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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() : AudioTrack(AttributionSourceState())
Colin Crossb8a9dbb2020-08-27 04:12:26 +0000237{
238}
239
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000240AudioTrack::AudioTrack(const AttributionSourceState& attributionSource)
Glenn Kasten87913512011-06-22 16:15:25 -0700241 : mStatus(NO_INIT),
Haynes Mathew George9b3359f2016-03-23 19:09:10 -0700242 mState(STATE_STOPPED),
John Grossman4ff14ba2012-02-08 16:37:41 -0800243 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800244 mPreviousSchedulingGroup(SP_DEFAULT),
Paul McLeanaa981192015-03-21 09:55:15 -0700245 mPausedPosition(0),
Eric Laurent20b9ef02016-12-05 11:03:16 -0800246 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
jiabinf6eb4c32020-02-25 14:06:25 -0800247 mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000248 mClientAttributionSource(attributionSource),
jiabinf6eb4c32020-02-25 14:06:25 -0800249 mAudioTrackCallback(new AudioTrackCallback())
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800250{
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700251 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
252 mAttributes.usage = AUDIO_USAGE_UNKNOWN;
Mikhail Naganov55773032020-10-01 15:08:13 -0700253 mAttributes.flags = AUDIO_FLAG_NONE;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700254 strcpy(mAttributes.tags, "");
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800255}
256
257AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800258 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800259 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800260 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700261 audio_channel_mask_t channelMask,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800262 size_t frameCount,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700263 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400264 const wp<IAudioTrackCallback> & callback,
Atneyaf86d2692021-10-14 14:02:36 -0400265 int32_t notificationFrames,
266 audio_session_t sessionId,
267 transfer_type transferType,
268 const audio_offload_info_t *offloadInfo,
269 const AttributionSourceState& attributionSource,
270 const audio_attributes_t* pAttributes,
271 bool doNotReconnect,
272 float maxRequiredSpeed,
273 audio_port_handle_t selectedDeviceId)
274 : mStatus(NO_INIT),
275 mState(STATE_STOPPED),
276 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
277 mPreviousSchedulingGroup(SP_DEFAULT),
278 mPausedPosition(0),
279 mAudioTrackCallback(new AudioTrackCallback())
280{
281 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Daniel Chapinf13b98a2021-10-25 21:58:31 +0000282
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500283 // make_unique does not aggregate init until c++20
284 mSetParams = std::unique_ptr<SetParams>{
285 new SetParams{streamType, sampleRate, format, channelMask, frameCount, flags, callback,
286 notificationFrames, 0 /*sharedBuffer*/, false /*threadCanCallJava*/,
287 sessionId, transferType, offloadInfo, attributionSource, pAttributes,
288 doNotReconnect, maxRequiredSpeed, selectedDeviceId}};
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400289}
290
291namespace {
292 class LegacyCallbackWrapper : public AudioTrack::IAudioTrackCallback {
293 const AudioTrack::legacy_callback_t mCallback;
294 void * const mData;
295 public:
296 LegacyCallbackWrapper(AudioTrack::legacy_callback_t callback, void* user)
297 : mCallback(callback), mData(user) {}
298 size_t onMoreData(const AudioTrack::Buffer & buffer) override {
299 AudioTrack::Buffer copy = buffer;
300 mCallback(AudioTrack::EVENT_MORE_DATA, mData, static_cast<void*>(&copy));
Atneya Nair03079272022-01-18 17:03:14 -0500301 return copy.size();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400302 }
303 void onUnderrun() override {
304 mCallback(AudioTrack::EVENT_UNDERRUN, mData, nullptr);
305 }
306 void onLoopEnd(int32_t loopsRemaining) override {
307 mCallback(AudioTrack::EVENT_LOOP_END, mData, &loopsRemaining);
308 }
309 void onMarker(uint32_t markerPosition) override {
310 mCallback(AudioTrack::EVENT_MARKER, mData, &markerPosition);
311 }
312 void onNewPos(uint32_t newPos) override {
313 mCallback(AudioTrack::EVENT_NEW_POS, mData, &newPos);
314 }
315 void onBufferEnd() override {
316 mCallback(AudioTrack::EVENT_BUFFER_END, mData, nullptr);
317 }
318 void onNewIAudioTrack() override {
319 mCallback(AudioTrack::EVENT_NEW_IAUDIOTRACK, mData, nullptr);
320 }
321 void onStreamEnd() override {
322 mCallback(AudioTrack::EVENT_STREAM_END, mData, nullptr);
323 }
324 size_t onCanWriteMoreData(const AudioTrack::Buffer & buffer) override {
325 AudioTrack::Buffer copy = buffer;
326 mCallback(AudioTrack::EVENT_CAN_WRITE_MORE_DATA, mData, static_cast<void*>(&copy));
Atneya Nair03079272022-01-18 17:03:14 -0500327 return copy.size();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400328 }
329 };
330}
Andreas Huberc8139852012-01-18 10:51:55 -0800331AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800332 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800333 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800334 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700335 audio_channel_mask_t channelMask,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800336 const sp<IMemory>& sharedBuffer,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700337 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400338 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700339 int32_t notificationFrames,
Glenn Kastend848eb42016-03-08 13:42:11 -0800340 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000341 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800342 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000343 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700344 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700345 bool doNotReconnect,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700346 float maxRequiredSpeed)
Glenn Kasten87913512011-06-22 16:15:25 -0700347 : mStatus(NO_INIT),
Haynes Mathew George9b3359f2016-03-23 19:09:10 -0700348 mState(STATE_STOPPED),
John Grossman4ff14ba2012-02-08 16:37:41 -0800349 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800350 mPreviousSchedulingGroup(SP_DEFAULT),
Paul McLeanaa981192015-03-21 09:55:15 -0700351 mPausedPosition(0),
jiabinf6eb4c32020-02-25 14:06:25 -0800352 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
353 mAudioTrackCallback(new AudioTrackCallback())
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800354{
François Gaffie393f0e02019-04-10 09:09:08 +0200355 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Ippei Murata5fa32ed2018-10-01 17:37:50 +0900356
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500357 mSetParams = std::unique_ptr<SetParams>{
358 new SetParams{streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags,
359 callback, notificationFrames, sharedBuffer, false /*threadCanCallJava*/,
360 sessionId, transferType, offloadInfo, attributionSource, pAttributes,
361 doNotReconnect, maxRequiredSpeed, AUDIO_PORT_HANDLE_NONE}};
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800362}
363
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500364void AudioTrack::onFirstRef() {
365 if (mSetParams) {
366 set(*mSetParams);
367 mSetParams.reset();
368 }
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400369}
370
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800371AudioTrack::~AudioTrack()
372{
Ray Essicked304702017-12-12 14:00:57 -0800373 // pull together the numbers, before we clean up our structures
374 mMediaMetrics.gather(this);
375
Andy Hungb68f5eb2019-12-03 16:49:17 -0800376 mediametrics::LogItem(mMetricsId)
377 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DTOR)
Phil Burkd3813f32020-04-23 16:26:15 -0700378 .set(AMEDIAMETRICS_PROP_CALLERNAME,
379 mCallerName.empty()
Andy Hunga6b27032020-04-27 10:34:24 -0700380 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
Phil Burkd3813f32020-04-23 16:26:15 -0700381 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800382 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
383 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)mStatus)
384 .record();
385
Phil Burk7a9577c2021-03-12 20:12:11 +0000386 stopAndJoinCallbacks(); // checks mStatus
387
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800388 if (mStatus == NO_ERROR) {
Marco Nelissenf8880202014-11-14 07:58:25 -0800389 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten53cec222013-08-29 09:01:02 -0700390 mAudioTrack.clear();
Eric Laurent3bcffa12014-06-12 18:38:45 -0700391 mCblkMemory.clear();
392 mSharedBuffer.clear();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800393 IPCThreadState::self()->flushCommands();
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000394 pid_t clientPid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
Andy Hungfb8ede22018-09-12 19:03:24 -0700395 ALOGV("%s(%d), releasing session id %d from %d on behalf of %d",
Eric Laurent973db022018-11-20 14:54:31 -0800396 __func__, mPortId,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700397 mSessionId, IPCThreadState::self()->getCallingPid(), clientPid);
398 AudioSystem::releaseAudioSessionId(mSessionId, clientPid);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800399 }
400}
401
Phil Burk7a9577c2021-03-12 20:12:11 +0000402void AudioTrack::stopAndJoinCallbacks() {
403 // Prevent nullptr crash if it did not open properly.
404 if (mStatus != NO_ERROR) return;
405
406 // Make sure that callback function exits in the case where
407 // it is looping on buffer full condition in obtainBuffer().
408 // Otherwise the callback thread will never exit.
409 stop();
410 if (mAudioTrackThread != 0) { // not thread safe
Phil Burk7a9577c2021-03-12 20:12:11 +0000411 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
Kuowei Lifdf8e0f2021-11-26 17:38:34 +0800412 mProxy->interrupt();
Phil Burk7a9577c2021-03-12 20:12:11 +0000413 mAudioTrackThread->requestExitAndWait();
414 mAudioTrackThread.clear();
415 }
zhenjun.zhang116df6a2021-12-08 09:17:13 +0800416
417 AutoMutex lock(mLock);
Phil Burk7a9577c2021-03-12 20:12:11 +0000418 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) {
419 // This may not stop all of these device callbacks!
420 // TODO: Add some sort of protection.
421 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
422 mDeviceCallback.clear();
423 }
424}
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400425status_t AudioTrack::set(
426 audio_stream_type_t streamType,
427 uint32_t sampleRate,
428 audio_format_t format,
429 audio_channel_mask_t channelMask,
430 size_t frameCount,
431 audio_output_flags_t flags,
432 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700433 int32_t notificationFrames,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800434 const sp<IMemory>& sharedBuffer,
Eric Laurentbe916aa2010-06-01 23:49:17 -0700435 bool threadCanCallJava,
Glenn Kastend848eb42016-03-08 13:42:11 -0800436 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000437 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800438 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000439 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700440 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700441 bool doNotReconnect,
jiabin156c6872017-10-06 09:47:15 -0700442 float maxRequiredSpeed,
443 audio_port_handle_t selectedDeviceId)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800444{
Atneya Nair14aabae2021-11-30 17:36:24 -0500445 LOG_ALWAYS_FATAL_IF(mInitialized, "%s: should not be called twice", __func__);
446 mInitialized = true;
Eric Laurentf32d7812017-11-30 14:44:07 -0800447 status_t status;
448 uint32_t channelCount;
449 pid_t callingPid;
450 pid_t myPid;
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000451 uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
452 pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
Andy Hung3acde2c2021-11-11 09:18:08 -0800453 std::string errorMessage;
Eric Laurent973db022018-11-20 14:54:31 -0800454 // Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
Andy Hungfb8ede22018-09-12 19:03:24 -0700455 ALOGV("%s(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
wanggang1471f644f2022-07-08 11:10:20 +0800456 "flags %#x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
Andy Hungfb8ede22018-09-12 19:03:24 -0700457 __func__,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800458 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000459 sessionId, transferType, attributionSource.uid, attributionSource.pid);
Glenn Kasten86f04662014-02-24 15:13:05 -0800460
Phil Burk33ff89b2015-11-30 11:16:01 -0800461 mThreadCanCallJava = threadCanCallJava;
Andy Hungde602302021-12-07 21:35:49 -0800462
463 // These variables are pulled in an error report, so we initialize them early.
jiabin156c6872017-10-06 09:47:15 -0700464 mSelectedDeviceId = selectedDeviceId;
Eric Laurent21da6472017-11-09 16:29:26 -0800465 mSessionId = sessionId;
Andy Hungde602302021-12-07 21:35:49 -0800466 mChannelMask = channelMask;
Andy Hungde602302021-12-07 21:35:49 -0800467 mReqFrameCount = mFrameCount = frameCount;
468 mSampleRate = sampleRate;
469 mOriginalSampleRate = sampleRate;
470 mAttributes = pAttributes != nullptr ? *pAttributes : AUDIO_ATTRIBUTES_INITIALIZER;
471 mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
Phil Burk33ff89b2015-11-30 11:16:01 -0800472
Eric Laurentd7f33c52022-01-06 13:54:56 +0100473 // update format and flags before storing them in mFormat, mOrigFlags and mFlags
474 if (pAttributes != NULL) {
475 // stream type shouldn't be looked at, this track has audio attributes
476 ALOGV("%s(): Building AudioTrack with attributes:"
477 " usage=%d content=%d flags=0x%x tags=[%s]",
478 __func__,
479 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
480 audio_flags_to_audio_output_flags(mAttributes.flags, &flags);
481 }
482
483 // these below should probably come from the audioFlinger too...
484 if (format == AUDIO_FORMAT_DEFAULT) {
485 format = AUDIO_FORMAT_PCM_16_BIT;
486 } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
487 flags = static_cast<audio_output_flags_t>(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
488 }
489
490 // force direct flag if format is not linear PCM
491 // or offload was requested
492 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
493 || !audio_is_linear_pcm(format)) {
494 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
495 ? "%s(): Offload request, forcing to Direct Output"
496 : "%s(): Not linear PCM, forcing to Direct Output",
497 __func__);
498 flags = (audio_output_flags_t)
499 // FIXME why can't we allow direct AND fast?
500 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
501 }
502
503 // force direct flag if HW A/V sync requested
504 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
505 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
506 }
507
508 mFormat = format;
509 mOrigFlags = mFlags = flags;
510
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800511 switch (transferType) {
512 case TRANSFER_DEFAULT:
513 if (sharedBuffer != 0) {
514 transferType = TRANSFER_SHARED;
Atneya Nairba809b82022-03-04 18:11:10 -0500515 } else if (callback == nullptr|| threadCanCallJava) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800516 transferType = TRANSFER_SYNC;
517 } else {
518 transferType = TRANSFER_CALLBACK;
519 }
520 break;
521 case TRANSFER_CALLBACK:
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700522 case TRANSFER_SYNC_NOTIF_CALLBACK:
Atneya Nairba809b82022-03-04 18:11:10 -0500523 if (callback == nullptr || sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800524 errorMessage = StringPrintf(
525 "%s: Transfer type %s but callback == nullptr || sharedBuffer != 0",
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700526 convertTransferToText(transferType), __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800527 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800528 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800529 }
530 break;
531 case TRANSFER_OBTAIN:
532 case TRANSFER_SYNC:
533 if (sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800534 errorMessage = StringPrintf(
535 "%s: Transfer type TRANSFER_OBTAIN but sharedBuffer != 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800536 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800537 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800538 }
539 break;
540 case TRANSFER_SHARED:
541 if (sharedBuffer == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800542 errorMessage = StringPrintf(
543 "%s: Transfer type TRANSFER_SHARED but sharedBuffer == 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800544 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800545 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800546 }
547 break;
548 default:
Andy Hung3acde2c2021-11-11 09:18:08 -0800549 errorMessage = StringPrintf("%s: Invalid transfer type %d", __func__, transferType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800550 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800551 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800552 }
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800553 mSharedBuffer = sharedBuffer;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800554 mTransfer = transferType;
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700555 mDoNotReconnect = doNotReconnect;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800556
Andy Hungfb8ede22018-09-12 19:03:24 -0700557 ALOGV_IF(sharedBuffer != 0, "%s(): sharedBuffer: %p, size: %zu",
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -0700558 __func__, sharedBuffer->unsecurePointer(), sharedBuffer->size());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800559
Glenn Kasten53cec222013-08-29 09:01:02 -0700560 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Eric Laurent1dd70b92009-04-21 07:56:33 -0700561 if (mAudioTrack != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800562 errorMessage = StringPrintf("%s: Track already in use", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800563 status = INVALID_OPERATION;
Andy Hung3acde2c2021-11-11 09:18:08 -0800564 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800565 }
566
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800567 // handle default values first.
