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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*/));
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800181 media::AudioAttributesInternal attributesAidl = VALUE_OR_RETURN(
182 legacy2aidl_audio_attributes_t_AudioAttributesInternal(attributes));
183 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
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001236status_t AudioTrack::setDualMonoMode(audio_dual_mono_mode_t mode)
1237{
1238 AutoMutex lock(mLock);
1239 return setDualMonoMode_l(mode);
1240}
1241
1242status_t AudioTrack::setDualMonoMode_l(audio_dual_mono_mode_t mode)
1243{
1244 const status_t status = statusTFromBinderStatus(
1245 mAudioTrack->setDualMonoMode(VALUE_OR_RETURN_STATUS(
1246 legacy2aidl_audio_dual_mono_mode_t_AudioDualMonoMode(mode))));
1247 if (status == NO_ERROR) mDualMonoMode = mode;
1248 return status;
1249}
1250
1251status_t AudioTrack::getDualMonoMode(audio_dual_mono_mode_t* mode) const
1252{
1253 AutoMutex lock(mLock);
1254 media::AudioDualMonoMode mediaMode;
1255 const status_t status = statusTFromBinderStatus(mAudioTrack->getDualMonoMode(&mediaMode));
1256 if (status == NO_ERROR) {
1257 *mode = VALUE_OR_RETURN_STATUS(
1258 aidl2legacy_AudioDualMonoMode_audio_dual_mono_mode_t(mediaMode));
1259 }
1260 return status;
1261}
1262
1263status_t AudioTrack::setAudioDescriptionMixLevel(float leveldB)
1264{
1265 AutoMutex lock(mLock);
1266 return setAudioDescriptionMixLevel_l(leveldB);
1267}
1268
1269status_t AudioTrack::setAudioDescriptionMixLevel_l(float leveldB)
1270{
1271 const status_t status = statusTFromBinderStatus(
1272 mAudioTrack->setAudioDescriptionMixLevel(leveldB));
1273 if (status == NO_ERROR) mAudioDescriptionMixLeveldB = leveldB;
1274 return status;
1275}
1276
1277status_t AudioTrack::getAudioDescriptionMixLevel(float* leveldB) const
1278{
1279 AutoMutex lock(mLock);
1280 return statusTFromBinderStatus(mAudioTrack->getAudioDescriptionMixLevel(leveldB));
1281}
1282
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001283status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
Andy Hung8edb8dc2015-03-26 19:13:55 -07001284{
Andy Hung8edb8dc2015-03-26 19:13:55 -07001285 AutoMutex lock(mLock);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001286 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
Andy Hung8edb8dc2015-03-26 19:13:55 -07001287 return NO_ERROR;
1288 }
Glenn Kastend79072e2016-01-06 08:41:20 -08001289 if (isOffloadedOrDirect_l()) {
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001290 const status_t status = statusTFromBinderStatus(mAudioTrack->setPlaybackRateParameters(
1291 VALUE_OR_RETURN_STATUS(
1292 legacy2aidl_audio_playback_rate_t_AudioPlaybackRate(playbackRate))));
1293 if (status == NO_ERROR) {
1294 mPlaybackRate = playbackRate;
Bertil Ã…kesson081fb742022-07-14 16:39:36 +02001295 } else if (status == INVALID_OPERATION
1296 && playbackRate.mSpeed == 1.0f && mPlaybackRate.mPitch == 1.0f) {
1297 mPlaybackRate = playbackRate;
1298 return NO_ERROR;
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001299 }
1300 return status;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001301 }
1302 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
1303 return INVALID_OPERATION;
1304 }
Andy Hungff874dc2016-04-11 16:49:09 -07001305
Andy Hungfb8ede22018-09-12 19:03:24 -07001306 ALOGV("%s(%d): mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001307 __func__, mPortId, mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001308 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001309 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
1310 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
1311 const float effectivePitch = adjustPitch(playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001312 AudioPlaybackRate playbackRateTemp = playbackRate;
1313 playbackRateTemp.mSpeed = effectiveSpeed;
1314 playbackRateTemp.mPitch = effectivePitch;
1315
Andy Hungfb8ede22018-09-12 19:03:24 -07001316 ALOGV("%s(%d) (effective) mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001317 __func__, mPortId, effectiveRate, effectiveSpeed, effectivePitch);
Andy Hungff874dc2016-04-11 16:49:09 -07001318
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001319 if (!isAudioPlaybackRateValid(playbackRateTemp)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001320 ALOGW("%s(%d) (%f, %f) failed (effective rate out of bounds)",
Eric Laurent973db022018-11-20 14:54:31 -08001321 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001322 return BAD_VALUE;
1323 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001324 // Check if the buffer size is compatible.
Andy Hung26145642015-04-15 21:56:53 -07001325 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001326 ALOGW("%s(%d) (%f, %f) failed (buffer size)",
Eric Laurent973db022018-11-20 14:54:31 -08001327 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung8edb8dc2015-03-26 19:13:55 -07001328 return BAD_VALUE;
1329 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001330
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001331 // Check resampler ratios are within bounds
Glenn Kastend3bb6452016-12-05 18:14:37 -08001332 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
1333 (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001334 ALOGW("%s(%d) (%f, %f) failed. Resample rate exceeds max accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001335 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001336 return BAD_VALUE;
1337 }
1338
Dan Austine34eae22015-10-27 16:14:52 -07001339 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001340 ALOGW("%s(%d) (%f, %f) failed. Resample rate below min accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001341 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001342 return BAD_VALUE;
1343 }
1344 mPlaybackRate = playbackRate;
1345 //set effective rates
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001346 mProxy->setPlaybackRate(playbackRateTemp);
Andy Hung26145642015-04-15 21:56:53 -07001347 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
Andy Hungb68f5eb2019-12-03 16:49:17 -08001348
1349 mediametrics::LogItem(mMetricsId)
1350 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYBACKPARAM)
1351 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
1352 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
1353 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
1354 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1355 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveRate)
1356 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1357 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)playbackRateTemp.mSpeed)
1358 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1359 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)playbackRateTemp.mPitch)
1360 .record();
1361
Andy Hung8edb8dc2015-03-26 19:13:55 -07001362 return NO_ERROR;
1363}
1364
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001365const AudioPlaybackRate& AudioTrack::getPlaybackRate()
Andy Hung8edb8dc2015-03-26 19:13:55 -07001366{
1367 AutoMutex lock(mLock);
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001368 if (isOffloadedOrDirect_l()) {
1369 media::AudioPlaybackRate playbackRateTemp;
1370 const status_t status = statusTFromBinderStatus(
1371 mAudioTrack->getPlaybackRateParameters(&playbackRateTemp));
1372 if (status == NO_ERROR) { // update local version if changed.
1373 mPlaybackRate =
1374 aidl2legacy_AudioPlaybackRate_audio_playback_rate_t(playbackRateTemp).value();
1375 }
1376 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001377 return mPlaybackRate;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001378}
1379
Phil Burkc0adecb2016-01-08 12:44:11 -08001380ssize_t AudioTrack::getBufferSizeInFrames()
1381{
1382 AutoMutex lock(mLock);
1383 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1384 return NO_INIT;
1385 }
Phil Burka9876702020-04-20 18:16:15 -07001386
Phil Burke8972b02016-03-04 11:29:57 -08001387 return (ssize_t) mProxy->getBufferSizeInFrames();
Phil Burkc0adecb2016-01-08 12:44:11 -08001388}
1389
Andy Hungf2c87b32016-04-07 19:49:29 -07001390status_t AudioTrack::getBufferDurationInUs(int64_t *duration)
1391{
1392 if (duration == nullptr) {
1393 return BAD_VALUE;
1394 }
1395 AutoMutex lock(mLock);
1396 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1397 return NO_INIT;
1398 }
1399 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames();
1400 if (bufferSizeInFrames < 0) {
1401 return (status_t)bufferSizeInFrames;
1402 }
1403 *duration = (int64_t)((double)bufferSizeInFrames * 1000000
1404 / ((double)mSampleRate * mPlaybackRate.mSpeed));
1405 return NO_ERROR;
1406}
1407
Phil Burkc0adecb2016-01-08 12:44:11 -08001408ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames)
1409{
1410 AutoMutex lock(mLock);
1411 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1412 return NO_INIT;
1413 }
Phil Burka9876702020-04-20 18:16:15 -07001414
1415 ssize_t originalBufferSize = mProxy->getBufferSizeInFrames();
1416 ssize_t finalBufferSize = mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
1417 if (originalBufferSize != finalBufferSize) {
Phil Burk64e16a72020-06-01 13:25:51 -07001418 android::mediametrics::LogItem(mMetricsId)
1419 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
1420 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1421 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t)getUnderrunCount_l())
1422 .record();
Phil Burka9876702020-04-20 18:16:15 -07001423 }
1424 return finalBufferSize;
Phil Burkc0adecb2016-01-08 12:44:11 -08001425}
1426
Andy Hung3c7f47a2021-03-16 17:30:09 -07001427ssize_t AudioTrack::getStartThresholdInFrames() const
1428{
1429 AutoMutex lock(mLock);
1430 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1431 return NO_INIT;
1432 }
1433 return (ssize_t) mProxy->getStartThresholdInFrames();
1434}
1435
1436ssize_t AudioTrack::setStartThresholdInFrames(size_t startThresholdInFrames)
1437{
1438 if (startThresholdInFrames > INT32_MAX || startThresholdInFrames == 0) {
1439 // contractually we could simply return the current threshold in frames
1440 // to indicate the request was ignored, but we return an error here.
1441 return BAD_VALUE;
1442 }
1443 AutoMutex lock(mLock);
1444 // We do not permit calling setStartThresholdInFrames() between the AudioTrack
1445 // default ctor AudioTrack() and set(...) but rather fail such an attempt.
1446 // (To do so would require a cached mOrigStartThresholdInFrames and we may
1447 // not have proper validation for the actual set value).
1448 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1449 return NO_INIT;
1450 }
1451 const uint32_t original = mProxy->getStartThresholdInFrames();
1452 const uint32_t final = mProxy->setStartThresholdInFrames(startThresholdInFrames);
1453 if (original != final) {
1454 android::mediametrics::LogItem(mMetricsId)
1455 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD)
1456 .set(AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES, (int32_t)final)
1457 .record();
1458 if (original > final) {
1459 // restart track if it was disabled by audioflinger due to previous underrun
1460 // and we reduced the number of frames for the threshold.
1461 restartIfDisabled();
1462 }
1463 }
1464 return final;
1465}
1466
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001467status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1468{
Glenn Kastend79072e2016-01-06 08:41:20 -08001469 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001470 return INVALID_OPERATION;
1471 }
1472
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001473 if (loopCount == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001474 ;
1475 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount &&
1476 loopEnd - loopStart >= MIN_LOOP) {
1477 ;
1478 } else {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001479 return BAD_VALUE;
1480 }
1481
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001482 AutoMutex lock(mLock);
1483 // See setPosition() regarding setting parameters such as loop points or position while active
1484 if (mState == STATE_ACTIVE) {
1485 return INVALID_OPERATION;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001486 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001487 setLoop_l(loopStart, loopEnd, loopCount);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001488 return NO_ERROR;
1489}
1490
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001491void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1492{
Andy Hung4ede21d2014-12-12 15:37:34 -08001493 // We do not update the periodic notification point.
1494 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
1495 mLoopCount = loopCount;
1496 mLoopEnd = loopEnd;
1497 mLoopStart = loopStart;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001498 mLoopCountNotified = loopCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001499 mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
Andy Hung3c09c782014-12-29 18:39:32 -08001500
1501 // Waking the AudioTrackThread is not needed as this cannot be called when active.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001502}
1503
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001504status_t AudioTrack::setMarkerPosition(uint32_t marker)
1505{
Atneya Nair14aabae2021-11-30 17:36:24 -05001506 AutoMutex lock(mLock);
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001507 // The only purpose of setting marker position is to get a callback
Atneya Nair14aabae2021-11-30 17:36:24 -05001508 if (!mCallback.promote() || isOffloadedOrDirect_l()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001509 return INVALID_OPERATION;
1510 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001511
1512 mMarkerPosition = marker;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001513 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001514
Andy Hung3c09c782014-12-29 18:39:32 -08001515 sp<AudioTrackThread> t = mAudioTrackThread;
1516 if (t != 0) {
1517 t->wake();
1518 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001519 return NO_ERROR;
1520}
1521
Glenn Kastena5224f32012-01-04 12:41:44 -08001522status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001523{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001524 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001525 return INVALID_OPERATION;
1526 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001527 if (marker == NULL) {
1528 return BAD_VALUE;
1529 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001530
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001531 AutoMutex lock(mLock);
Andy Hung90e8a972015-11-09 16:42:40 -08001532 mMarkerPosition.getValue(marker);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001533
1534 return NO_ERROR;
1535}
1536
1537status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
1538{
Atneya Nair14aabae2021-11-30 17:36:24 -05001539 AutoMutex lock(mLock);
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001540 // The only purpose of setting position update period is to get a callback
Atneya Nair14aabae2021-11-30 17:36:24 -05001541 if (!mCallback.promote() || isOffloadedOrDirect_l()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001542 return INVALID_OPERATION;
1543 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001544
Glenn Kasten200092b2014-08-15 15:13:30 -07001545 mNewPosition = updateAndGetPosition_l() + updatePeriod;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001546 mUpdatePeriod = updatePeriod;
Glenn Kasten2b2165c2014-01-13 08:53:36 -08001547
Andy Hung3c09c782014-12-29 18:39:32 -08001548 sp<AudioTrackThread> t = mAudioTrackThread;
1549 if (t != 0) {
1550 t->wake();
1551 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001552 return NO_ERROR;
1553}
1554
Glenn Kastena5224f32012-01-04 12:41:44 -08001555status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001556{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001557 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001558 return INVALID_OPERATION;
1559 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001560 if (updatePeriod == NULL) {
1561 return BAD_VALUE;
1562 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001563
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001564 AutoMutex lock(mLock);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001565 *updatePeriod = mUpdatePeriod;
1566
1567 return NO_ERROR;
1568}
1569
1570status_t AudioTrack::setPosition(uint32_t position)
1571{
Glenn Kastend79072e2016-01-06 08:41:20 -08001572 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001573 return INVALID_OPERATION;
1574 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001575 if (position > mFrameCount) {
1576 return BAD_VALUE;
1577 }
John Grossman4ff14ba2012-02-08 16:37:41 -08001578
Eric Laurent1703cdf2011-03-07 14:52:59 -08001579 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001580 // Currently we require that the player is inactive before setting parameters such as position
1581 // or loop points. Otherwise, there could be a race condition: the application could read the
1582 // current position, compute a new position or loop parameters, and then set that position or
1583 // loop parameters but it would do the "wrong" thing since the position has continued to advance
1584 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app
1585 // to specify how it wants to handle such scenarios.
