32d7812017-11-30 14:44:07 -08001815 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001816 }
1817
Eric Laurent21da6472017-11-09 16:29:26 -08001818 {
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08001819 // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
1820 // After fast request is denied, we will request again if IAudioTrack is re-created.
Glenn Kastend79072e2016-01-06 08:41:20 -08001821 // Client can only express a preference for FAST. Server will perform additional tests.
Phil Burk33ff89b2015-11-30 11:16:01 -08001822 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Phil Burkadbb75a2017-06-16 12:19:42 -07001823 // either of these use cases:
1824 // use case 1: shared buffer
1825 bool sharedBuffer = mSharedBuffer != 0;
1826 bool transferAllowed =
Glenn Kastenc6ba8232014-02-27 13:34:29 -08001827 // use case 2: callback transfer mode
Glenn Kasten1dfe2f92015-03-09 12:03:14 -07001828 (mTransfer == TRANSFER_CALLBACK) ||
1829 // use case 3: obtain/release mode
Phil Burk33ff89b2015-11-30 11:16:01 -08001830 (mTransfer == TRANSFER_OBTAIN) ||
1831 // use case 4: synchronous write
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001832 ((mTransfer == TRANSFER_SYNC || mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK)
1833 && mThreadCanCallJava);
Phil Burkadbb75a2017-06-16 12:19:42 -07001834
Eric Laurent21da6472017-11-09 16:29:26 -08001835 bool fastAllowed = sharedBuffer || transferAllowed;
1836 if (!fastAllowed) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001837 ALOGW("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by client,"
1838 " not shared buffer and transfer = %s",
Eric Laurent973db022018-11-20 14:54:31 -08001839 __func__, mPortId,
Phil Burkadbb75a2017-06-16 12:19:42 -07001840 convertTransferToText(mTransfer));
Phil Burk33ff89b2015-11-30 11:16:01 -08001841 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
1842 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001843 }
1844
Eric Laurent21da6472017-11-09 16:29:26 -08001845 IAudioFlinger::CreateTrackInput input;
Andy Hunga2159aa2021-07-20 13:01:52 -07001846 if (mOriginalStreamType != AUDIO_STREAM_DEFAULT) {
1847 // Legacy: This is based on original parameters even if the track is recreated.
1848 input.attr = AudioSystem::streamTypeToAttributes(mOriginalStreamType);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001849 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08001850 input.attr = mAttributes;
Eric Laurentd1b449a2010-05-14 03:26:45 -07001851 }
Eric Laurent21da6472017-11-09 16:29:26 -08001852 input.config = AUDIO_CONFIG_INITIALIZER;
1853 input.config.sample_rate = mSampleRate;
1854 input.config.channel_mask = mChannelMask;
1855 input.config.format = mFormat;
1856 input.config.offload_info = mOffloadInfoCopy;
Svet Ganov3e5f14f2021-05-13 22:51:08 +00001857 input.clientInfo.attributionSource = mClientAttributionSource;
Eric Laurent21da6472017-11-09 16:29:26 -08001858 input.clientInfo.clientTid = -1;
Glenn Kasten363fb752014-01-15 12:27:31 -08001859 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Glenn Kasten47d55172017-05-23 11:19:30 -07001860 // It is currently meaningless to request SCHED_FIFO for a Java thread. Even if the
1861 // application-level code follows all non-blocking design rules, the language runtime
1862 // doesn't also follow those rules, so the thread will not benefit overall.
Phil Burk33ff89b2015-11-30 11:16:01 -08001863 if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
Eric Laurent21da6472017-11-09 16:29:26 -08001864 input.clientInfo.clientTid = mAudioTrackThread->getTid();
Glenn Kasten3acbd052012-02-28 10:39:56 -08001865 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001866 }
Eric Laurent21da6472017-11-09 16:29:26 -08001867 input.sharedBuffer = mSharedBuffer;
1868 input.notificationsPerBuffer = mNotificationsPerBufferReq;
1869 input.speed = 1.0;
1870 if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
1871 (mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
1872 input.speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
1873 max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
1874 }
1875 input.flags = mFlags;
1876 input.frameCount = mReqFrameCount;
1877 input.notificationFrameCount = mNotificationFramesReq;
1878 input.selectedDeviceId = mSelectedDeviceId;
1879 input.sessionId = mSessionId;
jiabinf6eb4c32020-02-25 14:06:25 -08001880 input.audioTrackCallback = mAudioTrackCallback;
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001881
Ytai Ben-Tsvi4dfeb622020-11-02 12:47:30 -08001882 media::CreateTrackResponse response;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001883 status = audioFlinger->createTrack(VALUE_OR_FATAL(input.toAidl()), response);
Ytai Ben-Tsvi357e26a2021-01-05 13:21:19 -08001884
1885 IAudioFlinger::CreateTrackOutput output{};
1886 if (status == NO_ERROR) {
1887 output = VALUE_OR_FATAL(IAudioFlinger::CreateTrackOutput::fromAidl(response));
1888 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001889
Eric Laurent21da6472017-11-09 16:29:26 -08001890 if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001891 errorMessage = StringPrintf(
1892 "%s(%d): AudioFlinger could not create track, status: %d output %d",
Eric Laurent973db022018-11-20 14:54:31 -08001893 __func__, mPortId, status, output.outputId);
Eric Laurentf32d7812017-11-30 14:44:07 -08001894 if (status == NO_ERROR) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001895 status = INVALID_OPERATION; // device not ready
Eric Laurentf32d7812017-11-30 14:44:07 -08001896 }
1897 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001898 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001899 ALOG_ASSERT(output.audioTrack != 0);
Glenn Kastenc08d20b2014-02-24 15:21:10 -08001900
Eric Laurent21da6472017-11-09 16:29:26 -08001901 mFrameCount = output.frameCount;
1902 mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
1903 mRoutedDeviceId = output.selectedDeviceId;
1904 mSessionId = output.sessionId;
Andy Hunga2159aa2021-07-20 13:01:52 -07001905 mStreamType = output.streamType;
Eric Laurent21da6472017-11-09 16:29:26 -08001906
1907 mSampleRate = output.sampleRate;
1908 if (mOriginalSampleRate == 0) {
1909 mOriginalSampleRate = mSampleRate;
1910 }
1911
1912 mAfFrameCount = output.afFrameCount;
1913 mAfSampleRate = output.afSampleRate;
Robert Wu310037a2022-09-06 21:48:18 +00001914 mAfChannelCount = audio_channel_count_from_out_mask(output.afChannelMask);
1915 mAfFormat = output.afFormat;
Eric Laurent21da6472017-11-09 16:29:26 -08001916 mAfLatency = output.afLatencyMs;
jiabin94ed47c2023-07-27 23:34:20 +00001917 mAfTrackFlags = output.afTrackFlags;
Eric Laurent21da6472017-11-09 16:29:26 -08001918
1919 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
1920
Glenn Kasten38e905b2014-01-13 10:21:48 -08001921 // AudioFlinger now owns the reference to the I/O handle,
1922 // so we are no longer responsible for releasing it.
1923
Glenn Kasten7fd04222016-02-02 12:38:16 -08001924 // FIXME compare to AudioRecord
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001925 std::optional<media::SharedFileRegion> sfr;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001926 output.audioTrack->getCblk(&sfr);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001927 sp<IMemory> iMem = VALUE_OR_FATAL(aidl2legacy_NullableSharedFileRegion_IMemory(sfr));
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001928 if (iMem == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001929 errorMessage = StringPrintf("%s(%d): Could not get control block", __func__, mPortId);
1930 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001931 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001932 }
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001933 // TODO: Using unsecurePointer() has some associated security pitfalls
1934 // (see declaration for details).
1935 // Either document why it is safe in this case or address the
1936 // issue (e.g. by copying).
1937 void *iMemPointer = iMem->unsecurePointer();
Glenn Kasten0cde0762014-01-16 15:06:36 -08001938 if (iMemPointer == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001939 errorMessage = StringPrintf(
1940 "%s(%d): Could not get control block pointer", __func__, mPortId);
1941 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001942 goto exit;
Glenn Kasten0cde0762014-01-16 15:06:36 -08001943 }
Glenn Kasten53cec222013-08-29 09:01:02 -07001944 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001945 if (mAudioTrack != 0) {
Marco Nelissenf8880202014-11-14 07:58:25 -08001946 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001947 mDeathNotifier.clear();
1948 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001949 mAudioTrack = output.audioTrack;
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001950 mCblkMemory = iMem;
Eric Laurent3bcffa12014-06-12 18:38:45 -07001951 IPCThreadState::self()->flushCommands();
1952
Glenn Kasten0cde0762014-01-16 15:06:36 -08001953 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001954 mCblk = cblk;
Glenn Kasten5f631512014-02-24 15:16:07 -08001955
Glenn Kastena07f17c2013-04-23 12:39:37 -07001956 mAwaitBoost = false;
Glenn Kasten363fb752014-01-15 12:27:31 -08001957 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Eric Laurent21da6472017-11-09 16:29:26 -08001958 if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001959 ALOGI("%s(%d): AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08001960 __func__, mPortId, mReqFrameCount, mFrameCount);
Phil Burk33ff89b2015-11-30 11:16:01 -08001961 if (!mThreadCanCallJava) {
1962 mAwaitBoost = true;
1963 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08001964 } else {
Robert Wuc121cd12021-08-13 17:51:40 +00001965 ALOGV("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08001966 __func__, mPortId, mReqFrameCount, mFrameCount);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001967 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08001968 }
Eric Laurent21da6472017-11-09 16:29:26 -08001969 mFlags = output.flags;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001970
Eric Laurentad2e7b92017-09-14 20:06:42 -07001971 //mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
Eric Laurent09f1ed22019-04-24 17:45:17 -07001972 if (mDeviceCallback != 0) {
Eric Laurentad2e7b92017-09-14 20:06:42 -07001973 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07001974 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001975 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07001976 AudioSystem::addAudioDeviceCallback(this, output.outputId, output.portId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001977 callbackAdded = true;
1978 }
1979
Eric Laurent09f1ed22019-04-24 17:45:17 -07001980 mPortId = output.portId;
Vlad Popaad0fe922022-06-10 00:43:14 +02001981 // notify the upper layers about the new portId
1982 triggerPortIdUpdate_l();
1983
Glenn Kasten38e905b2014-01-13 10:21:48 -08001984 // We retain a copy of the I/O handle, but don't own the reference
Eric Laurent21da6472017-11-09 16:29:26 -08001985 mOutput = output.outputId;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001986 mRefreshRemaining = true;
1987
1988 // Starting address of buffers in shared memory. If there is a shared buffer, buffers
1989 // is the value of pointer() for the shared buffer, otherwise buffers points
1990 // immediately after the control block. This address is for the mapping within client
1991 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
1992 void* buffers;
Glenn Kasten363fb752014-01-15 12:27:31 -08001993 if (mSharedBuffer == 0) {
Glenn Kasten138d6f92015-03-20 10:54:51 -07001994 buffers = cblk + 1;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001995 } else {
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001996 // TODO: Using unsecurePointer() has some associated security pitfalls
1997 // (see declaration for details).
1998 // Either document why it is safe in this case or address the
1999 // issue (e.g. by copying).
2000 buffers = mSharedBuffer->unsecurePointer();
Glenn Kasten138d6f92015-03-20 10:54:51 -07002001 if (buffers == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08002002 errorMessage = StringPrintf(
2003 "%s(%d): Could not get buffer pointer", __func__, mPortId);
2004 ALOGE("%s", errorMessage.c_str());
2005 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08002006 goto exit;
Glenn Kasten138d6f92015-03-20 10:54:51 -07002007 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002008 }
2009
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002010 mAudioTrack->attachAuxEffect(mAuxEffectId, &status);
Glenn Kasten5f631512014-02-24 15:16:07 -08002011
Glenn Kasten093000f2012-05-03 09:35:36 -07002012 // If IAudioTrack is re-created, don't let the requested frameCount
2013 // decrease. This can confuse clients that cache frameCount().
Eric Laurent21da6472017-11-09 16:29:26 -08002014 if (mFrameCount > mReqFrameCount) {
2015 mReqFrameCount = mFrameCount;
Glenn Kasten093000f2012-05-03 09:35:36 -07002016 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08002017
Andy Hungd7bd69e2015-07-24 07:52:41 -07002018 // reset server position to 0 as we have new cblk.