Eric Laurente83b55d2014-11-14 10:06:21 -0800568 if (streamType == AUDIO_STREAM_DEFAULT) {
Dima Zavinfce7a472011-04-19 22:30:36 -0700569 streamType = AUDIO_STREAM_MUSIC;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800570 }
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700571 if (pAttributes == NULL) {
Eric Laurent223fd5c2014-11-11 13:43:36 -0800572 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800573 errorMessage = StringPrintf("%s: Invalid stream type %d", __func__, streamType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800574 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800575 goto error;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700576 }
Andy Hunga2159aa2021-07-20 13:01:52 -0700577 mOriginalStreamType = streamType;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700578 } else {
Andy Hunga2159aa2021-07-20 13:01:52 -0700579 mOriginalStreamType = AUDIO_STREAM_DEFAULT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800580 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800581
582 // validate parameters
Dima Zavinfce7a472011-04-19 22:30:36 -0700583 if (!audio_is_valid_format(format)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800584 errorMessage = StringPrintf("%s: Invalid format %#x", __func__, format);
Eric Laurentf32d7812017-11-30 14:44:07 -0800585 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800586 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800587 }
Eric Laurentc2f1f072009-07-17 12:17:14 -0700588
Glenn Kasten8ba90322013-10-30 11:29:27 -0700589 if (!audio_is_output_channel(channelMask)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800590 errorMessage = StringPrintf("%s: Invalid channel mask %#x", __func__, channelMask);
Eric Laurentf32d7812017-11-30 14:44:07 -0800591 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800592 goto error;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700593 }
Eric Laurentf32d7812017-11-30 14:44:07 -0800594 channelCount = audio_channel_count_from_out_mask(channelMask);
Glenn Kastene3247bf2014-02-24 15:19:07 -0800595 mChannelCount = channelCount;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700596
Eric Laurentd7f33c52022-01-06 13:54:56 +0100597 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
Phil Burkfdb3c072016-02-09 10:47:02 -0800598 if (audio_has_proportional_frames(format)) {
Glenn Kastenb7730382014-04-30 15:50:31 -0700599 mFrameSize = channelCount * audio_bytes_per_sample(format);
600 } else {
601 mFrameSize = sizeof(uint8_t);
602 }
Glenn Kastene3aa6592012-12-04 12:22:46 -0800603 } else {
Phil Burkfdb3c072016-02-09 10:47:02 -0800604 ALOG_ASSERT(audio_has_proportional_frames(format));
Glenn Kastenb7730382014-04-30 15:50:31 -0700605 mFrameSize = channelCount * audio_bytes_per_sample(format);
Glenn Kastenb7730382014-04-30 15:50:31 -0700606 // createTrack will return an error if PCM format is not supported by server,
607 // so no need to check for specific PCM formats here
Glenn Kastene3aa6592012-12-04 12:22:46 -0800608 }
609
Eric Laurent0d6db582014-11-12 18:39:44 -0800610 // sampling rate must be specified for direct outputs
Eric Laurentd7f33c52022-01-06 13:54:56 +0100611 if (sampleRate == 0 && (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800612 errorMessage = StringPrintf(
613 "%s: sample rate must be specified for direct outputs", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800614 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800615 goto error;
Eric Laurent0d6db582014-11-12 18:39:44 -0800616 }
Andy Hungff874dc2016-04-11 16:49:09 -0700617 // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
618 mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
Eric Laurent0d6db582014-11-12 18:39:44 -0800619
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800620 // Make copy of input parameter offloadInfo so that in the future:
621 // (a) createTrack_l doesn't need it as an input parameter
622 // (b) we can support re-creation of offloaded tracks
623 if (offloadInfo != NULL) {
624 mOffloadInfoCopy = *offloadInfo;
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800625 } else {
Eric Laurent20b9ef02016-12-05 11:03:16 -0800626 memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
Ytai Ben-Tsviffa2fd92020-10-20 09:13:53 -0700627 mOffloadInfoCopy = AUDIO_INFO_INITIALIZER;
ziyang zhangb3ec8aa2022-05-10 13:28:38 +0800628 mOffloadInfoCopy.format = format;
629 mOffloadInfoCopy.sample_rate = sampleRate;
630 mOffloadInfoCopy.channel_mask = channelMask;
631 mOffloadInfoCopy.stream_type = streamType;
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800632 }
633
Glenn Kasten66e46352014-01-16 17:44:23 -0800634 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
635 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
Glenn Kasten05632a52012-01-03 14:22:33 -0800636 mSendLevel = 0.0f;
Glenn Kasten396fabd2014-01-08 08:54:23 -0800637 // mFrameCount is initialized in createTrack_l
Glenn Kastenea38ee72016-04-18 11:08:01 -0700638 if (notificationFrames >= 0) {
639 mNotificationFramesReq = notificationFrames;
640 mNotificationsPerBufferReq = 0;
641 } else {
Eric Laurentd7f33c52022-01-06 13:54:56 +0100642 if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800643 errorMessage = StringPrintf(
644 "%s: notificationFrames=%d not permitted for non-fast track",
Andy Hungfb8ede22018-09-12 19:03:24 -0700645 __func__, notificationFrames);
Eric Laurentf32d7812017-11-30 14:44:07 -0800646 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800647 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700648 }
649 if (frameCount > 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700650 ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
651 __func__, notificationFrames, frameCount);
Eric Laurentf32d7812017-11-30 14:44:07 -0800652 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800653 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700654 }
655 mNotificationFramesReq = 0;
656 const uint32_t minNotificationsPerBuffer = 1;
657 const uint32_t maxNotificationsPerBuffer = 8;
658 mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
659 max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
660 ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
Andy Hungfb8ede22018-09-12 19:03:24 -0700661 "%s(): notificationFrames=%d clamped to the range -%u to -%u",
662 __func__,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700663 notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
664 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800665 mNotificationFramesAct = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700666 // TODO b/182392553: refactor or remove
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000667 mClientAttributionSource = AttributionSourceState(attributionSource);
Eric Laurentf32d7812017-11-30 14:44:07 -0800668 callingPid = IPCThreadState::self()->getCallingPid();
669 myPid = getpid();
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700670 if (uid == -1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000671 mClientAttributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700672 IPCThreadState::self()->getCallingUid()));
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800673 }
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700674 if (pid == (pid_t)-1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000675 mClientAttributionSource.pid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(callingPid));
Marco Nelissend457c972014-02-11 08:47:07 -0800676 }
Eric Laurent2beeb502010-07-16 07:43:46 -0700677 mAuxEffectId = 0;
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400678 mCallback = callback;
Eric Laurentbe916aa2010-06-01 23:49:17 -0700679
Atneya Nairba809b82022-03-04 18:11:10 -0500680 if (callback != nullptr) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400681 mAudioTrackThread = sp<AudioTrackThread>::make(*this);
Glenn Kastena997e7a2012-08-07 09:44:19 -0700682 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
Glenn Kastenbfd31842015-03-20 09:01:44 -0700683 // thread begins in paused state, and will not reference us until start()
Glenn Kastena997e7a2012-08-07 09:44:19 -0700684 }
685
Eric Laurent34f1d8e2009-11-04 08:27:26 -0800686 // create the IAudioTrack
Francois Gaffie24a9fb02019-01-18 17:51:34 +0100687 {
688 AutoMutex lock(mLock);
689 status = createTrack_l();
690 }
Glenn Kastena997e7a2012-08-07 09:44:19 -0700691 if (status != NO_ERROR) {
692 if (mAudioTrackThread != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100693 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
694 mAudioTrackThread->requestExitAndWait();
Glenn Kastena997e7a2012-08-07 09:44:19 -0700695 mAudioTrackThread.clear();
696 }
Andy Hung3acde2c2021-11-11 09:18:08 -0800697 // We do not goto error to prevent double-logging errors.
Eric Laurentf32d7812017-11-30 14:44:07 -0800698 goto exit;
Glenn Kasten5d464eb2012-06-22 17:19:53 -0700699 }
700
Andy Hung4ede21d2014-12-12 15:37:34 -0800701 mLoopCount = 0;
702 mLoopStart = 0;
703 mLoopEnd = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -0800704 mLoopCountNotified = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800705 mMarkerPosition = 0;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -0700706 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800707 mNewPosition = 0;
708 mUpdatePeriod = 0;
Glenn Kasten200092b2014-08-15 15:13:30 -0700709 mPosition = 0;
710 mReleased = 0;
Andy Hungffa36952017-08-17 10:41:51 -0700711 mStartNs = 0;
712 mStartFromZeroUs = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700713 AudioSystem::acquireAudioSessionId(mSessionId, pid, uid);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800714 mSequence = 1;
715 mObservedSequence = mSequence;
716 mInUnderrun = false;
Phil Burk1b420972015-04-22 10:52:21 -0700717 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700718 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700719 mTimestampRetrogradePositionReported = false;
720 mTimestampRetrogradeTimeReported = false;
721 mTimestampStallReported = false;
722 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700723 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Andy Hung65ffdfc2016-10-10 15:52:11 -0700724 mStartTs.mPosition = 0;
Phil Burk2812d9e2016-01-04 10:34:30 -0800725 mUnderrunCountOffset = 0;
Andy Hungea2b9c02016-02-12 17:06:53 -0800726 mFramesWritten = 0;
727 mFramesWrittenServerOffset = 0;
Andy Hungf20a4e92016-08-15 19:10:34 -0700728 mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
Ivan Lozano8cf3a072017-08-09 09:01:33 -0700729 mVolumeHandler = new media::VolumeHandler();
Eric Laurentf32d7812017-11-30 14:44:07 -0800730
Andy Hung3acde2c2021-11-11 09:18:08 -0800731error:
732 if (status != NO_ERROR) {
733 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
734 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
735 }
736 // fall through
Eric Laurentf32d7812017-11-30 14:44:07 -0800737exit:
738 mStatus = status;
739 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800740}
741
Mikhail Naganov55773032020-10-01 15:08:13 -0700742
743status_t AudioTrack::set(
744 audio_stream_type_t streamType,
745 uint32_t sampleRate,
746 audio_format_t format,
747 uint32_t channelMask,
748 size_t frameCount,
749 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400750 legacy_callback_t callback,
Mikhail Naganov55773032020-10-01 15:08:13 -0700751 void* user,
752 int32_t notificationFrames,
753 const sp<IMemory>& sharedBuffer,
754 bool threadCanCallJava,
755 audio_session_t sessionId,
756 transfer_type transferType,
757 const audio_offload_info_t *offloadInfo,
758 uid_t uid,
759 pid_t pid,
760 const audio_attributes_t* pAttributes,
761 bool doNotReconnect,
762 float maxRequiredSpeed,
763 audio_port_handle_t selectedDeviceId)
764{
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000765 AttributionSourceState attributionSource;
766 attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(uid));
767 attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(pid));
768 attributionSource.token = sp<BBinder>::make();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400769 if (callback) {
770 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
771 } else if (user) {
772 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
773 }
774 return set(streamType, sampleRate, format, static_cast<audio_channel_mask_t>(channelMask),
775 frameCount, flags, mLegacyCallbackWrapper, notificationFrames, sharedBuffer,
776 threadCanCallJava, sessionId, transferType, offloadInfo, attributionSource,
777 pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
Mikhail Naganov55773032020-10-01 15:08:13 -0700778}
779
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800780// -------------------------------------------------------------------------
781
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100782status_t AudioTrack::start()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800783{
Eric Laurentf5aafb22010-11-18 08:40:16 -0800784 AutoMutex lock(mLock);
Andy Hungb68f5eb2019-12-03 16:49:17 -0800785
Andy Hung10fb4be2020-05-27 22:22:22 -0700786 if (mState == STATE_ACTIVE) {
787 return INVALID_OPERATION;
788 }
789
790 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
791
792 // Defer logging here due to OpenSL ES repeated start calls.
793 // TODO(b/154868033) after fix, restore this logging back to the beginning of start().
794 const int64_t beginNs = systemTime();
Andy Hungb68f5eb2019-12-03 16:49:17 -0800795 status_t status = NO_ERROR; // logged: make sure to set this before returning.
Andy Hung06a730b2020-04-09 13:28:31 -0700796 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800797 mediametrics::LogItem(mMetricsId)
Andy Hunga6b27032020-04-27 10:34:24 -0700798 .set(AMEDIAMETRICS_PROP_CALLERNAME,
799 mCallerName.empty()
800 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
801 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800802 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_START)
Andy Hungea840382020-05-05 21:50:17 -0700803 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800804 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
805 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
806 .record(); });
807
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800808
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800809 mInUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800810
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800811 State previousState = mState;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100812 if (previousState == STATE_PAUSED_STOPPING) {
813 mState = STATE_STOPPING;
814 } else {
815 mState = STATE_ACTIVE;
816 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700817 (void) updateAndGetPosition_l();
Andy Hung65ffdfc2016-10-10 15:52:11 -0700818
819 // save start timestamp
820 if (isOffloadedOrDirect_l()) {
821 if (getTimestamp_l(mStartTs) != OK) {
822 mStartTs.mPosition = 0;
823 }
824 } else {
825 if (getTimestamp_l(&mStartEts) != OK) {
826 mStartEts.clear();
827 }
828 }
Andy Hungffa36952017-08-17 10:41:51 -0700829 mStartNs = systemTime(); // save this for timestamp adjustment after starting.
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800830 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
831 // reset current position as seen by client to 0
Glenn Kasten200092b2014-08-15 15:13:30 -0700832 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -0700833 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700834 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700835 mTimestampRetrogradePositionReported = false;
836 mTimestampRetrogradeTimeReported = false;
837 mTimestampStallReported = false;
838 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700839 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Phil Burk1b420972015-04-22 10:52:21 -0700840
Andy Hung65ffdfc2016-10-10 15:52:11 -0700841 if (!isOffloadedOrDirect_l()
842 && mStartEts.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) {
Andy Hunge1e98462016-04-12 10:18:51 -0700843 // Server side has consumed something, but is it finished consuming?
844 // It is possible since flush and stop are asynchronous that the server
845 // is still active at this point.
Andy Hungfb8ede22018-09-12 19:03:24 -0700846 ALOGV("%s(%d): server read:%lld cumulative flushed:%lld client written:%lld",
Eric Laurent973db022018-11-20 14:54:31 -0800847 __func__, mPortId,
Andy Hunge1e98462016-04-12 10:18:51 -0700848 (long long)(mFramesWrittenServerOffset
Andy Hung65ffdfc2016-10-10 15:52:11 -0700849 + mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]),
850 (long long)mStartEts.mFlushed,
Andy Hunge1e98462016-04-12 10:18:51 -0700851 (long long)mFramesWritten);
Andy Hungc4e60eb2017-09-06 11:14:57 -0700852 // mStartEts is already adjusted by mFramesWrittenServerOffset, so we delta adjust.
853 mFramesWrittenServerOffset -= mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER];
Andy Hung61be8412015-10-06 10:51:09 -0700854 }
Andy Hunge1e98462016-04-12 10:18:51 -0700855 mFramesWritten = 0;
856 mProxy->clearTimestamp(); // need new server push for valid timestamp
857 mMarkerReached = false;
Andy Hung61be8412015-10-06 10:51:09 -0700858
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700859 // For offloaded tracks, we don't know if the hardware counters are really zero here,
860 // since the flush is asynchronous and stop may not fully drain.
861 // We save the time when the track is started to later verify whether
862 // the counters are realistic (i.e. start from zero after this time).
Andy Hungffa36952017-08-17 10:41:51 -0700863 mStartFromZeroUs = mStartNs / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700864
Eric Laurentec9a0322013-08-28 10:23:01 -0700865 // force refresh of remaining frames by processAudioBuffer() as last
866 // write before stop could be partial.
867 mRefreshRemaining = true;
Tomoharu Kasahara6e8bf982017-06-09 16:17:33 +0900868
869 // for static track, clear the old flags when starting from stopped state
870 if (mSharedBuffer != 0) {
871 android_atomic_and(
872 ~(CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
873 &mCblk->mFlags);
874 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800875 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700876 mNewPosition = mPosition + mUpdatePeriod;
Andy Hung4be3b832016-10-13 17:51:43 -0700877 int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800878
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800879 if (!(flags & CBLK_INVALID)) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -0800880 mAudioTrack->start(&status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800881 if (status == DEAD_OBJECT) {
882 flags |= CBLK_INVALID;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800883 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800884 }
885 if (flags & CBLK_INVALID) {
886 status = restoreTrack_l("start");
887 }
888
Andy Hung79629f02016-03-24 13:57:40 -0700889 // resume or pause the callback thread as needed.
890 sp<AudioTrackThread> t = mAudioTrackThread;
891 if (status == NO_ERROR) {
892 if (t != 0) {
893 if (previousState == STATE_STOPPING) {
894 mProxy->interrupt();
895 } else {
896 t->resume();
897 }
898 } else {
899 mPreviousPriority = getpriority(PRIO_PROCESS, 0);
900 get_sched_policy(0, &mPreviousSchedulingGroup);
901 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
902 }
Andy Hung39399b62017-04-21 15:07:45 -0700903
904 // Start our local VolumeHandler for restoration purposes.
905 mVolumeHandler->setStarted();
Andy Hung79629f02016-03-24 13:57:40 -0700906 } else {
Eric Laurent973db022018-11-20 14:54:31 -0800907 ALOGE("%s(%d): status %d", __func__, mPortId, status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800908 mState = previousState;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800909 if (t != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100910 if (previousState != STATE_STOPPING) {
911 t->pause();
912 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800913 } else {
Glenn Kasten87913512011-06-22 16:15:25 -0700914 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
Glenn Kastena6364332012-04-19 09:35:04 -0700915 set_sched_policy(0, mPreviousSchedulingGroup);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800916 }
917 }
918
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100919 return status;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800920}
921
922void AudioTrack::stop()
923{
Andy Hungb68f5eb2019-12-03 16:49:17 -0800924 const int64_t beginNs = systemTime();
925
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800926 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -0700927 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800928 mediametrics::LogItem(mMetricsId)
929 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_STOP)
Andy Hungea840382020-05-05 21:50:17 -0700930 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800931 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Phil Burk64e16a72020-06-01 13:25:51 -0700932 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
933 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getUnderrunCount_l())
Phil Burka9876702020-04-20 18:16:15 -0700934 .record();
Phil Burka9876702020-04-20 18:16:15 -0700935 });
Andy Hungb68f5eb2019-12-03 16:49:17 -0800936
Eric Laurent973db022018-11-20 14:54:31 -0800937 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -0700938
Glenn Kasten397edb32013-08-30 15:10:13 -0700939 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800940 return;
941 }
942
Glenn Kasten23a75452014-01-13 10:37:17 -0800943 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100944 mState = STATE_STOPPING;
945 } else {
946 mState = STATE_STOPPED;
Andy Hung2148bf02016-11-28 19:01:02 -0800947 ALOGD_IF(mSharedBuffer == nullptr,
Eric Laurent973db022018-11-20 14:54:31 -0800948 "%s(%d): called with %u frames delivered", __func__, mPortId, mReleased.value());
Andy Hungc2813e52014-10-16 17:54:34 -0700949 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100950 }
951
Andy Hung1d3556d2018-03-29 16:30:14 -0700952 mProxy->stop(); // notify server not to read beyond current client position until start().
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800953 mProxy->interrupt();
954 mAudioTrack->stop();
Andy Hung61be8412015-10-06 10:51:09 -0700955
956 // Note: legacy handling - stop does not clear playback marker
957 // and periodic update counter, but flush does for streaming tracks.
Andy Hung9b461582014-12-01 17:56:29 -0800958
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800959 if (mSharedBuffer != 0) {
Andy Hung9b461582014-12-01 17:56:29 -0800960 // clear buffer position and loop count.
Andy Hung9b461582014-12-01 17:56:29 -0800961 mStaticProxy->setBufferPositionAndLoop(0 /* position */,
962 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800963 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100964
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800965 sp<AudioTrackThread> t = mAudioTrackThread;
966 if (t != 0) {
Glenn Kasten23a75452014-01-13 10:37:17 -0800967 if (!isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100968 t->pause();
Jean-Michel Trivia60e9a22018-11-15 16:13:36 -0800969 } else if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
Jean-Michel Trivi0248a2a2018-11-21 10:05:57 -0800970 // causes wake up of the playback thread, that will callback the client for
971 // EVENT_STREAM_END in processAudioBuffer()
972 t->wake();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100973 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800974 } else {
975 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
976 set_sched_policy(0, mPreviousSchedulingGroup);
977 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800978}
979
980bool AudioTrack::stopped() const
981{
Glenn Kasten9a2aaf92012-01-03 09:42:47 -0800982 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800983 return mState != STATE_ACTIVE;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800984}
985
986void AudioTrack::flush()
987{
Andy Hungb68f5eb2019-12-03 16:49:17 -0800988 const int64_t beginNs = systemTime();
Andy Hungfb8ede22018-09-12 19:03:24 -0700989 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -0700990 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800991 mediametrics::LogItem(mMetricsId)
992 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_FLUSH)
Andy Hungea840382020-05-05 21:50:17 -0700993 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800994 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
995 .record(); });
996
Eric Laurent973db022018-11-20 14:54:31 -0800997 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -0700998
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800999 if (mSharedBuffer != 0) {
1000 return;
Glenn Kasten4bae3642012-11-30 13:41:12 -08001001 }
Andy Hung4c5ed302018-05-09 11:16:21 -07001002 if (mState == STATE_ACTIVE) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001003 return;
1004 }
1005 flush_l();
Eric Laurent1703cdf2011-03-07 14:52:59 -08001006}
1007
Eric Laurent1703cdf2011-03-07 14:52:59 -08001008void AudioTrack::flush_l()
1009{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001010 ALOG_ASSERT(mState != STATE_ACTIVE);
Eric Laurentc2f1f072009-07-17 12:17:14 -07001011
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001012 // clear playback marker and periodic update counter
1013 mMarkerPosition = 0;
1014 mMarkerReached = false;
1015 mUpdatePeriod = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001016 mRefreshRemaining = true;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001017
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001018 mState = STATE_FLUSHED;
Andy Hungc2813e52014-10-16 17:54:34 -07001019 mReleased = 0;
Glenn Kasten23a75452014-01-13 10:37:17 -08001020 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001021 mProxy->interrupt();
1022 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001023 mProxy->flush();
Glenn Kasten4bae3642012-11-30 13:41:12 -08001024 mAudioTrack->flush();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001025}
1026
Andy Hung959b5b82021-09-24 10:46:20 -07001027bool AudioTrack::pauseAndWait(const std::chrono::milliseconds& timeout)
1028{
1029 using namespace std::chrono_literals;
1030
Andy Hungd87a53a2022-01-19 16:56:17 -08001031 // We use atomic access here for state variables - these are used as hints
1032 // to ensure we have ramped down audio.
1033 const int priorState = mProxy->getState();
1034 const uint32_t priorPosition = mProxy->getPosition().unsignedValue();
1035
Andy Hung959b5b82021-09-24 10:46:20 -07001036 pause();
1037
Andy Hungd87a53a2022-01-19 16:56:17 -08001038 // Only if we were previously active, do we wait to ramp down the audio.
1039 if (priorState != CBLK_STATE_ACTIVE) return true;
1040
Andy Hung959b5b82021-09-24 10:46:20 -07001041 AutoMutex lock(mLock);
1042 // offload and direct tracks do not wait because pause volume ramp is handled by hardware.