1586 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001587 return INVALID_OPERATION;
1588 }
Andy Hung9b461582014-12-01 17:56:29 -08001589 // After setting the position, use full update period before notification.
Glenn Kasten200092b2014-08-15 15:13:30 -07001590 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001591 mStaticProxy->setBufferPosition(position);
Andy Hung3c09c782014-12-29 18:39:32 -08001592
1593 // Waking the AudioTrackThread is not needed as this cannot be called when active.
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001594 return NO_ERROR;
1595}
1596
Glenn Kasten200092b2014-08-15 15:13:30 -07001597status_t AudioTrack::getPosition(uint32_t *position)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001598{
Glenn Kastend65d73c2012-06-22 17:21:07 -07001599 if (position == NULL) {
1600 return BAD_VALUE;
1601 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001602
Eric Laurent1703cdf2011-03-07 14:52:59 -08001603 AutoMutex lock(mLock);
Andy Hung7a490e72016-03-23 15:58:10 -07001604 // FIXME: offloaded and direct tracks call into the HAL for render positions
1605 // for compressed/synced data; however, we use proxy position for pure linear pcm data
1606 // as we do not know the capability of the HAL for pcm position support and standby.
1607 // There may be some latency differences between the HAL position and the proxy position.
1608 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001609 uint32_t dspFrames = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001610
Eric Laurentab5cdba2014-06-09 17:22:27 -07001611 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001612 ALOGV("%s(%d): called in paused state, return cached position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001613 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001614 *position = mPausedPosition;
1615 return NO_ERROR;
1616 }
1617
Glenn Kasten142f5192014-03-25 17:44:59 -07001618 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung1f1db832015-06-08 13:26:10 -07001619 uint32_t halFrames; // actually unused
1620 (void) AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames);
1621 // FIXME: on getRenderPosition() error, we return OK with frame position 0.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001622 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001623 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED)
1624 // due to hardware latency. We leave this behavior for now.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001625 *position = dspFrames;
1626 } else {
Eric Laurent275e8e92014-11-30 15:14:47 -08001627 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung1f1db832015-06-08 13:26:10 -07001628 (void) restoreTrack_l("getPosition");
1629 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l()
1630 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position.
Eric Laurent275e8e92014-11-30 15:14:47 -08001631 }
1632
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001633 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
Glenn Kasten200092b2014-08-15 15:13:30 -07001634 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ?
Andy Hung90e8a972015-11-09 16:42:40 -08001635 0 : updateAndGetPosition_l().value();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001636 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001637 return NO_ERROR;
1638}
1639
Kévin PETIT377b2ec2014-02-03 12:35:36 +00001640status_t AudioTrack::getBufferPosition(uint32_t *position)
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001641{
Glenn Kastend79072e2016-01-06 08:41:20 -08001642 if (mSharedBuffer == 0) {
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001643 return INVALID_OPERATION;
1644 }
1645 if (position == NULL) {
1646 return BAD_VALUE;
1647 }
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001648
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001649 AutoMutex lock(mLock);
1650 *position = mStaticProxy->getBufferPosition();
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001651 return NO_ERROR;
1652}
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001653
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001654status_t AudioTrack::reload()
1655{
Glenn Kastend79072e2016-01-06 08:41:20 -08001656 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001657 return INVALID_OPERATION;
1658 }
1659
Eric Laurent1703cdf2011-03-07 14:52:59 -08001660 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001661 // See setPosition() regarding setting parameters such as loop points or position while active
1662 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001663 return INVALID_OPERATION;
1664 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001665 mNewPosition = mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001666 (void) updateAndGetPosition_l();
1667 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07001668 mPreviousTimestampValid = false;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001669#if 0
Andy Hung9b461582014-12-01 17:56:29 -08001670 // The documentation is not clear on the behavior of reload() and the restoration
Andy Hung53c3b5f2014-12-15 16:42:05 -08001671 // of loop count. Historically we have not restored loop count, start, end,
1672 // but it makes sense if one desires to repeat playing a particular sound.
1673 if (mLoopCount != 0) {
1674 mLoopCountNotified = mLoopCount;
1675 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
1676 }
1677#endif
Andy Hung9b461582014-12-01 17:56:29 -08001678 mStaticProxy->setBufferPosition(0);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001679 return NO_ERROR;
1680}
1681
Glenn Kasten38e905b2014-01-13 10:21:48 -08001682audio_io_handle_t AudioTrack::getOutput() const
Eric Laurentc2f1f072009-07-17 12:17:14 -07001683{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001684 AutoMutex lock(mLock);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001685 return mOutput;
Eric Laurent1703cdf2011-03-07 14:52:59 -08001686}
1687
Paul McLeanaa981192015-03-21 09:55:15 -07001688status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) {
1689 AutoMutex lock(mLock);
Eric Laurent2f2c1982021-06-02 14:03:11 +02001690 ALOGV("%s(%d): deviceId=%d mSelectedDeviceId=%d",
1691 __func__, mPortId, deviceId, mSelectedDeviceId);
Paul McLeanaa981192015-03-21 09:55:15 -07001692 if (mSelectedDeviceId != deviceId) {
1693 mSelectedDeviceId = deviceId;
Eric Laurentfb00fc72017-05-25 18:17:12 -07001694 if (mStatus == NO_ERROR) {
1695 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
jiabin156c6872017-10-06 09:47:15 -07001696 mProxy->interrupt();
Eric Laurentfb00fc72017-05-25 18:17:12 -07001697 }
Paul McLeanaa981192015-03-21 09:55:15 -07001698 }
Eric Laurent493404d2015-04-21 15:07:36 -07001699 return NO_ERROR;
Paul McLeanaa981192015-03-21 09:55:15 -07001700}
1701
1702audio_port_handle_t AudioTrack::getOutputDevice() {
1703 AutoMutex lock(mLock);
1704 return mSelectedDeviceId;
1705}
1706
Eric Laurentad2e7b92017-09-14 20:06:42 -07001707// must be called with mLock held
1708void AudioTrack::updateRoutedDeviceId_l()
1709{
1710 // if the track is inactive, do not update actual device as the output stream maybe routed
1711 // to a device not relevant to this client because of other active use cases.
1712 if (mState != STATE_ACTIVE) {
1713 return;
1714 }
1715 if (mOutput != AUDIO_IO_HANDLE_NONE) {
1716 audio_port_handle_t deviceId = AudioSystem::getDeviceIdForIo(mOutput);
1717 if (deviceId != AUDIO_PORT_HANDLE_NONE) {
1718 mRoutedDeviceId = deviceId;
1719 }
1720 }
1721}
1722
Eric Laurent296fb132015-05-01 11:38:42 -07001723audio_port_handle_t AudioTrack::getRoutedDeviceId() {
1724 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001725 updateRoutedDeviceId_l();
1726 return mRoutedDeviceId;
Eric Laurent296fb132015-05-01 11:38:42 -07001727}
1728
Eric Laurentbe916aa2010-06-01 23:49:17 -07001729status_t AudioTrack::attachAuxEffect(int effectId)
1730{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001731 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001732 status_t status;
1733 mAudioTrack->attachAuxEffect(effectId, &status);
Eric Laurent2beeb502010-07-16 07:43:46 -07001734 if (status == NO_ERROR) {
1735 mAuxEffectId = effectId;
1736 }
1737 return status;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001738}
1739
Eric Laurente83b55d2014-11-14 10:06:21 -08001740audio_stream_type_t AudioTrack::streamType() const
1741{
Eric Laurente83b55d2014-11-14 10:06:21 -08001742 return mStreamType;
1743}
1744
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001745uint32_t AudioTrack::latency()
1746{
1747 AutoMutex lock(mLock);
1748 updateLatency_l();
1749 return mLatency;
1750}
1751
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001752// -------------------------------------------------------------------------
1753
Eric Laurent1703cdf2011-03-07 14:52:59 -08001754// must be called with mLock held
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001755void AudioTrack::updateLatency_l()
1756{
1757 status_t status = AudioSystem::getLatency(mOutput, &mAfLatency);
1758 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08001759 ALOGW("%s(%d): getLatency(%d) failed status %d", __func__, mPortId, mOutput, status);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001760 } else {
1761 // FIXME don't believe this lie
Andy Hung13969262017-09-11 17:24:21 -07001762 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001763 }
1764}
1765
Phil Burkadbb75a2017-06-16 12:19:42 -07001766// TODO Move this macro to a common header file for enum to string conversion in audio framework.
1767#define MEDIA_CASE_ENUM(name) case name: return #name
1768const char * AudioTrack::convertTransferToText(transfer_type transferType) {
1769 switch (transferType) {
1770 MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
1771 MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
1772 MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
1773 MEDIA_CASE_ENUM(TRANSFER_SYNC);
1774 MEDIA_CASE_ENUM(TRANSFER_SHARED);
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001775 MEDIA_CASE_ENUM(TRANSFER_SYNC_NOTIF_CALLBACK);
Phil Burkadbb75a2017-06-16 12:19:42 -07001776 default:
1777 return "UNRECOGNIZED";
1778 }
1779}
1780
Glenn Kasten200092b2014-08-15 15:13:30 -07001781status_t AudioTrack::createTrack_l()
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001782{
Eric Laurentf32d7812017-11-30 14:44:07 -08001783 status_t status;
1784 bool callbackAdded = false;
Andy Hung3acde2c2021-11-11 09:18:08 -08001785 std::string errorMessage;
Eric Laurentf32d7812017-11-30 14:44:07 -08001786
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001787 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
1788 if (audioFlinger == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001789 errorMessage = StringPrintf("%s(%d): Could not get audioflinger",
Eric Laurent973db022018-11-20 14:54:31 -08001790 __func__, mPortId);
Andy Hung3acde2c2021-11-11 09:18:08 -08001791 status = DEAD_OBJECT;
Eric Laurentf32d7812017-11-30 14:44:07 -08001792 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001793 }
1794
Eric Laurent21da6472017-11-09 16:29:26 -08001795 {
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08001796 // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
1797 // After fast request is denied, we will request again if IAudioTrack is re-created.
Glenn Kastend79072e2016-01-06 08:41:20 -08001798 // Client can only express a preference for FAST. Server will perform additional tests.
Phil Burk33ff89b2015-11-30 11:16:01 -08001799 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Phil Burkadbb75a2017-06-16 12:19:42 -07001800 // either of these use cases:
1801 // use case 1: shared buffer
1802 bool sharedBuffer = mSharedBuffer != 0;
1803 bool transferAllowed =
Glenn Kastenc6ba8232014-02-27 13:34:29 -08001804 // use case 2: callback transfer mode
Glenn Kasten1dfe2f92015-03-09 12:03:14 -07001805 (mTransfer == TRANSFER_CALLBACK) ||
1806 // use case 3: obtain/release mode
Phil Burk33ff89b2015-11-30 11:16:01 -08001807 (mTransfer == TRANSFER_OBTAIN) ||
1808 // use case 4: synchronous write
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001809 ((mTransfer == TRANSFER_SYNC || mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK)
1810 && mThreadCanCallJava);
Phil Burkadbb75a2017-06-16 12:19:42 -07001811
Eric Laurent21da6472017-11-09 16:29:26 -08001812 bool fastAllowed = sharedBuffer || transferAllowed;
1813 if (!fastAllowed) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001814 ALOGW("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by client,"
1815 " not shared buffer and transfer = %s",
Eric Laurent973db022018-11-20 14:54:31 -08001816 __func__, mPortId,
Phil Burkadbb75a2017-06-16 12:19:42 -07001817 convertTransferToText(mTransfer));
Phil Burk33ff89b2015-11-30 11:16:01 -08001818 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
1819 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001820 }
1821
Eric Laurent21da6472017-11-09 16:29:26 -08001822 IAudioFlinger::CreateTrackInput input;
Andy Hunga2159aa2021-07-20 13:01:52 -07001823 if (mOriginalStreamType != AUDIO_STREAM_DEFAULT) {
1824 // Legacy: This is based on original parameters even if the track is recreated.