2019 mServer = 0;
2020
Glenn Kastene3aa6592012-12-04 12:22:46 -08002021 // update proxy
Glenn Kasten363fb752014-01-15 12:27:31 -08002022 if (mSharedBuffer == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002023 mStaticProxy.clear();
Eric Laurent21da6472017-11-09 16:29:26 -08002024 mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002025 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08002026 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002027 mProxy = mStaticProxy;
2028 }
seunghak.hane6a9d6582014-11-22 15:22:35 +09002029
2030 mProxy->setVolumeLR(gain_minifloat_pack(
2031 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
2032 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
2033
Glenn Kastene3aa6592012-12-04 12:22:46 -08002034 mProxy->setSendLevel(mSendLevel);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002035 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
2036 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
2037 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07002038 mProxy->setSampleRate(effectiveSampleRate);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002039
2040 AudioPlaybackRate playbackRateTemp = mPlaybackRate;
2041 playbackRateTemp.mSpeed = effectiveSpeed;
2042 playbackRateTemp.mPitch = effectivePitch;
2043 mProxy->setPlaybackRate(playbackRateTemp);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002044 mProxy->setMinimum(mNotificationFramesAct);
2045
Kuowei Lid4adbdb2020-08-13 14:44:25 +08002046 if (mDualMonoMode != AUDIO_DUAL_MONO_MODE_OFF) {
2047 setDualMonoMode_l(mDualMonoMode);
2048 }
2049 if (mAudioDescriptionMixLeveldB != -std::numeric_limits<float>::infinity()) {
2050 setAudioDescriptionMixLevel_l(mAudioDescriptionMixLeveldB);
2051 }
2052
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002053 mDeathNotifier = new DeathNotifier(this);
Marco Nelissenf8880202014-11-14 07:58:25 -08002054 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
Glenn Kastene3aa6592012-12-04 12:22:46 -08002055
Andy Hungb68f5eb2019-12-03 16:49:17 -08002056 // This is the first log sent from the AudioTrack client.
2057 // The creation of the audio track by AudioFlinger (in the code above)
2058 // is the first log of the AudioTrack and must be present before
2059 // any AudioTrack client logs will be accepted.
Andy Hungea840382020-05-05 21:50:17 -07002060
Andy Hungb68f5eb2019-12-03 16:49:17 -08002061 mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(mPortId);
2062 mediametrics::LogItem(mMetricsId)
2063 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE)
2064 // the following are immutable
Andy Hunga629bd12020-06-05 16:03:53 -07002065 .set(AMEDIAMETRICS_PROP_FLAGS, toString(mFlags).c_str())
2066 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -08002067 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
Andy Hung3a5c2f32021-02-17 15:06:42 -08002068 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, mLogSessionId)
Andy Hung839a3062021-02-17 11:15:16 -08002069 .set(AMEDIAMETRICS_PROP_PLAYERIID, mPlayerIId)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002070 .set(AMEDIAMETRICS_PROP_TRACKID, mPortId) // dup from key
Andy Hungb68f5eb2019-12-03 16:49:17 -08002071 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2072 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2073 .set(AMEDIAMETRICS_PROP_THREADID, (int32_t)output.outputId)
2074 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2075 .set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)mRoutedDeviceId)
2076 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2077 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
2078 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
2079 // the following are NOT immutable
2080 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)mVolume[AUDIO_INTERLEAVE_LEFT])
2081 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)mVolume[AUDIO_INTERLEAVE_RIGHT])
2082 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Andy Hung73dc2f92021-12-07 21:50:04 -08002083 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)NO_ERROR)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002084 .set(AMEDIAMETRICS_PROP_AUXEFFECTID, (int32_t)mAuxEffectId)
2085 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2086 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2087 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2088 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2089 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveSampleRate)
2090 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2091 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)effectiveSpeed)
2092 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2093 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)effectivePitch)
2094 .record();
2095
2096 // mSendLevel
2097 // mReqFrameCount?
2098 // mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq
2099 // mLatency, mAfLatency, mAfFrameCount, mAfSampleRate
2100
Glenn Kasten38e905b2014-01-13 10:21:48 -08002101 }
2102
Eric Laurentf32d7812017-11-30 14:44:07 -08002103exit:
Andy Hung3acde2c2021-11-11 09:18:08 -08002104 if (status != NO_ERROR) {
2105 if (callbackAdded) {
2106 // note: mOutput is always valid is callbackAdded is true
2107 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
2108 }
2109 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
2110 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
Eric Laurentad2e7b92017-09-14 20:06:42 -07002111 }
Eric Laurentf32d7812017-11-30 14:44:07 -08002112 mStatus = status;
Eric Laurent21da6472017-11-09 16:29:26 -08002113
2114 // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
Glenn Kasten38e905b2014-01-13 10:21:48 -08002115 return status;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002116}
2117
Andy Hung3acde2c2021-11-11 09:18:08 -08002118void AudioTrack::reportError(status_t status, const char *event, const char *message) const
2119{
2120 if (status == NO_ERROR) return;
2121 // We report error on the native side because some callers do not come
2122 // from Java.
Andy Hungde602302021-12-07 21:35:49 -08002123 // Ensure these variables are initialized in set().
2124 mediametrics::LogItem(AMEDIAMETRICS_KEY_AUDIO_TRACK_ERROR)
Andy Hung3acde2c2021-11-11 09:18:08 -08002125 .set(AMEDIAMETRICS_PROP_EVENT, event)
Andy Hung73dc2f92021-12-07 21:50:04 -08002126 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
2127 .set(AMEDIAMETRICS_PROP_STATUSMESSAGE, message)
Andy Hung3acde2c2021-11-11 09:18:08 -08002128 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
2129 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
2130 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2131 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2132 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2133 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2134 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
Andy Hung3acde2c2021-11-11 09:18:08 -08002135 // the following are NOT immutable
Andy Hungde602302021-12-07 21:35:49 -08002136 // frame count is initially the requested frame count, but may be adjusted
2137 // by AudioFlinger after creation.
2138 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
Andy Hung3acde2c2021-11-11 09:18:08 -08002139 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2140 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2141 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2142 .record();
2143}
2144
Glenn Kastenb46f3942015-03-09 12:00:30 -07002145status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002146{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002147 if (audioBuffer == NULL) {
Glenn Kasten551b5352015-03-20 11:30:28 -07002148 if (nonContig != NULL) {
2149 *nonContig = 0;
2150 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002151 return BAD_VALUE;
Eric Laurent9b7d9502011-03-21 11:49:00 -07002152 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002153 if (mTransfer != TRANSFER_OBTAIN) {
2154 audioBuffer->frameCount = 0;
Atneya Nair03079272022-01-18 17:03:14 -05002155 audioBuffer->mSize = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002156 audioBuffer->raw = NULL;
Glenn Kasten551b5352015-03-20 11:30:28 -07002157 if (nonContig != NULL) {
2158 *nonContig = 0;
2159 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002160 return INVALID_OPERATION;
2161 }
Eric Laurent9b7d9502011-03-21 11:49:00 -07002162
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002163 const struct timespec *requested;
Eric Laurentdf576992014-01-27 18:13:39 -08002164 struct timespec timeout;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002165 if (waitCount == -1) {
2166 requested = &ClientProxy::kForever;
2167 } else if (waitCount == 0) {
2168 requested = &ClientProxy::kNonBlocking;
2169 } else if (waitCount > 0) {
Chih-Hung Hsiehbca74292018-08-10 16:06:07 -07002170 time_t ms = WAIT_PERIOD_MS * (time_t) waitCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002171 timeout.tv_sec = ms / 1000;
Andy Hung06a730b2020-04-09 13:28:31 -07002172 timeout.tv_nsec = (ms % 1000) * 1000000;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002173 requested = &timeout;
2174 } else {
Eric Laurent973db022018-11-20 14:54:31 -08002175 ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002176 requested = NULL;
2177 }
Glenn Kastenb46f3942015-03-09 12:00:30 -07002178 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002179}
Eric Laurent1703cdf2011-03-07 14:52:59 -08002180
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002181status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
2182 struct timespec *elapsed, size_t *nonContig)
2183{
2184 // previous and new IAudioTrack sequence numbers are used to detect track re-creation
2185 uint32_t oldSequence = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002186
2187 Proxy::Buffer buffer;
2188 status_t status = NO_ERROR;
2189
2190 static const int32_t kMaxTries = 5;
2191 int32_t tryCounter = kMaxTries;
2192
2193 do {
2194 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
2195 // keep them from going away if another thread re-creates the track during obtainBuffer()
2196 sp<AudioTrackClientProxy> proxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002197
2198 { // start of lock scope
2199 AutoMutex lock(mLock);
2200
Glenn Kasten305996c2020-01-27 08:03:37 -08002201 uint32_t newSequence = mSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002202 // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
2203 if (status == DEAD_OBJECT) {
2204 // re-create track, unless someone else has already done so
2205 if (newSequence == oldSequence) {
2206 status = restoreTrack_l("obtainBuffer");
2207 if (status != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002208 buffer.mFrameCount = 0;
2209 buffer.mRaw = NULL;
2210 buffer.mNonContig = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002211 break;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002212 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002213 }
2214 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002215 oldSequence = newSequence;
2216
Eric Laurent4d231dc2016-03-11 18:38:23 -08002217 if (status == NOT_ENOUGH_DATA) {
2218 restartIfDisabled();
2219 }
2220
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002221 // Keep the extra references
jiabind42567c2023-03-23 22:01:16 +00002222 mProxyObtainBufferRef = mProxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002223 proxy = mProxy;
jiabind42567c2023-03-23 22:01:16 +00002224 mCblkMemoryObtainBufferRef = mCblkMemory;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002225
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002226 if (mState == STATE_STOPPING) {
2227 status = -EINTR;
2228 buffer.mFrameCount = 0;
2229 buffer.mRaw = NULL;
2230 buffer.mNonContig = 0;
2231 break;
2232 }
2233
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002234 // Non-blocking if track is stopped or paused
2235 if (mState != STATE_ACTIVE) {
2236 requested = &ClientProxy::kNonBlocking;
2237 }
2238
2239 } // end of lock scope
2240
2241 buffer.mFrameCount = audioBuffer->frameCount;
2242 // FIXME starts the requested timeout and elapsed over from scratch
2243 status = proxy->obtainBuffer(&buffer, requested, elapsed);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002244 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002245
2246 audioBuffer->frameCount = buffer.mFrameCount;
Atneya Nair03079272022-01-18 17:03:14 -05002247 audioBuffer->mSize = buffer.mFrameCount * mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002248 audioBuffer->raw = buffer.mRaw;
Glenn Kasten305996c2020-01-27 08:03:37 -08002249 audioBuffer->sequence = oldSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002250 if (nonContig != NULL) {
2251 *nonContig = buffer.mNonContig;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002252 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002253 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002254}
2255
Glenn Kasten54a8a452015-03-09 12:03:00 -07002256void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002257{
Glenn Kasten3f02be22015-03-09 11:59:04 -07002258 // FIXME add error checking on mode, by adding an internal version
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002259 if (mTransfer == TRANSFER_SHARED) {
2260 return;
2261 }
2262
Atneya Nair03079272022-01-18 17:03:14 -05002263 size_t stepCount = audioBuffer->mSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002264 if (stepCount == 0) {
2265 return;
2266 }
2267
2268 Proxy::Buffer buffer;
2269 buffer.mFrameCount = stepCount;
2270 buffer.mRaw = audioBuffer->raw;
Glenn Kastene3aa6592012-12-04 12:22:46 -08002271
jiabind42567c2023-03-23 22:01:16 +00002272 sp<IMemory> tempMemory;
2273 sp<AudioTrackClientProxy> tempProxy;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002274 AutoMutex lock(mLock);
Glenn Kasten305996c2020-01-27 08:03:37 -08002275 if (audioBuffer->sequence != mSequence) {
2276 // This Buffer came from a different IAudioTrack instance, so ignore the releaseBuffer
2277 ALOGD("%s is no-op due to IAudioTrack sequence mismatch %u != %u",
2278 __func__, audioBuffer->sequence, mSequence);
2279 return;
2280 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002281 mReleased += stepCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002282 mInUnderrun = false;
jiabind42567c2023-03-23 22:01:16 +00002283 mProxyObtainBufferRef->releaseBuffer(&buffer);
2284 // The extra reference of shared memory and proxy from `obtainBuffer` is not used after
2285 // calling `releaseBuffer`. Move the extra reference to a temp strong pointer so that it
2286 // will be cleared outside `releaseBuffer`.
2287 tempMemory = std::move(mCblkMemoryObtainBufferRef);
2288 tempProxy = std::move(mProxyObtainBufferRef);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002289
2290 // restart track if it was disabled by audioflinger due to previous underrun
Eric Laurent4d231dc2016-03-11 18:38:23 -08002291 restartIfDisabled();
2292}
2293
2294void AudioTrack::restartIfDisabled()
2295{
2296 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
2297 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002298 ALOGW("%s(%d): releaseBuffer() track %p disabled due to previous underrun, restarting",
Eric Laurent973db022018-11-20 14:54:31 -08002299 __func__, mPortId, this);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002300 // FIXME ignoring status
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002301 status_t status;
2302 mAudioTrack->start(&status);
Eric Laurentdf839842012-05-31 14:27:14 -07002303 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002304}
2305
2306// -------------------------------------------------------------------------
2307
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002308ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002309{
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002310 if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07002311 return INVALID_OPERATION;
2312 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002313
Eric Laurentab5cdba2014-06-09 17:22:27 -07002314 if (isDirect()) {
2315 AutoMutex lock(mLock);
2316 int32_t flags = android_atomic_and(
2317 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
2318 &mCblk->mFlags);
2319 if (flags & CBLK_INVALID) {
2320 return DEAD_OBJECT;
2321 }
2322 }
2323
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002324 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
Jiabin Huang447cea72020-07-28 22:35:18 +00002325 // Validation: user is most-likely passing an error code, and it would
Glenn Kasten99e53b82012-01-19 08:59:58 -08002326 // make the return value ambiguous (actualSize vs error).