1043 if (isOffloadedOrDirect_l()) return true;
1044
1045 // Wait for the track state to be anything besides pausing.
1046 // This ensures that the volume has ramped down.
1047 constexpr auto SLEEP_INTERVAL_MS = 10ms;
Andy Hungd87a53a2022-01-19 16:56:17 -08001048 constexpr auto POSITION_TIMEOUT_MS = 40ms; // don't wait longer than this for position change.
Andy Hung959b5b82021-09-24 10:46:20 -07001049 auto begin = std::chrono::steady_clock::now();
1050 while (true) {
Andy Hungd87a53a2022-01-19 16:56:17 -08001051 // Wait for state and position to change.
1052 // After pause() the server state should be PAUSING, but that may immediately
1053 // convert to PAUSED by prepareTracks before data is read into the mixer.
1054 // Hence we check that the state is not PAUSING and that the server position
1055 // has advanced to be a more reliable estimate that the volume ramp has completed.
Andy Hung959b5b82021-09-24 10:46:20 -07001056 const int state = mProxy->getState();
Andy Hungd87a53a2022-01-19 16:56:17 -08001057 const uint32_t position = mProxy->getPosition().unsignedValue();
Andy Hung959b5b82021-09-24 10:46:20 -07001058
1059 mLock.unlock(); // only local variables accessed until lock.
1060 auto elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(
1061 std::chrono::steady_clock::now() - begin);
Andy Hungd87a53a2022-01-19 16:56:17 -08001062 if (state != CBLK_STATE_PAUSING &&
1063 (elapsed >= POSITION_TIMEOUT_MS || position != priorPosition)) {
1064 ALOGV("%s: success state:%d, position:%u after %lld ms"
1065 " (prior state:%d prior position:%u)",
1066 __func__, state, position, elapsed.count(), priorState, priorPosition);
Andy Hung959b5b82021-09-24 10:46:20 -07001067 return true;
1068 }
1069 std::chrono::milliseconds remaining = timeout - elapsed;
1070 if (remaining.count() <= 0) {
1071 ALOGW("%s: timeout expired state:%d still pausing:%d after %lld ms",
1072 __func__, state, CBLK_STATE_PAUSING, elapsed.count());
1073 return false;
1074 }
1075 // It is conceivable that the track is restored while sleeping;
1076 // as this logic is advisory, we allow that.
1077 std::this_thread::sleep_for(std::min(remaining, SLEEP_INTERVAL_MS));
1078 mLock.lock();
1079 }
1080}
1081
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001082void AudioTrack::pause()
1083{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001084 const int64_t beginNs = systemTime();
Eric Laurentf5aafb22010-11-18 08:40:16 -08001085 AutoMutex lock(mLock);
Andy Hunga6b27032020-04-27 10:34:24 -07001086 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001087 mediametrics::LogItem(mMetricsId)
1088 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_PAUSE)
Andy Hungea840382020-05-05 21:50:17 -07001089 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001090 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
1091 .record(); });
1092
Eric Laurent973db022018-11-20 14:54:31 -08001093 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001094
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001095 if (mState == STATE_ACTIVE) {
1096 mState = STATE_PAUSED;
1097 } else if (mState == STATE_STOPPING) {
1098 mState = STATE_PAUSED_STOPPING;
1099 } else {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001100 return;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001101 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001102 mProxy->interrupt();
1103 mAudioTrack->pause();
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001104
Marco Nelissen3a90f282014-03-10 11:21:43 -07001105 if (isOffloaded_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001106 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001107 // An offload output can be re-used between two audio tracks having
1108 // the same configuration. A timestamp query for a paused track
1109 // while the other is running would return an incorrect time.
1110 // To fix this, cache the playback position on a pause() and return
1111 // this time when requested until the track is resumed.
1112
1113 // OffloadThread sends HAL pause in its threadLoop. Time saved
1114 // here can be slightly off.
1115
1116 // TODO: check return code for getRenderPosition.
1117
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001118 uint32_t halFrames;
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001119 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition);
Andy Hungfb8ede22018-09-12 19:03:24 -07001120 ALOGV("%s(%d): for offload, cache current position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001121 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001122 }
1123 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001124}
1125
Eric Laurentbe916aa2010-06-01 23:49:17 -07001126status_t AudioTrack::setVolume(float left, float right)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001127{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001128 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1129 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY ||
1130 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001131 return BAD_VALUE;
1132 }
1133
Andy Hungb68f5eb2019-12-03 16:49:17 -08001134 mediametrics::LogItem(mMetricsId)
1135 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOLUME)
1136 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)left)
1137 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)right)
1138 .record();
1139
Eric Laurent1703cdf2011-03-07 14:52:59 -08001140 AutoMutex lock(mLock);
Glenn Kasten66e46352014-01-16 17:44:23 -08001141 mVolume[AUDIO_INTERLEAVE_LEFT] = left;
1142 mVolume[AUDIO_INTERLEAVE_RIGHT] = right;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001143
Glenn Kastenc56f3422014-03-21 17:53:17 -07001144 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right)));
Eric Laurentbe916aa2010-06-01 23:49:17 -07001145
Glenn Kasten23a75452014-01-13 10:37:17 -08001146 if (isOffloaded_l()) {
Eric Laurent59fe0102013-09-27 18:48:26 -07001147 mAudioTrack->signal();
1148 }
Eric Laurentbe916aa2010-06-01 23:49:17 -07001149 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001150}
1151
Glenn Kastenb1c09932012-02-27 16:21:04 -08001152status_t AudioTrack::setVolume(float volume)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001153{
Glenn Kastenb1c09932012-02-27 16:21:04 -08001154 return setVolume(volume, volume);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001155}
1156
Eric Laurent2beeb502010-07-16 07:43:46 -07001157status_t AudioTrack::setAuxEffectSendLevel(float level)
Eric Laurentbe916aa2010-06-01 23:49:17 -07001158{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001159 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1160 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001161 return BAD_VALUE;
1162 }
1163
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001164 AutoMutex lock(mLock);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001165 mSendLevel = level;
Glenn Kastene3aa6592012-12-04 12:22:46 -08001166 mProxy->setSendLevel(level);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001167
1168 return NO_ERROR;
1169}
1170
Glenn Kastena5224f32012-01-04 12:41:44 -08001171void AudioTrack::getAuxEffectSendLevel(float* level) const
Eric Laurentbe916aa2010-06-01 23:49:17 -07001172{
1173 if (level != NULL) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001174 *level = mSendLevel;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001175 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001176}
1177
Glenn Kasten3b16c762012-11-14 08:44:39 -08001178status_t AudioTrack::setSampleRate(uint32_t rate)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001179{
Andy Hung5cbb5782015-03-27 18:39:59 -07001180 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08001181 ALOGV("%s(%d): prior state:%s rate:%u", __func__, mPortId, stateToString(mState), rate);
Andy Hungfb8ede22018-09-12 19:03:24 -07001182
Andy Hung5cbb5782015-03-27 18:39:59 -07001183 if (rate == mSampleRate) {
1184 return NO_ERROR;
1185 }
jiabinf4de6112018-12-19 12:40:08 -08001186 if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)
1187 || (mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL)) {
John Grossman4ff14ba2012-02-08 16:37:41 -08001188 return INVALID_OPERATION;
1189 }
Eric Laurent0d6db582014-11-12 18:39:44 -08001190 if (mOutput == AUDIO_IO_HANDLE_NONE) {
1191 return NO_INIT;
1192 }
Andy Hung5cbb5782015-03-27 18:39:59 -07001193 // NOTE: it is theoretically possible, but highly unlikely, that a device change
1194 // could mean a previously allowed sampling rate is no longer allowed.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001195 uint32_t afSamplingRate;
Eric Laurent0d6db582014-11-12 18:39:44 -08001196 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
Eric Laurent57326622009-07-07 07:10:45 -07001197 return NO_INIT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001198 }
Andy Hung26145642015-04-15 21:56:53 -07001199 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001200 const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001201 if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001202 return BAD_VALUE;
1203 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001204 // TODO: Should we also check if the buffer size is compatible?
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001205
Glenn Kastene3aa6592012-12-04 12:22:46 -08001206 mSampleRate = rate;
Andy Hung26145642015-04-15 21:56:53 -07001207 mProxy->setSampleRate(effectiveSampleRate);
Glenn Kastene3aa6592012-12-04 12:22:46 -08001208
Eric Laurent57326622009-07-07 07:10:45 -07001209 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001210}
1211
Glenn Kastena5224f32012-01-04 12:41:44 -08001212uint32_t AudioTrack::getSampleRate() const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001213{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001214 AutoMutex lock(mLock);
Eric Laurent6f59db12013-07-26 17:16:50 -07001215
1216 // sample rate can be updated during playback by the offloaded decoder so we need to
1217 // query the HAL and update if needed.
1218// FIXME use Proxy return channel to update the rate from server and avoid polling here
Eric Laurentab5cdba2014-06-09 17:22:27 -07001219 if (isOffloadedOrDirect_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001220 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent6f59db12013-07-26 17:16:50 -07001221 uint32_t sampleRate = 0;
Jean-Michel Trivib7f24b12014-06-11 10:05:30 -07001222 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
Eric Laurent6f59db12013-07-26 17:16:50 -07001223 if (status == NO_ERROR) {
1224 mSampleRate = sampleRate;
1225 }
1226 }
1227 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001228 return mSampleRate;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001229}
1230
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001231uint32_t AudioTrack::getOriginalSampleRate() const
1232{
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001233 return mOriginalSampleRate;
1234}
1235
Robert Wu310037a2022-09-06 21:48:18 +00001236uint32_t AudioTrack::getHalSampleRate() const
1237{
1238 return mAfSampleRate;
1239}
1240
1241uint32_t AudioTrack::getHalChannelCount() const
1242{
1243 return mAfChannelCount;
1244}
1245
1246audio_format_t AudioTrack::getHalFormat() const
1247{
1248 return mAfFormat;
1249}
1250
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001251status_t AudioTrack::setDualMonoMode(audio_dual_mono_mode_t mode)
1252{
1253 AutoMutex lock(mLock);
1254 return setDualMonoMode_l(mode);
1255}
1256
1257status_t AudioTrack::setDualMonoMode_l(audio_dual_mono_mode_t mode)
1258{
1259 const status_t status = statusTFromBinderStatus(
1260 mAudioTrack->setDualMonoMode(VALUE_OR_RETURN_STATUS(
1261 legacy2aidl_audio_dual_mono_mode_t_AudioDualMonoMode(mode))));
1262 if (status == NO_ERROR) mDualMonoMode = mode;
1263 return status;
1264}
1265
1266status_t AudioTrack::getDualMonoMode(audio_dual_mono_mode_t* mode) const
1267{
1268 AutoMutex lock(mLock);
Mikhail Naganovf53e1822022-12-18 02:48:14 +00001269 media::audio::common::AudioDualMonoMode mediaMode;
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001270 const status_t status = statusTFromBinderStatus(mAudioTrack->getDualMonoMode(&mediaMode));
1271 if (status == NO_ERROR) {
1272 *mode = VALUE_OR_RETURN_STATUS(
1273 aidl2legacy_AudioDualMonoMode_audio_dual_mono_mode_t(mediaMode));
1274 }
1275 return status;
1276}
1277
1278status_t AudioTrack::setAudioDescriptionMixLevel(float leveldB)
1279{
1280 AutoMutex lock(mLock);
1281 return setAudioDescriptionMixLevel_l(leveldB);
1282}
1283
1284status_t AudioTrack::setAudioDescriptionMixLevel_l(float leveldB)
1285{
1286 const status_t status = statusTFromBinderStatus(
1287 mAudioTrack->setAudioDescriptionMixLevel(leveldB));
1288 if (status == NO_ERROR) mAudioDescriptionMixLeveldB = leveldB;
1289 return status;
1290}
1291
1292status_t AudioTrack::getAudioDescriptionMixLevel(float* leveldB) const
1293{
1294 AutoMutex lock(mLock);
1295 return statusTFromBinderStatus(mAudioTrack->getAudioDescriptionMixLevel(leveldB));
1296}
1297
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001298status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
Andy Hung8edb8dc2015-03-26 19:13:55 -07001299{
Andy Hung8edb8dc2015-03-26 19:13:55 -07001300 AutoMutex lock(mLock);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001301 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
Andy Hung8edb8dc2015-03-26 19:13:55 -07001302 return NO_ERROR;
1303 }
Glenn Kastend79072e2016-01-06 08:41:20 -08001304 if (isOffloadedOrDirect_l()) {
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001305 const status_t status = statusTFromBinderStatus(mAudioTrack->setPlaybackRateParameters(
1306 VALUE_OR_RETURN_STATUS(
1307 legacy2aidl_audio_playback_rate_t_AudioPlaybackRate(playbackRate))));
1308 if (status == NO_ERROR) {
1309 mPlaybackRate = playbackRate;
Bertil Ã…kesson081fb742022-07-14 16:39:36 +02001310 } else if (status == INVALID_OPERATION
1311 && playbackRate.mSpeed == 1.0f && mPlaybackRate.mPitch == 1.0f) {
1312 mPlaybackRate = playbackRate;
1313 return NO_ERROR;
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001314 }
1315 return status;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001316 }
1317 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
1318 return INVALID_OPERATION;
1319 }
Andy Hungff874dc2016-04-11 16:49:09 -07001320
Andy Hungfb8ede22018-09-12 19:03:24 -07001321 ALOGV("%s(%d): mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001322 __func__, mPortId, mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001323 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001324 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
1325 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
1326 const float effectivePitch = adjustPitch(playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001327 AudioPlaybackRate playbackRateTemp = playbackRate;
1328 playbackRateTemp.mSpeed = effectiveSpeed;
1329 playbackRateTemp.mPitch = effectivePitch;
1330
Andy Hungfb8ede22018-09-12 19:03:24 -07001331 ALOGV("%s(%d) (effective) mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001332 __func__, mPortId, effectiveRate, effectiveSpeed, effectivePitch);
Andy Hungff874dc2016-04-11 16:49:09 -07001333
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001334 if (!isAudioPlaybackRateValid(playbackRateTemp)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001335 ALOGW("%s(%d) (%f, %f) failed (effective rate out of bounds)",
Eric Laurent973db022018-11-20 14:54:31 -08001336 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001337 return BAD_VALUE;
1338 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001339 // Check if the buffer size is compatible.
Andy Hung26145642015-04-15 21:56:53 -07001340 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001341 ALOGW("%s(%d) (%f, %f) failed (buffer size)",
Eric Laurent973db022018-11-20 14:54:31 -08001342 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung8edb8dc2015-03-26 19:13:55 -07001343 return BAD_VALUE;
1344 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001345
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001346 // Check resampler ratios are within bounds
Glenn Kastend3bb6452016-12-05 18:14:37 -08001347 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
1348 (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001349 ALOGW("%s(%d) (%f, %f) failed. Resample rate exceeds max accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001350 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001351 return BAD_VALUE;
1352 }
1353
Dan Austine34eae22015-10-27 16:14:52 -07001354 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001355 ALOGW("%s(%d) (%f, %f) failed. Resample rate below min accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001356 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001357 return BAD_VALUE;
1358 }
1359 mPlaybackRate = playbackRate;
1360 //set effective rates
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001361 mProxy->setPlaybackRate(playbackRateTemp);
Andy Hung26145642015-04-15 21:56:53 -07001362 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
Andy Hungb68f5eb2019-12-03 16:49:17 -08001363
1364 mediametrics::LogItem(mMetricsId)
1365 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYBACKPARAM)
1366 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
1367 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
1368 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
1369 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1370 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveRate)
1371 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1372 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)playbackRateTemp.mSpeed)
1373 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1374 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)playbackRateTemp.mPitch)
1375 .record();
1376
Andy Hung8edb8dc2015-03-26 19:13:55 -07001377 return NO_ERROR;
1378}
1379
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001380const AudioPlaybackRate& AudioTrack::getPlaybackRate()
Andy Hung8edb8dc2015-03-26 19:13:55 -07001381{
1382 AutoMutex lock(mLock);
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001383 if (isOffloadedOrDirect_l()) {
Mikhail Naganovf53e1822022-12-18 02:48:14 +00001384 media::audio::common::AudioPlaybackRate playbackRateTemp;
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001385 const status_t status = statusTFromBinderStatus(
1386 mAudioTrack->getPlaybackRateParameters(&playbackRateTemp));
1387 if (status == NO_ERROR) { // update local version if changed.
1388 mPlaybackRate =
1389 aidl2legacy_AudioPlaybackRate_audio_playback_rate_t(playbackRateTemp).value();
1390 }
1391 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001392 return mPlaybackRate;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001393}
1394
Phil Burkc0adecb2016-01-08 12:44:11 -08001395ssize_t AudioTrack::getBufferSizeInFrames()
1396{
1397 AutoMutex lock(mLock);
1398 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1399 return NO_INIT;
1400 }
Phil Burka9876702020-04-20 18:16:15 -07001401
Phil Burke8972b02016-03-04 11:29:57 -08001402 return (ssize_t) mProxy->getBufferSizeInFrames();
Phil Burkc0adecb2016-01-08 12:44:11 -08001403}
1404
Andy Hungf2c87b32016-04-07 19:49:29 -07001405status_t AudioTrack::getBufferDurationInUs(int64_t *duration)
1406{
1407 if (duration == nullptr) {
1408 return BAD_VALUE;
1409 }
1410 AutoMutex lock(mLock);
1411 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1412 return NO_INIT;
1413 }
1414 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames();
1415 if (bufferSizeInFrames < 0) {
1416 return (status_t)bufferSizeInFrames;
1417 }
1418 *duration = (int64_t)((double)bufferSizeInFrames * 1000000
1419 / ((double)mSampleRate * mPlaybackRate.mSpeed));
1420 return NO_ERROR;
1421}
1422
Phil Burkc0adecb2016-01-08 12:44:11 -08001423ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames)
1424{
1425 AutoMutex lock(mLock);
1426 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1427 return NO_INIT;
1428 }
Phil Burka9876702020-04-20 18:16:15 -07001429
1430 ssize_t originalBufferSize = mProxy->getBufferSizeInFrames();
1431 ssize_t finalBufferSize = mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
1432 if (originalBufferSize != finalBufferSize) {
Phil Burk64e16a72020-06-01 13:25:51 -07001433 android::mediametrics::LogItem(mMetricsId)
1434 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
1435 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1436 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t)getUnderrunCount_l())
1437 .record();
Phil Burka9876702020-04-20 18:16:15 -07001438 }
1439 return finalBufferSize;
Phil Burkc0adecb2016-01-08 12:44:11 -08001440}
1441
Andy Hung3c7f47a2021-03-16 17:30:09 -07001442ssize_t AudioTrack::getStartThresholdInFrames() const
1443{
1444 AutoMutex lock(mLock);
1445 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1446 return NO_INIT;
1447 }
1448 return (ssize_t) mProxy->getStartThresholdInFrames();
1449}
1450
1451ssize_t AudioTrack::setStartThresholdInFrames(size_t startThresholdInFrames)
1452{
1453 if (startThresholdInFrames > INT32_MAX || startThresholdInFrames == 0) {
1454 // contractually we could simply return the current threshold in frames
1455 // to indicate the request was ignored, but we return an error here.
1456 return BAD_VALUE;
1457 }
1458 AutoMutex lock(mLock);
1459 // We do not permit calling setStartThresholdInFrames() between the AudioTrack
1460 // default ctor AudioTrack() and set(...) but rather fail such an attempt.