1825 input.attr = AudioSystem::streamTypeToAttributes(mOriginalStreamType);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001826 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08001827 input.attr = mAttributes;
Eric Laurentd1b449a2010-05-14 03:26:45 -07001828 }
Eric Laurent21da6472017-11-09 16:29:26 -08001829 input.config = AUDIO_CONFIG_INITIALIZER;
1830 input.config.sample_rate = mSampleRate;
1831 input.config.channel_mask = mChannelMask;
1832 input.config.format = mFormat;
1833 input.config.offload_info = mOffloadInfoCopy;
Svet Ganov3e5f14f2021-05-13 22:51:08 +00001834 input.clientInfo.attributionSource = mClientAttributionSource;
Eric Laurent21da6472017-11-09 16:29:26 -08001835 input.clientInfo.clientTid = -1;
Glenn Kasten363fb752014-01-15 12:27:31 -08001836 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Glenn Kasten47d55172017-05-23 11:19:30 -07001837 // It is currently meaningless to request SCHED_FIFO for a Java thread. Even if the
1838 // application-level code follows all non-blocking design rules, the language runtime
1839 // doesn't also follow those rules, so the thread will not benefit overall.
Phil Burk33ff89b2015-11-30 11:16:01 -08001840 if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
Eric Laurent21da6472017-11-09 16:29:26 -08001841 input.clientInfo.clientTid = mAudioTrackThread->getTid();
Glenn Kasten3acbd052012-02-28 10:39:56 -08001842 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001843 }
Eric Laurent21da6472017-11-09 16:29:26 -08001844 input.sharedBuffer = mSharedBuffer;
1845 input.notificationsPerBuffer = mNotificationsPerBufferReq;
1846 input.speed = 1.0;
1847 if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
1848 (mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
1849 input.speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
1850 max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
1851 }
1852 input.flags = mFlags;
1853 input.frameCount = mReqFrameCount;
1854 input.notificationFrameCount = mNotificationFramesReq;
1855 input.selectedDeviceId = mSelectedDeviceId;
1856 input.sessionId = mSessionId;
jiabinf6eb4c32020-02-25 14:06:25 -08001857 input.audioTrackCallback = mAudioTrackCallback;
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001858
Ytai Ben-Tsvi4dfeb622020-11-02 12:47:30 -08001859 media::CreateTrackResponse response;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001860 status = audioFlinger->createTrack(VALUE_OR_FATAL(input.toAidl()), response);
Ytai Ben-Tsvi357e26a2021-01-05 13:21:19 -08001861
1862 IAudioFlinger::CreateTrackOutput output{};
1863 if (status == NO_ERROR) {
1864 output = VALUE_OR_FATAL(IAudioFlinger::CreateTrackOutput::fromAidl(response));
1865 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001866
Eric Laurent21da6472017-11-09 16:29:26 -08001867 if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001868 errorMessage = StringPrintf(
1869 "%s(%d): AudioFlinger could not create track, status: %d output %d",
Eric Laurent973db022018-11-20 14:54:31 -08001870 __func__, mPortId, status, output.outputId);
Eric Laurentf32d7812017-11-30 14:44:07 -08001871 if (status == NO_ERROR) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001872 status = INVALID_OPERATION; // device not ready
Eric Laurentf32d7812017-11-30 14:44:07 -08001873 }
1874 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001875 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001876 ALOG_ASSERT(output.audioTrack != 0);
Glenn Kastenc08d20b2014-02-24 15:21:10 -08001877
Eric Laurent21da6472017-11-09 16:29:26 -08001878 mFrameCount = output.frameCount;
1879 mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
1880 mRoutedDeviceId = output.selectedDeviceId;
1881 mSessionId = output.sessionId;
Andy Hunga2159aa2021-07-20 13:01:52 -07001882 mStreamType = output.streamType;
Eric Laurent21da6472017-11-09 16:29:26 -08001883
1884 mSampleRate = output.sampleRate;
1885 if (mOriginalSampleRate == 0) {
1886 mOriginalSampleRate = mSampleRate;
1887 }
1888
1889 mAfFrameCount = output.afFrameCount;
1890 mAfSampleRate = output.afSampleRate;
1891 mAfLatency = output.afLatencyMs;
1892
1893 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
1894
Glenn Kasten38e905b2014-01-13 10:21:48 -08001895 // AudioFlinger now owns the reference to the I/O handle,
1896 // so we are no longer responsible for releasing it.
1897
Glenn Kasten7fd04222016-02-02 12:38:16 -08001898 // FIXME compare to AudioRecord
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001899 std::optional<media::SharedFileRegion> sfr;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001900 output.audioTrack->getCblk(&sfr);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001901 sp<IMemory> iMem = VALUE_OR_FATAL(aidl2legacy_NullableSharedFileRegion_IMemory(sfr));
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001902 if (iMem == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001903 errorMessage = StringPrintf("%s(%d): Could not get control block", __func__, mPortId);
1904 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001905 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001906 }
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001907 // TODO: Using unsecurePointer() has some associated security pitfalls
1908 // (see declaration for details).
1909 // Either document why it is safe in this case or address the
1910 // issue (e.g. by copying).
1911 void *iMemPointer = iMem->unsecurePointer();
Glenn Kasten0cde0762014-01-16 15:06:36 -08001912 if (iMemPointer == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001913 errorMessage = StringPrintf(
1914 "%s(%d): Could not get control block pointer", __func__, mPortId);
1915 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001916 goto exit;
Glenn Kasten0cde0762014-01-16 15:06:36 -08001917 }
Glenn Kasten53cec222013-08-29 09:01:02 -07001918 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001919 if (mAudioTrack != 0) {
Marco Nelissenf8880202014-11-14 07:58:25 -08001920 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001921 mDeathNotifier.clear();
1922 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001923 mAudioTrack = output.audioTrack;
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001924 mCblkMemory = iMem;
Eric Laurent3bcffa12014-06-12 18:38:45 -07001925 IPCThreadState::self()->flushCommands();
1926
Glenn Kasten0cde0762014-01-16 15:06:36 -08001927 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001928 mCblk = cblk;
Glenn Kasten5f631512014-02-24 15:16:07 -08001929
Glenn Kastena07f17c2013-04-23 12:39:37 -07001930 mAwaitBoost = false;
Glenn Kasten363fb752014-01-15 12:27:31 -08001931 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Eric Laurent21da6472017-11-09 16:29:26 -08001932 if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001933 ALOGI("%s(%d): AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08001934 __func__, mPortId, mReqFrameCount, mFrameCount);
Phil Burk33ff89b2015-11-30 11:16:01 -08001935 if (!mThreadCanCallJava) {
1936 mAwaitBoost = true;
1937 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08001938 } else {
Robert Wuc121cd12021-08-13 17:51:40 +00001939 ALOGV("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08001940 __func__, mPortId, mReqFrameCount, mFrameCount);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001941 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08001942 }
Eric Laurent21da6472017-11-09 16:29:26 -08001943 mFlags = output.flags;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001944
Eric Laurentad2e7b92017-09-14 20:06:42 -07001945 //mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
Eric Laurent09f1ed22019-04-24 17:45:17 -07001946 if (mDeviceCallback != 0) {
Eric Laurentad2e7b92017-09-14 20:06:42 -07001947 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07001948 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001949 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07001950 AudioSystem::addAudioDeviceCallback(this, output.outputId, output.portId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001951 callbackAdded = true;
1952 }
1953
Eric Laurent09f1ed22019-04-24 17:45:17 -07001954 mPortId = output.portId;
Vlad Popaad0fe922022-06-10 00:43:14 +02001955 // notify the upper layers about the new portId
1956 triggerPortIdUpdate_l();
1957
Glenn Kasten38e905b2014-01-13 10:21:48 -08001958 // We retain a copy of the I/O handle, but don't own the reference
Eric Laurent21da6472017-11-09 16:29:26 -08001959 mOutput = output.outputId;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001960 mRefreshRemaining = true;
1961
1962 // Starting address of buffers in shared memory. If there is a shared buffer, buffers
1963 // is the value of pointer() for the shared buffer, otherwise buffers points
1964 // immediately after the control block. This address is for the mapping within client
1965 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
1966 void* buffers;
Glenn Kasten363fb752014-01-15 12:27:31 -08001967 if (mSharedBuffer == 0) {
Glenn Kasten138d6f92015-03-20 10:54:51 -07001968 buffers = cblk + 1;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001969 } else {
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001970 // TODO: Using unsecurePointer() has some associated security pitfalls
1971 // (see declaration for details).
1972 // Either document why it is safe in this case or address the
1973 // issue (e.g. by copying).
1974 buffers = mSharedBuffer->unsecurePointer();
Glenn Kasten138d6f92015-03-20 10:54:51 -07001975 if (buffers == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001976 errorMessage = StringPrintf(
1977 "%s(%d): Could not get buffer pointer", __func__, mPortId);
1978 ALOGE("%s", errorMessage.c_str());
1979 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001980 goto exit;
Glenn Kasten138d6f92015-03-20 10:54:51 -07001981 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001982 }
1983
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001984 mAudioTrack->attachAuxEffect(mAuxEffectId, &status);
Glenn Kasten5f631512014-02-24 15:16:07 -08001985
Glenn Kasten093000f2012-05-03 09:35:36 -07001986 // If IAudioTrack is re-created, don't let the requested frameCount
1987 // decrease. This can confuse clients that cache frameCount().
Eric Laurent21da6472017-11-09 16:29:26 -08001988 if (mFrameCount > mReqFrameCount) {
1989 mReqFrameCount = mFrameCount;
Glenn Kasten093000f2012-05-03 09:35:36 -07001990 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001991
Andy Hungd7bd69e2015-07-24 07:52:41 -07001992 // reset server position to 0 as we have new cblk.
1993 mServer = 0;
1994
Glenn Kastene3aa6592012-12-04 12:22:46 -08001995 // update proxy
Glenn Kasten363fb752014-01-15 12:27:31 -08001996 if (mSharedBuffer == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001997 mStaticProxy.clear();
Eric Laurent21da6472017-11-09 16:29:26 -08001998 mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001999 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08002000 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002001 mProxy = mStaticProxy;
2002 }
seunghak.hane6a9d6582014-11-22 15:22:35 +09002003
2004 mProxy->setVolumeLR(gain_minifloat_pack(
2005 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
2006 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
2007
Glenn Kastene3aa6592012-12-04 12:22:46 -08002008 mProxy->setSendLevel(mSendLevel);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002009 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
2010 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
2011 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07002012 mProxy->setSampleRate(effectiveSampleRate);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002013
2014 AudioPlaybackRate playbackRateTemp = mPlaybackRate;
2015 playbackRateTemp.mSpeed = effectiveSpeed;
2016 playbackRateTemp.mPitch = effectivePitch;
2017 mProxy->setPlaybackRate(playbackRateTemp);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002018 mProxy->setMinimum(mNotificationFramesAct);
2019
Kuowei Lid4adbdb2020-08-13 14:44:25 +08002020 if (mDualMonoMode != AUDIO_DUAL_MONO_MODE_OFF) {
2021 setDualMonoMode_l(mDualMonoMode);
2022 }
2023 if (mAudioDescriptionMixLeveldB != -std::numeric_limits<float>::infinity()) {
2024 setAudioDescriptionMixLevel_l(mAudioDescriptionMixLeveldB);
2025 }
2026
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002027 mDeathNotifier = new DeathNotifier(this);
Marco Nelissenf8880202014-11-14 07:58:25 -08002028 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
Glenn Kastene3aa6592012-12-04 12:22:46 -08002029
Andy Hungb68f5eb2019-12-03 16:49:17 -08002030 // This is the first log sent from the AudioTrack client.
2031 // The creation of the audio track by AudioFlinger (in the code above)
2032 // is the first log of the AudioTrack and must be present before
2033 // any AudioTrack client logs will be accepted.