Andy Hungfb8ede22018-09-12 19:03:24 -07002327 ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
Eric Laurent973db022018-11-20 14:54:31 -08002328 __func__, mPortId, buffer, userSize, userSize);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002329 return BAD_VALUE;
2330 }
2331
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002332 size_t written = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002333 Buffer audioBuffer;
2334
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002335 while (userSize >= mFrameSize) {
2336 audioBuffer.frameCount = userSize / mFrameSize;
Eric Laurentc2f1f072009-07-17 12:17:14 -07002337
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002338 status_t err = obtainBuffer(&audioBuffer,
2339 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002340 if (err < 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002341 if (written > 0) {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002342 break;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002343 }
Glenn Kasten0a2f1512016-07-22 08:06:37 -07002344 if (err == TIMED_OUT || err == -EINTR) {
2345 err = WOULD_BLOCK;
2346 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002347 return ssize_t(err);
2348 }
2349
Atneya Nair03079272022-01-18 17:03:14 -05002350 size_t toWrite = audioBuffer.size();
2351 memcpy(audioBuffer.raw, buffer, toWrite);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002352 buffer = ((const char *) buffer) + toWrite;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002353 userSize -= toWrite;
2354 written += toWrite;
2355
2356 releaseBuffer(&audioBuffer);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002357 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002358
Andy Hungea2b9c02016-02-12 17:06:53 -08002359 if (written > 0) {
2360 mFramesWritten += written / mFrameSize;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002361
2362 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2363 const sp<AudioTrackThread> t = mAudioTrackThread;
2364 if (t != 0) {
2365 // causes wake up of the playback thread, that will callback the client for
2366 // more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
2367 t->wake();
2368 }
2369 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002370 }
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002371
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002372 return written;
2373}
2374
2375// -------------------------------------------------------------------------
2376
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08002377nsecs_t AudioTrack::processAudioBuffer()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002378{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002379 // Currently the AudioTrack thread is not created if there are no callbacks.
2380 // Would it ever make sense to run the thread, even without callbacks?
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002381 // If so, then replace this by checks at each use for mCallback != NULL.
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002382 LOG_ALWAYS_FATAL_IF(mCblk == NULL);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002383 mLock.lock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002384 sp<IAudioTrackCallback> callback = mCallback.promote();
2385 if (!callback) {
2386 mCallback = nullptr;
Atneya Naire260f5a2022-05-03 17:02:20 -04002387 mLock.unlock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002388 return NS_NEVER;
2389 }
Glenn Kastena07f17c2013-04-23 12:39:37 -07002390 if (mAwaitBoost) {
2391 mAwaitBoost = false;
2392 mLock.unlock();
2393 static const int32_t kMaxTries = 5;
2394 int32_t tryCounter = kMaxTries;
2395 uint32_t pollUs = 10000;
2396 do {
Glenn Kasten8255ba72016-08-23 13:54:23 -07002397 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002398 if (policy == SCHED_FIFO || policy == SCHED_RR) {
2399 break;
2400 }
2401 usleep(pollUs);
2402 pollUs <<= 1;
2403 } while (tryCounter-- > 0);
2404 if (tryCounter < 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002405 ALOGE("%s(%d): did not receive expected priority boost on time",
Eric Laurent973db022018-11-20 14:54:31 -08002406 __func__, mPortId);
Glenn Kastena07f17c2013-04-23 12:39:37 -07002407 }
Glenn Kastenb0dfd462013-07-10 16:52:47 -07002408 // Run again immediately
2409 return 0;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002410 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002411
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002412 // Can only reference mCblk while locked
2413 int32_t flags = android_atomic_and(
Glenn Kasten96f60d82013-07-12 10:21:18 -07002414 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);
Glenn Kastena47f3162012-11-07 10:13:08 -08002415
Andy Hunge8273252024-08-07 16:42:42 -07002416 const bool isOffloaded = isOffloaded_l();
2417 const bool isOffloadedOrDirect = isOffloadedOrDirect_l();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002418 // Check for track invalidation
2419 if (flags & CBLK_INVALID) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002420 // for offloaded tracks restoreTrack_l() will just update the sequence and clear
2421 // AudioSystem cache. We should not exit here but after calling the callback so
2422 // that the upper layers can recreate the track
Andy Hunge8273252024-08-07 16:42:42 -07002423 if (!isOffloadedOrDirect || (mSequence == mObservedSequence)) {
Lajos Molnarf1063e22015-04-17 15:19:42 -07002424 status_t status __unused = restoreTrack_l("processAudioBuffer");
2425 // FIXME unused status
Andy Hung53c3b5f2014-12-15 16:42:05 -08002426 // after restoration, continue below to make sure that the loop and buffer events
2427 // are notified because they have been cleared from mCblk->mFlags above.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002428 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002429 }
2430
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002431 bool waitStreamEnd = mState == STATE_STOPPING;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002432 bool active = mState == STATE_ACTIVE;
2433
2434 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
2435 bool newUnderrun = false;
2436 if (flags & CBLK_UNDERRUN) {
2437#if 0
2438 // Currently in shared buffer mode, when the server reaches the end of buffer,
2439 // the track stays active in continuous underrun state. It's up to the application
2440 // to pause or stop the track, or set the position to a new offset within buffer.
2441 // This was some experimental code to auto-pause on underrun. Keeping it here
2442 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
2443 if (mTransfer == TRANSFER_SHARED) {
2444 mState = STATE_PAUSED;
2445 active = false;
2446 }
2447#endif
2448 if (!mInUnderrun) {
2449 mInUnderrun = true;
2450 newUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002451 }
2452 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002453
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002454 // Get current position of server
Andy Hung90e8a972015-11-09 16:42:40 -08002455 Modulo<uint32_t> position(updateAndGetPosition_l());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002456
2457 // Manage marker callback
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002458 bool markerReached = false;
Andy Hung90e8a972015-11-09 16:42:40 -08002459 Modulo<uint32_t> markerPosition(mMarkerPosition);
2460 // uses 32 bit wraparound for comparison with position.
2461 if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002462 mMarkerReached = markerReached = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002463 }
2464
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002465 // Determine number of new position callback(s) that will be needed, while locked
2466 size_t newPosCount = 0;
Andy Hung90e8a972015-11-09 16:42:40 -08002467 Modulo<uint32_t> newPosition(mNewPosition);
2468 uint32_t updatePeriod = mUpdatePeriod;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002469 // FIXME fails for wraparound, need 64 bits
2470 if (updatePeriod > 0 && position >= newPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002471 newPosCount = ((position - newPosition).value() / updatePeriod) + 1;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002472 mNewPosition += updatePeriod * newPosCount;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002473 }
2474
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002475 // Cache other fields that will be needed soon
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002476 uint32_t sampleRate = mSampleRate;
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002477 float speed = mPlaybackRate.mSpeed;
Andy Hunga7f03352015-05-31 21:54:49 -07002478 const uint32_t notificationFrames = mNotificationFramesAct;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002479 if (mRefreshRemaining) {
2480 mRefreshRemaining = false;
2481 mRemainingFrames = notificationFrames;
2482 mRetryOnPartialBuffer = false;
2483 }
2484 size_t misalignment = mProxy->getMisalignment();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002485 uint32_t sequence = mSequence;
Glenn Kasten96f04882013-09-20 09:28:56 -07002486 sp<AudioTrackClientProxy> proxy = mProxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002487
Andy Hung53c3b5f2014-12-15 16:42:05 -08002488 // Determine the number of new loop callback(s) that will be needed, while locked.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002489 uint32_t loopCountNotifications = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -08002490 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
2491
2492 if (mLoopCount > 0) {
2493 int loopCount;
2494 size_t bufferPosition;
2495 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
2496 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
2497 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
2498 mLoopCountNotified = loopCount; // discard any excess notifications
2499 } else if (mLoopCount < 0) {
2500 // FIXME: We're not accurate with notification count and position with infinite looping
2501 // since loopCount from server side will always return -1 (we could decrement it).
2502 size_t bufferPosition = mStaticProxy->getBufferPosition();
2503 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
2504 loopPeriod = mLoopEnd - bufferPosition;
2505 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
2506 size_t bufferPosition = mStaticProxy->getBufferPosition();
2507 loopPeriod = mFrameCount - bufferPosition;
2508 }
2509
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002510 // These fields don't need to be cached, because they are assigned only by set():
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002511 // mTransfer, mCallback, mUserData, mFormat, mFrameSize, mFlags
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002512 // mFlags is also assigned by createTrack_l(), but not the bit we care about.
2513
2514 mLock.unlock();
2515
Andy Hunga7f03352015-05-31 21:54:49 -07002516 // get anchor time to account for callbacks.
2517 const nsecs_t timeBeforeCallbacks = systemTime();
2518
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002519 if (waitStreamEnd) {
Andy Hunga7f03352015-05-31 21:54:49 -07002520 // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread
2521 // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
2522 // (and make sure we don't callback for more data while we're stopping).
2523 // This helps with position, marker notifications, and track invalidation.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002524 struct timespec timeout;
2525 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
2526 timeout.tv_nsec = 0;
2527
Andy Hung45b8cbe2023-03-29 20:31:47 -07002528 // Use timestamp progress to safeguard we don't falsely time out.
2529 AudioTimestamp timestamp{};
2530 const bool isTimestampValid = getTimestamp(timestamp) == OK;
2531 const auto frameCount = isTimestampValid ? timestamp.mPosition : 0;
2532
Glenn Kasten96f04882013-09-20 09:28:56 -07002533 status_t status = proxy->waitStreamEndDone(&timeout);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002534 switch (status) {
Andy Hung45b8cbe2023-03-29 20:31:47 -07002535 case TIMED_OUT:
2536 if (isTimestampValid
2537 && getTimestamp(timestamp) == OK && frameCount != timestamp.mPosition) {
2538 ALOGD("%s: waitStreamEndDone retrying", __func__);
2539 break; // we retry again (and recheck possible state change).
2540 }
2541 [[fallthrough]];
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002542 case NO_ERROR:
2543 case DEAD_OBJECT:
Andy Hung39609a02015-09-03 16:38:38 -07002544 if (status != DEAD_OBJECT) {
2545 // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop();
2546 // instead, the application should handle the EVENT_NEW_IAUDIOTRACK.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002547 callback->onStreamEnd();
Andy Hung39609a02015-09-03 16:38:38 -07002548 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002549 {
2550 AutoMutex lock(mLock);
2551 // The previously assigned value of waitStreamEnd is no longer valid,
2552 // since the mutex has been unlocked and either the callback handler
2553 // or another thread could have re-started the AudioTrack during that time.
2554 waitStreamEnd = mState == STATE_STOPPING;
2555 if (waitStreamEnd) {
2556 mState = STATE_STOPPED;
Andy Hungc2813e52014-10-16 17:54:34 -07002557 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002558 }
2559 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002560 if (waitStreamEnd && status != DEAD_OBJECT) {
Andy Hung45b8cbe2023-03-29 20:31:47 -07002561 ALOGV("%s: waitStreamEndDone complete", __func__);
Glenn Kasten96f04882013-09-20 09:28:56 -07002562 return NS_INACTIVE;
2563 }
2564 break;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002565 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002566 return 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002567 }
2568
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002569 // perform callbacks while unlocked
2570 if (newUnderrun) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002571 callback->onUnderrun();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002572 }
Andy Hung53c3b5f2014-12-15 16:42:05 -08002573 while (loopCountNotifications > 0) {
Andy Hung53c3b5f2014-12-15 16:42:05 -08002574 --loopCountNotifications;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002575 callback->onLoopEnd(mLoopCount > 0 ? loopCountNotifications + mLoopCountNotified : -1);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002576 }
2577 if (flags & CBLK_BUFFER_END) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002578 callback->onBufferEnd();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002579 }
2580 if (markerReached) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002581 callback->onMarker(markerPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002582 }
2583 while (newPosCount > 0) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002584 callback->onNewPos(newPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002585 newPosition += updatePeriod;
2586 newPosCount--;
2587 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002588
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002589 if (mObservedSequence != sequence) {
2590 mObservedSequence = sequence;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002591 callback->onNewIAudioTrack();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002592 // for offloaded tracks, just wait for the upper layers to recreate the track
Andy Hunge8273252024-08-07 16:42:42 -07002593 if (isOffloadedOrDirect) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002594 return NS_INACTIVE;
2595 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002596 }
2597
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002598 // if inactive, then don't run me again until re-started
2599 if (!active) {
2600 return NS_INACTIVE;
Eric Laurent2267ba12011-09-07 11:13:23 -07002601 }
2602
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002603 // Compute the estimated time until the next timed event (position, markers, loops)
2604 // FIXME only for non-compressed audio
2605 uint32_t minFrames = ~0;
2606 if (!markerReached && position < markerPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002607 minFrames = (markerPosition - position).value();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002608 }
2609 if (loopPeriod > 0 && loopPeriod < minFrames) {
Andy Hung2d85f092015-01-07 12:45:13 -08002610 // loopPeriod is already adjusted for actual position.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002611 minFrames = loopPeriod;
2612 }
Andy Hung2d85f092015-01-07 12:45:13 -08002613 if (updatePeriod > 0) {
Andy Hung90e8a972015-11-09 16:42:40 -08002614 minFrames = min(minFrames, (newPosition - position).value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002615 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002616
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002617 // If > 0, poll periodically to recover from a stuck server. A good value is 2.