1461 // (To do so would require a cached mOrigStartThresholdInFrames and we may
1462 // not have proper validation for the actual set value).
1463 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1464 return NO_INIT;
1465 }
1466 const uint32_t original = mProxy->getStartThresholdInFrames();
1467 const uint32_t final = mProxy->setStartThresholdInFrames(startThresholdInFrames);
1468 if (original != final) {
1469 android::mediametrics::LogItem(mMetricsId)
1470 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD)
1471 .set(AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES, (int32_t)final)
1472 .record();
1473 if (original > final) {
1474 // restart track if it was disabled by audioflinger due to previous underrun
1475 // and we reduced the number of frames for the threshold.
1476 restartIfDisabled();
1477 }
1478 }
1479 return final;
1480}
1481
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001482status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1483{
Glenn Kastend79072e2016-01-06 08:41:20 -08001484 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001485 return INVALID_OPERATION;
1486 }
1487
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001488 if (loopCount == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001489 ;
1490 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount &&
1491 loopEnd - loopStart >= MIN_LOOP) {
1492 ;
1493 } else {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001494 return BAD_VALUE;
1495 }
1496
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001497 AutoMutex lock(mLock);
1498 // See setPosition() regarding setting parameters such as loop points or position while active
1499 if (mState == STATE_ACTIVE) {
1500 return INVALID_OPERATION;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001501 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001502 setLoop_l(loopStart, loopEnd, loopCount);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001503 return NO_ERROR;
1504}
1505
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001506void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1507{
Andy Hung4ede21d2014-12-12 15:37:34 -08001508 // We do not update the periodic notification point.
1509 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
1510 mLoopCount = loopCount;
1511 mLoopEnd = loopEnd;
1512 mLoopStart = loopStart;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001513 mLoopCountNotified = loopCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001514 mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
Andy Hung3c09c782014-12-29 18:39:32 -08001515
1516 // Waking the AudioTrackThread is not needed as this cannot be called when active.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001517}
1518
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001519status_t AudioTrack::setMarkerPosition(uint32_t marker)
1520{
Atneya Nair14aabae2021-11-30 17:36:24 -05001521 AutoMutex lock(mLock);
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001522 // The only purpose of setting marker position is to get a callback
Atneya Nair14aabae2021-11-30 17:36:24 -05001523 if (!mCallback.promote() || isOffloadedOrDirect_l()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001524 return INVALID_OPERATION;
1525 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001526
1527 mMarkerPosition = marker;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001528 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001529
Andy Hung3c09c782014-12-29 18:39:32 -08001530 sp<AudioTrackThread> t = mAudioTrackThread;
1531 if (t != 0) {
1532 t->wake();
1533 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001534 return NO_ERROR;
1535}
1536
Glenn Kastena5224f32012-01-04 12:41:44 -08001537status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001538{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001539 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001540 return INVALID_OPERATION;
1541 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001542 if (marker == NULL) {
1543 return BAD_VALUE;
1544 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001545
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001546 AutoMutex lock(mLock);
Andy Hung90e8a972015-11-09 16:42:40 -08001547 mMarkerPosition.getValue(marker);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001548
1549 return NO_ERROR;
1550}
1551
1552status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
1553{
Atneya Nair14aabae2021-11-30 17:36:24 -05001554 AutoMutex lock(mLock);
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001555 // The only purpose of setting position update period is to get a callback
Atneya Nair14aabae2021-11-30 17:36:24 -05001556 if (!mCallback.promote() || isOffloadedOrDirect_l()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001557 return INVALID_OPERATION;
1558 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001559
Glenn Kasten200092b2014-08-15 15:13:30 -07001560 mNewPosition = updateAndGetPosition_l() + updatePeriod;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001561 mUpdatePeriod = updatePeriod;
Glenn Kasten2b2165c2014-01-13 08:53:36 -08001562
Andy Hung3c09c782014-12-29 18:39:32 -08001563 sp<AudioTrackThread> t = mAudioTrackThread;
1564 if (t != 0) {
1565 t->wake();
1566 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001567 return NO_ERROR;
1568}
1569
Glenn Kastena5224f32012-01-04 12:41:44 -08001570status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001571{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001572 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001573 return INVALID_OPERATION;
1574 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001575 if (updatePeriod == NULL) {
1576 return BAD_VALUE;
1577 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001578
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001579 AutoMutex lock(mLock);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001580 *updatePeriod = mUpdatePeriod;
1581
1582 return NO_ERROR;
1583}
1584
1585status_t AudioTrack::setPosition(uint32_t position)
1586{
Glenn Kastend79072e2016-01-06 08:41:20 -08001587 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001588 return INVALID_OPERATION;
1589 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001590 if (position > mFrameCount) {
1591 return BAD_VALUE;
1592 }
John Grossman4ff14ba2012-02-08 16:37:41 -08001593
Eric Laurent1703cdf2011-03-07 14:52:59 -08001594 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001595 // Currently we require that the player is inactive before setting parameters such as position
1596 // or loop points. Otherwise, there could be a race condition: the application could read the
1597 // current position, compute a new position or loop parameters, and then set that position or
1598 // loop parameters but it would do the "wrong" thing since the position has continued to advance
1599 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app
1600 // to specify how it wants to handle such scenarios.
1601 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001602 return INVALID_OPERATION;
1603 }
Andy Hung9b461582014-12-01 17:56:29 -08001604 // After setting the position, use full update period before notification.
Glenn Kasten200092b2014-08-15 15:13:30 -07001605 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001606 mStaticProxy->setBufferPosition(position);
Andy Hung3c09c782014-12-29 18:39:32 -08001607
1608 // Waking the AudioTrackThread is not needed as this cannot be called when active.
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001609 return NO_ERROR;
1610}
1611
Glenn Kasten200092b2014-08-15 15:13:30 -07001612status_t AudioTrack::getPosition(uint32_t *position)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001613{
Glenn Kastend65d73c2012-06-22 17:21:07 -07001614 if (position == NULL) {
1615 return BAD_VALUE;
1616 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001617
Eric Laurent1703cdf2011-03-07 14:52:59 -08001618 AutoMutex lock(mLock);
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001619 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
1620 if (mState == STATE_STOPPED || mState == STATE_FLUSHED) {
1621 *position = 0;
1622 return NO_ERROR;
1623 }
Andy Hung7a490e72016-03-23 15:58:10 -07001624 // FIXME: offloaded and direct tracks call into the HAL for render positions
1625 // for compressed/synced data; however, we use proxy position for pure linear pcm data
1626 // as we do not know the capability of the HAL for pcm position support and standby.
1627 // There may be some latency differences between the HAL position and the proxy position.
1628 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) {
Eric Laurentab5cdba2014-06-09 17:22:27 -07001629 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001630 ALOGV("%s(%d): called in paused state, return cached position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001631 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001632 *position = mPausedPosition;
1633 return NO_ERROR;
1634 }
1635
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001636 uint32_t dspFrames = 0;
Glenn Kasten142f5192014-03-25 17:44:59 -07001637 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung1f1db832015-06-08 13:26:10 -07001638 uint32_t halFrames; // actually unused
Andy Hung1f1db832015-06-08 13:26:10 -07001639 // FIXME: on getRenderPosition() error, we return OK with frame position 0.
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001640 if (AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames) != NO_ERROR) {
1641 *position = 0;
1642 return NO_ERROR;
1643 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001644 }
1645 *position = dspFrames;
1646 } else {
Eric Laurent275e8e92014-11-30 15:14:47 -08001647 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung1f1db832015-06-08 13:26:10 -07001648 (void) restoreTrack_l("getPosition");
1649 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l()
1650 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position.
Eric Laurent275e8e92014-11-30 15:14:47 -08001651 }
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001652 *position = updateAndGetPosition_l().value();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001653 }
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001654
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001655 return NO_ERROR;
1656}
1657
Kévin PETIT377b2ec2014-02-03 12:35:36 +00001658status_t AudioTrack::getBufferPosition(uint32_t *position)
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001659{
Glenn Kastend79072e2016-01-06 08:41:20 -08001660 if (mSharedBuffer == 0) {
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001661 return INVALID_OPERATION;
1662 }
1663 if (position == NULL) {
1664 return BAD_VALUE;
1665 }
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001666
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001667 AutoMutex lock(mLock);
1668 *position = mStaticProxy->getBufferPosition();
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001669 return NO_ERROR;
1670}
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001671
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001672status_t AudioTrack::reload()
1673{
Glenn Kastend79072e2016-01-06 08:41:20 -08001674 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001675 return INVALID_OPERATION;
1676 }
1677
Eric Laurent1703cdf2011-03-07 14:52:59 -08001678 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001679 // See setPosition() regarding setting parameters such as loop points or position while active
1680 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001681 return INVALID_OPERATION;
1682 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001683 mNewPosition = mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001684 (void) updateAndGetPosition_l();
1685 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07001686 mPreviousTimestampValid = false;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001687#if 0
Andy Hung9b461582014-12-01 17:56:29 -08001688 // The documentation is not clear on the behavior of reload() and the restoration
Andy Hung53c3b5f2014-12-15 16:42:05 -08001689 // of loop count. Historically we have not restored loop count, start, end,
1690 // but it makes sense if one desires to repeat playing a particular sound.
1691 if (mLoopCount != 0) {
1692 mLoopCountNotified = mLoopCount;
1693 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
1694 }
1695#endif
Andy Hung9b461582014-12-01 17:56:29 -08001696 mStaticProxy->setBufferPosition(0);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001697 return NO_ERROR;
1698}
1699
Glenn Kasten38e905b2014-01-13 10:21:48 -08001700audio_io_handle_t AudioTrack::getOutput() const
Eric Laurentc2f1f072009-07-17 12:17:14 -07001701{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001702 AutoMutex lock(mLock);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001703 return mOutput;
Eric Laurent1703cdf2011-03-07 14:52:59 -08001704}
1705
Paul McLeanaa981192015-03-21 09:55:15 -07001706status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) {
1707 AutoMutex lock(mLock);
Eric Laurent72af8012023-03-15 17:36:22 +01001708 ALOGV("%s(%d): deviceId=%d mSelectedDeviceId=%d mRoutedDeviceId %d",
1709 __func__, mPortId, deviceId, mSelectedDeviceId, mRoutedDeviceId);
Paul McLeanaa981192015-03-21 09:55:15 -07001710 if (mSelectedDeviceId != deviceId) {
1711 mSelectedDeviceId = deviceId;
Eric Laurent72af8012023-03-15 17:36:22 +01001712 if (mStatus == NO_ERROR && mSelectedDeviceId != mRoutedDeviceId) {
1713 if (isPlaying_l()) {
1714 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
1715 mProxy->interrupt();
1716 } else {
1717 // if the track is idle, try to restore now and
1718 // defer to next start if not possible
1719 if (restoreTrack_l("setOutputDevice") != OK) {
1720 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
1721 }
1722 }
Eric Laurentfb00fc72017-05-25 18:17:12 -07001723 }
Paul McLeanaa981192015-03-21 09:55:15 -07001724 }
Eric Laurent493404d2015-04-21 15:07:36 -07001725 return NO_ERROR;
Paul McLeanaa981192015-03-21 09:55:15 -07001726}
1727
1728audio_port_handle_t AudioTrack::getOutputDevice() {
1729 AutoMutex lock(mLock);
1730 return mSelectedDeviceId;
1731}
1732
Eric Laurentad2e7b92017-09-14 20:06:42 -07001733// must be called with mLock held
1734void AudioTrack::updateRoutedDeviceId_l()
1735{
1736 // if the track is inactive, do not update actual device as the output stream maybe routed
1737 // to a device not relevant to this client because of other active use cases.
1738 if (mState != STATE_ACTIVE) {
1739 return;
1740 }
1741 if (mOutput != AUDIO_IO_HANDLE_NONE) {
1742 audio_port_handle_t deviceId = AudioSystem::getDeviceIdForIo(mOutput);
1743 if (deviceId != AUDIO_PORT_HANDLE_NONE) {
1744 mRoutedDeviceId = deviceId;
1745 }
1746 }
1747}
1748
Eric Laurent296fb132015-05-01 11:38:42 -07001749audio_port_handle_t AudioTrack::getRoutedDeviceId() {
1750 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001751 updateRoutedDeviceId_l();
1752 return mRoutedDeviceId;
Eric Laurent296fb132015-05-01 11:38:42 -07001753}
1754
Eric Laurentbe916aa2010-06-01 23:49:17 -07001755status_t AudioTrack::attachAuxEffect(int effectId)
1756{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001757 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001758 status_t status;
1759 mAudioTrack->attachAuxEffect(effectId, &status);
Eric Laurent2beeb502010-07-16 07:43:46 -07001760 if (status == NO_ERROR) {
1761 mAuxEffectId = effectId;
1762 }
1763 return status;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001764}
1765
Eric Laurente83b55d2014-11-14 10:06:21 -08001766audio_stream_type_t AudioTrack::streamType() const
1767{
Eric Laurente83b55d2014-11-14 10:06:21 -08001768 return mStreamType;
1769}
1770
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001771uint32_t AudioTrack::latency()
1772{
1773 AutoMutex lock(mLock);
1774 updateLatency_l();
1775 return mLatency;
1776}
1777
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001778// -------------------------------------------------------------------------
1779
Eric Laurent1703cdf2011-03-07 14:52:59 -08001780// must be called with mLock held
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001781void AudioTrack::updateLatency_l()
1782{
1783 status_t status = AudioSystem::getLatency(mOutput, &mAfLatency);
1784 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08001785 ALOGW("%s(%d): getLatency(%d) failed status %d", __func__, mPortId, mOutput, status);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001786 } else {
1787 // FIXME don't believe this lie
Andy Hung13969262017-09-11 17:24:21 -07001788 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001789 }
1790}
1791
Phil Burkadbb75a2017-06-16 12:19:42 -07001792// TODO Move this macro to a common header file for enum to string conversion in audio framework.
1793#define MEDIA_CASE_ENUM(name) case name: return #name
1794const char * AudioTrack::convertTransferToText(transfer_type transferType) {
1795 switch (transferType) {
1796 MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
1797 MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
1798 MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
1799 MEDIA_CASE_ENUM(TRANSFER_SYNC);
1800 MEDIA_CASE_ENUM(TRANSFER_SHARED);
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001801 MEDIA_CASE_ENUM(TRANSFER_SYNC_NOTIF_CALLBACK);
Phil Burkadbb75a2017-06-16 12:19:42 -07001802 default:
1803 return "UNRECOGNIZED";
1804 }
1805}
1806
Glenn Kasten200092b2014-08-15 15:13:30 -07001807status_t AudioTrack::createTrack_l()
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001808{
Eric Laurentf32d7812017-11-30 14:44:07 -08001809 status_t status;
1810 bool callbackAdded = false;
Andy Hung3acde2c2021-11-11 09:18:08 -08001811 std::string errorMessage;
Eric Laurentf32d7812017-11-30 14:44:07 -08001812
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001813 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
1814 if (audioFlinger == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001815 errorMessage = StringPrintf("%s(%d): Could not get audioflinger",
Eric Laurent973db022018-11-20 14:54:31 -08001816 __func__, mPortId);
Andy Hung3acde2c2021-11-11 09:18:08 -08001817 status = DEAD_OBJECT;
Eric Laurentf32d7812017-11-30 14:44:07 -08001818 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001819 }
1820
Eric Laurent21da6472017-11-09 16:29:26 -08001821 {
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08001822 // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
1823 // After fast request is denied, we will request again if IAudioTrack is re-created.
Glenn Kastend79072e2016-01-06 08:41:20 -08001824 // Client can only express a preference for FAST. Server will perform additional tests.
Phil Burk33ff89b2015-11-30 11:16:01 -08001825 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Phil Burkadbb75a2017-06-16 12:19:42 -07001826 // either of these use cases:
1827 // use case 1: shared buffer
1828 bool sharedBuffer = mSharedBuffer != 0;
1829 bool transferAllowed =
Glenn Kastenc6ba8232014-02-27 13:34:29 -08001830 // use case 2: callback transfer mode
Glenn Kasten1dfe2f92015-03-09 12:03:14 -07001831 (mTransfer == TRANSFER_CALLBACK) ||
1832 // use case 3: obtain/release mode
Phil Burk33ff89b2015-11-30 11:16:01 -08001833 (mTransfer == TRANSFER_OBTAIN) ||
1834 // use case 4: synchronous write
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001835 ((mTransfer == TRANSFER_SYNC || mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK)
1836 && mThreadCanCallJava);
Phil Burkadbb75a2017-06-16 12:19:42 -07001837
Eric Laurent21da6472017-11-09 16:29:26 -08001838 bool fastAllowed = sharedBuffer || transferAllowed;
1839 if (!fastAllowed) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001840 ALOGW("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by client,"
1841 " not shared buffer and transfer = %s",
Eric Laurent973db022018-11-20 14:54:31 -08001842 __func__, mPortId,
Phil Burkadbb75a2017-06-16 12:19:42 -07001843 convertTransferToText(mTransfer));
Phil Burk33ff89b2015-11-30 11:16:01 -08001844 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
1845 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001846 }
1847
Eric Laurent21da6472017-11-09 16:29:26 -08001848 IAudioFlinger::CreateTrackInput input;
Andy Hunga2159aa2021-07-20 13:01:52 -07001849 if (mOriginalStreamType != AUDIO_STREAM_DEFAULT) {
1850 // Legacy: This is based on original parameters even if the track is recreated.
1851 input.attr = AudioSystem::streamTypeToAttributes(mOriginalStreamType);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001852 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08001853 input.attr = mAttributes;
Eric Laurentd1b449a2010-05-14 03:26:45 -07001854 }
Eric Laurent21da6472017-11-09 16:29:26 -08001855 input.config = AUDIO_CONFIG_INITIALIZER;
1856 input.config.sample_rate = mSampleRate;
1857 input.config.channel_mask = mChannelMask;
1858 input.config.format = mFormat;
1859 input.config.offload_info = mOffloadInfoCopy;
Svet Ganov3e5f14f2021-05-13 22:51:08 +00001860 input.clientInfo.attributionSource = mClientAttributionSource;
Eric Laurent21da6472017-11-09 16:29:26 -08001861 input.clientInfo.clientTid = -1;
Glenn Kasten363fb752014-01-15 12:27:31 -08001862 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Glenn Kasten47d55172017-05-23 11:19:30 -07001863 // It is currently meaningless to request SCHED_FIFO for a Java thread. Even if the
1864 // application-level code follows all non-blocking design rules, the language runtime
1865 // doesn't also follow those rules, so the thread will not benefit overall.