Andy Hungea840382020-05-05 21:50:17 -07002034
Andy Hungb68f5eb2019-12-03 16:49:17 -08002035 mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(mPortId);
2036 mediametrics::LogItem(mMetricsId)
2037 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE)
2038 // the following are immutable
Andy Hunga629bd12020-06-05 16:03:53 -07002039 .set(AMEDIAMETRICS_PROP_FLAGS, toString(mFlags).c_str())
2040 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -08002041 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
Andy Hung3a5c2f32021-02-17 15:06:42 -08002042 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, mLogSessionId)
Andy Hung839a3062021-02-17 11:15:16 -08002043 .set(AMEDIAMETRICS_PROP_PLAYERIID, mPlayerIId)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002044 .set(AMEDIAMETRICS_PROP_TRACKID, mPortId) // dup from key
Andy Hungb68f5eb2019-12-03 16:49:17 -08002045 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2046 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2047 .set(AMEDIAMETRICS_PROP_THREADID, (int32_t)output.outputId)
2048 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2049 .set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)mRoutedDeviceId)
2050 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2051 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
2052 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
2053 // the following are NOT immutable
2054 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)mVolume[AUDIO_INTERLEAVE_LEFT])
2055 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)mVolume[AUDIO_INTERLEAVE_RIGHT])
2056 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Andy Hung73dc2f92021-12-07 21:50:04 -08002057 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)NO_ERROR)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002058 .set(AMEDIAMETRICS_PROP_AUXEFFECTID, (int32_t)mAuxEffectId)
2059 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2060 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2061 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2062 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2063 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveSampleRate)
2064 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2065 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)effectiveSpeed)
2066 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2067 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)effectivePitch)
2068 .record();
2069
2070 // mSendLevel
2071 // mReqFrameCount?
2072 // mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq
2073 // mLatency, mAfLatency, mAfFrameCount, mAfSampleRate
2074
Glenn Kasten38e905b2014-01-13 10:21:48 -08002075 }
2076
Eric Laurentf32d7812017-11-30 14:44:07 -08002077exit:
Andy Hung3acde2c2021-11-11 09:18:08 -08002078 if (status != NO_ERROR) {
2079 if (callbackAdded) {
2080 // note: mOutput is always valid is callbackAdded is true
2081 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
2082 }
2083 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
2084 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
Eric Laurentad2e7b92017-09-14 20:06:42 -07002085 }
Eric Laurentf32d7812017-11-30 14:44:07 -08002086 mStatus = status;
Eric Laurent21da6472017-11-09 16:29:26 -08002087
2088 // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
Glenn Kasten38e905b2014-01-13 10:21:48 -08002089 return status;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002090}
2091
Andy Hung3acde2c2021-11-11 09:18:08 -08002092void AudioTrack::reportError(status_t status, const char *event, const char *message) const
2093{
2094 if (status == NO_ERROR) return;
2095 // We report error on the native side because some callers do not come
2096 // from Java.
Andy Hungde602302021-12-07 21:35:49 -08002097 // Ensure these variables are initialized in set().
2098 mediametrics::LogItem(AMEDIAMETRICS_KEY_AUDIO_TRACK_ERROR)
Andy Hung3acde2c2021-11-11 09:18:08 -08002099 .set(AMEDIAMETRICS_PROP_EVENT, event)
Andy Hung73dc2f92021-12-07 21:50:04 -08002100 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
2101 .set(AMEDIAMETRICS_PROP_STATUSMESSAGE, message)
Andy Hung3acde2c2021-11-11 09:18:08 -08002102 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
2103 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
2104 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2105 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2106 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2107 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2108 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
Andy Hung3acde2c2021-11-11 09:18:08 -08002109 // the following are NOT immutable
Andy Hungde602302021-12-07 21:35:49 -08002110 // frame count is initially the requested frame count, but may be adjusted
2111 // by AudioFlinger after creation.
2112 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
Andy Hung3acde2c2021-11-11 09:18:08 -08002113 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2114 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2115 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2116 .record();
2117}
2118
Glenn Kastenb46f3942015-03-09 12:00:30 -07002119status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002120{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002121 if (audioBuffer == NULL) {
Glenn Kasten551b5352015-03-20 11:30:28 -07002122 if (nonContig != NULL) {
2123 *nonContig = 0;
2124 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002125 return BAD_VALUE;
Eric Laurent9b7d9502011-03-21 11:49:00 -07002126 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002127 if (mTransfer != TRANSFER_OBTAIN) {
2128 audioBuffer->frameCount = 0;
Atneya Nair03079272022-01-18 17:03:14 -05002129 audioBuffer->mSize = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002130 audioBuffer->raw = NULL;
Glenn Kasten551b5352015-03-20 11:30:28 -07002131 if (nonContig != NULL) {
2132 *nonContig = 0;
2133 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002134 return INVALID_OPERATION;
2135 }
Eric Laurent9b7d9502011-03-21 11:49:00 -07002136
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002137 const struct timespec *requested;
Eric Laurentdf576992014-01-27 18:13:39 -08002138 struct timespec timeout;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002139 if (waitCount == -1) {
2140 requested = &ClientProxy::kForever;
2141 } else if (waitCount == 0) {
2142 requested = &ClientProxy::kNonBlocking;
2143 } else if (waitCount > 0) {
Chih-Hung Hsiehbca74292018-08-10 16:06:07 -07002144 time_t ms = WAIT_PERIOD_MS * (time_t) waitCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002145 timeout.tv_sec = ms / 1000;
Andy Hung06a730b2020-04-09 13:28:31 -07002146 timeout.tv_nsec = (ms % 1000) * 1000000;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002147 requested = &timeout;
2148 } else {
Eric Laurent973db022018-11-20 14:54:31 -08002149 ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002150 requested = NULL;
2151 }
Glenn Kastenb46f3942015-03-09 12:00:30 -07002152 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002153}
Eric Laurent1703cdf2011-03-07 14:52:59 -08002154
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002155status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
2156 struct timespec *elapsed, size_t *nonContig)
2157{
2158 // previous and new IAudioTrack sequence numbers are used to detect track re-creation
2159 uint32_t oldSequence = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002160
2161 Proxy::Buffer buffer;
2162 status_t status = NO_ERROR;
2163
2164 static const int32_t kMaxTries = 5;
2165 int32_t tryCounter = kMaxTries;
2166
2167 do {
2168 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
2169 // keep them from going away if another thread re-creates the track during obtainBuffer()
2170 sp<AudioTrackClientProxy> proxy;
2171 sp<IMemory> iMem;
2172
2173 { // start of lock scope
2174 AutoMutex lock(mLock);
2175
Glenn Kasten305996c2020-01-27 08:03:37 -08002176 uint32_t newSequence = mSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002177 // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
2178 if (status == DEAD_OBJECT) {
2179 // re-create track, unless someone else has already done so
2180 if (newSequence == oldSequence) {
2181 status = restoreTrack_l("obtainBuffer");
2182 if (status != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002183 buffer.mFrameCount = 0;
2184 buffer.mRaw = NULL;
2185 buffer.mNonContig = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002186 break;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002187 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002188 }
2189 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002190 oldSequence = newSequence;
2191
Eric Laurent4d231dc2016-03-11 18:38:23 -08002192 if (status == NOT_ENOUGH_DATA) {
2193 restartIfDisabled();
2194 }
2195
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002196 // Keep the extra references
2197 proxy = mProxy;
2198 iMem = mCblkMemory;
2199
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002200 if (mState == STATE_STOPPING) {
2201 status = -EINTR;
2202 buffer.mFrameCount = 0;
2203 buffer.mRaw = NULL;
2204 buffer.mNonContig = 0;
2205 break;
2206 }
2207
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002208 // Non-blocking if track is stopped or paused
2209 if (mState != STATE_ACTIVE) {
2210 requested = &ClientProxy::kNonBlocking;
2211 }
2212
2213 } // end of lock scope
2214
2215 buffer.mFrameCount = audioBuffer->frameCount;
2216 // FIXME starts the requested timeout and elapsed over from scratch
2217 status = proxy->obtainBuffer(&buffer, requested, elapsed);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002218 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002219
2220 audioBuffer->frameCount = buffer.mFrameCount;
Atneya Nair03079272022-01-18 17:03:14 -05002221 audioBuffer->mSize = buffer.mFrameCount * mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002222 audioBuffer->raw = buffer.mRaw;
Glenn Kasten305996c2020-01-27 08:03:37 -08002223 audioBuffer->sequence = oldSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002224 if (nonContig != NULL) {
2225 *nonContig = buffer.mNonContig;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002226 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002227 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002228}
2229
Glenn Kasten54a8a452015-03-09 12:03:00 -07002230void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002231{
Glenn Kasten3f02be22015-03-09 11:59:04 -07002232 // FIXME add error checking on mode, by adding an internal version
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002233 if (mTransfer == TRANSFER_SHARED) {
2234 return;
2235 }
2236
Atneya Nair03079272022-01-18 17:03:14 -05002237 size_t stepCount = audioBuffer->mSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002238 if (stepCount == 0) {
2239 return;
2240 }
2241
2242 Proxy::Buffer buffer;
2243 buffer.mFrameCount = stepCount;
2244 buffer.mRaw = audioBuffer->raw;
Glenn Kastene3aa6592012-12-04 12:22:46 -08002245
Eric Laurent1703cdf2011-03-07 14:52:59 -08002246 AutoMutex lock(mLock);
Glenn Kasten305996c2020-01-27 08:03:37 -08002247 if (audioBuffer->sequence != mSequence) {
2248 // This Buffer came from a different IAudioTrack instance, so ignore the releaseBuffer
2249 ALOGD("%s is no-op due to IAudioTrack sequence mismatch %u != %u",
2250 __func__, audioBuffer->sequence, mSequence);
2251 return;
2252 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002253 mReleased += stepCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002254 mInUnderrun = false;
2255 mProxy->releaseBuffer(&buffer);
2256
2257 // restart track if it was disabled by audioflinger due to previous underrun
Eric Laurent4d231dc2016-03-11 18:38:23 -08002258 restartIfDisabled();
2259}
2260
2261void AudioTrack::restartIfDisabled()
2262{
2263 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
2264 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002265 ALOGW("%s(%d): releaseBuffer() track %p disabled due to previous underrun, restarting",
Eric Laurent973db022018-11-20 14:54:31 -08002266 __func__, mPortId, this);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002267 // FIXME ignoring status
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002268 status_t status;
2269 mAudioTrack->start(&status);
Eric Laurentdf839842012-05-31 14:27:14 -07002270 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002271}
2272
2273// -------------------------------------------------------------------------
2274
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002275ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002276{
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002277 if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07002278 return INVALID_OPERATION;
2279 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002280
Eric Laurentab5cdba2014-06-09 17:22:27 -07002281 if (isDirect()) {
2282 AutoMutex lock(mLock);
2283 int32_t flags = android_atomic_and(
2284 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
2285 &mCblk->mFlags);
2286 if (flags & CBLK_INVALID) {
2287 return DEAD_OBJECT;
2288 }
2289 }
2290
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002291 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
Jiabin Huang447cea72020-07-28 22:35:18 +00002292 // Validation: user is most-likely passing an error code, and it would
Glenn Kasten99e53b82012-01-19 08:59:58 -08002293 // make the return value ambiguous (actualSize vs error).
Andy Hungfb8ede22018-09-12 19:03:24 -07002294 ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
Eric Laurent973db022018-11-20 14:54:31 -08002295 __func__, mPortId, buffer, userSize, userSize);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002296 return BAD_VALUE;
2297 }
2298
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002299 size_t written = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002300 Buffer audioBuffer;
2301
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002302 while (userSize >= mFrameSize) {
2303 audioBuffer.frameCount = userSize / mFrameSize;
Eric Laurentc2f1f072009-07-17 12:17:14 -07002304
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002305 status_t err = obtainBuffer(&audioBuffer,
2306 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002307 if (err < 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002308 if (written > 0) {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002309 break;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002310 }
Glenn Kasten0a2f1512016-07-22 08:06:37 -07002311 if (err == TIMED_OUT || err == -EINTR) {
2312 err = WOULD_BLOCK;
2313 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002314 return ssize_t(err);
2315 }
2316
Atneya Nair03079272022-01-18 17:03:14 -05002317 size_t toWrite = audioBuffer.size();
2318 memcpy(audioBuffer.raw, buffer, toWrite);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002319 buffer = ((const char *) buffer) + toWrite;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002320 userSize -= toWrite;
2321 written += toWrite;
2322
2323 releaseBuffer(&audioBuffer);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002324 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002325
Andy Hungea2b9c02016-02-12 17:06:53 -08002326 if (written > 0) {
2327 mFramesWritten += written / mFrameSize;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002328
2329 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2330 const sp<AudioTrackThread> t = mAudioTrackThread;
2331 if (t != 0) {
2332 // causes wake up of the playback thread, that will callback the client for
2333 // more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
2334 t->wake();
2335 }
2336 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002337 }
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002338
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002339 return written;
2340}
2341
2342// -------------------------------------------------------------------------
2343
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08002344nsecs_t AudioTrack::processAudioBuffer()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002345{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002346 // Currently the AudioTrack thread is not created if there are no callbacks.
2347 // Would it ever make sense to run the thread, even without callbacks?
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002348 // If so, then replace this by checks at each use for mCallback != NULL.