2618 static const uint32_t kPoll = 0;
2619 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
2620 minFrames = kPoll * notificationFrames;
2621 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002622
Andy Hunga7f03352015-05-31 21:54:49 -07002623 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
2624 static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL;
2625 const nsecs_t timeAfterCallbacks = systemTime();
2626
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002627 // Convert frame units to time units
2628 nsecs_t ns = NS_WHENEVER;
2629 if (minFrames != (uint32_t) ~0) {
jiabinc7bb8322017-09-06 18:20:11 -07002630 // AudioFlinger consumption of client data may be irregular when coming out of device
2631 // standby since the kernel buffers require filling. This is throttled to no more than 2x
2632 // the expected rate in the MixerThread. Hence, we reduce the estimated time to wait by one
2633 // half (but no more than half a second) to improve callback accuracy during these temporary
2634 // data surges.
2635 const nsecs_t estimatedNs = framesToNanoseconds(minFrames, sampleRate, speed);
2636 constexpr nsecs_t maxThrottleCompensationNs = 500000000LL;
2637 ns = estimatedNs - min(estimatedNs / 2, maxThrottleCompensationNs) + kWaitPeriodNs;
Andy Hunga7f03352015-05-31 21:54:49 -07002638 ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time
2639 // TODO: Should we warn if the callback time is too long?
2640 if (ns < 0) ns = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002641 }
2642
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002643 // If not supplying data by EVENT_MORE_DATA or EVENT_CAN_WRITE_MORE_DATA, then we're done
2644 if (mTransfer != TRANSFER_CALLBACK && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002645 return ns;
2646 }
2647
Andy Hunga7f03352015-05-31 21:54:49 -07002648 // EVENT_MORE_DATA callback handling.
2649 // Timing for linear pcm audio data formats can be derived directly from the
2650 // buffer fill level.
2651 // Timing for compressed data is not directly available from the buffer fill level,
2652 // rather indirectly from waiting for blocking mode callbacks or waiting for obtain()
2653 // to return a certain fill level.
2654
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002655 struct timespec timeout;
2656 const struct timespec *requested = &ClientProxy::kForever;
2657 if (ns != NS_WHENEVER) {
2658 timeout.tv_sec = ns / 1000000000LL;
2659 timeout.tv_nsec = ns % 1000000000LL;
Andy Hungfb8ede22018-09-12 19:03:24 -07002660 ALOGV("%s(%d): timeout %ld.%03d",
Eric Laurent973db022018-11-20 14:54:31 -08002661 __func__, mPortId, timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002662 requested = &timeout;
2663 }
2664
Andy Hungea2b9c02016-02-12 17:06:53 -08002665 size_t writtenFrames = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002666 while (mRemainingFrames > 0) {
2667
2668 Buffer audioBuffer;
2669 audioBuffer.frameCount = mRemainingFrames;
2670 size_t nonContig;
2671 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
2672 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
Andy Hungfb8ede22018-09-12 19:03:24 -07002673 "%s(%d): obtainBuffer() err=%d frameCount=%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002674 __func__, mPortId, err, audioBuffer.frameCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002675 requested = &ClientProxy::kNonBlocking;
2676 size_t avail = audioBuffer.frameCount + nonContig;
Andy Hungfb8ede22018-09-12 19:03:24 -07002677 ALOGV("%s(%d): obtainBuffer(%u) returned %zu = %zu + %zu err %d",
Eric Laurent973db022018-11-20 14:54:31 -08002678 __func__, mPortId, mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002679 if (err != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002680 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
Andy Hunge8273252024-08-07 16:42:42 -07002681 (isOffloaded && (err == DEAD_OBJECT))) {
Glenn Kasten606fbc12015-10-22 15:28:15 -07002682 // FIXME bug 25195759
2683 return 1000000;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002684 }
Andy Hungfb8ede22018-09-12 19:03:24 -07002685 ALOGE("%s(%d): Error %d obtaining an audio buffer, giving up.",
Eric Laurent973db022018-11-20 14:54:31 -08002686 __func__, mPortId, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002687 return NS_NEVER;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002688 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002689
Phil Burkfdb3c072016-02-09 10:47:02 -08002690 if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002691 mRetryOnPartialBuffer = false;
2692 if (avail < mRemainingFrames) {
Andy Hunga7f03352015-05-31 21:54:49 -07002693 if (ns > 0) { // account for obtain time
2694 const nsecs_t timeNow = systemTime();
2695 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2696 }
Andy Hungeb6e6502019-04-29 13:47:40 -07002697
2698 // delayNs is first computed by the additional frames required in the buffer.
2699 nsecs_t delayNs = framesToNanoseconds(
2700 mRemainingFrames - avail, sampleRate, speed);
2701
2702 // afNs is the AudioFlinger mixer period in ns.
2703 const nsecs_t afNs = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2704
2705 // If the AudioTrack is double buffered based on the AudioFlinger mixer period,
2706 // we may have a race if we wait based on the number of frames desired.
2707 // This is a possible issue with resampling and AAudio.
2708 //
2709 // The granularity of audioflinger processing is one mixer period; if
2710 // our wait time is less than one mixer period, wait at most half the period.
2711 if (delayNs < afNs) {
2712 delayNs = std::min(delayNs, afNs / 2);
2713 }
2714
2715 // adjust our ns wait by delayNs.
2716 if (ns < 0 /* NS_WHENEVER */ || delayNs < ns) {
2717 ns = delayNs;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002718 }
2719 return ns;
2720 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07002721 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002722
Atneya Nair03079272022-01-18 17:03:14 -05002723 size_t reqSize = audioBuffer.size();
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002724 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2725 // when notifying client it can write more data, pass the total size that can be
2726 // written in the next write() call, since it's not passed through the callback
Atneya Nair03079272022-01-18 17:03:14 -05002727 audioBuffer.mSize += nonContig;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002728 }
Atneya Nairc2dd1272021-10-26 19:39:51 -04002729 const size_t writtenSize = (mTransfer == TRANSFER_CALLBACK)
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002730 ? callback->onMoreData(audioBuffer)
2731 : callback->onCanWriteMoreData(audioBuffer);
Jiabin Huang447cea72020-07-28 22:35:18 +00002732 // Validate on returned size
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002733 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002734 ALOGE("%s(%d): EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
Eric Laurent973db022018-11-20 14:54:31 -08002735 __func__, mPortId, reqSize, ssize_t(writtenSize));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002736 return NS_NEVER;
2737 }
2738
2739 if (writtenSize == 0) {
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002740 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2741 // The callback EVENT_CAN_WRITE_MORE_DATA was processed in the JNI of
2742 // android.media.AudioTrack. The JNI is not using the callback to provide data,
2743 // it only signals to the Java client that it can provide more data, which
2744 // this track is read to accept now.
2745 // The playback thread will be awaken at the next ::write()
2746 return NS_WHENEVER;
2747 }
The Android Open Source Project8555d082009-03-05 14:34:35 -08002748 // The callback is done filling buffers
2749 // Keep this thread going to handle timed events and
2750 // still try to get more data in intervals of WAIT_PERIOD_MS
2751 // but don't just loop and block the CPU, so wait
Andy Hunga7f03352015-05-31 21:54:49 -07002752
2753 // mCbf(EVENT_MORE_DATA, ...) might either
2754 // (1) Block until it can fill the buffer, returning 0 size on EOS.
2755 // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS.
2756 // (3) Return 0 size when no data is available, does not wait for more data.
2757 //
2758 // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer.
2759 // We try to compute the wait time to avoid a tight sleep-wait cycle,
2760 // especially for case (3).
2761 //
2762 // The decision to support (1) and (2) affect the sizing of mRemainingFrames
2763 // and this loop; whereas for case (3) we could simply check once with the full
2764 // buffer size and skip the loop entirely.
2765
2766 nsecs_t myns;
Andy Hunge8273252024-08-07 16:42:42 -07002767 if (!isOffloaded && audio_has_proportional_frames(mFormat)) {
Andy Hunga7f03352015-05-31 21:54:49 -07002768 // time to wait based on buffer occupancy
2769 const nsecs_t datans = mRemainingFrames <= avail ? 0 :
2770 framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
2771 // audio flinger thread buffer size (TODO: adjust for fast tracks)
Glenn Kastenea38ee72016-04-18 11:08:01 -07002772 // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks.
Andy Hunga7f03352015-05-31 21:54:49 -07002773 const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2774 // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0.
2775 myns = datans + (afns / 2);
2776 } else {
2777 // FIXME: This could ping quite a bit if the buffer isn't full.
2778 // Note that when mState is stopping we waitStreamEnd, so it never gets here.
2779 myns = kWaitPeriodNs;
2780 }
2781 if (ns > 0) { // account for obtain and callback time
2782 const nsecs_t timeNow = systemTime();
2783 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2784 }
2785 if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
2786 ns = myns;
2787 }
2788 return ns;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002789 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002790
Atneya Nairc2dd1272021-10-26 19:39:51 -04002791 // releaseBuffer reads from audioBuffer.size
Atneya Nair03079272022-01-18 17:03:14 -05002792 audioBuffer.mSize = writtenSize;
Atneya Nairc2dd1272021-10-26 19:39:51 -04002793
Glenn Kasten138d6f92015-03-20 10:54:51 -07002794 size_t releasedFrames = writtenSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002795 audioBuffer.frameCount = releasedFrames;
2796 mRemainingFrames -= releasedFrames;
2797 if (misalignment >= releasedFrames) {
2798 misalignment -= releasedFrames;
2799 } else {
2800 misalignment = 0;
2801 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002802
2803 releaseBuffer(&audioBuffer);
Andy Hungea2b9c02016-02-12 17:06:53 -08002804 writtenFrames += releasedFrames;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002805
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002806 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
2807 // if callback doesn't like to accept the full chunk
2808 if (writtenSize < reqSize) {
2809 continue;
2810 }
2811
2812 // There could be enough non-contiguous frames available to satisfy the remaining request
2813 if (mRemainingFrames <= nonContig) {
2814 continue;
2815 }
2816
2817#if 0
2818 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
2819 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
2820 // that total to a sum == notificationFrames.
2821 if (0 < misalignment && misalignment <= mRemainingFrames) {
2822 mRemainingFrames = misalignment;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002823 return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002824 }
2825#endif
2826
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002827 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002828 if (writtenFrames > 0) {
2829 AutoMutex lock(mLock);
2830 mFramesWritten += writtenFrames;
2831 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002832 mRemainingFrames = notificationFrames;
2833 mRetryOnPartialBuffer = true;
2834
2835 // A lot has transpired since ns was calculated, so run again immediately and re-calculate
2836 return 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002837}
2838
Kuowei Li72c8b062023-08-31 13:38:32 +08002839status_t AudioTrack::restoreTrack_l(const char *from, bool forceRestore)
Eric Laurent1703cdf2011-03-07 14:52:59 -08002840{
Andy Hungb68f5eb2019-12-03 16:49:17 -08002841 status_t result = NO_ERROR; // logged: make sure to set this before returning.
2842 const int64_t beginNs = systemTime();
Andy Hunga6b27032020-04-27 10:34:24 -07002843 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -08002844 mediametrics::LogItem(mMetricsId)
2845 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_RESTORE)
Andy Hungea840382020-05-05 21:50:17 -07002846 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08002847 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
2848 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
2849 .set(AMEDIAMETRICS_PROP_WHERE, from)
2850 .record(); });
2851
Andy Hungfb8ede22018-09-12 19:03:24 -07002852 ALOGW("%s(%d): dead IAudioTrack, %s, creating a new one from %s()",
Eric Laurent973db022018-11-20 14:54:31 -08002853 __func__, mPortId, isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002854 ++mSequence;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002855
Glenn Kastena47f3162012-11-07 10:13:08 -08002856 // refresh the audio configuration cache in this process to make sure we get new
Glenn Kastend2d089f2014-11-05 11:48:12 -08002857 // output parameters and new IAudioFlinger in createTrack_l()
Glenn Kastena47f3162012-11-07 10:13:08 -08002858 AudioSystem::clearAudioConfigCache();
Eric Laurent9f6530f2011-08-30 10:18:54 -07002859
Kuowei Li72c8b062023-08-31 13:38:32 +08002860 if (!forceRestore &&
2861 (isOffloadedOrDirect_l() || mDoNotReconnect)) {
Andy Hung1f1db832015-06-08 13:26:10 -07002862 // FIXME re-creation of offloaded and direct tracks is not yet implemented;
Atneya Nairb16666a2023-12-11 20:18:33 -08002863 // Disabled since (1) timestamp correction is not implemented for non-PCM and
2864 // (2) We pre-empt existing direct tracks on resource constraint, so these tracks
2865 // shouldn't reconnect.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002866 result = DEAD_OBJECT;
2867 return result;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002868 }
2869
Phil Burk2812d9e2016-01-04 10:34:30 -08002870 // Save so we can return count since creation.