Phil Burk33ff89b2015-11-30 11:16:01 -08001866 if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
Eric Laurent21da6472017-11-09 16:29:26 -08001867 input.clientInfo.clientTid = mAudioTrackThread->getTid();
Glenn Kasten3acbd052012-02-28 10:39:56 -08001868 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001869 }
Eric Laurent21da6472017-11-09 16:29:26 -08001870 input.sharedBuffer = mSharedBuffer;
1871 input.notificationsPerBuffer = mNotificationsPerBufferReq;
1872 input.speed = 1.0;
1873 if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
1874 (mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
1875 input.speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
1876 max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
1877 }
1878 input.flags = mFlags;
1879 input.frameCount = mReqFrameCount;
1880 input.notificationFrameCount = mNotificationFramesReq;
1881 input.selectedDeviceId = mSelectedDeviceId;
1882 input.sessionId = mSessionId;
jiabinf6eb4c32020-02-25 14:06:25 -08001883 input.audioTrackCallback = mAudioTrackCallback;
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001884
Ytai Ben-Tsvi4dfeb622020-11-02 12:47:30 -08001885 media::CreateTrackResponse response;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001886 status = audioFlinger->createTrack(VALUE_OR_FATAL(input.toAidl()), response);
Ytai Ben-Tsvi357e26a2021-01-05 13:21:19 -08001887
1888 IAudioFlinger::CreateTrackOutput output{};
1889 if (status == NO_ERROR) {
1890 output = VALUE_OR_FATAL(IAudioFlinger::CreateTrackOutput::fromAidl(response));
1891 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001892
Eric Laurent21da6472017-11-09 16:29:26 -08001893 if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001894 errorMessage = StringPrintf(
1895 "%s(%d): AudioFlinger could not create track, status: %d output %d",
Eric Laurent973db022018-11-20 14:54:31 -08001896 __func__, mPortId, status, output.outputId);
Eric Laurentf32d7812017-11-30 14:44:07 -08001897 if (status == NO_ERROR) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001898 status = INVALID_OPERATION; // device not ready
Eric Laurentf32d7812017-11-30 14:44:07 -08001899 }
1900 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001901 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001902 ALOG_ASSERT(output.audioTrack != 0);
Glenn Kastenc08d20b2014-02-24 15:21:10 -08001903
Eric Laurent21da6472017-11-09 16:29:26 -08001904 mFrameCount = output.frameCount;
1905 mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
1906 mRoutedDeviceId = output.selectedDeviceId;
1907 mSessionId = output.sessionId;
Andy Hunga2159aa2021-07-20 13:01:52 -07001908 mStreamType = output.streamType;
Eric Laurent21da6472017-11-09 16:29:26 -08001909
1910 mSampleRate = output.sampleRate;
1911 if (mOriginalSampleRate == 0) {
1912 mOriginalSampleRate = mSampleRate;
1913 }
1914
1915 mAfFrameCount = output.afFrameCount;
1916 mAfSampleRate = output.afSampleRate;
Robert Wu310037a2022-09-06 21:48:18 +00001917 mAfChannelCount = audio_channel_count_from_out_mask(output.afChannelMask);
1918 mAfFormat = output.afFormat;
Eric Laurent21da6472017-11-09 16:29:26 -08001919 mAfLatency = output.afLatencyMs;
1920
1921 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
1922
Glenn Kasten38e905b2014-01-13 10:21:48 -08001923 // AudioFlinger now owns the reference to the I/O handle,
1924 // so we are no longer responsible for releasing it.
1925
Glenn Kasten7fd04222016-02-02 12:38:16 -08001926 // FIXME compare to AudioRecord
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001927 std::optional<media::SharedFileRegion> sfr;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001928 output.audioTrack->getCblk(&sfr);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001929 sp<IMemory> iMem = VALUE_OR_FATAL(aidl2legacy_NullableSharedFileRegion_IMemory(sfr));
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001930 if (iMem == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001931 errorMessage = StringPrintf("%s(%d): Could not get control block", __func__, mPortId);
1932 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001933 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001934 }
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001935 // TODO: Using unsecurePointer() has some associated security pitfalls
1936 // (see declaration for details).
1937 // Either document why it is safe in this case or address the
1938 // issue (e.g. by copying).
1939 void *iMemPointer = iMem->unsecurePointer();
Glenn Kasten0cde0762014-01-16 15:06:36 -08001940 if (iMemPointer == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001941 errorMessage = StringPrintf(
1942 "%s(%d): Could not get control block pointer", __func__, mPortId);
1943 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001944 goto exit;
Glenn Kasten0cde0762014-01-16 15:06:36 -08001945 }
Glenn Kasten53cec222013-08-29 09:01:02 -07001946 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001947 if (mAudioTrack != 0) {
Marco Nelissenf8880202014-11-14 07:58:25 -08001948 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001949 mDeathNotifier.clear();
1950 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001951 mAudioTrack = output.audioTrack;
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001952 mCblkMemory = iMem;
Eric Laurent3bcffa12014-06-12 18:38:45 -07001953 IPCThreadState::self()->flushCommands();
1954
Glenn Kasten0cde0762014-01-16 15:06:36 -08001955 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001956 mCblk = cblk;
Glenn Kasten5f631512014-02-24 15:16:07 -08001957
Glenn Kastena07f17c2013-04-23 12:39:37 -07001958 mAwaitBoost = false;
Glenn Kasten363fb752014-01-15 12:27:31 -08001959 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Eric Laurent21da6472017-11-09 16:29:26 -08001960 if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001961 ALOGI("%s(%d): AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08001962 __func__, mPortId, mReqFrameCount, mFrameCount);
Phil Burk33ff89b2015-11-30 11:16:01 -08001963 if (!mThreadCanCallJava) {
1964 mAwaitBoost = true;
1965 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08001966 } else {
Robert Wuc121cd12021-08-13 17:51:40 +00001967 ALOGV("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08001968 __func__, mPortId, mReqFrameCount, mFrameCount);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001969 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08001970 }
Eric Laurent21da6472017-11-09 16:29:26 -08001971 mFlags = output.flags;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001972
Eric Laurentad2e7b92017-09-14 20:06:42 -07001973 //mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
Eric Laurent09f1ed22019-04-24 17:45:17 -07001974 if (mDeviceCallback != 0) {
Eric Laurentad2e7b92017-09-14 20:06:42 -07001975 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07001976 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001977 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07001978 AudioSystem::addAudioDeviceCallback(this, output.outputId, output.portId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001979 callbackAdded = true;
1980 }
1981
Eric Laurent09f1ed22019-04-24 17:45:17 -07001982 mPortId = output.portId;
Vlad Popaad0fe922022-06-10 00:43:14 +02001983 // notify the upper layers about the new portId
1984 triggerPortIdUpdate_l();
1985
Glenn Kasten38e905b2014-01-13 10:21:48 -08001986 // We retain a copy of the I/O handle, but don't own the reference
Eric Laurent21da6472017-11-09 16:29:26 -08001987 mOutput = output.outputId;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001988 mRefreshRemaining = true;
1989
1990 // Starting address of buffers in shared memory. If there is a shared buffer, buffers
1991 // is the value of pointer() for the shared buffer, otherwise buffers points
1992 // immediately after the control block. This address is for the mapping within client
1993 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
1994 void* buffers;
Glenn Kasten363fb752014-01-15 12:27:31 -08001995 if (mSharedBuffer == 0) {
Glenn Kasten138d6f92015-03-20 10:54:51 -07001996 buffers = cblk + 1;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001997 } else {
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001998 // TODO: Using unsecurePointer() has some associated security pitfalls
1999 // (see declaration for details).
2000 // Either document why it is safe in this case or address the
2001 // issue (e.g. by copying).
2002 buffers = mSharedBuffer->unsecurePointer();
Glenn Kasten138d6f92015-03-20 10:54:51 -07002003 if (buffers == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08002004 errorMessage = StringPrintf(
2005 "%s(%d): Could not get buffer pointer", __func__, mPortId);
2006 ALOGE("%s", errorMessage.c_str());
2007 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08002008 goto exit;
Glenn Kasten138d6f92015-03-20 10:54:51 -07002009 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002010 }
2011
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002012 mAudioTrack->attachAuxEffect(mAuxEffectId, &status);
Glenn Kasten5f631512014-02-24 15:16:07 -08002013
Glenn Kasten093000f2012-05-03 09:35:36 -07002014 // If IAudioTrack is re-created, don't let the requested frameCount
2015 // decrease. This can confuse clients that cache frameCount().
Eric Laurent21da6472017-11-09 16:29:26 -08002016 if (mFrameCount > mReqFrameCount) {
2017 mReqFrameCount = mFrameCount;
Glenn Kasten093000f2012-05-03 09:35:36 -07002018 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08002019
Andy Hungd7bd69e2015-07-24 07:52:41 -07002020 // reset server position to 0 as we have new cblk.
2021 mServer = 0;
2022
Glenn Kastene3aa6592012-12-04 12:22:46 -08002023 // update proxy
Glenn Kasten363fb752014-01-15 12:27:31 -08002024 if (mSharedBuffer == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002025 mStaticProxy.clear();
Eric Laurent21da6472017-11-09 16:29:26 -08002026 mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002027 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08002028 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002029 mProxy = mStaticProxy;
2030 }
seunghak.hane6a9d6582014-11-22 15:22:35 +09002031
2032 mProxy->setVolumeLR(gain_minifloat_pack(
2033 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
2034 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
2035
Glenn Kastene3aa6592012-12-04 12:22:46 -08002036 mProxy->setSendLevel(mSendLevel);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002037 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
2038 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
2039 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07002040 mProxy->setSampleRate(effectiveSampleRate);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002041
2042 AudioPlaybackRate playbackRateTemp = mPlaybackRate;
2043 playbackRateTemp.mSpeed = effectiveSpeed;
2044 playbackRateTemp.mPitch = effectivePitch;
2045 mProxy->setPlaybackRate(playbackRateTemp);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002046 mProxy->setMinimum(mNotificationFramesAct);
2047
Kuowei Lid4adbdb2020-08-13 14:44:25 +08002048 if (mDualMonoMode != AUDIO_DUAL_MONO_MODE_OFF) {
2049 setDualMonoMode_l(mDualMonoMode);
2050 }
2051 if (mAudioDescriptionMixLeveldB != -std::numeric_limits<float>::infinity()) {
2052 setAudioDescriptionMixLevel_l(mAudioDescriptionMixLeveldB);
2053 }
2054
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002055 mDeathNotifier = new DeathNotifier(this);
Marco Nelissenf8880202014-11-14 07:58:25 -08002056 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
Glenn Kastene3aa6592012-12-04 12:22:46 -08002057
Andy Hungb68f5eb2019-12-03 16:49:17 -08002058 // This is the first log sent from the AudioTrack client.
2059 // The creation of the audio track by AudioFlinger (in the code above)
2060 // is the first log of the AudioTrack and must be present before
2061 // any AudioTrack client logs will be accepted.
Andy Hungea840382020-05-05 21:50:17 -07002062
Andy Hungb68f5eb2019-12-03 16:49:17 -08002063 mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(mPortId);
2064 mediametrics::LogItem(mMetricsId)
2065 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE)
2066 // the following are immutable
Andy Hunga629bd12020-06-05 16:03:53 -07002067 .set(AMEDIAMETRICS_PROP_FLAGS, toString(mFlags).c_str())
2068 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -08002069 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
Andy Hung3a5c2f32021-02-17 15:06:42 -08002070 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, mLogSessionId)
Andy Hung839a3062021-02-17 11:15:16 -08002071 .set(AMEDIAMETRICS_PROP_PLAYERIID, mPlayerIId)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002072 .set(AMEDIAMETRICS_PROP_TRACKID, mPortId) // dup from key
Andy Hungb68f5eb2019-12-03 16:49:17 -08002073 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2074 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2075 .set(AMEDIAMETRICS_PROP_THREADID, (int32_t)output.outputId)
2076 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2077 .set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)mRoutedDeviceId)
2078 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2079 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
2080 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
2081 // the following are NOT immutable
2082 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)mVolume[AUDIO_INTERLEAVE_LEFT])
2083 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)mVolume[AUDIO_INTERLEAVE_RIGHT])
2084 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Andy Hung73dc2f92021-12-07 21:50:04 -08002085 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)NO_ERROR)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002086 .set(AMEDIAMETRICS_PROP_AUXEFFECTID, (int32_t)mAuxEffectId)
2087 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2088 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2089 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2090 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2091 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveSampleRate)
2092 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2093 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)effectiveSpeed)
2094 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2095 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)effectivePitch)
2096 .record();
2097
2098 // mSendLevel
2099 // mReqFrameCount?
2100 // mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq
2101 // mLatency, mAfLatency, mAfFrameCount, mAfSampleRate
2102
Glenn Kasten38e905b2014-01-13 10:21:48 -08002103 }
2104
Eric Laurentf32d7812017-11-30 14:44:07 -08002105exit:
Andy Hung3acde2c2021-11-11 09:18:08 -08002106 if (status != NO_ERROR) {
2107 if (callbackAdded) {
2108 // note: mOutput is always valid is callbackAdded is true
2109 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
2110 }
2111 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
2112 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
Eric Laurentad2e7b92017-09-14 20:06:42 -07002113 }
Eric Laurentf32d7812017-11-30 14:44:07 -08002114 mStatus = status;
Eric Laurent21da6472017-11-09 16:29:26 -08002115
2116 // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
Glenn Kasten38e905b2014-01-13 10:21:48 -08002117 return status;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002118}
2119
Andy Hung3acde2c2021-11-11 09:18:08 -08002120void AudioTrack::reportError(status_t status, const char *event, const char *message) const
2121{
2122 if (status == NO_ERROR) return;
2123 // We report error on the native side because some callers do not come
2124 // from Java.
Andy Hungde602302021-12-07 21:35:49 -08002125 // Ensure these variables are initialized in set().
2126 mediametrics::LogItem(AMEDIAMETRICS_KEY_AUDIO_TRACK_ERROR)
Andy Hung3acde2c2021-11-11 09:18:08 -08002127 .set(AMEDIAMETRICS_PROP_EVENT, event)
Andy Hung73dc2f92021-12-07 21:50:04 -08002128 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
2129 .set(AMEDIAMETRICS_PROP_STATUSMESSAGE, message)
Andy Hung3acde2c2021-11-11 09:18:08 -08002130 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
2131 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
2132 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2133 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2134 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2135 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2136 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
Andy Hung3acde2c2021-11-11 09:18:08 -08002137 // the following are NOT immutable
Andy Hungde602302021-12-07 21:35:49 -08002138 // frame count is initially the requested frame count, but may be adjusted
2139 // by AudioFlinger after creation.
2140 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
Andy Hung3acde2c2021-11-11 09:18:08 -08002141 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2142 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2143 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2144 .record();
2145}
2146
Glenn Kastenb46f3942015-03-09 12:00:30 -07002147status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002148{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002149 if (audioBuffer == NULL) {
Glenn Kasten551b5352015-03-20 11:30:28 -07002150 if (nonContig != NULL) {
2151 *nonContig = 0;
2152 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002153 return BAD_VALUE;
Eric Laurent9b7d9502011-03-21 11:49:00 -07002154 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002155 if (mTransfer != TRANSFER_OBTAIN) {
2156 audioBuffer->frameCount = 0;
Atneya Nair03079272022-01-18 17:03:14 -05002157 audioBuffer->mSize = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002158 audioBuffer->raw = NULL;
Glenn Kasten551b5352015-03-20 11:30:28 -07002159 if (nonContig != NULL) {
2160 *nonContig = 0;
2161 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002162 return INVALID_OPERATION;
2163 }
Eric Laurent9b7d9502011-03-21 11:49:00 -07002164
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002165 const struct timespec *requested;
Eric Laurentdf576992014-01-27 18:13:39 -08002166 struct timespec timeout;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002167 if (waitCount == -1) {
2168 requested = &ClientProxy::kForever;
2169 } else if (waitCount == 0) {
2170 requested = &ClientProxy::kNonBlocking;
2171 } else if (waitCount > 0) {
Chih-Hung Hsiehbca74292018-08-10 16:06:07 -07002172 time_t ms = WAIT_PERIOD_MS * (time_t) waitCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002173 timeout.tv_sec = ms / 1000;
Andy Hung06a730b2020-04-09 13:28:31 -07002174 timeout.tv_nsec = (ms % 1000) * 1000000;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002175 requested = &timeout;
2176 } else {
Eric Laurent973db022018-11-20 14:54:31 -08002177 ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002178 requested = NULL;
2179 }
Glenn Kastenb46f3942015-03-09 12:00:30 -07002180 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002181}
Eric Laurent1703cdf2011-03-07 14:52:59 -08002182
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002183status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
2184 struct timespec *elapsed, size_t *nonContig)
2185{
2186 // previous and new IAudioTrack sequence numbers are used to detect track re-creation
2187 uint32_t oldSequence = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002188
2189 Proxy::Buffer buffer;
2190 status_t status = NO_ERROR;
2191
2192 static const int32_t kMaxTries = 5;
2193 int32_t tryCounter = kMaxTries;
2194
2195 do {
2196 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
2197 // keep them from going away if another thread re-creates the track during obtainBuffer()
2198 sp<AudioTrackClientProxy> proxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002199
2200 { // start of lock scope
2201 AutoMutex lock(mLock);
2202
Glenn Kasten305996c2020-01-27 08:03:37 -08002203 uint32_t newSequence = mSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002204 // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
2205 if (status == DEAD_OBJECT) {
2206 // re-create track, unless someone else has already done so
2207 if (newSequence == oldSequence) {
2208 status = restoreTrack_l("obtainBuffer");
2209 if (status != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002210 buffer.mFrameCount = 0;
2211 buffer.mRaw = NULL;
2212 buffer.mNonContig = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002213 break;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002214 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002215 }
2216 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002217 oldSequence = newSequence;
2218
Eric Laurent4d231dc2016-03-11 18:38:23 -08002219 if (status == NOT_ENOUGH_DATA) {
2220 restartIfDisabled();
2221 }
2222
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002223 // Keep the extra references
jiabind42567c2023-03-23 22:01:16 +00002224 mProxyObtainBufferRef = mProxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002225 proxy = mProxy;
jiabind42567c2023-03-23 22:01:16 +00002226 mCblkMemoryObtainBufferRef = mCblkMemory;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002227
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002228 if (mState == STATE_STOPPING) {
2229 status = -EINTR;
2230 buffer.mFrameCount = 0;
2231 buffer.mRaw = NULL;
2232 buffer.mNonContig = 0;
2233 break;
2234 }
2235
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002236 // Non-blocking if track is stopped or paused
2237 if (mState != STATE_ACTIVE) {
2238 requested = &ClientProxy::kNonBlocking;
2239 }
2240
2241 } // end of lock scope
2242
2243 buffer.mFrameCount = audioBuffer->frameCount;
2244 // FIXME starts the requested timeout and elapsed over from scratch
2245 status = proxy->obtainBuffer(&buffer, requested, elapsed);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002246 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002247
2248 audioBuffer->frameCount = buffer.mFrameCount;
Atneya Nair03079272022-01-18 17:03:14 -05002249 audioBuffer->mSize = buffer.mFrameCount * mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002250 audioBuffer->raw = buffer.mRaw;
Glenn Kasten305996c2020-01-27 08:03:37 -08002251 audioBuffer->sequence = oldSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002252 if (nonContig != NULL) {
2253 *nonContig = buffer.mNonContig;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002254 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002255 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002256}
2257
Glenn Kasten54a8a452015-03-09 12:03:00 -07002258void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002259{
Glenn Kasten3f02be22015-03-09 11:59:04 -07002260 // FIXME add error checking on mode, by adding an internal version
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002261 if (mTransfer == TRANSFER_SHARED) {
2262 return;
2263 }
2264
Atneya Nair03079272022-01-18 17:03:14 -05002265 size_t stepCount = audioBuffer->mSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002266 if (stepCount == 0) {
2267 return;
2268 }
2269
2270 Proxy::Buffer buffer;
2271 buffer.mFrameCount = stepCount;
2272 buffer.mRaw = audioBuffer->raw;
Glenn Kastene3aa6592012-12-04 12:22:46 -08002273
jiabind42567c2023-03-23 22:01:16 +00002274 sp<IMemory> tempMemory;
2275 sp<AudioTrackClientProxy> tempProxy;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002276 AutoMutex lock(mLock);
Glenn Kasten305996c2020-01-27 08:03:37 -08002277 if (audioBuffer->sequence != mSequence) {
2278 // This Buffer came from a different IAudioTrack instance, so ignore the releaseBuffer
2279 ALOGD("%s is no-op due to IAudioTrack sequence mismatch %u != %u",
2280 __func__, audioBuffer->sequence, mSequence);
2281 return;
2282 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002283 mReleased += stepCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002284 mInUnderrun = false;
jiabind42567c2023-03-23 22:01:16 +00002285 mProxyObtainBufferRef->releaseBuffer(&buffer);
2286 // The extra reference of shared memory and proxy from `obtainBuffer` is not used after
2287 // calling `releaseBuffer`. Move the extra reference to a temp strong pointer so that it
2288 // will be cleared outside `releaseBuffer`.