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002349 LOG_ALWAYS_FATAL_IF(mCblk == NULL);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002350 mLock.lock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002351 sp<IAudioTrackCallback> callback = mCallback.promote();
2352 if (!callback) {
2353 mCallback = nullptr;
Atneya Naire260f5a2022-05-03 17:02:20 -04002354 mLock.unlock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002355 return NS_NEVER;
2356 }
Glenn Kastena07f17c2013-04-23 12:39:37 -07002357 if (mAwaitBoost) {
2358 mAwaitBoost = false;
2359 mLock.unlock();
2360 static const int32_t kMaxTries = 5;
2361 int32_t tryCounter = kMaxTries;
2362 uint32_t pollUs = 10000;
2363 do {
Glenn Kasten8255ba72016-08-23 13:54:23 -07002364 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002365 if (policy == SCHED_FIFO || policy == SCHED_RR) {
2366 break;
2367 }
2368 usleep(pollUs);
2369 pollUs <<= 1;
2370 } while (tryCounter-- > 0);
2371 if (tryCounter < 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002372 ALOGE("%s(%d): did not receive expected priority boost on time",
Eric Laurent973db022018-11-20 14:54:31 -08002373 __func__, mPortId);
Glenn Kastena07f17c2013-04-23 12:39:37 -07002374 }
Glenn Kastenb0dfd462013-07-10 16:52:47 -07002375 // Run again immediately
2376 return 0;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002377 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002378
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002379 // Can only reference mCblk while locked
2380 int32_t flags = android_atomic_and(
Glenn Kasten96f60d82013-07-12 10:21:18 -07002381 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);
Glenn Kastena47f3162012-11-07 10:13:08 -08002382
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002383 // Check for track invalidation
2384 if (flags & CBLK_INVALID) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002385 // for offloaded tracks restoreTrack_l() will just update the sequence and clear
2386 // AudioSystem cache. We should not exit here but after calling the callback so
2387 // that the upper layers can recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002388 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
Lajos Molnarf1063e22015-04-17 15:19:42 -07002389 status_t status __unused = restoreTrack_l("processAudioBuffer");
2390 // FIXME unused status
Andy Hung53c3b5f2014-12-15 16:42:05 -08002391 // after restoration, continue below to make sure that the loop and buffer events
2392 // are notified because they have been cleared from mCblk->mFlags above.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002393 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002394 }
2395
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002396 bool waitStreamEnd = mState == STATE_STOPPING;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002397 bool active = mState == STATE_ACTIVE;
2398
2399 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
2400 bool newUnderrun = false;
2401 if (flags & CBLK_UNDERRUN) {
2402#if 0
2403 // Currently in shared buffer mode, when the server reaches the end of buffer,
2404 // the track stays active in continuous underrun state. It's up to the application
2405 // to pause or stop the track, or set the position to a new offset within buffer.
2406 // This was some experimental code to auto-pause on underrun. Keeping it here
2407 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
2408 if (mTransfer == TRANSFER_SHARED) {
2409 mState = STATE_PAUSED;
2410 active = false;
2411 }
2412#endif
2413 if (!mInUnderrun) {
2414 mInUnderrun = true;
2415 newUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002416 }
2417 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002418
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002419 // Get current position of server
Andy Hung90e8a972015-11-09 16:42:40 -08002420 Modulo<uint32_t> position(updateAndGetPosition_l());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002421
2422 // Manage marker callback
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002423 bool markerReached = false;
Andy Hung90e8a972015-11-09 16:42:40 -08002424 Modulo<uint32_t> markerPosition(mMarkerPosition);
2425 // uses 32 bit wraparound for comparison with position.
2426 if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002427 mMarkerReached = markerReached = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002428 }
2429
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002430 // Determine number of new position callback(s) that will be needed, while locked
2431 size_t newPosCount = 0;
Andy Hung90e8a972015-11-09 16:42:40 -08002432 Modulo<uint32_t> newPosition(mNewPosition);
2433 uint32_t updatePeriod = mUpdatePeriod;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002434 // FIXME fails for wraparound, need 64 bits
2435 if (updatePeriod > 0 && position >= newPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002436 newPosCount = ((position - newPosition).value() / updatePeriod) + 1;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002437 mNewPosition += updatePeriod * newPosCount;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002438 }
2439
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002440 // Cache other fields that will be needed soon
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002441 uint32_t sampleRate = mSampleRate;
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002442 float speed = mPlaybackRate.mSpeed;
Andy Hunga7f03352015-05-31 21:54:49 -07002443 const uint32_t notificationFrames = mNotificationFramesAct;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002444 if (mRefreshRemaining) {
2445 mRefreshRemaining = false;
2446 mRemainingFrames = notificationFrames;
2447 mRetryOnPartialBuffer = false;
2448 }
2449 size_t misalignment = mProxy->getMisalignment();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002450 uint32_t sequence = mSequence;
Glenn Kasten96f04882013-09-20 09:28:56 -07002451 sp<AudioTrackClientProxy> proxy = mProxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002452
Andy Hung53c3b5f2014-12-15 16:42:05 -08002453 // Determine the number of new loop callback(s) that will be needed, while locked.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002454 uint32_t loopCountNotifications = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -08002455 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
2456
2457 if (mLoopCount > 0) {
2458 int loopCount;
2459 size_t bufferPosition;
2460 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
2461 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
2462 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
2463 mLoopCountNotified = loopCount; // discard any excess notifications
2464 } else if (mLoopCount < 0) {
2465 // FIXME: We're not accurate with notification count and position with infinite looping
2466 // since loopCount from server side will always return -1 (we could decrement it).
2467 size_t bufferPosition = mStaticProxy->getBufferPosition();
2468 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
2469 loopPeriod = mLoopEnd - bufferPosition;
2470 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
2471 size_t bufferPosition = mStaticProxy->getBufferPosition();
2472 loopPeriod = mFrameCount - bufferPosition;
2473 }
2474
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002475 // These fields don't need to be cached, because they are assigned only by set():
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002476 // mTransfer, mCallback, mUserData, mFormat, mFrameSize, mFlags
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002477 // mFlags is also assigned by createTrack_l(), but not the bit we care about.
2478
2479 mLock.unlock();
2480
Andy Hunga7f03352015-05-31 21:54:49 -07002481 // get anchor time to account for callbacks.
2482 const nsecs_t timeBeforeCallbacks = systemTime();
2483
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002484 if (waitStreamEnd) {
Andy Hunga7f03352015-05-31 21:54:49 -07002485 // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread
2486 // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
2487 // (and make sure we don't callback for more data while we're stopping).
2488 // This helps with position, marker notifications, and track invalidation.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002489 struct timespec timeout;
2490 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
2491 timeout.tv_nsec = 0;
2492
Glenn Kasten96f04882013-09-20 09:28:56 -07002493 status_t status = proxy->waitStreamEndDone(&timeout);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002494 switch (status) {
2495 case NO_ERROR:
2496 case DEAD_OBJECT:
2497 case TIMED_OUT:
Andy Hung39609a02015-09-03 16:38:38 -07002498 if (status != DEAD_OBJECT) {
2499 // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop();
2500 // instead, the application should handle the EVENT_NEW_IAUDIOTRACK.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002501 callback->onStreamEnd();
Andy Hung39609a02015-09-03 16:38:38 -07002502 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002503 {
2504 AutoMutex lock(mLock);
2505 // The previously assigned value of waitStreamEnd is no longer valid,
2506 // since the mutex has been unlocked and either the callback handler
2507 // or another thread could have re-started the AudioTrack during that time.
2508 waitStreamEnd = mState == STATE_STOPPING;
2509 if (waitStreamEnd) {
2510 mState = STATE_STOPPED;
Andy Hungc2813e52014-10-16 17:54:34 -07002511 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002512 }
2513 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002514 if (waitStreamEnd && status != DEAD_OBJECT) {
2515 return NS_INACTIVE;
2516 }
2517 break;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002518 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002519 return 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002520 }
2521
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002522 // perform callbacks while unlocked
2523 if (newUnderrun) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002524 callback->onUnderrun();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002525 }
Andy Hung53c3b5f2014-12-15 16:42:05 -08002526 while (loopCountNotifications > 0) {
Andy Hung53c3b5f2014-12-15 16:42:05 -08002527 --loopCountNotifications;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002528 callback->onLoopEnd(mLoopCount > 0 ? loopCountNotifications + mLoopCountNotified : -1);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002529 }
2530 if (flags & CBLK_BUFFER_END) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002531 callback->onBufferEnd();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002532 }
2533 if (markerReached) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002534 callback->onMarker(markerPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002535 }
2536 while (newPosCount > 0) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002537 callback->onNewPos(newPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002538 newPosition += updatePeriod;
2539 newPosCount--;
2540 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002541
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002542 if (mObservedSequence != sequence) {
2543 mObservedSequence = sequence;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002544 callback->onNewIAudioTrack();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002545 // for offloaded tracks, just wait for the upper layers to recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002546 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002547 return NS_INACTIVE;
2548 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002549 }
2550
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002551 // if inactive, then don't run me again until re-started
2552 if (!active) {
2553 return NS_INACTIVE;
Eric Laurent2267ba12011-09-07 11:13:23 -07002554 }
2555
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002556 // Compute the estimated time until the next timed event (position, markers, loops)
2557 // FIXME only for non-compressed audio
2558 uint32_t minFrames = ~0;
2559 if (!markerReached && position < markerPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002560 minFrames = (markerPosition - position).value();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002561 }
2562 if (loopPeriod > 0 && loopPeriod < minFrames) {
Andy Hung2d85f092015-01-07 12:45:13 -08002563 // loopPeriod is already adjusted for actual position.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002564 minFrames = loopPeriod;
2565 }
Andy Hung2d85f092015-01-07 12:45:13 -08002566 if (updatePeriod > 0) {
Andy Hung90e8a972015-11-09 16:42:40 -08002567 minFrames = min(minFrames, (newPosition - position).value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002568 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002569
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002570 // If > 0, poll periodically to recover from a stuck server. A good value is 2.
2571 static const uint32_t kPoll = 0;
2572 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
2573 minFrames = kPoll * notificationFrames;
2574 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002575
Andy Hunga7f03352015-05-31 21:54:49 -07002576 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
2577 static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL;
2578 const nsecs_t timeAfterCallbacks = systemTime();
2579
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002580 // Convert frame units to time units
2581 nsecs_t ns = NS_WHENEVER;
2582 if (minFrames != (uint32_t) ~0) {
jiabinc7bb8322017-09-06 18:20:11 -07002583 // AudioFlinger consumption of client data may be irregular when coming out of device
2584 // standby since the kernel buffers require filling. This is throttled to no more than 2x
2585 // the expected rate in the MixerThread. Hence, we reduce the estimated time to wait by one
2586 // half (but no more than half a second) to improve callback accuracy during these temporary
2587 // data surges.
2588 const nsecs_t estimatedNs = framesToNanoseconds(minFrames, sampleRate, speed);
2589 constexpr nsecs_t maxThrottleCompensationNs = 500000000LL;
2590 ns = estimatedNs - min(estimatedNs / 2, maxThrottleCompensationNs) + kWaitPeriodNs;
Andy Hunga7f03352015-05-31 21:54:49 -07002591 ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time
2592 // TODO: Should we warn if the callback time is too long?
2593 if (ns < 0) ns = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002594 }
2595
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002596 // If not supplying data by EVENT_MORE_DATA or EVENT_CAN_WRITE_MORE_DATA, then we're done
2597 if (mTransfer != TRANSFER_CALLBACK && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002598 return ns;
2599 }
2600
Andy Hunga7f03352015-05-31 21:54:49 -07002601 // EVENT_MORE_DATA callback handling.
2602 // Timing for linear pcm audio data formats can be derived directly from the
2603 // buffer fill level.
2604 // Timing for compressed data is not directly available from the buffer fill level,
2605 // rather indirectly from waiting for blocking mode callbacks or waiting for obtain()
2606 // to return a certain fill level.
2607
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002608 struct timespec timeout;
2609 const struct timespec *requested = &ClientProxy::kForever;
2610 if (ns != NS_WHENEVER) {
2611 timeout.tv_sec = ns / 1000000000LL;
2612 timeout.tv_nsec = ns % 1000000000LL;
Andy Hungfb8ede22018-09-12 19:03:24 -07002613 ALOGV("%s(%d): timeout %ld.%03d",
Eric Laurent973db022018-11-20 14:54:31 -08002614 __func__, mPortId, timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002615 requested = &timeout;
2616 }
2617
Andy Hungea2b9c02016-02-12 17:06:53 -08002618 size_t writtenFrames = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002619 while (mRemainingFrames > 0) {
2620
2621 Buffer audioBuffer;
2622 audioBuffer.frameCount = mRemainingFrames;
2623 size_t nonContig;
2624 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
2625 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
Andy Hungfb8ede22018-09-12 19:03:24 -07002626 "%s(%d): obtainBuffer() err=%d frameCount=%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002627 __func__, mPortId, err, audioBuffer.frameCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002628 requested = &ClientProxy::kNonBlocking;
2629 size_t avail = audioBuffer.frameCount + nonContig;
Andy Hungfb8ede22018-09-12 19:03:24 -07002630 ALOGV("%s(%d): obtainBuffer(%u) returned %zu = %zu + %zu err %d",
Eric Laurent973db022018-11-20 14:54:31 -08002631 __func__, mPortId, mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002632 if (err != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002633 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
2634 (isOffloaded() && (err == DEAD_OBJECT))) {
Glenn Kasten606fbc12015-10-22 15:28:15 -07002635 // FIXME bug 25195759
2636 return 1000000;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002637 }
Andy Hungfb8ede22018-09-12 19:03:24 -07002638 ALOGE("%s(%d): Error %d obtaining an audio buffer, giving up.",
Eric Laurent973db022018-11-20 14:54:31 -08002639 __func__, mPortId, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002640 return NS_NEVER;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002641 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002642
Phil Burkfdb3c072016-02-09 10:47:02 -08002643 if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002644 mRetryOnPartialBuffer = false;
2645 if (avail < mRemainingFrames) {
Andy Hunga7f03352015-05-31 21:54:49 -07002646 if (ns > 0) { // account for obtain time
2647 const nsecs_t timeNow = systemTime();
2648 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2649 }
Andy Hungeb6e6502019-04-29 13:47:40 -07002650
2651 // delayNs is first computed by the additional frames required in the buffer.