2871 mUnderrunCountOffset = getUnderrunCount_l();
2872
Glenn Kasten200092b2014-08-15 15:13:30 -07002873 // save the old static buffer position
Andy Hungf20a4e92016-08-15 19:10:34 -07002874 uint32_t staticPosition = 0;
Andy Hung4ede21d2014-12-12 15:37:34 -08002875 size_t bufferPosition = 0;
2876 int loopCount = 0;
2877 if (mStaticProxy != 0) {
2878 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
Andy Hungf20a4e92016-08-15 19:10:34 -07002879 staticPosition = mStaticProxy->getPosition().unsignedValue();
Andy Hung4ede21d2014-12-12 15:37:34 -08002880 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002881
Andy Hung3c7f47a2021-03-16 17:30:09 -07002882 // save the old startThreshold and framecount
2883 const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
2884 const uint32_t originalFrameCount = mProxy->frameCount();
2885
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002886 // See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
2887 // causes a lot of churn on the service side, and it can reject starting
2888 // playback of a previously created track. May also apply to other cases.
2889 const int INITIAL_RETRIES = 3;
2890 int retries = INITIAL_RETRIES;
2891retry:
2892 if (retries < INITIAL_RETRIES) {
2893 // See the comment for clearAudioConfigCache at the start of the function.
2894 AudioSystem::clearAudioConfigCache();
2895 }
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08002896 mFlags = mOrigFlags;
2897
Glenn Kasten200092b2014-08-15 15:13:30 -07002898 // If a new IAudioTrack is successfully created, createTrack_l() will modify the
Glenn Kastena47f3162012-11-07 10:13:08 -08002899 // following member variables: mAudioTrack, mCblkMemory and mCblk.
Glenn Kasten200092b2014-08-15 15:13:30 -07002900 // It will also delete the strong references on previous IAudioTrack and IMemory.
2901 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002902 result = createTrack_l();
Eric Laurentcc21e4f2013-10-16 15:12:32 -07002903
Eric Laurent6ec546d2018-10-10 16:52:14 -07002904 if (result == NO_ERROR) {
Andy Hungd7bd69e2015-07-24 07:52:41 -07002905 // take the frames that will be lost by track recreation into account in saved position
2906 // For streaming tracks, this is the amount we obtained from the user/client
2907 // (not the number actually consumed at the server - those are already lost).
2908 if (mStaticProxy == 0) {
2909 mPosition = mReleased;
2910 }
Andy Hung4ede21d2014-12-12 15:37:34 -08002911 // Continue playback from last known position and restore loop.
2912 if (mStaticProxy != 0) {
2913 if (loopCount != 0) {
2914 mStaticProxy->setBufferPositionAndLoop(bufferPosition,
2915 mLoopStart, mLoopEnd, loopCount);
2916 } else {
2917 mStaticProxy->setBufferPosition(bufferPosition);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002918 if (bufferPosition == mFrameCount) {
Eric Laurent973db022018-11-20 14:54:31 -08002919 ALOGD("%s(%d): restoring track at end of static buffer", __func__, mPortId);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002920 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002921 }
2922 }
Andy Hung4ef88d72017-02-21 19:47:53 -08002923 // restore volume handler
Andy Hung39399b62017-04-21 15:07:45 -07002924 mVolumeHandler->forall([this](const VolumeShaper &shaper) -> VolumeShaper::Status {
2925 sp<VolumeShaper::Operation> operationToEnd =
2926 new VolumeShaper::Operation(shaper.mOperation);
Andy Hung4ef88d72017-02-21 19:47:53 -08002927 // TODO: Ideally we would restore to the exact xOffset position
2928 // as returned by getVolumeShaperState(), but we don't have that
2929 // information when restoring at the client unless we periodically poll
2930 // the server or create shared memory state.
2931 //
Andy Hung39399b62017-04-21 15:07:45 -07002932 // For now, we simply advance to the end of the VolumeShaper effect
2933 // if it has been started.
2934 if (shaper.isStarted()) {
Andy Hungf3702642017-05-05 17:33:32 -07002935 operationToEnd->setNormalizedTime(1.f);
Andy Hung39399b62017-04-21 15:07:45 -07002936 }
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002937 media::VolumeShaperConfiguration config;
2938 shaper.mConfiguration->writeToParcelable(&config);
2939 media::VolumeShaperOperation operation;
2940 operationToEnd->writeToParcelable(&operation);
2941 status_t status;
2942 mAudioTrack->applyVolumeShaper(config, operation, &status);
2943 return status;
Andy Hung4ef88d72017-02-21 19:47:53 -08002944 });
2945
Andy Hung3c7f47a2021-03-16 17:30:09 -07002946 // restore the original start threshold if different than frameCount.
2947 if (originalStartThresholdInFrames != originalFrameCount) {
2948 // Note: mProxy->setStartThresholdInFrames() call is in the Proxy
2949 // and does not trigger a restart.
2950 // (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
2951 // Any start would be triggered on the mState == ACTIVE check below.
2952 const uint32_t currentThreshold =
2953 mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
2954 ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
2955 "%s(%d) startThresholdInFrames changing from %u to %u",
2956 __func__, mPortId, originalStartThresholdInFrames, currentThreshold);
2957 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002958 if (mState == STATE_ACTIVE) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002959 mAudioTrack->start(&result);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002960 }
Andy Hungf20a4e92016-08-15 19:10:34 -07002961 // server resets to zero so we offset
2962 mFramesWrittenServerOffset =
2963 mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten;
2964 mFramesWrittenAtRestore = mFramesWrittenServerOffset;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002965 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002966 if (result != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08002967 ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002968 if (--retries > 0) {
Eric Laurent6ec546d2018-10-10 16:52:14 -07002969 // leave time for an eventual race condition to clear before retrying
2970 usleep(500000);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002971 goto retry;
2972 }
Eric Laurent6ec546d2018-10-10 16:52:14 -07002973 // if no retries left, set invalid bit to force restoring at next occasion
2974 // and avoid inconsistent active state on client and server sides
2975 if (mCblk != nullptr) {
2976 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
2977 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002978 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002979 return result;
2980}
2981
Andy Hung90e8a972015-11-09 16:42:40 -08002982Modulo<uint32_t> AudioTrack::updateAndGetPosition_l()
Glenn Kasten200092b2014-08-15 15:13:30 -07002983{
2984 // This is the sole place to read server consumed frames
Andy Hung90e8a972015-11-09 16:42:40 -08002985 Modulo<uint32_t> newServer(mProxy->getPosition());
2986 const int32_t delta = (newServer - mServer).signedValue();
Glenn Kasten200092b2014-08-15 15:13:30 -07002987 // TODO There is controversy about whether there can be "negative jitter" in server position.
2988 // This should be investigated further, and if possible, it should be addressed.
2989 // A more definite failure mode is infrequent polling by client.
2990 // One could call (void)getPosition_l() in releaseBuffer(),
2991 // so mReleased and mPosition are always lock-step as best possible.
2992 // That should ensure delta never goes negative for infrequent polling
2993 // unless the server has more than 2^31 frames in its buffer,
2994 // in which case the use of uint32_t for these counters has bigger issues.
Andy Hung90e8a972015-11-09 16:42:40 -08002995 ALOGE_IF(delta < 0,
Andy Hungfb8ede22018-09-12 19:03:24 -07002996 "%s(%d): detected illegal retrograde motion by the server: mServer advanced by %d",
Eric Laurent973db022018-11-20 14:54:31 -08002997 __func__, mPortId, delta);
Chad Brubaker039c27a2015-09-23 15:17:29 -07002998 mServer = newServer;
Andy Hung90e8a972015-11-09 16:42:40 -08002999 if (delta > 0) { // avoid retrograde
3000 mPosition += delta;
3001 }
3002 return mPosition;
Glenn Kasten200092b2014-08-15 15:13:30 -07003003}
3004
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003005bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed)
Andy Hung8edb8dc2015-03-26 19:13:55 -07003006{
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003007 updateLatency_l();
Andy Hung8edb8dc2015-03-26 19:13:55 -07003008 // applicable for mixing tracks only (not offloaded or direct)
3009 if (mStaticProxy != 0) {
3010 return true; // static tracks do not have issues with buffer sizing.
3011 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07003012 const size_t minFrameCount =
Eric Laurent21da6472017-11-09 16:29:26 -08003013 AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
3014 sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003015 const bool allowed = mFrameCount >= minFrameCount;
3016 ALOGD_IF(!allowed,
Andy Hungfb8ede22018-09-12 19:03:24 -07003017 "%s(%d): denied "
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003018 "mAfLatency:%u mAfFrameCount:%zu mAfSampleRate:%u sampleRate:%u speed:%f "
3019 "mFrameCount:%zu < minFrameCount:%zu",
Eric Laurent973db022018-11-20 14:54:31 -08003020 __func__, mPortId,
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003021 mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed,
Andy Hung8edb8dc2015-03-26 19:13:55 -07003022 mFrameCount, minFrameCount);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003023 return allowed;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003024}
3025
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003026status_t AudioTrack::setParameters(const String8& keyValuePairs)
3027{
3028 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003029 status_t status;
3030 mAudioTrack->setParameters(keyValuePairs.c_str(), &status);
3031 return status;
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003032}
3033
Dean Wheatleya70eef72018-01-04 14:23:50 +11003034status_t AudioTrack::selectPresentation(int presentationId, int programId)
3035{
3036 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08003037 AudioParameter param = AudioParameter();
3038 param.addInt(String8(AudioParameter::keyPresentationId), presentationId);
3039 param.addInt(String8(AudioParameter::keyProgramId), programId);
3040 ALOGV("%s(%d): PresentationId/ProgramId[%s]",
Tomasz Wasilczyk833345b2023-08-15 20:59:35 +00003041 __func__, mPortId, param.toString().c_str());
Eric Laurent973db022018-11-20 14:54:31 -08003042
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003043 status_t status;
3044 mAudioTrack->setParameters(param.toString().c_str(), &status);
3045 return status;
Dean Wheatleya70eef72018-01-04 14:23:50 +11003046}
3047
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003048VolumeShaper::Status AudioTrack::applyVolumeShaper(
3049 const sp<VolumeShaper::Configuration>& configuration,
3050 const sp<VolumeShaper::Operation>& operation)
3051{
Andy Hung23f81622024-06-07 18:48:49 -07003052 const int64_t beginNs = systemTime();
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003053 AutoMutex lock(mLock);
Andy Hung4ef88d72017-02-21 19:47:53 -08003054 mVolumeHandler->setIdIfNecessary(configuration);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003055 media::VolumeShaperConfiguration config;
3056 configuration->writeToParcelable(&config);
3057 media::VolumeShaperOperation op;
3058 operation->writeToParcelable(&op);
3059 VolumeShaper::Status status;
Andy Hung23f81622024-06-07 18:48:49 -07003060
3061 mediametrics::Defer defer([&] {
3062 mediametrics::LogItem(mMetricsId)
3063 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_APPLYVOLUMESHAPER)
3064 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
3065 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
3066 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
3067 .set(AMEDIAMETRICS_PROP_TOSTRING, configuration->toString()
3068 .append(" ")
3069 .append(operation->toString()))
3070 .record(); });
3071
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003072 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003073
3074 if (status == DEAD_OBJECT) {
3075 if (restoreTrack_l("applyVolumeShaper") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003076 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003077 }
3078 }
Andy Hung4ef88d72017-02-21 19:47:53 -08003079 if (status >= 0) {
3080 // save VolumeShaper for restore
3081 mVolumeHandler->applyVolumeShaper(configuration, operation);
Andy Hung39399b62017-04-21 15:07:45 -07003082 if (mState == STATE_ACTIVE || mState == STATE_STOPPING) {
3083 mVolumeHandler->setStarted();
3084 }
3085 } else {
3086 // warn only if not an expected restore failure.
3087 ALOGW_IF(!((isOffloadedOrDirect_l() || mDoNotReconnect) && status == DEAD_OBJECT),
Eric Laurent973db022018-11-20 14:54:31 -08003088 "%s(%d): applyVolumeShaper failed: %d", __func__, mPortId, status);
Andy Hung4ef88d72017-02-21 19:47:53 -08003089 }
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003090 return status;
3091}
3092
3093sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id)
3094{
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003095 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003096 std::optional<media::VolumeShaperState> vss;
3097 mAudioTrack->getVolumeShaperState(id, &vss);
3098 sp<VolumeShaper::State> state;
3099 if (vss.has_value()) {
3100 state = new VolumeShaper::State();
3101 state->readFromParcelable(vss.value());
3102 }
Andy Hung39399b62017-04-21 15:07:45 -07003103 if (state.get() == nullptr && (mCblk->mFlags & CBLK_INVALID) != 0) {
3104 if (restoreTrack_l("getVolumeShaperState") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003105 mAudioTrack->getVolumeShaperState(id, &vss);
3106 if (vss.has_value()) {
3107 state = new VolumeShaper::State();
3108 state->readFromParcelable(vss.value());
3109 }
Andy Hung39399b62017-04-21 15:07:45 -07003110 }
3111 }
3112 return state;
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003113}
3114
Andy Hungea2b9c02016-02-12 17:06:53 -08003115status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp)
3116{
3117 if (timestamp == nullptr) {
3118 return BAD_VALUE;
3119 }
3120 AutoMutex lock(mLock);
Andy Hunge13f8a62016-03-30 14:20:42 -07003121 return getTimestamp_l(timestamp);
3122}
3123
3124status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp)
3125{
Andy Hungea2b9c02016-02-12 17:06:53 -08003126 if (mCblk->mFlags & CBLK_INVALID) {
3127 const status_t status = restoreTrack_l("getTimestampExtended");
3128 if (status != OK) {
3129 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3130 // recommending that the track be recreated.