2289 tempMemory = std::move(mCblkMemoryObtainBufferRef);
2290 tempProxy = std::move(mProxyObtainBufferRef);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002291
2292 // restart track if it was disabled by audioflinger due to previous underrun
Eric Laurent4d231dc2016-03-11 18:38:23 -08002293 restartIfDisabled();
2294}
2295
2296void AudioTrack::restartIfDisabled()
2297{
2298 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
2299 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002300 ALOGW("%s(%d): releaseBuffer() track %p disabled due to previous underrun, restarting",
Eric Laurent973db022018-11-20 14:54:31 -08002301 __func__, mPortId, this);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002302 // FIXME ignoring status
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002303 status_t status;
2304 mAudioTrack->start(&status);
Eric Laurentdf839842012-05-31 14:27:14 -07002305 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002306}
2307
2308// -------------------------------------------------------------------------
2309
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002310ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002311{
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002312 if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07002313 return INVALID_OPERATION;
2314 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002315
Eric Laurentab5cdba2014-06-09 17:22:27 -07002316 if (isDirect()) {
2317 AutoMutex lock(mLock);
2318 int32_t flags = android_atomic_and(
2319 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
2320 &mCblk->mFlags);
2321 if (flags & CBLK_INVALID) {
2322 return DEAD_OBJECT;
2323 }
2324 }
2325
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002326 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
Jiabin Huang447cea72020-07-28 22:35:18 +00002327 // Validation: user is most-likely passing an error code, and it would
Glenn Kasten99e53b82012-01-19 08:59:58 -08002328 // make the return value ambiguous (actualSize vs error).
Andy Hungfb8ede22018-09-12 19:03:24 -07002329 ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
Eric Laurent973db022018-11-20 14:54:31 -08002330 __func__, mPortId, buffer, userSize, userSize);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002331 return BAD_VALUE;
2332 }
2333
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002334 size_t written = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002335 Buffer audioBuffer;
2336
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002337 while (userSize >= mFrameSize) {
2338 audioBuffer.frameCount = userSize / mFrameSize;
Eric Laurentc2f1f072009-07-17 12:17:14 -07002339
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002340 status_t err = obtainBuffer(&audioBuffer,
2341 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002342 if (err < 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002343 if (written > 0) {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002344 break;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002345 }
Glenn Kasten0a2f1512016-07-22 08:06:37 -07002346 if (err == TIMED_OUT || err == -EINTR) {
2347 err = WOULD_BLOCK;
2348 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002349 return ssize_t(err);
2350 }
2351
Atneya Nair03079272022-01-18 17:03:14 -05002352 size_t toWrite = audioBuffer.size();
2353 memcpy(audioBuffer.raw, buffer, toWrite);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002354 buffer = ((const char *) buffer) + toWrite;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002355 userSize -= toWrite;
2356 written += toWrite;
2357
2358 releaseBuffer(&audioBuffer);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002359 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002360
Andy Hungea2b9c02016-02-12 17:06:53 -08002361 if (written > 0) {
2362 mFramesWritten += written / mFrameSize;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002363
2364 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2365 const sp<AudioTrackThread> t = mAudioTrackThread;
2366 if (t != 0) {
2367 // causes wake up of the playback thread, that will callback the client for
2368 // more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
2369 t->wake();
2370 }
2371 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002372 }
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002373
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002374 return written;
2375}
2376
2377// -------------------------------------------------------------------------
2378
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08002379nsecs_t AudioTrack::processAudioBuffer()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002380{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002381 // Currently the AudioTrack thread is not created if there are no callbacks.
2382 // Would it ever make sense to run the thread, even without callbacks?
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002383 // If so, then replace this by checks at each use for mCallback != NULL.
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002384 LOG_ALWAYS_FATAL_IF(mCblk == NULL);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002385 mLock.lock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002386 sp<IAudioTrackCallback> callback = mCallback.promote();
2387 if (!callback) {
2388 mCallback = nullptr;
Atneya Naire260f5a2022-05-03 17:02:20 -04002389 mLock.unlock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002390 return NS_NEVER;
2391 }
Glenn Kastena07f17c2013-04-23 12:39:37 -07002392 if (mAwaitBoost) {
2393 mAwaitBoost = false;
2394 mLock.unlock();
2395 static const int32_t kMaxTries = 5;
2396 int32_t tryCounter = kMaxTries;
2397 uint32_t pollUs = 10000;
2398 do {
Glenn Kasten8255ba72016-08-23 13:54:23 -07002399 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002400 if (policy == SCHED_FIFO || policy == SCHED_RR) {
2401 break;
2402 }
2403 usleep(pollUs);
2404 pollUs <<= 1;
2405 } while (tryCounter-- > 0);
2406 if (tryCounter < 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002407 ALOGE("%s(%d): did not receive expected priority boost on time",
Eric Laurent973db022018-11-20 14:54:31 -08002408 __func__, mPortId);
Glenn Kastena07f17c2013-04-23 12:39:37 -07002409 }
Glenn Kastenb0dfd462013-07-10 16:52:47 -07002410 // Run again immediately
2411 return 0;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002412 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002413
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002414 // Can only reference mCblk while locked
2415 int32_t flags = android_atomic_and(
Glenn Kasten96f60d82013-07-12 10:21:18 -07002416 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);
Glenn Kastena47f3162012-11-07 10:13:08 -08002417
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
Eric Laurentab5cdba2014-06-09 17:22:27 -07002423 if (!isOffloadedOrDirect_l() || (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
Eric Laurentab5cdba2014-06-09 17:22:27 -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 ||
2681 (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;
Phil Burkfdb3c072016-02-09 10:47:02 -08002767 if (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
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002839status_t AudioTrack::restoreTrack_l(const char *from)
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
Ronghua Wufaeb0f22015-05-21 12:20:21 -07002860 if (isOffloadedOrDirect_l() || mDoNotReconnect) {
Andy Hung1f1db832015-06-08 13:26:10 -07002861 // FIXME re-creation of offloaded and direct tracks is not yet implemented;
2862 // reconsider enabling for linear PCM encodings when position can be preserved.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002863 result = DEAD_OBJECT;
2864 return result;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002865 }
2866
Phil Burk2812d9e2016-01-04 10:34:30 -08002867 // Save so we can return count since creation.
2868 mUnderrunCountOffset = getUnderrunCount_l();
2869
Glenn Kasten200092b2014-08-15 15:13:30 -07002870 // save the old static buffer position
Andy Hungf20a4e92016-08-15 19:10:34 -07002871 uint32_t staticPosition = 0;
Andy Hung4ede21d2014-12-12 15:37:34 -08002872 size_t bufferPosition = 0;
2873 int loopCount = 0;
2874 if (mStaticProxy != 0) {
2875 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
Andy Hungf20a4e92016-08-15 19:10:34 -07002876 staticPosition = mStaticProxy->getPosition().unsignedValue();
Andy Hung4ede21d2014-12-12 15:37:34 -08002877 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002878
Andy Hung3c7f47a2021-03-16 17:30:09 -07002879 // save the old startThreshold and framecount
2880 const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
2881 const uint32_t originalFrameCount = mProxy->frameCount();
2882
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002883 // See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
2884 // causes a lot of churn on the service side, and it can reject starting
2885 // playback of a previously created track. May also apply to other cases.
2886 const int INITIAL_RETRIES = 3;
2887 int retries = INITIAL_RETRIES;
2888retry:
2889 if (retries < INITIAL_RETRIES) {
2890 // See the comment for clearAudioConfigCache at the start of the function.
2891 AudioSystem::clearAudioConfigCache();
2892 }
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08002893 mFlags = mOrigFlags;
2894
Glenn Kasten200092b2014-08-15 15:13:30 -07002895 // If a new IAudioTrack is successfully created, createTrack_l() will modify the
Glenn Kastena47f3162012-11-07 10:13:08 -08002896 // following member variables: mAudioTrack, mCblkMemory and mCblk.
Glenn Kasten200092b2014-08-15 15:13:30 -07002897 // It will also delete the strong references on previous IAudioTrack and IMemory.
2898 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002899 result = createTrack_l();
Eric Laurentcc21e4f2013-10-16 15:12:32 -07002900
Eric Laurent6ec546d2018-10-10 16:52:14 -07002901 if (result == NO_ERROR) {
Andy Hungd7bd69e2015-07-24 07:52:41 -07002902 // take the frames that will be lost by track recreation into account in saved position
2903 // For streaming tracks, this is the amount we obtained from the user/client
2904 // (not the number actually consumed at the server - those are already lost).
2905 if (mStaticProxy == 0) {
2906 mPosition = mReleased;
2907 }
Andy Hung4ede21d2014-12-12 15:37:34 -08002908 // Continue playback from last known position and restore loop.
2909 if (mStaticProxy != 0) {
2910 if (loopCount != 0) {
2911 mStaticProxy->setBufferPositionAndLoop(bufferPosition,
2912 mLoopStart, mLoopEnd, loopCount);
2913 } else {
2914 mStaticProxy->setBufferPosition(bufferPosition);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002915 if (bufferPosition == mFrameCount) {
Eric Laurent973db022018-11-20 14:54:31 -08002916 ALOGD("%s(%d): restoring track at end of static buffer", __func__, mPortId);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002917 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002918 }
2919 }
Andy Hung4ef88d72017-02-21 19:47:53 -08002920 // restore volume handler
Andy Hung39399b62017-04-21 15:07:45 -07002921 mVolumeHandler->forall([this](const VolumeShaper &shaper) -> VolumeShaper::Status {
2922 sp<VolumeShaper::Operation> operationToEnd =
2923 new VolumeShaper::Operation(shaper.mOperation);
Andy Hung4ef88d72017-02-21 19:47:53 -08002924 // TODO: Ideally we would restore to the exact xOffset position
2925 // as returned by getVolumeShaperState(), but we don't have that
2926 // information when restoring at the client unless we periodically poll
2927 // the server or create shared memory state.
2928 //
Andy Hung39399b62017-04-21 15:07:45 -07002929 // For now, we simply advance to the end of the VolumeShaper effect
2930 // if it has been started.
2931 if (shaper.isStarted()) {
Andy Hungf3702642017-05-05 17:33:32 -07002932 operationToEnd->setNormalizedTime(1.f);
Andy Hung39399b62017-04-21 15:07:45 -07002933 }
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002934 media::VolumeShaperConfiguration config;
2935 shaper.mConfiguration->writeToParcelable(&config);
2936 media::VolumeShaperOperation operation;
2937 operationToEnd->writeToParcelable(&operation);
2938 status_t status;
2939 mAudioTrack->applyVolumeShaper(config, operation, &status);
2940 return status;
Andy Hung4ef88d72017-02-21 19:47:53 -08002941 });
2942
Andy Hung3c7f47a2021-03-16 17:30:09 -07002943 // restore the original start threshold if different than frameCount.
2944 if (originalStartThresholdInFrames != originalFrameCount) {
2945 // Note: mProxy->setStartThresholdInFrames() call is in the Proxy
2946 // and does not trigger a restart.
2947 // (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
2948 // Any start would be triggered on the mState == ACTIVE check below.
2949 const uint32_t currentThreshold =
2950 mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
2951 ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
2952 "%s(%d) startThresholdInFrames changing from %u to %u",
2953 __func__, mPortId, originalStartThresholdInFrames, currentThreshold);
2954 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002955 if (mState == STATE_ACTIVE) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002956 mAudioTrack->start(&result);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002957 }
Andy Hungf20a4e92016-08-15 19:10:34 -07002958 // server resets to zero so we offset
2959 mFramesWrittenServerOffset =
2960 mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten;
2961 mFramesWrittenAtRestore = mFramesWrittenServerOffset;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002962 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002963 if (result != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08002964 ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002965 if (--retries > 0) {
Eric Laurent6ec546d2018-10-10 16:52:14 -07002966 // leave time for an eventual race condition to clear before retrying
2967 usleep(500000);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002968 goto retry;
2969 }
Eric Laurent6ec546d2018-10-10 16:52:14 -07002970 // if no retries left, set invalid bit to force restoring at next occasion
2971 // and avoid inconsistent active state on client and server sides
2972 if (mCblk != nullptr) {
2973 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
2974 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002975 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002976 return result;
2977}
2978
Andy Hung90e8a972015-11-09 16:42:40 -08002979Modulo<uint32_t> AudioTrack::updateAndGetPosition_l()
Glenn Kasten200092b2014-08-15 15:13:30 -07002980{
2981 // This is the sole place to read server consumed frames
Andy Hung90e8a972015-11-09 16:42:40 -08002982 Modulo<uint32_t> newServer(mProxy->getPosition());
2983 const int32_t delta = (newServer - mServer).signedValue();
Glenn Kasten200092b2014-08-15 15:13:30 -07002984 // TODO There is controversy about whether there can be "negative jitter" in server position.
2985 // This should be investigated further, and if possible, it should be addressed.
2986 // A more definite failure mode is infrequent polling by client.
2987 // One could call (void)getPosition_l() in releaseBuffer(),
2988 // so mReleased and mPosition are always lock-step as best possible.
2989 // That should ensure delta never goes negative for infrequent polling
2990 // unless the server has more than 2^31 frames in its buffer,
2991 // in which case the use of uint32_t for these counters has bigger issues.
Andy Hung90e8a972015-11-09 16:42:40 -08002992 ALOGE_IF(delta < 0,
Andy Hungfb8ede22018-09-12 19:03:24 -07002993 "%s(%d): detected illegal retrograde motion by the server: mServer advanced by %d",
Eric Laurent973db022018-11-20 14:54:31 -08002994 __func__, mPortId, delta);
Chad Brubaker039c27a2015-09-23 15:17:29 -07002995 mServer = newServer;
Andy Hung90e8a972015-11-09 16:42:40 -08002996 if (delta > 0) { // avoid retrograde
2997 mPosition += delta;
2998 }
2999 return mPosition;
Glenn Kasten200092b2014-08-15 15:13:30 -07003000}
3001
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003002bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed)
Andy Hung8edb8dc2015-03-26 19:13:55 -07003003{
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003004 updateLatency_l();
Andy Hung8edb8dc2015-03-26 19:13:55 -07003005 // applicable for mixing tracks only (not offloaded or direct)
3006 if (mStaticProxy != 0) {
3007 return true; // static tracks do not have issues with buffer sizing.
3008 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07003009 const size_t minFrameCount =
Eric Laurent21da6472017-11-09 16:29:26 -08003010 AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
3011 sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003012 const bool allowed = mFrameCount >= minFrameCount;
3013 ALOGD_IF(!allowed,
Andy Hungfb8ede22018-09-12 19:03:24 -07003014 "%s(%d): denied "
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003015 "mAfLatency:%u mAfFrameCount:%zu mAfSampleRate:%u sampleRate:%u speed:%f "
3016 "mFrameCount:%zu < minFrameCount:%zu",
Eric Laurent973db022018-11-20 14:54:31 -08003017 __func__, mPortId,
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003018 mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed,
Andy Hung8edb8dc2015-03-26 19:13:55 -07003019 mFrameCount, minFrameCount);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003020 return allowed;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003021}
3022
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003023status_t AudioTrack::setParameters(const String8& keyValuePairs)
3024{
3025 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003026 status_t status;
3027 mAudioTrack->setParameters(keyValuePairs.c_str(), &status);
3028 return status;
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003029}
3030
Dean Wheatleya70eef72018-01-04 14:23:50 +11003031status_t AudioTrack::selectPresentation(int presentationId, int programId)
3032{
3033 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08003034 AudioParameter param = AudioParameter();
3035 param.addInt(String8(AudioParameter::keyPresentationId), presentationId);
3036 param.addInt(String8(AudioParameter::keyProgramId), programId);
3037 ALOGV("%s(%d): PresentationId/ProgramId[%s]",
3038 __func__, mPortId, param.toString().string());
3039
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003040 status_t status;
3041 mAudioTrack->setParameters(param.toString().c_str(), &status);
3042 return status;
Dean Wheatleya70eef72018-01-04 14:23:50 +11003043}
3044
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003045VolumeShaper::Status AudioTrack::applyVolumeShaper(
3046 const sp<VolumeShaper::Configuration>& configuration,
3047 const sp<VolumeShaper::Operation>& operation)
3048{
3049 AutoMutex lock(mLock);
Andy Hung4ef88d72017-02-21 19:47:53 -08003050 mVolumeHandler->setIdIfNecessary(configuration);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003051 media::VolumeShaperConfiguration config;
3052 configuration->writeToParcelable(&config);
3053 media::VolumeShaperOperation op;
3054 operation->writeToParcelable(&op);
3055 VolumeShaper::Status status;
3056 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003057
3058 if (status == DEAD_OBJECT) {
3059 if (restoreTrack_l("applyVolumeShaper") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003060 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003061 }
3062 }
Andy Hung4ef88d72017-02-21 19:47:53 -08003063 if (status >= 0) {
3064 // save VolumeShaper for restore
3065 mVolumeHandler->applyVolumeShaper(configuration, operation);
Andy Hung39399b62017-04-21 15:07:45 -07003066 if (mState == STATE_ACTIVE || mState == STATE_STOPPING) {
3067 mVolumeHandler->setStarted();
3068 }
3069 } else {
3070 // warn only if not an expected restore failure.
3071 ALOGW_IF(!((isOffloadedOrDirect_l() || mDoNotReconnect) && status == DEAD_OBJECT),
Eric Laurent973db022018-11-20 14:54:31 -08003072 "%s(%d): applyVolumeShaper failed: %d", __func__, mPortId, status);
Andy Hung4ef88d72017-02-21 19:47:53 -08003073 }
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003074 return status;
3075}
3076
3077sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id)
3078{
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003079 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003080 std::optional<media::VolumeShaperState> vss;
3081 mAudioTrack->getVolumeShaperState(id, &vss);
3082 sp<VolumeShaper::State> state;
3083 if (vss.has_value()) {
3084 state = new VolumeShaper::State();
3085 state->readFromParcelable(vss.value());
3086 }
Andy Hung39399b62017-04-21 15:07:45 -07003087 if (state.get() == nullptr && (mCblk->mFlags & CBLK_INVALID) != 0) {
3088 if (restoreTrack_l("getVolumeShaperState") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003089 mAudioTrack->getVolumeShaperState(id, &vss);
3090 if (vss.has_value()) {
3091 state = new VolumeShaper::State();
3092 state->readFromParcelable(vss.value());
3093 }
Andy Hung39399b62017-04-21 15:07:45 -07003094 }
3095 }
3096 return state;
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003097}
3098
Andy Hungea2b9c02016-02-12 17:06:53 -08003099status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp)
3100{
3101 if (timestamp == nullptr) {
3102 return BAD_VALUE;
3103 }
3104 AutoMutex lock(mLock);
Andy Hunge13f8a62016-03-30 14:20:42 -07003105 return getTimestamp_l(timestamp);
3106}
3107
3108status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp)
3109{
Andy Hungea2b9c02016-02-12 17:06:53 -08003110 if (mCblk->mFlags & CBLK_INVALID) {
3111 const status_t status = restoreTrack_l("getTimestampExtended");
3112 if (status != OK) {
3113 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3114 // recommending that the track be recreated.