2652 nsecs_t delayNs = framesToNanoseconds(
2653 mRemainingFrames - avail, sampleRate, speed);
2654
2655 // afNs is the AudioFlinger mixer period in ns.
2656 const nsecs_t afNs = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2657
2658 // If the AudioTrack is double buffered based on the AudioFlinger mixer period,
2659 // we may have a race if we wait based on the number of frames desired.
2660 // This is a possible issue with resampling and AAudio.
2661 //
2662 // The granularity of audioflinger processing is one mixer period; if
2663 // our wait time is less than one mixer period, wait at most half the period.
2664 if (delayNs < afNs) {
2665 delayNs = std::min(delayNs, afNs / 2);
2666 }
2667
2668 // adjust our ns wait by delayNs.
2669 if (ns < 0 /* NS_WHENEVER */ || delayNs < ns) {
2670 ns = delayNs;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002671 }
2672 return ns;
2673 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07002674 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002675
Atneya Nair03079272022-01-18 17:03:14 -05002676 size_t reqSize = audioBuffer.size();
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002677 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2678 // when notifying client it can write more data, pass the total size that can be
2679 // written in the next write() call, since it's not passed through the callback
Atneya Nair03079272022-01-18 17:03:14 -05002680 audioBuffer.mSize += nonContig;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002681 }
Atneya Nairc2dd1272021-10-26 19:39:51 -04002682 const size_t writtenSize = (mTransfer == TRANSFER_CALLBACK)
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002683 ? callback->onMoreData(audioBuffer)
2684 : callback->onCanWriteMoreData(audioBuffer);
Jiabin Huang447cea72020-07-28 22:35:18 +00002685 // Validate on returned size
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002686 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002687 ALOGE("%s(%d): EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
Eric Laurent973db022018-11-20 14:54:31 -08002688 __func__, mPortId, reqSize, ssize_t(writtenSize));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002689 return NS_NEVER;
2690 }
2691
2692 if (writtenSize == 0) {
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002693 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2694 // The callback EVENT_CAN_WRITE_MORE_DATA was processed in the JNI of
2695 // android.media.AudioTrack. The JNI is not using the callback to provide data,
2696 // it only signals to the Java client that it can provide more data, which
2697 // this track is read to accept now.
2698 // The playback thread will be awaken at the next ::write()
2699 return NS_WHENEVER;
2700 }
The Android Open Source Project8555d082009-03-05 14:34:35 -08002701 // The callback is done filling buffers
2702 // Keep this thread going to handle timed events and
2703 // still try to get more data in intervals of WAIT_PERIOD_MS
2704 // but don't just loop and block the CPU, so wait
Andy Hunga7f03352015-05-31 21:54:49 -07002705
2706 // mCbf(EVENT_MORE_DATA, ...) might either
2707 // (1) Block until it can fill the buffer, returning 0 size on EOS.
2708 // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS.
2709 // (3) Return 0 size when no data is available, does not wait for more data.
2710 //
2711 // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer.
2712 // We try to compute the wait time to avoid a tight sleep-wait cycle,
2713 // especially for case (3).
2714 //
2715 // The decision to support (1) and (2) affect the sizing of mRemainingFrames
2716 // and this loop; whereas for case (3) we could simply check once with the full
2717 // buffer size and skip the loop entirely.
2718
2719 nsecs_t myns;
Phil Burkfdb3c072016-02-09 10:47:02 -08002720 if (audio_has_proportional_frames(mFormat)) {
Andy Hunga7f03352015-05-31 21:54:49 -07002721 // time to wait based on buffer occupancy
2722 const nsecs_t datans = mRemainingFrames <= avail ? 0 :
2723 framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
2724 // audio flinger thread buffer size (TODO: adjust for fast tracks)
Glenn Kastenea38ee72016-04-18 11:08:01 -07002725 // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks.
Andy Hunga7f03352015-05-31 21:54:49 -07002726 const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2727 // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0.
2728 myns = datans + (afns / 2);
2729 } else {
2730 // FIXME: This could ping quite a bit if the buffer isn't full.
2731 // Note that when mState is stopping we waitStreamEnd, so it never gets here.
2732 myns = kWaitPeriodNs;
2733 }
2734 if (ns > 0) { // account for obtain and callback time
2735 const nsecs_t timeNow = systemTime();
2736 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2737 }
2738 if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
2739 ns = myns;
2740 }
2741 return ns;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002742 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002743
Atneya Nairc2dd1272021-10-26 19:39:51 -04002744 // releaseBuffer reads from audioBuffer.size
Atneya Nair03079272022-01-18 17:03:14 -05002745 audioBuffer.mSize = writtenSize;
Atneya Nairc2dd1272021-10-26 19:39:51 -04002746
Glenn Kasten138d6f92015-03-20 10:54:51 -07002747 size_t releasedFrames = writtenSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002748 audioBuffer.frameCount = releasedFrames;
2749 mRemainingFrames -= releasedFrames;
2750 if (misalignment >= releasedFrames) {
2751 misalignment -= releasedFrames;
2752 } else {
2753 misalignment = 0;
2754 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002755
2756 releaseBuffer(&audioBuffer);
Andy Hungea2b9c02016-02-12 17:06:53 -08002757 writtenFrames += releasedFrames;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002758
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002759 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
2760 // if callback doesn't like to accept the full chunk
2761 if (writtenSize < reqSize) {
2762 continue;
2763 }
2764
2765 // There could be enough non-contiguous frames available to satisfy the remaining request
2766 if (mRemainingFrames <= nonContig) {
2767 continue;
2768 }
2769
2770#if 0
2771 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
2772 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
2773 // that total to a sum == notificationFrames.
2774 if (0 < misalignment && misalignment <= mRemainingFrames) {
2775 mRemainingFrames = misalignment;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002776 return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002777 }
2778#endif
2779
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002780 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002781 if (writtenFrames > 0) {
2782 AutoMutex lock(mLock);
2783 mFramesWritten += writtenFrames;
2784 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002785 mRemainingFrames = notificationFrames;
2786 mRetryOnPartialBuffer = true;
2787
2788 // A lot has transpired since ns was calculated, so run again immediately and re-calculate
2789 return 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002790}
2791
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002792status_t AudioTrack::restoreTrack_l(const char *from)
Eric Laurent1703cdf2011-03-07 14:52:59 -08002793{
Andy Hungb68f5eb2019-12-03 16:49:17 -08002794 status_t result = NO_ERROR; // logged: make sure to set this before returning.
2795 const int64_t beginNs = systemTime();
Andy Hunga6b27032020-04-27 10:34:24 -07002796 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -08002797 mediametrics::LogItem(mMetricsId)
2798 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_RESTORE)
Andy Hungea840382020-05-05 21:50:17 -07002799 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08002800 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
2801 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
2802 .set(AMEDIAMETRICS_PROP_WHERE, from)
2803 .record(); });
2804
Andy Hungfb8ede22018-09-12 19:03:24 -07002805 ALOGW("%s(%d): dead IAudioTrack, %s, creating a new one from %s()",
Eric Laurent973db022018-11-20 14:54:31 -08002806 __func__, mPortId, isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002807 ++mSequence;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002808
Glenn Kastena47f3162012-11-07 10:13:08 -08002809 // refresh the audio configuration cache in this process to make sure we get new
Glenn Kastend2d089f2014-11-05 11:48:12 -08002810 // output parameters and new IAudioFlinger in createTrack_l()
Glenn Kastena47f3162012-11-07 10:13:08 -08002811 AudioSystem::clearAudioConfigCache();
Eric Laurent9f6530f2011-08-30 10:18:54 -07002812
Ronghua Wufaeb0f22015-05-21 12:20:21 -07002813 if (isOffloadedOrDirect_l() || mDoNotReconnect) {
Andy Hung1f1db832015-06-08 13:26:10 -07002814 // FIXME re-creation of offloaded and direct tracks is not yet implemented;
2815 // reconsider enabling for linear PCM encodings when position can be preserved.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002816 result = DEAD_OBJECT;
2817 return result;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002818 }
2819
Phil Burk2812d9e2016-01-04 10:34:30 -08002820 // Save so we can return count since creation.
2821 mUnderrunCountOffset = getUnderrunCount_l();
2822
Glenn Kasten200092b2014-08-15 15:13:30 -07002823 // save the old static buffer position
Andy Hungf20a4e92016-08-15 19:10:34 -07002824 uint32_t staticPosition = 0;
Andy Hung4ede21d2014-12-12 15:37:34 -08002825 size_t bufferPosition = 0;
2826 int loopCount = 0;
2827 if (mStaticProxy != 0) {
2828 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
Andy Hungf20a4e92016-08-15 19:10:34 -07002829 staticPosition = mStaticProxy->getPosition().unsignedValue();
Andy Hung4ede21d2014-12-12 15:37:34 -08002830 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002831
Andy Hung3c7f47a2021-03-16 17:30:09 -07002832 // save the old startThreshold and framecount
2833 const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
2834 const uint32_t originalFrameCount = mProxy->frameCount();
2835
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002836 // See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
2837 // causes a lot of churn on the service side, and it can reject starting
2838 // playback of a previously created track. May also apply to other cases.
2839 const int INITIAL_RETRIES = 3;
2840 int retries = INITIAL_RETRIES;
2841retry:
2842 if (retries < INITIAL_RETRIES) {
2843 // See the comment for clearAudioConfigCache at the start of the function.
2844 AudioSystem::clearAudioConfigCache();
2845 }
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08002846 mFlags = mOrigFlags;
2847
Glenn Kasten200092b2014-08-15 15:13:30 -07002848 // If a new IAudioTrack is successfully created, createTrack_l() will modify the
Glenn Kastena47f3162012-11-07 10:13:08 -08002849 // following member variables: mAudioTrack, mCblkMemory and mCblk.
Glenn Kasten200092b2014-08-15 15:13:30 -07002850 // It will also delete the strong references on previous IAudioTrack and IMemory.
2851 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002852 result = createTrack_l();
Eric Laurentcc21e4f2013-10-16 15:12:32 -07002853
Eric Laurent6ec546d2018-10-10 16:52:14 -07002854 if (result == NO_ERROR) {
Andy Hungd7bd69e2015-07-24 07:52:41 -07002855 // take the frames that will be lost by track recreation into account in saved position
2856 // For streaming tracks, this is the amount we obtained from the user/client
2857 // (not the number actually consumed at the server - those are already lost).
2858 if (mStaticProxy == 0) {
2859 mPosition = mReleased;
2860 }
Andy Hung4ede21d2014-12-12 15:37:34 -08002861 // Continue playback from last known position and restore loop.
2862 if (mStaticProxy != 0) {
2863 if (loopCount != 0) {
2864 mStaticProxy->setBufferPositionAndLoop(bufferPosition,
2865 mLoopStart, mLoopEnd, loopCount);
2866 } else {
2867 mStaticProxy->setBufferPosition(bufferPosition);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002868 if (bufferPosition == mFrameCount) {
Eric Laurent973db022018-11-20 14:54:31 -08002869 ALOGD("%s(%d): restoring track at end of static buffer", __func__, mPortId);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002870 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002871 }
2872 }
Andy Hung4ef88d72017-02-21 19:47:53 -08002873 // restore volume handler
Andy Hung39399b62017-04-21 15:07:45 -07002874 mVolumeHandler->forall([this](const VolumeShaper &shaper) -> VolumeShaper::Status {
2875 sp<VolumeShaper::Operation> operationToEnd =
2876 new VolumeShaper::Operation(shaper.mOperation);
Andy Hung4ef88d72017-02-21 19:47:53 -08002877 // TODO: Ideally we would restore to the exact xOffset position
2878 // as returned by getVolumeShaperState(), but we don't have that
2879 // information when restoring at the client unless we periodically poll
2880 // the server or create shared memory state.
2881 //
Andy Hung39399b62017-04-21 15:07:45 -07002882 // For now, we simply advance to the end of the VolumeShaper effect
2883 // if it has been started.
2884 if (shaper.isStarted()) {
Andy Hungf3702642017-05-05 17:33:32 -07002885 operationToEnd->setNormalizedTime(1.f);
Andy Hung39399b62017-04-21 15:07:45 -07002886 }
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002887 media::VolumeShaperConfiguration config;
2888 shaper.mConfiguration->writeToParcelable(&config);
2889 media::VolumeShaperOperation operation;
2890 operationToEnd->writeToParcelable(&operation);
2891 status_t status;
2892 mAudioTrack->applyVolumeShaper(config, operation, &status);
2893 return status;
Andy Hung4ef88d72017-02-21 19:47:53 -08002894 });
2895
Andy Hung3c7f47a2021-03-16 17:30:09 -07002896 // restore the original start threshold if different than frameCount.