3131 return DEAD_OBJECT;
3132 }
3133 }
3134 // check for offloaded/direct here in case restoring somehow changed those flags.
3135 if (isOffloadedOrDirect_l()) {
3136 return INVALID_OPERATION; // not supported
3137 }
3138 status_t status = mProxy->getTimestamp(timestamp);
Andy Hungfb8ede22018-09-12 19:03:24 -07003139 LOG_ALWAYS_FATAL_IF(status != OK, "%s(%d): status %d not allowed from proxy getTimestamp",
Eric Laurent973db022018-11-20 14:54:31 -08003140 __func__, mPortId, status);
Andy Hungea2b9c02016-02-12 17:06:53 -08003141 bool found = false;
Andy Hunge1e98462016-04-12 10:18:51 -07003142 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten;
3143 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
3144 // server side frame offset in case AudioTrack has been restored.
3145 for (int i = ExtendedTimestamp::LOCATION_SERVER;
3146 i < ExtendedTimestamp::LOCATION_MAX; ++i) {
3147 if (timestamp->mTimeNs[i] >= 0) {
3148 // apply server offset (frames flushed is ignored
3149 // so we don't report the jump when the flush occurs).
3150 timestamp->mPosition[i] += mFramesWrittenServerOffset;
3151 found = true;
Andy Hungea2b9c02016-02-12 17:06:53 -08003152 }
3153 }
3154 return found ? OK : WOULD_BLOCK;
3155}
3156
Glenn Kastence703742013-07-19 16:33:58 -07003157status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
3158{
Glenn Kasten53cec222013-08-29 09:01:02 -07003159 AutoMutex lock(mLock);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003160 return getTimestamp_l(timestamp);
3161}
Phil Burk1b420972015-04-22 10:52:21 -07003162
Andy Hung65ffdfc2016-10-10 15:52:11 -07003163status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp)
3164{
Phil Burk1b420972015-04-22 10:52:21 -07003165 bool previousTimestampValid = mPreviousTimestampValid;
3166 // Set false here to cover all the error return cases.
3167 mPreviousTimestampValid = false;
3168
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003169 switch (mState) {
3170 case STATE_ACTIVE:
3171 case STATE_PAUSED:
3172 break; // handle below
3173 case STATE_FLUSHED:
3174 case STATE_STOPPED:
3175 return WOULD_BLOCK;
3176 case STATE_STOPPING:
3177 case STATE_PAUSED_STOPPING:
3178 if (!isOffloaded_l()) {
3179 return INVALID_OPERATION;
3180 }
3181 break; // offloaded tracks handled below
3182 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003183 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in getTimestamp(): %d",
Eric Laurent973db022018-11-20 14:54:31 -08003184 __func__, mPortId, mState);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003185 break;
Glenn Kastenfe346c72013-08-30 13:28:22 -07003186 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003187
Eric Laurent275e8e92014-11-30 15:14:47 -08003188 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung6653c932015-06-08 13:27:48 -07003189 const status_t status = restoreTrack_l("getTimestamp");
3190 if (status != OK) {
3191 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3192 // recommending that the track be recreated.
3193 return DEAD_OBJECT;
3194 }
Eric Laurent275e8e92014-11-30 15:14:47 -08003195 }
3196
Glenn Kasten200092b2014-08-15 15:13:30 -07003197 // The presented frame count must always lag behind the consumed frame count.
3198 // To avoid a race, read the presented frames first. This ensures that presented <= consumed.
Andy Hung6ae58432016-02-16 18:32:24 -08003199
3200 status_t status;
jiabin94ed47c2023-07-27 23:34:20 +00003201 if (isAfTrackOffloadedOrDirect_l()) {
Andy Hung6ae58432016-02-16 18:32:24 -08003202 // use Binder to get timestamp
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003203 media::AudioTimestampInternal ts;
3204 mAudioTrack->getTimestamp(&ts, &status);
3205 if (status == OK) {
Andy Hung973638a2020-12-08 20:47:45 -08003206 timestamp = VALUE_OR_FATAL(aidl2legacy_AudioTimestampInternal_AudioTimestamp(ts));
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003207 }
Andy Hung6ae58432016-02-16 18:32:24 -08003208 } else {
3209 // read timestamp from shared memory
3210 ExtendedTimestamp ets;
3211 status = mProxy->getTimestamp(&ets);
3212 if (status == OK) {
Andy Hungb01faa32016-04-27 12:51:32 -07003213 ExtendedTimestamp::Location location;
3214 status = ets.getBestTimestamp(&timestamp, &location);
3215
3216 if (status == OK) {
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003217 updateLatency_l();
Andy Hungb01faa32016-04-27 12:51:32 -07003218 // It is possible that the best location has moved from the kernel to the server.
3219 // In this case we adjust the position from the previous computed latency.
3220 if (location == ExtendedTimestamp::LOCATION_SERVER) {
3221 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL,
Andy Hungfb8ede22018-09-12 19:03:24 -07003222 "%s(%d): location moved from kernel to server",
Eric Laurent973db022018-11-20 14:54:31 -08003223 __func__, mPortId);
Andy Hung07eee802016-06-21 16:47:16 -07003224 // check that the last kernel OK time info exists and the positions
3225 // are valid (if they predate the current track, the positions may
3226 // be zero or negative).
Andy Hungb01faa32016-04-27 12:51:32 -07003227 const int64_t frames =
Andy Hung6d7b1192016-05-07 22:59:48 -07003228 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
Andy Hung07eee802016-06-21 16:47:16 -07003229 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 ||
3230 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 ||
3231 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0)
Andy Hung6d7b1192016-05-07 22:59:48 -07003232 ?
3233 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed
3234 / 1000)
3235 :
3236 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3237 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]);
Andy Hungfb8ede22018-09-12 19:03:24 -07003238 ALOGV("%s(%d): frame adjustment:%lld timestamp:%s",
Eric Laurent973db022018-11-20 14:54:31 -08003239 __func__, mPortId, (long long)frames, ets.toString().c_str());
Andy Hungb01faa32016-04-27 12:51:32 -07003240 if (frames >= ets.mPosition[location]) {
3241 timestamp.mPosition = 0;
3242 } else {
3243 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames);
3244 }
Andy Hung69488c42016-05-16 18:43:33 -07003245 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) {
3246 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER,
Andy Hungfb8ede22018-09-12 19:03:24 -07003247 "%s(%d): location moved from server to kernel",
Eric Laurent973db022018-11-20 14:54:31 -08003248 __func__, mPortId);
Andy Hung98731a22019-04-08 19:19:07 -07003249
3250 if (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER] ==
3251 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL]) {
3252 // In Q, we don't return errors as an invalid time
3253 // but instead we leave the last kernel good timestamp alone.
3254 //
3255 // If server is identical to kernel, the device data pipeline is idle.
3256 // A better start time is now. The retrograde check ensures
3257 // timestamp monotonicity.
3258 const int64_t nowNs = systemTime();
Andy Hungcf3b7152019-04-19 18:29:21 -07003259 if (!mTimestampStallReported) {
3260 ALOGD("%s(%d): device stall time corrected using current time %lld",
3261 __func__, mPortId, (long long)nowNs);
3262 mTimestampStallReported = true;
3263 }
Andy Hung98731a22019-04-08 19:19:07 -07003264 timestamp.mTime = convertNsToTimespec(nowNs);
Andy Hungcf3b7152019-04-19 18:29:21 -07003265 } else {
3266 mTimestampStallReported = false;
Andy Hung98731a22019-04-08 19:19:07 -07003267 }
Andy Hungb01faa32016-04-27 12:51:32 -07003268 }
Andy Hung5d313802016-10-10 15:09:39 -07003269
3270 // We update the timestamp time even when paused.
3271 if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) {
3272 const int64_t now = systemTime();
Andy Hung2b01f002017-07-05 12:01:36 -07003273 const int64_t at = audio_utils_ns_from_timespec(&timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003274 const int64_t lag =
3275 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
3276 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0)
3277 ? int64_t(mAfLatency * 1000000LL)
3278 : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3279 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK])
3280 * NANOS_PER_SECOND / mSampleRate;
3281 const int64_t limit = now - lag; // no earlier than this limit
3282 if (at < limit) {
3283 ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld",
3284 (long long)lag, (long long)at, (long long)limit);
Andy Hungffa36952017-08-17 10:41:51 -07003285 timestamp.mTime = convertNsToTimespec(limit);
Andy Hung5d313802016-10-10 15:09:39 -07003286 }
3287 }
Andy Hungb01faa32016-04-27 12:51:32 -07003288 mPreviousLocation = location;
3289 } else {
3290 // right after AudioTrack is started, one may not find a timestamp
Eric Laurent973db022018-11-20 14:54:31 -08003291 ALOGV("%s(%d): getBestTimestamp did not find timestamp", __func__, mPortId);
Andy Hungb01faa32016-04-27 12:51:32 -07003292 }
Andy Hung6ae58432016-02-16 18:32:24 -08003293 }
3294 if (status == INVALID_OPERATION) {
Andy Hungf20a4e92016-08-15 19:10:34 -07003295 // INVALID_OPERATION occurs when no timestamp has been issued by the server;
3296 // other failures are signaled by a negative time.
3297 // If we come out of FLUSHED or STOPPED where the position is known
3298 // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of
3299 // "zero" for NuPlayer). We don't convert for track restoration as position
3300 // does not reset.
Andy Hungfb8ede22018-09-12 19:03:24 -07003301 ALOGV("%s(%d): timestamp server offset:%lld restore frames:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003302 __func__, mPortId,
Andy Hungf20a4e92016-08-15 19:10:34 -07003303 (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore);
3304 if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) {
3305 status = WOULD_BLOCK;
3306 }
Andy Hung6ae58432016-02-16 18:32:24 -08003307 }
3308 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003309 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08003310 ALOGV_IF(status != WOULD_BLOCK, "%s(%d): getTimestamp error:%#x", __func__, mPortId, status);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003311 return status;
3312 }
jiabin94ed47c2023-07-27 23:34:20 +00003313 if (isAfTrackOffloadedOrDirect_l()) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003314 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) {
3315 // use cached paused position in case another offloaded track is running.
3316 timestamp.mPosition = mPausedPosition;
3317 clock_gettime(CLOCK_MONOTONIC, &timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003318 // TODO: adjust for delay
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003319 return NO_ERROR;
3320 }
3321
3322 // Check whether a pending flush or stop has completed, as those commands may
Andy Hungc8e09c62015-06-03 23:43:36 -07003323 // be asynchronous or return near finish or exhibit glitchy behavior.
3324 //
3325 // Originally this showed up as the first timestamp being a continuation of
3326 // the previous song under gapless playback.
3327 // However, we sometimes see zero timestamps, then a glitch of
3328 // the previous song's position, and then correct timestamps afterwards.
Andy Hungffa36952017-08-17 10:41:51 -07003329 if (mStartFromZeroUs != 0 && mSampleRate != 0) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003330 static const int kTimeJitterUs = 100000; // 100 ms
3331 static const int k1SecUs = 1000000;
3332
3333 const int64_t timeNow = getNowUs();
3334
Andy Hungffa36952017-08-17 10:41:51 -07003335 if (timeNow < mStartFromZeroUs + k1SecUs) { // within first second of starting
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003336 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003337 if (timestampTimeUs < mStartFromZeroUs) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003338 return WOULD_BLOCK; // stale timestamp time, occurs before start.
3339 }
Andy Hungffa36952017-08-17 10:41:51 -07003340 const int64_t deltaTimeUs = timestampTimeUs - mStartFromZeroUs;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003341 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07003342 / ((double)mSampleRate * mPlaybackRate.mSpeed);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003343
3344 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
3345 // Verify that the counter can't count faster than the sample rate
Andy Hungc8e09c62015-06-03 23:43:36 -07003346 // since the start time. If greater, then that means we may have failed
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003347 // to completely flush or stop the previous playing track.
Andy Hungc8e09c62015-06-03 23:43:36 -07003348 ALOGW_IF(!mTimestampStartupGlitchReported,
Andy Hungfb8ede22018-09-12 19:03:24 -07003349 "%s(%d): startup glitch detected"
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003350 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)",
Eric Laurent973db022018-11-20 14:54:31 -08003351 __func__, mPortId,
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003352 (long long)deltaTimeUs, (long long)deltaPositionByUs,
3353 timestamp.mPosition);
Andy Hungc8e09c62015-06-03 23:43:36 -07003354 mTimestampStartupGlitchReported = true;
3355 if (previousTimestampValid
3356 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) {
3357 timestamp = mPreviousTimestamp;
3358 mPreviousTimestampValid = true;
3359 return NO_ERROR;
3360 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003361 return WOULD_BLOCK;
3362 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003363 if (deltaPositionByUs != 0) {
Andy Hungffa36952017-08-17 10:41:51 -07003364 mStartFromZeroUs = 0; // don't check again, we got valid nonzero position.
Andy Hungc8e09c62015-06-03 23:43:36 -07003365 }
3366 } else {
Andy Hungffa36952017-08-17 10:41:51 -07003367 mStartFromZeroUs = 0; // don't check again, start time expired.