3115 return DEAD_OBJECT;
3116 }
3117 }
3118 // check for offloaded/direct here in case restoring somehow changed those flags.
3119 if (isOffloadedOrDirect_l()) {
3120 return INVALID_OPERATION; // not supported
3121 }
3122 status_t status = mProxy->getTimestamp(timestamp);
Andy Hungfb8ede22018-09-12 19:03:24 -07003123 LOG_ALWAYS_FATAL_IF(status != OK, "%s(%d): status %d not allowed from proxy getTimestamp",
Eric Laurent973db022018-11-20 14:54:31 -08003124 __func__, mPortId, status);
Andy Hungea2b9c02016-02-12 17:06:53 -08003125 bool found = false;
Andy Hunge1e98462016-04-12 10:18:51 -07003126 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten;
3127 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
3128 // server side frame offset in case AudioTrack has been restored.
3129 for (int i = ExtendedTimestamp::LOCATION_SERVER;
3130 i < ExtendedTimestamp::LOCATION_MAX; ++i) {
3131 if (timestamp->mTimeNs[i] >= 0) {
3132 // apply server offset (frames flushed is ignored
3133 // so we don't report the jump when the flush occurs).
3134 timestamp->mPosition[i] += mFramesWrittenServerOffset;
3135 found = true;
Andy Hungea2b9c02016-02-12 17:06:53 -08003136 }
3137 }
3138 return found ? OK : WOULD_BLOCK;
3139}
3140
Glenn Kastence703742013-07-19 16:33:58 -07003141status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
3142{
Glenn Kasten53cec222013-08-29 09:01:02 -07003143 AutoMutex lock(mLock);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003144 return getTimestamp_l(timestamp);
3145}
Phil Burk1b420972015-04-22 10:52:21 -07003146
Andy Hung65ffdfc2016-10-10 15:52:11 -07003147status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp)
3148{
Phil Burk1b420972015-04-22 10:52:21 -07003149 bool previousTimestampValid = mPreviousTimestampValid;
3150 // Set false here to cover all the error return cases.
3151 mPreviousTimestampValid = false;
3152
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003153 switch (mState) {
3154 case STATE_ACTIVE:
3155 case STATE_PAUSED:
3156 break; // handle below
3157 case STATE_FLUSHED:
3158 case STATE_STOPPED:
3159 return WOULD_BLOCK;
3160 case STATE_STOPPING:
3161 case STATE_PAUSED_STOPPING:
3162 if (!isOffloaded_l()) {
3163 return INVALID_OPERATION;
3164 }
3165 break; // offloaded tracks handled below
3166 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003167 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in getTimestamp(): %d",
Eric Laurent973db022018-11-20 14:54:31 -08003168 __func__, mPortId, mState);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003169 break;
Glenn Kastenfe346c72013-08-30 13:28:22 -07003170 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003171
Eric Laurent275e8e92014-11-30 15:14:47 -08003172 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung6653c932015-06-08 13:27:48 -07003173 const status_t status = restoreTrack_l("getTimestamp");
3174 if (status != OK) {
3175 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3176 // recommending that the track be recreated.
3177 return DEAD_OBJECT;
3178 }
Eric Laurent275e8e92014-11-30 15:14:47 -08003179 }
3180
Glenn Kasten200092b2014-08-15 15:13:30 -07003181 // The presented frame count must always lag behind the consumed frame count.
3182 // To avoid a race, read the presented frames first. This ensures that presented <= consumed.
Andy Hung6ae58432016-02-16 18:32:24 -08003183
3184 status_t status;
Andy Hung818e7a32016-02-16 18:08:07 -08003185 if (isOffloadedOrDirect_l()) {
Andy Hung6ae58432016-02-16 18:32:24 -08003186 // use Binder to get timestamp
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003187 media::AudioTimestampInternal ts;
3188 mAudioTrack->getTimestamp(&ts, &status);
3189 if (status == OK) {
Andy Hung973638a2020-12-08 20:47:45 -08003190 timestamp = VALUE_OR_FATAL(aidl2legacy_AudioTimestampInternal_AudioTimestamp(ts));
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003191 }
Andy Hung6ae58432016-02-16 18:32:24 -08003192 } else {
3193 // read timestamp from shared memory
3194 ExtendedTimestamp ets;
3195 status = mProxy->getTimestamp(&ets);
3196 if (status == OK) {
Andy Hungb01faa32016-04-27 12:51:32 -07003197 ExtendedTimestamp::Location location;
3198 status = ets.getBestTimestamp(&timestamp, &location);
3199
3200 if (status == OK) {
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003201 updateLatency_l();
Andy Hungb01faa32016-04-27 12:51:32 -07003202 // It is possible that the best location has moved from the kernel to the server.
3203 // In this case we adjust the position from the previous computed latency.
3204 if (location == ExtendedTimestamp::LOCATION_SERVER) {
3205 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL,
Andy Hungfb8ede22018-09-12 19:03:24 -07003206 "%s(%d): location moved from kernel to server",
Eric Laurent973db022018-11-20 14:54:31 -08003207 __func__, mPortId);
Andy Hung07eee802016-06-21 16:47:16 -07003208 // check that the last kernel OK time info exists and the positions
3209 // are valid (if they predate the current track, the positions may
3210 // be zero or negative).
Andy Hungb01faa32016-04-27 12:51:32 -07003211 const int64_t frames =
Andy Hung6d7b1192016-05-07 22:59:48 -07003212 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
Andy Hung07eee802016-06-21 16:47:16 -07003213 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 ||
3214 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 ||
3215 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0)
Andy Hung6d7b1192016-05-07 22:59:48 -07003216 ?
3217 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed
3218 / 1000)
3219 :
3220 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3221 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]);
Andy Hungfb8ede22018-09-12 19:03:24 -07003222 ALOGV("%s(%d): frame adjustment:%lld timestamp:%s",
Eric Laurent973db022018-11-20 14:54:31 -08003223 __func__, mPortId, (long long)frames, ets.toString().c_str());
Andy Hungb01faa32016-04-27 12:51:32 -07003224 if (frames >= ets.mPosition[location]) {
3225 timestamp.mPosition = 0;
3226 } else {
3227 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames);
3228 }
Andy Hung69488c42016-05-16 18:43:33 -07003229 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) {
3230 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER,
Andy Hungfb8ede22018-09-12 19:03:24 -07003231 "%s(%d): location moved from server to kernel",
Eric Laurent973db022018-11-20 14:54:31 -08003232 __func__, mPortId);
Andy Hung98731a22019-04-08 19:19:07 -07003233
3234 if (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER] ==
3235 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL]) {
3236 // In Q, we don't return errors as an invalid time
3237 // but instead we leave the last kernel good timestamp alone.
3238 //
3239 // If server is identical to kernel, the device data pipeline is idle.
3240 // A better start time is now. The retrograde check ensures
3241 // timestamp monotonicity.
3242 const int64_t nowNs = systemTime();
Andy Hungcf3b7152019-04-19 18:29:21 -07003243 if (!mTimestampStallReported) {
3244 ALOGD("%s(%d): device stall time corrected using current time %lld",
3245 __func__, mPortId, (long long)nowNs);
3246 mTimestampStallReported = true;
3247 }
Andy Hung98731a22019-04-08 19:19:07 -07003248 timestamp.mTime = convertNsToTimespec(nowNs);
Andy Hungcf3b7152019-04-19 18:29:21 -07003249 } else {
3250 mTimestampStallReported = false;
Andy Hung98731a22019-04-08 19:19:07 -07003251 }
Andy Hungb01faa32016-04-27 12:51:32 -07003252 }
Andy Hung5d313802016-10-10 15:09:39 -07003253
3254 // We update the timestamp time even when paused.
3255 if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) {
3256 const int64_t now = systemTime();
Andy Hung2b01f002017-07-05 12:01:36 -07003257 const int64_t at = audio_utils_ns_from_timespec(&timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003258 const int64_t lag =
3259 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
3260 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0)
3261 ? int64_t(mAfLatency * 1000000LL)
3262 : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3263 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK])
3264 * NANOS_PER_SECOND / mSampleRate;
3265 const int64_t limit = now - lag; // no earlier than this limit
3266 if (at < limit) {
3267 ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld",
3268 (long long)lag, (long long)at, (long long)limit);
Andy Hungffa36952017-08-17 10:41:51 -07003269 timestamp.mTime = convertNsToTimespec(limit);
Andy Hung5d313802016-10-10 15:09:39 -07003270 }
3271 }
Andy Hungb01faa32016-04-27 12:51:32 -07003272 mPreviousLocation = location;
3273 } else {
3274 // right after AudioTrack is started, one may not find a timestamp
Eric Laurent973db022018-11-20 14:54:31 -08003275 ALOGV("%s(%d): getBestTimestamp did not find timestamp", __func__, mPortId);
Andy Hungb01faa32016-04-27 12:51:32 -07003276 }
Andy Hung6ae58432016-02-16 18:32:24 -08003277 }
3278 if (status == INVALID_OPERATION) {
Andy Hungf20a4e92016-08-15 19:10:34 -07003279 // INVALID_OPERATION occurs when no timestamp has been issued by the server;
3280 // other failures are signaled by a negative time.
3281 // If we come out of FLUSHED or STOPPED where the position is known
3282 // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of
3283 // "zero" for NuPlayer). We don't convert for track restoration as position
3284 // does not reset.
Andy Hungfb8ede22018-09-12 19:03:24 -07003285 ALOGV("%s(%d): timestamp server offset:%lld restore frames:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003286 __func__, mPortId,
Andy Hungf20a4e92016-08-15 19:10:34 -07003287 (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore);
3288 if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) {
3289 status = WOULD_BLOCK;
3290 }
Andy Hung6ae58432016-02-16 18:32:24 -08003291 }
3292 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003293 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08003294 ALOGV_IF(status != WOULD_BLOCK, "%s(%d): getTimestamp error:%#x", __func__, mPortId, status);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003295 return status;
3296 }
3297 if (isOffloadedOrDirect_l()) {
3298 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) {
3299 // use cached paused position in case another offloaded track is running.
3300 timestamp.mPosition = mPausedPosition;
3301 clock_gettime(CLOCK_MONOTONIC, &timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003302 // TODO: adjust for delay
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003303 return NO_ERROR;
3304 }
3305
3306 // Check whether a pending flush or stop has completed, as those commands may
Andy Hungc8e09c62015-06-03 23:43:36 -07003307 // be asynchronous or return near finish or exhibit glitchy behavior.
3308 //
3309 // Originally this showed up as the first timestamp being a continuation of
3310 // the previous song under gapless playback.
3311 // However, we sometimes see zero timestamps, then a glitch of
3312 // the previous song's position, and then correct timestamps afterwards.
Andy Hungffa36952017-08-17 10:41:51 -07003313 if (mStartFromZeroUs != 0 && mSampleRate != 0) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003314 static const int kTimeJitterUs = 100000; // 100 ms
3315 static const int k1SecUs = 1000000;
3316
3317 const int64_t timeNow = getNowUs();
3318
Andy Hungffa36952017-08-17 10:41:51 -07003319 if (timeNow < mStartFromZeroUs + k1SecUs) { // within first second of starting
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003320 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003321 if (timestampTimeUs < mStartFromZeroUs) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003322 return WOULD_BLOCK; // stale timestamp time, occurs before start.
3323 }
Andy Hungffa36952017-08-17 10:41:51 -07003324 const int64_t deltaTimeUs = timestampTimeUs - mStartFromZeroUs;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003325 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07003326 / ((double)mSampleRate * mPlaybackRate.mSpeed);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003327
3328 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
3329 // Verify that the counter can't count faster than the sample rate
Andy Hungc8e09c62015-06-03 23:43:36 -07003330 // since the start time. If greater, then that means we may have failed
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003331 // to completely flush or stop the previous playing track.
Andy Hungc8e09c62015-06-03 23:43:36 -07003332 ALOGW_IF(!mTimestampStartupGlitchReported,
Andy Hungfb8ede22018-09-12 19:03:24 -07003333 "%s(%d): startup glitch detected"
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003334 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)",
Eric Laurent973db022018-11-20 14:54:31 -08003335 __func__, mPortId,
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003336 (long long)deltaTimeUs, (long long)deltaPositionByUs,
3337 timestamp.mPosition);
Andy Hungc8e09c62015-06-03 23:43:36 -07003338 mTimestampStartupGlitchReported = true;
3339 if (previousTimestampValid
3340 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) {
3341 timestamp = mPreviousTimestamp;
3342 mPreviousTimestampValid = true;
3343 return NO_ERROR;
3344 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003345 return WOULD_BLOCK;
3346 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003347 if (deltaPositionByUs != 0) {
Andy Hungffa36952017-08-17 10:41:51 -07003348 mStartFromZeroUs = 0; // don't check again, we got valid nonzero position.
Andy Hungc8e09c62015-06-03 23:43:36 -07003349 }
3350 } else {
Andy Hungffa36952017-08-17 10:41:51 -07003351 mStartFromZeroUs = 0; // don't check again, start time expired.
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003352 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003353 mTimestampStartupGlitchReported = false;
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003354 }
3355 } else {
Glenn Kasten200092b2014-08-15 15:13:30 -07003356 // Update the mapping between local consumed (mPosition) and server consumed (mServer)
3357 (void) updateAndGetPosition_l();
3358 // Server consumed (mServer) and presented both use the same server time base,
3359 // and server consumed is always >= presented.
3360 // The delta between these represents the number of frames in the buffer pipeline.
3361 // If this delta between these is greater than the client position, it means that
3362 // actually presented is still stuck at the starting line (figuratively speaking),
3363 // waiting for the first frame to go by. So we can't report a valid timestamp yet.
Andy Hung90e8a972015-11-09 16:42:40 -08003364 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when
3365 // mPosition exceeds 32 bits.
3366 // TODO Remove when timestamp is updated to contain pipeline status info.
3367 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue();
3368 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */
3369 && (uint32_t)pipelineDepthInFrames > mPosition.value()) {
Glenn Kasten200092b2014-08-15 15:13:30 -07003370 return INVALID_OPERATION;
3371 }
3372 // Convert timestamp position from server time base to client time base.
3373 // TODO The following code should work OK now because timestamp.mPosition is 32-bit.
3374 // But if we change it to 64-bit then this could fail.
Andy Hung90e8a972015-11-09 16:42:40 -08003375 // Use Modulo computation here.
3376 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value();
Glenn Kasten200092b2014-08-15 15:13:30 -07003377 // Immediately after a call to getPosition_l(), mPosition and
3378 // mServer both represent the same frame position. mPosition is
3379 // in client's point of view, and mServer is in server's point of
3380 // view. So the difference between them is the "fudge factor"
3381 // between client and server views due to stop() and/or new
3382 // IAudioTrack. And timestamp.mPosition is initially in server's
3383 // point of view, so we need to apply the same fudge factor to it.
Glenn Kastenfe346c72013-08-30 13:28:22 -07003384 }
Phil Burk1b420972015-04-22 10:52:21 -07003385
3386 // Prevent retrograde motion in timestamp.
3387 // This is sometimes caused by erratic reports of the available space in the ALSA drivers.
3388 if (status == NO_ERROR) {
Andy Hung3b8c6332019-04-03 19:29:36 -07003389 // Fix stale time when checking timestamp right after start().
3390 // The position is at the last reported location but the time can be stale
3391 // due to pause or standby or cold start latency.
3392 //
3393 // We keep advancing the time (but not the position) to ensure that the
3394 // stale value does not confuse the application.
3395 //
3396 // For offload compatibility, use a default lag value here.
3397 // Any time discrepancy between this update and the pause timestamp is handled
3398 // by the retrograde check afterwards.
3399 int64_t currentTimeNanos = audio_utils_ns_from_timespec(&timestamp.mTime);
3400 const int64_t lagNs = int64_t(mAfLatency * 1000000LL);
3401 const int64_t limitNs = mStartNs - lagNs;
3402 if (currentTimeNanos < limitNs) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003403 if (!mTimestampStaleTimeReported) {
3404 ALOGD("%s(%d): stale timestamp time corrected, "
3405 "currentTimeNanos: %lld < limitNs: %lld < mStartNs: %lld",
3406 __func__, mPortId,
3407 (long long)currentTimeNanos, (long long)limitNs, (long long)mStartNs);
3408 mTimestampStaleTimeReported = true;
3409 }
Andy Hung3b8c6332019-04-03 19:29:36 -07003410 timestamp.mTime = convertNsToTimespec(limitNs);
3411 currentTimeNanos = limitNs;
Andy Hungcf3b7152019-04-19 18:29:21 -07003412 } else {
3413 mTimestampStaleTimeReported = false;
Andy Hung3b8c6332019-04-03 19:29:36 -07003414 }
3415
Andy Hungffa36952017-08-17 10:41:51 -07003416 // previousTimestampValid is set to false when starting after a stop or flush.
Phil Burk1b420972015-04-22 10:52:21 -07003417 if (previousTimestampValid) {
Andy Hung2b01f002017-07-05 12:01:36 -07003418 const int64_t previousTimeNanos =
3419 audio_utils_ns_from_timespec(&mPreviousTimestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003420
3421 // retrograde check
Phil Burk1b420972015-04-22 10:52:21 -07003422 if (currentTimeNanos < previousTimeNanos) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003423 if (!mTimestampRetrogradeTimeReported) {
3424 ALOGW("%s(%d): retrograde timestamp time corrected, %lld < %lld",
3425 __func__, mPortId,
3426 (long long)currentTimeNanos, (long long)previousTimeNanos);
3427 mTimestampRetrogradeTimeReported = true;
3428 }
Andy Hung5d313802016-10-10 15:09:39 -07003429 timestamp.mTime = mPreviousTimestamp.mTime;
Andy Hungcf3b7152019-04-19 18:29:21 -07003430 } else {
3431 mTimestampRetrogradeTimeReported = false;
Phil Burk1b420972015-04-22 10:52:21 -07003432 }
3433
3434 // Looking at signed delta will work even when the timestamps
3435 // are wrapping around.
Andy Hung90e8a972015-11-09 16:42:40 -08003436 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition)
3437 - mPreviousTimestamp.mPosition).signedValue();
Phil Burk4c5a3672015-04-30 16:18:53 -07003438 if (deltaPosition < 0) {
3439 // Only report once per position instead of spamming the log.
Andy Hungcf3b7152019-04-19 18:29:21 -07003440 if (!mTimestampRetrogradePositionReported) {
Andy Hungfb8ede22018-09-12 19:03:24 -07003441 ALOGW("%s(%d): retrograde timestamp position corrected, %d = %u - %u",
Eric Laurent973db022018-11-20 14:54:31 -08003442 __func__, mPortId,
Phil Burk4c5a3672015-04-30 16:18:53 -07003443 deltaPosition,
3444 timestamp.mPosition,
3445 mPreviousTimestamp.mPosition);
Andy Hungcf3b7152019-04-19 18:29:21 -07003446 mTimestampRetrogradePositionReported = true;
Phil Burk4c5a3672015-04-30 16:18:53 -07003447 }
3448 } else {
Andy Hungcf3b7152019-04-19 18:29:21 -07003449 mTimestampRetrogradePositionReported = false;
Phil Burk4c5a3672015-04-30 16:18:53 -07003450 }
Andy Hung5d313802016-10-10 15:09:39 -07003451 if (deltaPosition < 0) {
3452 timestamp.mPosition = mPreviousTimestamp.mPosition;
3453 deltaPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07003454 }
Andy Hung5d313802016-10-10 15:09:39 -07003455#if 0
3456 // Uncomment this to verify audio timestamp rate.