2897 if (originalStartThresholdInFrames != originalFrameCount) {
2898 // Note: mProxy->setStartThresholdInFrames() call is in the Proxy
2899 // and does not trigger a restart.
2900 // (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
2901 // Any start would be triggered on the mState == ACTIVE check below.
2902 const uint32_t currentThreshold =
2903 mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
2904 ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
2905 "%s(%d) startThresholdInFrames changing from %u to %u",
2906 __func__, mPortId, originalStartThresholdInFrames, currentThreshold);
2907 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002908 if (mState == STATE_ACTIVE) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002909 mAudioTrack->start(&result);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002910 }
Andy Hungf20a4e92016-08-15 19:10:34 -07002911 // server resets to zero so we offset
2912 mFramesWrittenServerOffset =
2913 mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten;
2914 mFramesWrittenAtRestore = mFramesWrittenServerOffset;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002915 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002916 if (result != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08002917 ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002918 if (--retries > 0) {
Eric Laurent6ec546d2018-10-10 16:52:14 -07002919 // leave time for an eventual race condition to clear before retrying
2920 usleep(500000);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002921 goto retry;
2922 }
Eric Laurent6ec546d2018-10-10 16:52:14 -07002923 // if no retries left, set invalid bit to force restoring at next occasion
2924 // and avoid inconsistent active state on client and server sides
2925 if (mCblk != nullptr) {
2926 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
2927 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002928 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002929 return result;
2930}
2931
Andy Hung90e8a972015-11-09 16:42:40 -08002932Modulo<uint32_t> AudioTrack::updateAndGetPosition_l()
Glenn Kasten200092b2014-08-15 15:13:30 -07002933{
2934 // This is the sole place to read server consumed frames
Andy Hung90e8a972015-11-09 16:42:40 -08002935 Modulo<uint32_t> newServer(mProxy->getPosition());
2936 const int32_t delta = (newServer - mServer).signedValue();
Glenn Kasten200092b2014-08-15 15:13:30 -07002937 // TODO There is controversy about whether there can be "negative jitter" in server position.
2938 // This should be investigated further, and if possible, it should be addressed.
2939 // A more definite failure mode is infrequent polling by client.
2940 // One could call (void)getPosition_l() in releaseBuffer(),
2941 // so mReleased and mPosition are always lock-step as best possible.
2942 // That should ensure delta never goes negative for infrequent polling
2943 // unless the server has more than 2^31 frames in its buffer,
2944 // in which case the use of uint32_t for these counters has bigger issues.
Andy Hung90e8a972015-11-09 16:42:40 -08002945 ALOGE_IF(delta < 0,
Andy Hungfb8ede22018-09-12 19:03:24 -07002946 "%s(%d): detected illegal retrograde motion by the server: mServer advanced by %d",
Eric Laurent973db022018-11-20 14:54:31 -08002947 __func__, mPortId, delta);
Chad Brubaker039c27a2015-09-23 15:17:29 -07002948 mServer = newServer;
Andy Hung90e8a972015-11-09 16:42:40 -08002949 if (delta > 0) { // avoid retrograde
2950 mPosition += delta;
2951 }
2952 return mPosition;
Glenn Kasten200092b2014-08-15 15:13:30 -07002953}
2954
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002955bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed)
Andy Hung8edb8dc2015-03-26 19:13:55 -07002956{
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002957 updateLatency_l();
Andy Hung8edb8dc2015-03-26 19:13:55 -07002958 // applicable for mixing tracks only (not offloaded or direct)
2959 if (mStaticProxy != 0) {
2960 return true; // static tracks do not have issues with buffer sizing.
2961 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07002962 const size_t minFrameCount =
Eric Laurent21da6472017-11-09 16:29:26 -08002963 AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
2964 sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002965 const bool allowed = mFrameCount >= minFrameCount;
2966 ALOGD_IF(!allowed,
Andy Hungfb8ede22018-09-12 19:03:24 -07002967 "%s(%d): denied "
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002968 "mAfLatency:%u mAfFrameCount:%zu mAfSampleRate:%u sampleRate:%u speed:%f "
2969 "mFrameCount:%zu < minFrameCount:%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002970 __func__, mPortId,
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002971 mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed,
Andy Hung8edb8dc2015-03-26 19:13:55 -07002972 mFrameCount, minFrameCount);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002973 return allowed;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002974}
2975
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00002976status_t AudioTrack::setParameters(const String8& keyValuePairs)
2977{
2978 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002979 status_t status;
2980 mAudioTrack->setParameters(keyValuePairs.c_str(), &status);
2981 return status;
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00002982}
2983
Dean Wheatleya70eef72018-01-04 14:23:50 +11002984status_t AudioTrack::selectPresentation(int presentationId, int programId)
2985{
2986 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08002987 AudioParameter param = AudioParameter();
2988 param.addInt(String8(AudioParameter::keyPresentationId), presentationId);
2989 param.addInt(String8(AudioParameter::keyProgramId), programId);
2990 ALOGV("%s(%d): PresentationId/ProgramId[%s]",
2991 __func__, mPortId, param.toString().string());
2992
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002993 status_t status;
2994 mAudioTrack->setParameters(param.toString().c_str(), &status);
2995 return status;
Dean Wheatleya70eef72018-01-04 14:23:50 +11002996}
2997
Andy Hung9fc8b5c2017-01-24 13:36:48 -08002998VolumeShaper::Status AudioTrack::applyVolumeShaper(
2999 const sp<VolumeShaper::Configuration>& configuration,
3000 const sp<VolumeShaper::Operation>& operation)
3001{
3002 AutoMutex lock(mLock);
Andy Hung4ef88d72017-02-21 19:47:53 -08003003 mVolumeHandler->setIdIfNecessary(configuration);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003004 media::VolumeShaperConfiguration config;
3005 configuration->writeToParcelable(&config);
3006 media::VolumeShaperOperation op;
3007 operation->writeToParcelable(&op);
3008 VolumeShaper::Status status;
3009 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003010
3011 if (status == DEAD_OBJECT) {
3012 if (restoreTrack_l("applyVolumeShaper") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003013 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003014 }
3015 }
Andy Hung4ef88d72017-02-21 19:47:53 -08003016 if (status >= 0) {
3017 // save VolumeShaper for restore
3018 mVolumeHandler->applyVolumeShaper(configuration, operation);
Andy Hung39399b62017-04-21 15:07:45 -07003019 if (mState == STATE_ACTIVE || mState == STATE_STOPPING) {
3020 mVolumeHandler->setStarted();
3021 }
3022 } else {
3023 // warn only if not an expected restore failure.
3024 ALOGW_IF(!((isOffloadedOrDirect_l() || mDoNotReconnect) && status == DEAD_OBJECT),
Eric Laurent973db022018-11-20 14:54:31 -08003025 "%s(%d): applyVolumeShaper failed: %d", __func__, mPortId, status);
Andy Hung4ef88d72017-02-21 19:47:53 -08003026 }
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003027 return status;
3028}
3029
3030sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id)
3031{
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003032 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003033 std::optional<media::VolumeShaperState> vss;
3034 mAudioTrack->getVolumeShaperState(id, &vss);
3035 sp<VolumeShaper::State> state;
3036 if (vss.has_value()) {
3037 state = new VolumeShaper::State();
3038 state->readFromParcelable(vss.value());
3039 }
Andy Hung39399b62017-04-21 15:07:45 -07003040 if (state.get() == nullptr && (mCblk->mFlags & CBLK_INVALID) != 0) {
3041 if (restoreTrack_l("getVolumeShaperState") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003042 mAudioTrack->getVolumeShaperState(id, &vss);
3043 if (vss.has_value()) {
3044 state = new VolumeShaper::State();
3045 state->readFromParcelable(vss.value());
3046 }
Andy Hung39399b62017-04-21 15:07:45 -07003047 }
3048 }
3049 return state;
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003050}
3051
Andy Hungea2b9c02016-02-12 17:06:53 -08003052status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp)
3053{
3054 if (timestamp == nullptr) {
3055 return BAD_VALUE;
3056 }
3057 AutoMutex lock(mLock);
Andy Hunge13f8a62016-03-30 14:20:42 -07003058 return getTimestamp_l(timestamp);
3059}
3060
3061status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp)
3062{
Andy Hungea2b9c02016-02-12 17:06:53 -08003063 if (mCblk->mFlags & CBLK_INVALID) {
3064 const status_t status = restoreTrack_l("getTimestampExtended");
3065 if (status != OK) {
3066 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3067 // recommending that the track be recreated.
3068 return DEAD_OBJECT;
3069 }
3070 }
3071 // check for offloaded/direct here in case restoring somehow changed those flags.
3072 if (isOffloadedOrDirect_l()) {
3073 return INVALID_OPERATION; // not supported
3074 }
3075 status_t status = mProxy->getTimestamp(timestamp);
Andy Hungfb8ede22018-09-12 19:03:24 -07003076 LOG_ALWAYS_FATAL_IF(status != OK, "%s(%d): status %d not allowed from proxy getTimestamp",
Eric Laurent973db022018-11-20 14:54:31 -08003077 __func__, mPortId, status);
Andy Hungea2b9c02016-02-12 17:06:53 -08003078 bool found = false;
Andy Hunge1e98462016-04-12 10:18:51 -07003079 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten;
3080 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
3081 // server side frame offset in case AudioTrack has been restored.
3082 for (int i = ExtendedTimestamp::LOCATION_SERVER;
3083 i < ExtendedTimestamp::LOCATION_MAX; ++i) {
3084 if (timestamp->mTimeNs[i] >= 0) {
3085 // apply server offset (frames flushed is ignored
3086 // so we don't report the jump when the flush occurs).
3087 timestamp->mPosition[i] += mFramesWrittenServerOffset;
3088 found = true;
Andy Hungea2b9c02016-02-12 17:06:53 -08003089 }
3090 }
3091 return found ? OK : WOULD_BLOCK;
3092}
3093
Glenn Kastence703742013-07-19 16:33:58 -07003094status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
3095{
Glenn Kasten53cec222013-08-29 09:01:02 -07003096 AutoMutex lock(mLock);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003097 return getTimestamp_l(timestamp);
3098}
Phil Burk1b420972015-04-22 10:52:21 -07003099
Andy Hung65ffdfc2016-10-10 15:52:11 -07003100status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp)
3101{
Phil Burk1b420972015-04-22 10:52:21 -07003102 bool previousTimestampValid = mPreviousTimestampValid;
3103 // Set false here to cover all the error return cases.
3104 mPreviousTimestampValid = false;
3105
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003106 switch (mState) {
3107 case STATE_ACTIVE:
3108 case STATE_PAUSED:
3109 break; // handle below
3110 case STATE_FLUSHED:
3111 case STATE_STOPPED:
3112 return WOULD_BLOCK;
3113 case STATE_STOPPING:
3114 case STATE_PAUSED_STOPPING:
3115 if (!isOffloaded_l()) {
3116 return INVALID_OPERATION;
3117 }
3118 break; // offloaded tracks handled below
3119 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003120 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in getTimestamp(): %d",
Eric Laurent973db022018-11-20 14:54:31 -08003121 __func__, mPortId, mState);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003122 break;
Glenn Kastenfe346c72013-08-30 13:28:22 -07003123 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003124
Eric Laurent275e8e92014-11-30 15:14:47 -08003125 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung6653c932015-06-08 13:27:48 -07003126 const status_t status = restoreTrack_l("getTimestamp");
3127 if (status != OK) {
3128 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3129 // recommending that the track be recreated.
3130 return DEAD_OBJECT;
3131 }
Eric Laurent275e8e92014-11-30 15:14:47 -08003132 }
3133
Glenn Kasten200092b2014-08-15 15:13:30 -07003134 // The presented frame count must always lag behind the consumed frame count.
3135 // To avoid a race, read the presented frames first. This ensures that presented <= consumed.
Andy Hung6ae58432016-02-16 18:32:24 -08003136
3137 status_t status;
Andy Hung818e7a32016-02-16 18:08:07 -08003138 if (isOffloadedOrDirect_l()) {
Andy Hung6ae58432016-02-16 18:32:24 -08003139 // use Binder to get timestamp
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003140 media::AudioTimestampInternal ts;
3141 mAudioTrack->getTimestamp(&ts, &status);
3142 if (status == OK) {
Andy Hung973638a2020-12-08 20:47:45 -08003143 timestamp = VALUE_OR_FATAL(aidl2legacy_AudioTimestampInternal_AudioTimestamp(ts));
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003144 }
Andy Hung6ae58432016-02-16 18:32:24 -08003145 } else {
3146 // read timestamp from shared memory
3147 ExtendedTimestamp ets;
3148 status = mProxy->getTimestamp(&ets);
3149 if (status == OK) {
Andy Hungb01faa32016-04-27 12:51:32 -07003150 ExtendedTimestamp::Location location;
3151 status = ets.getBestTimestamp(&timestamp, &location);
3152
3153 if (status == OK) {
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003154 updateLatency_l();
Andy Hungb01faa32016-04-27 12:51:32 -07003155 // It is possible that the best location has moved from the kernel to the server.
3156 // In this case we adjust the position from the previous computed latency.