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003368 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003369 mTimestampStartupGlitchReported = false;
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003370 }
3371 } else {
Glenn Kasten200092b2014-08-15 15:13:30 -07003372 // Update the mapping between local consumed (mPosition) and server consumed (mServer)
3373 (void) updateAndGetPosition_l();
3374 // Server consumed (mServer) and presented both use the same server time base,
3375 // and server consumed is always >= presented.
3376 // The delta between these represents the number of frames in the buffer pipeline.
3377 // If this delta between these is greater than the client position, it means that
3378 // actually presented is still stuck at the starting line (figuratively speaking),
3379 // waiting for the first frame to go by. So we can't report a valid timestamp yet.
Andy Hung90e8a972015-11-09 16:42:40 -08003380 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when
3381 // mPosition exceeds 32 bits.
3382 // TODO Remove when timestamp is updated to contain pipeline status info.
3383 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue();
3384 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */
3385 && (uint32_t)pipelineDepthInFrames > mPosition.value()) {
Glenn Kasten200092b2014-08-15 15:13:30 -07003386 return INVALID_OPERATION;
3387 }
3388 // Convert timestamp position from server time base to client time base.
3389 // TODO The following code should work OK now because timestamp.mPosition is 32-bit.
3390 // But if we change it to 64-bit then this could fail.
Andy Hung90e8a972015-11-09 16:42:40 -08003391 // Use Modulo computation here.
3392 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value();
Glenn Kasten200092b2014-08-15 15:13:30 -07003393 // Immediately after a call to getPosition_l(), mPosition and
3394 // mServer both represent the same frame position. mPosition is
3395 // in client's point of view, and mServer is in server's point of
3396 // view. So the difference between them is the "fudge factor"
3397 // between client and server views due to stop() and/or new
3398 // IAudioTrack. And timestamp.mPosition is initially in server's
3399 // point of view, so we need to apply the same fudge factor to it.
Glenn Kastenfe346c72013-08-30 13:28:22 -07003400 }
Phil Burk1b420972015-04-22 10:52:21 -07003401
3402 // Prevent retrograde motion in timestamp.
3403 // This is sometimes caused by erratic reports of the available space in the ALSA drivers.
3404 if (status == NO_ERROR) {
Andy Hung3b8c6332019-04-03 19:29:36 -07003405 // Fix stale time when checking timestamp right after start().
3406 // The position is at the last reported location but the time can be stale
3407 // due to pause or standby or cold start latency.
3408 //
3409 // We keep advancing the time (but not the position) to ensure that the
3410 // stale value does not confuse the application.
3411 //
3412 // For offload compatibility, use a default lag value here.
3413 // Any time discrepancy between this update and the pause timestamp is handled
3414 // by the retrograde check afterwards.
3415 int64_t currentTimeNanos = audio_utils_ns_from_timespec(&timestamp.mTime);
3416 const int64_t lagNs = int64_t(mAfLatency * 1000000LL);
3417 const int64_t limitNs = mStartNs - lagNs;
3418 if (currentTimeNanos < limitNs) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003419 if (!mTimestampStaleTimeReported) {
3420 ALOGD("%s(%d): stale timestamp time corrected, "
3421 "currentTimeNanos: %lld < limitNs: %lld < mStartNs: %lld",
3422 __func__, mPortId,
3423 (long long)currentTimeNanos, (long long)limitNs, (long long)mStartNs);
3424 mTimestampStaleTimeReported = true;
3425 }
Andy Hung3b8c6332019-04-03 19:29:36 -07003426 timestamp.mTime = convertNsToTimespec(limitNs);
3427 currentTimeNanos = limitNs;
Andy Hungcf3b7152019-04-19 18:29:21 -07003428 } else {
3429 mTimestampStaleTimeReported = false;
Andy Hung3b8c6332019-04-03 19:29:36 -07003430 }
3431
Andy Hungffa36952017-08-17 10:41:51 -07003432 // previousTimestampValid is set to false when starting after a stop or flush.
Phil Burk1b420972015-04-22 10:52:21 -07003433 if (previousTimestampValid) {
Andy Hung2b01f002017-07-05 12:01:36 -07003434 const int64_t previousTimeNanos =
3435 audio_utils_ns_from_timespec(&mPreviousTimestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003436
3437 // retrograde check
Phil Burk1b420972015-04-22 10:52:21 -07003438 if (currentTimeNanos < previousTimeNanos) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003439 if (!mTimestampRetrogradeTimeReported) {
3440 ALOGW("%s(%d): retrograde timestamp time corrected, %lld < %lld",
3441 __func__, mPortId,
3442 (long long)currentTimeNanos, (long long)previousTimeNanos);
3443 mTimestampRetrogradeTimeReported = true;
3444 }
Andy Hung5d313802016-10-10 15:09:39 -07003445 timestamp.mTime = mPreviousTimestamp.mTime;
Andy Hungcf3b7152019-04-19 18:29:21 -07003446 } else {
3447 mTimestampRetrogradeTimeReported = false;
Phil Burk1b420972015-04-22 10:52:21 -07003448 }
3449
3450 // Looking at signed delta will work even when the timestamps
3451 // are wrapping around.
Andy Hung90e8a972015-11-09 16:42:40 -08003452 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition)
3453 - mPreviousTimestamp.mPosition).signedValue();
Phil Burk4c5a3672015-04-30 16:18:53 -07003454 if (deltaPosition < 0) {
3455 // Only report once per position instead of spamming the log.
Andy Hungcf3b7152019-04-19 18:29:21 -07003456 if (!mTimestampRetrogradePositionReported) {
Andy Hungfb8ede22018-09-12 19:03:24 -07003457 ALOGW("%s(%d): retrograde timestamp position corrected, %d = %u - %u",
Eric Laurent973db022018-11-20 14:54:31 -08003458 __func__, mPortId,
Phil Burk4c5a3672015-04-30 16:18:53 -07003459 deltaPosition,
3460 timestamp.mPosition,
3461 mPreviousTimestamp.mPosition);
Andy Hungcf3b7152019-04-19 18:29:21 -07003462 mTimestampRetrogradePositionReported = true;
Phil Burk4c5a3672015-04-30 16:18:53 -07003463 }
3464 } else {
Andy Hungcf3b7152019-04-19 18:29:21 -07003465 mTimestampRetrogradePositionReported = false;
Phil Burk4c5a3672015-04-30 16:18:53 -07003466 }
Andy Hung5d313802016-10-10 15:09:39 -07003467 if (deltaPosition < 0) {
3468 timestamp.mPosition = mPreviousTimestamp.mPosition;
3469 deltaPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07003470 }
Andy Hung5d313802016-10-10 15:09:39 -07003471#if 0
3472 // Uncomment this to verify audio timestamp rate.
3473 const int64_t deltaTime =
Andy Hung2b01f002017-07-05 12:01:36 -07003474 audio_utils_ns_from_timespec(&timestamp.mTime) - previousTimeNanos;
Andy Hung5d313802016-10-10 15:09:39 -07003475 if (deltaTime != 0) {
3476 const int64_t computedSampleRate =
3477 deltaPosition * (long long)NANOS_PER_SECOND / deltaTime;
Andy Hungfb8ede22018-09-12 19:03:24 -07003478 ALOGD("%s(%d): computedSampleRate:%u sampleRate:%u",
Eric Laurent973db022018-11-20 14:54:31 -08003479 __func__, mPortId,
Andy Hung5d313802016-10-10 15:09:39 -07003480 (unsigned)computedSampleRate, mSampleRate);
3481 }
3482#endif
Phil Burk1b420972015-04-22 10:52:21 -07003483 }
3484 mPreviousTimestamp = timestamp;
3485 mPreviousTimestampValid = true;
3486 }
3487
Glenn Kastenfe346c72013-08-30 13:28:22 -07003488 return status;
Glenn Kastence703742013-07-19 16:33:58 -07003489}
3490
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003491String8 AudioTrack::getParameters(const String8& keys)
3492{
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003493 audio_io_handle_t output = getOutput();
Glenn Kasten142f5192014-03-25 17:44:59 -07003494 if (output != AUDIO_IO_HANDLE_NONE) {
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003495 return AudioSystem::getParameters(output, keys);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01003496 } else {
Tomasz Wasilczyk53ce3af2023-08-14 16:16:55 +00003497 return String8();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01003498 }
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003499}
3500
Glenn Kasten23a75452014-01-13 10:37:17 -08003501bool AudioTrack::isOffloaded() const
3502{
3503 AutoMutex lock(mLock);
3504 return isOffloaded_l();
3505}
3506
Eric Laurentab5cdba2014-06-09 17:22:27 -07003507bool AudioTrack::isDirect() const
3508{
3509 AutoMutex lock(mLock);
3510 return isDirect_l();
3511}
3512
3513bool AudioTrack::isOffloadedOrDirect() const
3514{
3515 AutoMutex lock(mLock);
3516 return isOffloadedOrDirect_l();
3517}
3518
3519
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003520status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003521{
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003522 String8 result;
3523
3524 result.append(" AudioTrack::dump\n");
Andy Hungfb8ede22018-09-12 19:03:24 -07003525 result.appendFormat(" id(%d) status(%d), state(%d), session Id(%d), flags(%#x)\n",
Eric Laurent973db022018-11-20 14:54:31 -08003526 mPortId, mStatus, mState, mSessionId, mFlags);
Eric Laurentd114b622017-11-27 18:37:04 -08003527 result.appendFormat(" stream type(%d), left - right volume(%f, %f)\n",
François Gaffie58d4be52018-11-06 15:30:12 +01003528 mStreamType,
Eric Laurentd114b622017-11-27 18:37:04 -08003529 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
Glenn Kasten49f36ba2017-12-06 13:02:02 -08003530 result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
Eric Laurentd114b622017-11-27 18:37:04 -08003531 mFormat, mChannelMask, mChannelCount);
3532 result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
3533 mSampleRate, mOriginalSampleRate, mPlaybackRate.mSpeed);
3534 result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
3535 mFrameCount, mReqFrameCount);
3536 result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
3537 " req. notif. per buff(%u)\n",
3538 mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
3539 result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
3540 mLatency, mSelectedDeviceId, mRoutedDeviceId);
3541 result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
3542 mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
Tomasz Wasilczyk833345b2023-08-15 20:59:35 +00003543 ::write(fd, result.c_str(), result.size());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003544 return NO_ERROR;
3545}
3546
Phil Burk2812d9e2016-01-04 10:34:30 -08003547uint32_t AudioTrack::getUnderrunCount() const
3548{
3549 AutoMutex lock(mLock);
3550 return getUnderrunCount_l();
3551}
3552
3553uint32_t AudioTrack::getUnderrunCount_l() const
3554{
3555 return mProxy->getUnderrunCount() + mUnderrunCountOffset;
3556}
3557
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003558uint32_t AudioTrack::getUnderrunFrames() const
3559{
3560 AutoMutex lock(mLock);
3561 return mProxy->getUnderrunFrames();
3562}
3563
Andy Hung3a5c2f32021-02-17 15:06:42 -08003564void AudioTrack::setLogSessionId(const char *logSessionId)
3565{
3566 AutoMutex lock(mLock);
Andy Hung1a9c21b2021-02-25 20:43:18 -08003567 if (logSessionId == nullptr) logSessionId = ""; // an empty string is an unset session id.