3457 const int64_t deltaTime =
Andy Hung2b01f002017-07-05 12:01:36 -07003458 audio_utils_ns_from_timespec(&timestamp.mTime) - previousTimeNanos;
Andy Hung5d313802016-10-10 15:09:39 -07003459 if (deltaTime != 0) {
3460 const int64_t computedSampleRate =
3461 deltaPosition * (long long)NANOS_PER_SECOND / deltaTime;
Andy Hungfb8ede22018-09-12 19:03:24 -07003462 ALOGD("%s(%d): computedSampleRate:%u sampleRate:%u",
Eric Laurent973db022018-11-20 14:54:31 -08003463 __func__, mPortId,
Andy Hung5d313802016-10-10 15:09:39 -07003464 (unsigned)computedSampleRate, mSampleRate);
3465 }
3466#endif
Phil Burk1b420972015-04-22 10:52:21 -07003467 }
3468 mPreviousTimestamp = timestamp;
3469 mPreviousTimestampValid = true;
3470 }
3471
Glenn Kastenfe346c72013-08-30 13:28:22 -07003472 return status;
Glenn Kastence703742013-07-19 16:33:58 -07003473}
3474
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003475String8 AudioTrack::getParameters(const String8& keys)
3476{
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003477 audio_io_handle_t output = getOutput();
Glenn Kasten142f5192014-03-25 17:44:59 -07003478 if (output != AUDIO_IO_HANDLE_NONE) {
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003479 return AudioSystem::getParameters(output, keys);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01003480 } else {
3481 return String8::empty();
3482 }
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003483}
3484
Glenn Kasten23a75452014-01-13 10:37:17 -08003485bool AudioTrack::isOffloaded() const
3486{
3487 AutoMutex lock(mLock);
3488 return isOffloaded_l();
3489}
3490
Eric Laurentab5cdba2014-06-09 17:22:27 -07003491bool AudioTrack::isDirect() const
3492{
3493 AutoMutex lock(mLock);
3494 return isDirect_l();
3495}
3496
3497bool AudioTrack::isOffloadedOrDirect() const
3498{
3499 AutoMutex lock(mLock);
3500 return isOffloadedOrDirect_l();
3501}
3502
3503
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003504status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003505{
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003506 String8 result;
3507
3508 result.append(" AudioTrack::dump\n");
Andy Hungfb8ede22018-09-12 19:03:24 -07003509 result.appendFormat(" id(%d) status(%d), state(%d), session Id(%d), flags(%#x)\n",
Eric Laurent973db022018-11-20 14:54:31 -08003510 mPortId, mStatus, mState, mSessionId, mFlags);
Eric Laurentd114b622017-11-27 18:37:04 -08003511 result.appendFormat(" stream type(%d), left - right volume(%f, %f)\n",
François Gaffie58d4be52018-11-06 15:30:12 +01003512 mStreamType,
Eric Laurentd114b622017-11-27 18:37:04 -08003513 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
Glenn Kasten49f36ba2017-12-06 13:02:02 -08003514 result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
Eric Laurentd114b622017-11-27 18:37:04 -08003515 mFormat, mChannelMask, mChannelCount);
3516 result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
3517 mSampleRate, mOriginalSampleRate, mPlaybackRate.mSpeed);
3518 result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
3519 mFrameCount, mReqFrameCount);
3520 result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
3521 " req. notif. per buff(%u)\n",
3522 mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
3523 result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
3524 mLatency, mSelectedDeviceId, mRoutedDeviceId);
3525 result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
3526 mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003527 ::write(fd, result.string(), result.size());
3528 return NO_ERROR;
3529}
3530
Phil Burk2812d9e2016-01-04 10:34:30 -08003531uint32_t AudioTrack::getUnderrunCount() const
3532{
3533 AutoMutex lock(mLock);
3534 return getUnderrunCount_l();
3535}
3536
3537uint32_t AudioTrack::getUnderrunCount_l() const
3538{
3539 return mProxy->getUnderrunCount() + mUnderrunCountOffset;
3540}
3541
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003542uint32_t AudioTrack::getUnderrunFrames() const
3543{
3544 AutoMutex lock(mLock);
3545 return mProxy->getUnderrunFrames();
3546}
3547
Andy Hung3a5c2f32021-02-17 15:06:42 -08003548void AudioTrack::setLogSessionId(const char *logSessionId)
3549{
3550 AutoMutex lock(mLock);
Andy Hung1a9c21b2021-02-25 20:43:18 -08003551 if (logSessionId == nullptr) logSessionId = ""; // an empty string is an unset session id.
Andy Hung3a5c2f32021-02-17 15:06:42 -08003552 if (mLogSessionId == logSessionId) return;
3553
3554 mLogSessionId = logSessionId;
3555 mediametrics::LogItem(mMetricsId)
3556 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETLOGSESSIONID)
3557 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, logSessionId)
3558 .record();
3559}
3560
Andy Hung839a3062021-02-17 11:15:16 -08003561void AudioTrack::setPlayerIId(int playerIId)
3562{
3563 AutoMutex lock(mLock);
3564 if (mPlayerIId == playerIId) return;
3565
3566 mPlayerIId = playerIId;
Vlad Popaad0fe922022-06-10 00:43:14 +02003567 triggerPortIdUpdate_l();
Andy Hung839a3062021-02-17 11:15:16 -08003568 mediametrics::LogItem(mMetricsId)
3569 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYERIID)
3570 .set(AMEDIAMETRICS_PROP_PLAYERIID, playerIId)
3571 .record();
3572}
3573
Vlad Popaad0fe922022-06-10 00:43:14 +02003574void AudioTrack::triggerPortIdUpdate_l() {
3575 if (mAudioManager == nullptr) {
3576 // use checkService() to avoid blocking if audio service is not up yet
3577 sp<IBinder> binder =
3578 defaultServiceManager()->checkService(String16(kAudioServiceName));
3579 if (binder == nullptr) {
3580 ALOGE("%s(%d): binding to audio service failed.",
3581 __func__,
3582 mPlayerIId);
3583 return;
3584 }
3585
3586 mAudioManager = interface_cast<IAudioManager>(binder);
3587 }
3588
3589 // first time when the track is created we do not have a valid piid
3590 if (mPlayerIId != PLAYER_PIID_INVALID) {
3591 mAudioManager->playerEvent(mPlayerIId, PLAYER_UPDATE_PORT_ID, mPortId);
3592 }
3593}
3594
Eric Laurent296fb132015-05-01 11:38:42 -07003595status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback)
3596{
Eric Laurent09f1ed22019-04-24 17:45:17 -07003597
Eric Laurent296fb132015-05-01 11:38:42 -07003598 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003599 ALOGW("%s(%d): adding NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003600 return BAD_VALUE;
3601 }
3602 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07003603 if (mDeviceCallback.unsafe_get() == callback.get()) {
Eric Laurent973db022018-11-20 14:54:31 -08003604 ALOGW("%s(%d): adding same callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003605 return INVALID_OPERATION;
3606 }
3607 status_t status = NO_ERROR;
3608 if (mOutput != AUDIO_IO_HANDLE_NONE) {
3609 if (mDeviceCallback != 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003610 ALOGW("%s(%d): callback already present!", __func__, mPortId);
Eric Laurent09f1ed22019-04-24 17:45:17 -07003611 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003612 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003613 status = AudioSystem::addAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003614 }
3615 mDeviceCallback = callback;
3616 return status;
3617}
3618
3619status_t AudioTrack::removeAudioDeviceCallback(
3620 const sp<AudioSystem::AudioDeviceCallback>& callback)
3621{
3622 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003623 ALOGW("%s(%d): removing NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003624 return BAD_VALUE;
3625 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003626 AutoMutex lock(mLock);
3627 if (mDeviceCallback.unsafe_get() != callback.get()) {
3628 ALOGW("%s removing different callback!", __FUNCTION__);
3629 return INVALID_OPERATION;
Eric Laurent296fb132015-05-01 11:38:42 -07003630 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003631 mDeviceCallback.clear();
Eric Laurent296fb132015-05-01 11:38:42 -07003632 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07003633 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003634 }
Eric Laurent296fb132015-05-01 11:38:42 -07003635 return NO_ERROR;
3636}
3637
Eric Laurentad2e7b92017-09-14 20:06:42 -07003638
3639void AudioTrack::onAudioDeviceUpdate(audio_io_handle_t audioIo,
3640 audio_port_handle_t deviceId)
3641{
3642 sp<AudioSystem::AudioDeviceCallback> callback;
3643 {
3644 AutoMutex lock(mLock);
3645 if (audioIo != mOutput) {
3646 return;
3647 }
3648 callback = mDeviceCallback.promote();
3649 // only update device if the track is active as route changes due to other use cases are
3650 // irrelevant for this client
3651 if (mState == STATE_ACTIVE) {
3652 mRoutedDeviceId = deviceId;
3653 }
3654 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003655
Eric Laurentad2e7b92017-09-14 20:06:42 -07003656 if (callback.get() != nullptr) {
3657 callback->onAudioDeviceUpdate(mOutput, mRoutedDeviceId);
3658 }
3659}
3660
Andy Hunge13f8a62016-03-30 14:20:42 -07003661status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location)
3662{
3663 if (msec == nullptr ||
3664 (location != ExtendedTimestamp::LOCATION_SERVER
3665 && location != ExtendedTimestamp::LOCATION_KERNEL)) {
3666 return BAD_VALUE;
3667 }
3668 AutoMutex lock(mLock);
3669 // inclusive of offloaded and direct tracks.
3670 //
3671 // It is possible, but not enabled, to allow duration computation for non-pcm
3672 // audio_has_proportional_frames() formats because currently they have
3673 // the drain rate equivalent to the pcm sample rate * framesize.
3674 if (!isPurePcmData_l()) {
3675 return INVALID_OPERATION;
3676 }
3677 ExtendedTimestamp ets;
3678 if (getTimestamp_l(&ets) == OK
3679 && ets.mTimeNs[location] > 0) {
3680 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
3681 - ets.mPosition[location];
3682 if (diff < 0) {
3683 *msec = 0;
3684 } else {
3685 // ms is the playback time by frames
3686 int64_t ms = (int64_t)((double)diff * 1000 /
3687 ((double)mSampleRate * mPlaybackRate.mSpeed));
3688 // clockdiff is the timestamp age (negative)
3689 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
3690 ets.mTimeNs[location]
3691 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
3692 - systemTime(SYSTEM_TIME_MONOTONIC);
3693
3694 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
3695 static const int NANOS_PER_MILLIS = 1000000;
3696 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
3697 }
3698 return NO_ERROR;
3699 }
3700 if (location != ExtendedTimestamp::LOCATION_SERVER) {
3701 return INVALID_OPERATION; // LOCATION_KERNEL is not available
3702 }
3703 // use server position directly (offloaded and direct arrive here)
3704 updateAndGetPosition_l();
3705 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue();
3706 *msec = (diff <= 0) ? 0
3707 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed));
3708 return NO_ERROR;
3709}
3710
Andy Hung65ffdfc2016-10-10 15:52:11 -07003711bool AudioTrack::hasStarted()
3712{
3713 AutoMutex lock(mLock);
3714 switch (mState) {
3715 case STATE_STOPPED:
3716 if (isOffloadedOrDirect_l()) {
3717 // check if we have started in the past to return true.
Andy Hungffa36952017-08-17 10:41:51 -07003718 return mStartFromZeroUs > 0;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003719 }
3720 // A normal audio track may still be draining, so
3721 // check if stream has ended. This covers fasttrack position
3722 // instability and start/stop without any data written.
3723 if (mProxy->getStreamEndDone()) {
3724 return true;
3725 }
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003726 FALLTHROUGH_INTENDED;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003727 case STATE_ACTIVE:
3728 case STATE_STOPPING:
3729 break;
3730 case STATE_PAUSED:
3731 case STATE_PAUSED_STOPPING:
3732 case STATE_FLUSHED:
3733 return false; // we're not active
3734 default:
Eric Laurent973db022018-11-20 14:54:31 -08003735 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in hasStarted(): %d", __func__, mPortId, mState);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003736 break;
3737 }
3738
3739 // wait indicates whether we need to wait for a timestamp.
3740 // This is conservatively figured - if we encounter an unexpected error
3741 // then we will not wait.
3742 bool wait = false;
3743 if (isOffloadedOrDirect_l()) {
3744 AudioTimestamp ts;
3745 status_t status = getTimestamp_l(ts);
3746 if (status == WOULD_BLOCK) {
3747 wait = true;
3748 } else if (status == OK) {
3749 wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition);
3750 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003751 ALOGV("%s(%d): hasStarted wait:%d ts:%u start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003752 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003753 (int)wait,
3754 ts.mPosition,
3755 (long long)mStartTs.mPosition);
3756 } else {
3757 int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG
3758 ExtendedTimestamp ets;
3759 status_t status = getTimestamp_l(&ets);
3760 if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets
3761 wait = true;
3762 } else if (status == OK) {
3763 for (location = ExtendedTimestamp::LOCATION_KERNEL;
3764 location >= ExtendedTimestamp::LOCATION_SERVER; --location) {
3765 if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) {
3766 continue;
3767 }
3768 wait = ets.mPosition[location] == 0
3769 || ets.mPosition[location] == mStartEts.mPosition[location];
3770 break;
3771 }
3772 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003773 ALOGV("%s(%d): hasStarted wait:%d ets:%lld start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003774 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003775 (int)wait,
3776 (long long)ets.mPosition[location],
3777 (long long)mStartEts.mPosition[location]);
3778 }
3779 return !wait;
3780}
3781
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003782// =========================================================================
3783
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003784void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003785{
3786 sp<AudioTrack> audioTrack = mAudioTrack.promote();
3787 if (audioTrack != 0) {
3788 AutoMutex lock(audioTrack->mLock);
3789 audioTrack->mProxy->binderDied();
3790 }
3791}
3792
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003793// =========================================================================
3794
Andy Hungca353672019-03-06 11:54:38 -08003795AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver)
Andy Hungeb46cf92019-03-06 10:13:38 -08003796 : Thread(true /* bCanCallJava */) // binder recursion on restoreTrack_l() may call Java.
3797 , mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
Glenn Kasten598de6c2013-10-16 17:02:13 -07003798 mIgnoreNextPausedInt(false)
Glenn Kasten3acbd052012-02-28 10:39:56 -08003799{
3800}
3801
3802AudioTrack::AudioTrackThread::~AudioTrackThread()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003803{
3804}
3805
3806bool AudioTrack::AudioTrackThread::threadLoop()
3807{
Glenn Kasten3acbd052012-02-28 10:39:56 -08003808 {
3809 AutoMutex _l(mMyLock);
3810 if (mPaused) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003811 // TODO check return value and handle or log
Glenn Kasten3acbd052012-02-28 10:39:56 -08003812 mMyCond.wait(mMyLock);
3813 // caller will check for exitPending()
3814 return true;
3815 }
Glenn Kasten598de6c2013-10-16 17:02:13 -07003816 if (mIgnoreNextPausedInt) {
3817 mIgnoreNextPausedInt = false;
3818 mPausedInt = false;
3819 }
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003820 if (mPausedInt) {
Glenn Kasten47d55172017-05-23 11:19:30 -07003821 // TODO use futex instead of condition, for event flag "or"
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003822 if (mPausedNs > 0) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003823 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003824 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
3825 } else {
Glenn Kasten75f79032017-05-23 11:22:44 -07003826 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003827 mMyCond.wait(mMyLock);
3828 }
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003829 mPausedInt = false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003830 return true;
3831 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08003832 }
Eric Laurent7985dcb2014-10-07 15:45:14 -07003833 if (exitPending()) {
3834 return false;
3835 }
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003836 nsecs_t ns = mReceiver.processAudioBuffer();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003837 switch (ns) {
3838 case 0:
3839 return true;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003840 case NS_INACTIVE:
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003841 pauseInternal();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003842 return true;
3843 case NS_NEVER:
3844 return false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003845 case NS_WHENEVER:
Andy Hung3c09c782014-12-29 18:39:32 -08003846 // Event driven: call wake() when callback notifications conditions change.
3847 ns = INT64_MAX;
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003848 FALLTHROUGH_INTENDED;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003849 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003850 LOG_ALWAYS_FATAL_IF(ns < 0, "%s(%d): processAudioBuffer() returned %lld",
Eric Laurent973db022018-11-20 14:54:31 -08003851 __func__, mReceiver.mPortId, (long long)ns);
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003852 pauseInternal(ns);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003853 return true;
Glenn Kastenca8b2802012-04-23 13:58:16 -07003854 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003855}
3856
Glenn Kasten3acbd052012-02-28 10:39:56 -08003857void AudioTrack::AudioTrackThread::requestExit()
3858{
3859 // must be in this order to avoid a race condition
3860 Thread::requestExit();
Glenn Kasten598de6c2013-10-16 17:02:13 -07003861 resume();
Glenn Kasten3acbd052012-02-28 10:39:56 -08003862}
3863
3864void AudioTrack::AudioTrackThread::pause()
3865{
3866 AutoMutex _l(mMyLock);
3867 mPaused = true;
3868}
3869
3870void AudioTrack::AudioTrackThread::resume()
3871{
3872 AutoMutex _l(mMyLock);
Glenn Kasten598de6c2013-10-16 17:02:13 -07003873 mIgnoreNextPausedInt = true;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003874 if (mPaused || mPausedInt) {
Glenn Kasten3acbd052012-02-28 10:39:56 -08003875 mPaused = false;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003876 mPausedInt = false;
Glenn Kasten3acbd052012-02-28 10:39:56 -08003877 mMyCond.signal();
3878 }
3879}
3880
Andy Hung3c09c782014-12-29 18:39:32 -08003881void AudioTrack::AudioTrackThread::wake()
3882{
3883 AutoMutex _l(mMyLock);
Andy Hunga8d08902015-07-22 11:52:13 -07003884 if (!mPaused) {
3885 // wake() might be called while servicing a callback - ignore the next
3886 // pause time and call processAudioBuffer.
Andy Hung3c09c782014-12-29 18:39:32 -08003887 mIgnoreNextPausedInt = true;
Andy Hunga8d08902015-07-22 11:52:13 -07003888 if (mPausedInt && mPausedNs > 0) {
3889 // audio track is active and internally paused with timeout.
3890 mPausedInt = false;
3891 mMyCond.signal();
3892 }
Andy Hung3c09c782014-12-29 18:39:32 -08003893 }
3894}
3895
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003896void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
3897{
3898 AutoMutex _l(mMyLock);
3899 mPausedInt = true;
3900 mPausedNs = ns;
3901}
3902
jiabinf6eb4c32020-02-25 14:06:25 -08003903binder::Status AudioTrack::AudioTrackCallback::onCodecFormatChanged(
3904 const std::vector<uint8_t>& audioMetadata)
3905{
3906 AutoMutex _l(mAudioTrackCbLock);
3907 sp<media::IAudioTrackCallback> callback = mCallback.promote();
3908 if (callback.get() != nullptr) {
3909 callback->onCodecFormatChanged(audioMetadata);
3910 } else {
3911 mCallback.clear();
3912 }
3913 return binder::Status::ok();
3914}
3915
3916void AudioTrack::AudioTrackCallback::setAudioTrackCallback(
3917 const sp<media::IAudioTrackCallback> &callback) {
3918 AutoMutex lock(mAudioTrackCbLock);
3919 mCallback = callback;
3920}
3921
Glenn Kasten40bc9062015-03-20 09:09:33 -07003922} // namespace android