Andy Hung3a5c2f32021-02-17 15:06:42 -08003568 if (mLogSessionId == logSessionId) return;
3569
3570 mLogSessionId = logSessionId;
3571 mediametrics::LogItem(mMetricsId)
3572 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETLOGSESSIONID)
3573 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, logSessionId)
3574 .record();
3575}
3576
Andy Hung839a3062021-02-17 11:15:16 -08003577void AudioTrack::setPlayerIId(int playerIId)
3578{
3579 AutoMutex lock(mLock);
3580 if (mPlayerIId == playerIId) return;
3581
3582 mPlayerIId = playerIId;
Vlad Popaad0fe922022-06-10 00:43:14 +02003583 triggerPortIdUpdate_l();
Andy Hung839a3062021-02-17 11:15:16 -08003584 mediametrics::LogItem(mMetricsId)
3585 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYERIID)
3586 .set(AMEDIAMETRICS_PROP_PLAYERIID, playerIId)
3587 .record();
3588}
3589
Vlad Popaad0fe922022-06-10 00:43:14 +02003590void AudioTrack::triggerPortIdUpdate_l() {
3591 if (mAudioManager == nullptr) {
3592 // use checkService() to avoid blocking if audio service is not up yet
3593 sp<IBinder> binder =
3594 defaultServiceManager()->checkService(String16(kAudioServiceName));
3595 if (binder == nullptr) {
3596 ALOGE("%s(%d): binding to audio service failed.",
3597 __func__,
3598 mPlayerIId);
3599 return;
3600 }
3601
3602 mAudioManager = interface_cast<IAudioManager>(binder);
3603 }
3604
3605 // first time when the track is created we do not have a valid piid
3606 if (mPlayerIId != PLAYER_PIID_INVALID) {
3607 mAudioManager->playerEvent(mPlayerIId, PLAYER_UPDATE_PORT_ID, mPortId);
3608 }
3609}
3610
Eric Laurent296fb132015-05-01 11:38:42 -07003611status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback)
3612{
Eric Laurent09f1ed22019-04-24 17:45:17 -07003613
Eric Laurent296fb132015-05-01 11:38:42 -07003614 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003615 ALOGW("%s(%d): adding NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003616 return BAD_VALUE;
3617 }
3618 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07003619 if (mDeviceCallback.unsafe_get() == callback.get()) {
Eric Laurent973db022018-11-20 14:54:31 -08003620 ALOGW("%s(%d): adding same callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003621 return INVALID_OPERATION;
3622 }
3623 status_t status = NO_ERROR;
3624 if (mOutput != AUDIO_IO_HANDLE_NONE) {
3625 if (mDeviceCallback != 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003626 ALOGW("%s(%d): callback already present!", __func__, mPortId);
Eric Laurent09f1ed22019-04-24 17:45:17 -07003627 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003628 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003629 status = AudioSystem::addAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003630 }
3631 mDeviceCallback = callback;
3632 return status;
3633}
3634
3635status_t AudioTrack::removeAudioDeviceCallback(
3636 const sp<AudioSystem::AudioDeviceCallback>& callback)
3637{
3638 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003639 ALOGW("%s(%d): removing NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003640 return BAD_VALUE;
3641 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003642 AutoMutex lock(mLock);
3643 if (mDeviceCallback.unsafe_get() != callback.get()) {
3644 ALOGW("%s removing different callback!", __FUNCTION__);
3645 return INVALID_OPERATION;
Eric Laurent296fb132015-05-01 11:38:42 -07003646 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003647 mDeviceCallback.clear();
Eric Laurent296fb132015-05-01 11:38:42 -07003648 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07003649 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003650 }
Eric Laurent296fb132015-05-01 11:38:42 -07003651 return NO_ERROR;
3652}
3653
Eric Laurentad2e7b92017-09-14 20:06:42 -07003654
3655void AudioTrack::onAudioDeviceUpdate(audio_io_handle_t audioIo,
3656 audio_port_handle_t deviceId)
3657{
3658 sp<AudioSystem::AudioDeviceCallback> callback;
3659 {
3660 AutoMutex lock(mLock);
3661 if (audioIo != mOutput) {
3662 return;
3663 }
3664 callback = mDeviceCallback.promote();
3665 // only update device if the track is active as route changes due to other use cases are
3666 // irrelevant for this client
3667 if (mState == STATE_ACTIVE) {
3668 mRoutedDeviceId = deviceId;
3669 }
3670 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003671
Eric Laurentad2e7b92017-09-14 20:06:42 -07003672 if (callback.get() != nullptr) {
3673 callback->onAudioDeviceUpdate(mOutput, mRoutedDeviceId);
3674 }
3675}
3676
Andy Hunge13f8a62016-03-30 14:20:42 -07003677status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location)
3678{
3679 if (msec == nullptr ||
3680 (location != ExtendedTimestamp::LOCATION_SERVER
3681 && location != ExtendedTimestamp::LOCATION_KERNEL)) {
3682 return BAD_VALUE;
3683 }
3684 AutoMutex lock(mLock);
3685 // inclusive of offloaded and direct tracks.
3686 //
3687 // It is possible, but not enabled, to allow duration computation for non-pcm
3688 // audio_has_proportional_frames() formats because currently they have
3689 // the drain rate equivalent to the pcm sample rate * framesize.
3690 if (!isPurePcmData_l()) {
3691 return INVALID_OPERATION;
3692 }
3693 ExtendedTimestamp ets;
3694 if (getTimestamp_l(&ets) == OK
3695 && ets.mTimeNs[location] > 0) {
3696 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
3697 - ets.mPosition[location];
3698 if (diff < 0) {
3699 *msec = 0;
3700 } else {
3701 // ms is the playback time by frames
3702 int64_t ms = (int64_t)((double)diff * 1000 /
3703 ((double)mSampleRate * mPlaybackRate.mSpeed));
3704 // clockdiff is the timestamp age (negative)
3705 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
3706 ets.mTimeNs[location]
3707 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
3708 - systemTime(SYSTEM_TIME_MONOTONIC);
3709
3710 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
3711 static const int NANOS_PER_MILLIS = 1000000;
3712 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
3713 }
3714 return NO_ERROR;
3715 }
3716 if (location != ExtendedTimestamp::LOCATION_SERVER) {
3717 return INVALID_OPERATION; // LOCATION_KERNEL is not available
3718 }
3719 // use server position directly (offloaded and direct arrive here)
3720 updateAndGetPosition_l();
3721 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue();
3722 *msec = (diff <= 0) ? 0
3723 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed));
3724 return NO_ERROR;
3725}
3726
Andy Hung65ffdfc2016-10-10 15:52:11 -07003727bool AudioTrack::hasStarted()
3728{
3729 AutoMutex lock(mLock);
3730 switch (mState) {
3731 case STATE_STOPPED:
3732 if (isOffloadedOrDirect_l()) {
3733 // check if we have started in the past to return true.
Andy Hungffa36952017-08-17 10:41:51 -07003734 return mStartFromZeroUs > 0;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003735 }
3736 // A normal audio track may still be draining, so
3737 // check if stream has ended. This covers fasttrack position
3738 // instability and start/stop without any data written.
3739 if (mProxy->getStreamEndDone()) {
3740 return true;
3741 }
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003742 FALLTHROUGH_INTENDED;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003743 case STATE_ACTIVE:
3744 case STATE_STOPPING:
3745 break;
3746 case STATE_PAUSED:
3747 case STATE_PAUSED_STOPPING:
3748 case STATE_FLUSHED:
3749 return false; // we're not active
3750 default:
Eric Laurent973db022018-11-20 14:54:31 -08003751 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in hasStarted(): %d", __func__, mPortId, mState);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003752 break;
3753 }
3754
3755 // wait indicates whether we need to wait for a timestamp.
3756 // This is conservatively figured - if we encounter an unexpected error
3757 // then we will not wait.
3758 bool wait = false;
jiabin94ed47c2023-07-27 23:34:20 +00003759 if (isAfTrackOffloadedOrDirect_l()) {
Andy Hung65ffdfc2016-10-10 15:52:11 -07003760 AudioTimestamp ts;
3761 status_t status = getTimestamp_l(ts);
3762 if (status == WOULD_BLOCK) {
3763 wait = true;
3764 } else if (status == OK) {
3765 wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition);
3766 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003767 ALOGV("%s(%d): hasStarted wait:%d ts:%u start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003768 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003769 (int)wait,
3770 ts.mPosition,
3771 (long long)mStartTs.mPosition);
3772 } else {
3773 int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG
3774 ExtendedTimestamp ets;
3775 status_t status = getTimestamp_l(&ets);
3776 if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets
3777 wait = true;
3778 } else if (status == OK) {
3779 for (location = ExtendedTimestamp::LOCATION_KERNEL;
3780 location >= ExtendedTimestamp::LOCATION_SERVER; --location) {
3781 if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) {
3782 continue;
3783 }
3784 wait = ets.mPosition[location] == 0
3785 || ets.mPosition[location] == mStartEts.mPosition[location];
3786 break;
3787 }
3788 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003789 ALOGV("%s(%d): hasStarted wait:%d ets:%lld start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003790 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003791 (int)wait,
3792 (long long)ets.mPosition[location],
3793 (long long)mStartEts.mPosition[location]);
3794 }
3795 return !wait;
3796}
3797
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003798// =========================================================================
3799
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003800void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003801{
3802 sp<AudioTrack> audioTrack = mAudioTrack.promote();
3803 if (audioTrack != 0) {
3804 AutoMutex lock(audioTrack->mLock);
3805 audioTrack->mProxy->binderDied();
3806 }
3807}
3808
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003809// =========================================================================
3810
Andy Hungca353672019-03-06 11:54:38 -08003811AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver)
Andy Hungeb46cf92019-03-06 10:13:38 -08003812 : Thread(true /* bCanCallJava */) // binder recursion on restoreTrack_l() may call Java.
3813 , mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
Glenn Kasten598de6c2013-10-16 17:02:13 -07003814 mIgnoreNextPausedInt(false)
Glenn Kasten3acbd052012-02-28 10:39:56 -08003815{
3816}
3817
3818AudioTrack::AudioTrackThread::~AudioTrackThread()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003819{
3820}
3821
3822bool AudioTrack::AudioTrackThread::threadLoop()
3823{
Glenn Kasten3acbd052012-02-28 10:39:56 -08003824 {
3825 AutoMutex _l(mMyLock);
3826 if (mPaused) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003827 // TODO check return value and handle or log
Glenn Kasten3acbd052012-02-28 10:39:56 -08003828 mMyCond.wait(mMyLock);
3829 // caller will check for exitPending()
3830 return true;
3831 }
Glenn Kasten598de6c2013-10-16 17:02:13 -07003832 if (mIgnoreNextPausedInt) {
3833 mIgnoreNextPausedInt = false;
3834 mPausedInt = false;
3835 }
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003836 if (mPausedInt) {
Glenn Kasten47d55172017-05-23 11:19:30 -07003837 // TODO use futex instead of condition, for event flag "or"
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003838 if (mPausedNs > 0) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003839 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003840 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
3841 } else {
Glenn Kasten75f79032017-05-23 11:22:44 -07003842 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003843 mMyCond.wait(mMyLock);
3844 }
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003845 mPausedInt = false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003846 return true;
3847 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08003848 }
Eric Laurent7985dcb2014-10-07 15:45:14 -07003849 if (exitPending()) {
3850 return false;
3851 }
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003852 nsecs_t ns = mReceiver.processAudioBuffer();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003853 switch (ns) {
3854 case 0:
3855 return true;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003856 case NS_INACTIVE:
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003857 pauseInternal();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003858 return true;
3859 case NS_NEVER:
3860 return false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003861 case NS_WHENEVER:
Andy Hung3c09c782014-12-29 18:39:32 -08003862 // Event driven: call wake() when callback notifications conditions change.
3863 ns = INT64_MAX;
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003864 FALLTHROUGH_INTENDED;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003865 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003866 LOG_ALWAYS_FATAL_IF(ns < 0, "%s(%d): processAudioBuffer() returned %lld",
Eric Laurent973db022018-11-20 14:54:31 -08003867 __func__, mReceiver.mPortId, (long long)ns);
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003868 pauseInternal(ns);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003869 return true;
Glenn Kastenca8b2802012-04-23 13:58:16 -07003870 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003871}
3872
Glenn Kasten3acbd052012-02-28 10:39:56 -08003873void AudioTrack::AudioTrackThread::requestExit()
3874{
3875 // must be in this order to avoid a race condition
3876 Thread::requestExit();
Glenn Kasten598de6c2013-10-16 17:02:13 -07003877 resume();
Glenn Kasten3acbd052012-02-28 10:39:56 -08003878}
3879
3880void AudioTrack::AudioTrackThread::pause()
3881{
3882 AutoMutex _l(mMyLock);
3883 mPaused = true;
3884}
3885
3886void AudioTrack::AudioTrackThread::resume()
3887{
3888 AutoMutex _l(mMyLock);
Glenn Kasten598de6c2013-10-16 17:02:13 -07003889 mIgnoreNextPausedInt = true;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003890 if (mPaused || mPausedInt) {
Glenn Kasten3acbd052012-02-28 10:39:56 -08003891 mPaused = false;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003892 mPausedInt = false;
Glenn Kasten3acbd052012-02-28 10:39:56 -08003893 mMyCond.signal();
3894 }
3895}
3896
Andy Hung3c09c782014-12-29 18:39:32 -08003897void AudioTrack::AudioTrackThread::wake()
3898{
3899 AutoMutex _l(mMyLock);
Andy Hunga8d08902015-07-22 11:52:13 -07003900 if (!mPaused) {
3901 // wake() might be called while servicing a callback - ignore the next
3902 // pause time and call processAudioBuffer.
Andy Hung3c09c782014-12-29 18:39:32 -08003903 mIgnoreNextPausedInt = true;
Andy Hunga8d08902015-07-22 11:52:13 -07003904 if (mPausedInt && mPausedNs > 0) {
3905 // audio track is active and internally paused with timeout.
3906 mPausedInt = false;
3907 mMyCond.signal();
3908 }
Andy Hung3c09c782014-12-29 18:39:32 -08003909 }
3910}
3911
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003912void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
3913{
3914 AutoMutex _l(mMyLock);
3915 mPausedInt = true;
3916 mPausedNs = ns;
3917}
3918
jiabinf6eb4c32020-02-25 14:06:25 -08003919binder::Status AudioTrack::AudioTrackCallback::onCodecFormatChanged(
3920 const std::vector<uint8_t>& audioMetadata)
3921{
3922 AutoMutex _l(mAudioTrackCbLock);
3923 sp<media::IAudioTrackCallback> callback = mCallback.promote();
3924 if (callback.get() != nullptr) {
3925 callback->onCodecFormatChanged(audioMetadata);
3926 } else {
3927 mCallback.clear();
3928 }
3929 return binder::Status::ok();
3930}
3931
3932void AudioTrack::AudioTrackCallback::setAudioTrackCallback(
3933 const sp<media::IAudioTrackCallback> &callback) {
3934 AutoMutex lock(mAudioTrackCbLock);
3935 mCallback = callback;
3936}
3937
Glenn Kasten40bc9062015-03-20 09:09:33 -07003938} // namespace android