3157 if (location == ExtendedTimestamp::LOCATION_SERVER) {
3158 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL,
Andy Hungfb8ede22018-09-12 19:03:24 -07003159 "%s(%d): location moved from kernel to server",
Eric Laurent973db022018-11-20 14:54:31 -08003160 __func__, mPortId);
Andy Hung07eee802016-06-21 16:47:16 -07003161 // check that the last kernel OK time info exists and the positions
3162 // are valid (if they predate the current track, the positions may
3163 // be zero or negative).
Andy Hungb01faa32016-04-27 12:51:32 -07003164 const int64_t frames =
Andy Hung6d7b1192016-05-07 22:59:48 -07003165 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
Andy Hung07eee802016-06-21 16:47:16 -07003166 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 ||
3167 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 ||
3168 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0)
Andy Hung6d7b1192016-05-07 22:59:48 -07003169 ?
3170 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed
3171 / 1000)
3172 :
3173 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3174 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]);
Andy Hungfb8ede22018-09-12 19:03:24 -07003175 ALOGV("%s(%d): frame adjustment:%lld timestamp:%s",
Eric Laurent973db022018-11-20 14:54:31 -08003176 __func__, mPortId, (long long)frames, ets.toString().c_str());
Andy Hungb01faa32016-04-27 12:51:32 -07003177 if (frames >= ets.mPosition[location]) {
3178 timestamp.mPosition = 0;
3179 } else {
3180 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames);
3181 }
Andy Hung69488c42016-05-16 18:43:33 -07003182 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) {
3183 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER,
Andy Hungfb8ede22018-09-12 19:03:24 -07003184 "%s(%d): location moved from server to kernel",
Eric Laurent973db022018-11-20 14:54:31 -08003185 __func__, mPortId);
Andy Hung98731a22019-04-08 19:19:07 -07003186
3187 if (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER] ==
3188 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL]) {
3189 // In Q, we don't return errors as an invalid time
3190 // but instead we leave the last kernel good timestamp alone.
3191 //
3192 // If server is identical to kernel, the device data pipeline is idle.
3193 // A better start time is now. The retrograde check ensures
3194 // timestamp monotonicity.
3195 const int64_t nowNs = systemTime();
Andy Hungcf3b7152019-04-19 18:29:21 -07003196 if (!mTimestampStallReported) {
3197 ALOGD("%s(%d): device stall time corrected using current time %lld",
3198 __func__, mPortId, (long long)nowNs);
3199 mTimestampStallReported = true;
3200 }
Andy Hung98731a22019-04-08 19:19:07 -07003201 timestamp.mTime = convertNsToTimespec(nowNs);
Andy Hungcf3b7152019-04-19 18:29:21 -07003202 } else {
3203 mTimestampStallReported = false;
Andy Hung98731a22019-04-08 19:19:07 -07003204 }
Andy Hungb01faa32016-04-27 12:51:32 -07003205 }
Andy Hung5d313802016-10-10 15:09:39 -07003206
3207 // We update the timestamp time even when paused.
3208 if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) {
3209 const int64_t now = systemTime();
Andy Hung2b01f002017-07-05 12:01:36 -07003210 const int64_t at = audio_utils_ns_from_timespec(&timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003211 const int64_t lag =
3212 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
3213 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0)
3214 ? int64_t(mAfLatency * 1000000LL)
3215 : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3216 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK])
3217 * NANOS_PER_SECOND / mSampleRate;
3218 const int64_t limit = now - lag; // no earlier than this limit
3219 if (at < limit) {
3220 ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld",
3221 (long long)lag, (long long)at, (long long)limit);
Andy Hungffa36952017-08-17 10:41:51 -07003222 timestamp.mTime = convertNsToTimespec(limit);
Andy Hung5d313802016-10-10 15:09:39 -07003223 }
3224 }
Andy Hungb01faa32016-04-27 12:51:32 -07003225 mPreviousLocation = location;
3226 } else {
3227 // right after AudioTrack is started, one may not find a timestamp
Eric Laurent973db022018-11-20 14:54:31 -08003228 ALOGV("%s(%d): getBestTimestamp did not find timestamp", __func__, mPortId);
Andy Hungb01faa32016-04-27 12:51:32 -07003229 }
Andy Hung6ae58432016-02-16 18:32:24 -08003230 }
3231 if (status == INVALID_OPERATION) {
Andy Hungf20a4e92016-08-15 19:10:34 -07003232 // INVALID_OPERATION occurs when no timestamp has been issued by the server;
3233 // other failures are signaled by a negative time.
3234 // If we come out of FLUSHED or STOPPED where the position is known
3235 // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of
3236 // "zero" for NuPlayer). We don't convert for track restoration as position
3237 // does not reset.
Andy Hungfb8ede22018-09-12 19:03:24 -07003238 ALOGV("%s(%d): timestamp server offset:%lld restore frames:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003239 __func__, mPortId,
Andy Hungf20a4e92016-08-15 19:10:34 -07003240 (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore);
3241 if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) {
3242 status = WOULD_BLOCK;
3243 }
Andy Hung6ae58432016-02-16 18:32:24 -08003244 }
3245 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003246 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08003247 ALOGV_IF(status != WOULD_BLOCK, "%s(%d): getTimestamp error:%#x", __func__, mPortId, status);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003248 return status;
3249 }
3250 if (isOffloadedOrDirect_l()) {
3251 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) {
3252 // use cached paused position in case another offloaded track is running.
3253 timestamp.mPosition = mPausedPosition;
3254 clock_gettime(CLOCK_MONOTONIC, &timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003255 // TODO: adjust for delay
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003256 return NO_ERROR;
3257 }
3258
3259 // Check whether a pending flush or stop has completed, as those commands may
Andy Hungc8e09c62015-06-03 23:43:36 -07003260 // be asynchronous or return near finish or exhibit glitchy behavior.
3261 //
3262 // Originally this showed up as the first timestamp being a continuation of
3263 // the previous song under gapless playback.
3264 // However, we sometimes see zero timestamps, then a glitch of
3265 // the previous song's position, and then correct timestamps afterwards.
Andy Hungffa36952017-08-17 10:41:51 -07003266 if (mStartFromZeroUs != 0 && mSampleRate != 0) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003267 static const int kTimeJitterUs = 100000; // 100 ms
3268 static const int k1SecUs = 1000000;
3269
3270 const int64_t timeNow = getNowUs();
3271
Andy Hungffa36952017-08-17 10:41:51 -07003272 if (timeNow < mStartFromZeroUs + k1SecUs) { // within first second of starting
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003273 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003274 if (timestampTimeUs < mStartFromZeroUs) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003275 return WOULD_BLOCK; // stale timestamp time, occurs before start.
3276 }
Andy Hungffa36952017-08-17 10:41:51 -07003277 const int64_t deltaTimeUs = timestampTimeUs - mStartFromZeroUs;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003278 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07003279 / ((double)mSampleRate * mPlaybackRate.mSpeed);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003280
3281 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
3282 // Verify that the counter can't count faster than the sample rate
Andy Hungc8e09c62015-06-03 23:43:36 -07003283 // since the start time. If greater, then that means we may have failed
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003284 // to completely flush or stop the previous playing track.
Andy Hungc8e09c62015-06-03 23:43:36 -07003285 ALOGW_IF(!mTimestampStartupGlitchReported,
Andy Hungfb8ede22018-09-12 19:03:24 -07003286 "%s(%d): startup glitch detected"
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003287 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)",
Eric Laurent973db022018-11-20 14:54:31 -08003288 __func__, mPortId,
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003289 (long long)deltaTimeUs, (long long)deltaPositionByUs,
3290 timestamp.mPosition);
Andy Hungc8e09c62015-06-03 23:43:36 -07003291 mTimestampStartupGlitchReported = true;
3292 if (previousTimestampValid
3293 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) {
3294 timestamp = mPreviousTimestamp;
3295 mPreviousTimestampValid = true;
3296 return NO_ERROR;
3297 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003298 return WOULD_BLOCK;
3299 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003300 if (deltaPositionByUs != 0) {
Andy Hungffa36952017-08-17 10:41:51 -07003301 mStartFromZeroUs = 0; // don't check again, we got valid nonzero position.
Andy Hungc8e09c62015-06-03 23:43:36 -07003302 }
3303 } else {
Andy Hungffa36952017-08-17 10:41:51 -07003304 mStartFromZeroUs = 0; // don't check again, start time expired.
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003305 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003306 mTimestampStartupGlitchReported = false;
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003307 }
3308 } else {
Glenn Kasten200092b2014-08-15 15:13:30 -07003309 // Update the mapping between local consumed (mPosition) and server consumed (mServer)
3310 (void) updateAndGetPosition_l();
3311 // Server consumed (mServer) and presented both use the same server time base,
3312 // and server consumed is always >= presented.
3313 // The delta between these represents the number of frames in the buffer pipeline.
3314 // If this delta between these is greater than the client position, it means that
3315 // actually presented is still stuck at the starting line (figuratively speaking),
3316 // waiting for the first frame to go by. So we can't report a valid timestamp yet.
Andy Hung90e8a972015-11-09 16:42:40 -08003317 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when
3318 // mPosition exceeds 32 bits.
3319 // TODO Remove when timestamp is updated to contain pipeline status info.
3320 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue();
3321 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */
3322 && (uint32_t)pipelineDepthInFrames > mPosition.value()) {
Glenn Kasten200092b2014-08-15 15:13:30 -07003323 return INVALID_OPERATION;
3324 }
3325 // Convert timestamp position from server time base to client time base.
3326 // TODO The following code should work OK now because timestamp.mPosition is 32-bit.
3327 // But if we change it to 64-bit then this could fail.
Andy Hung90e8a972015-11-09 16:42:40 -08003328 // Use Modulo computation here.
3329 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value();
Glenn Kasten200092b2014-08-15 15:13:30 -07003330 // Immediately after a call to getPosition_l(), mPosition and
3331 // mServer both represent the same frame position. mPosition is
3332 // in client's point of view, and mServer is in server's point of
3333 // view. So the difference between them is the "fudge factor"
3334 // between client and server views due to stop() and/or new
3335 // IAudioTrack. And timestamp.mPosition is initially in server's
3336 // point of view, so we need to apply the same fudge factor to it.
Glenn Kastenfe346c72013-08-30 13:28:22 -07003337 }
Phil Burk1b420972015-04-22 10:52:21 -07003338
3339 // Prevent retrograde motion in timestamp.
3340 // This is sometimes caused by erratic reports of the available space in the ALSA drivers.
3341 if (status == NO_ERROR) {
Andy Hung3b8c6332019-04-03 19:29:36 -07003342 // Fix stale time when checking timestamp right after start().
3343 // The position is at the last reported location but the time can be stale
3344 // due to pause or standby or cold start latency.
3345 //
3346 // We keep advancing the time (but not the position) to ensure that the
3347 // stale value does not confuse the application.
3348 //
3349 // For offload compatibility, use a default lag value here.
3350 // Any time discrepancy between this update and the pause timestamp is handled
3351 // by the retrograde check afterwards.
3352 int64_t currentTimeNanos = audio_utils_ns_from_timespec(&timestamp.mTime);
3353 const int64_t lagNs = int64_t(mAfLatency * 1000000LL);
3354 const int64_t limitNs = mStartNs - lagNs;
3355 if (currentTimeNanos < limitNs) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003356 if (!mTimestampStaleTimeReported) {
3357 ALOGD("%s(%d): stale timestamp time corrected, "
3358 "currentTimeNanos: %lld < limitNs: %lld < mStartNs: %lld",
3359 __func__, mPortId,
3360 (long long)currentTimeNanos, (long long)limitNs, (long long)mStartNs);
3361 mTimestampStaleTimeReported = true;
3362 }
Andy Hung3b8c6332019-04-03 19:29:36 -07003363 timestamp.mTime = convertNsToTimespec(limitNs);
3364 currentTimeNanos = limitNs;
Andy Hungcf3b7152019-04-19 18:29:21 -07003365 } else {
3366 mTimestampStaleTimeReported = false;
Andy Hung3b8c6332019-04-03 19:29:36 -07003367 }
3368
Andy Hungffa36952017-08-17 10:41:51 -07003369 // previousTimestampValid is set to false when starting after a stop or flush.
Phil Burk1b420972015-04-22 10:52:21 -07003370 if (previousTimestampValid) {
Andy Hung2b01f002017-07-05 12:01:36 -07003371 const int64_t previousTimeNanos =
3372 audio_utils_ns_from_timespec(&mPreviousTimestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003373
3374 // retrograde check
Phil Burk1b420972015-04-22 10:52:21 -07003375 if (currentTimeNanos < previousTimeNanos) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003376 if (!mTimestampRetrogradeTimeReported) {
3377 ALOGW("%s(%d): retrograde timestamp time corrected, %lld < %lld",
3378 __func__, mPortId,
3379 (long long)currentTimeNanos, (long long)previousTimeNanos);
3380 mTimestampRetrogradeTimeReported = true;
3381 }
Andy Hung5d313802016-10-10 15:09:39 -07003382 timestamp.mTime = mPreviousTimestamp.mTime;
Andy Hungcf3b7152019-04-19 18:29:21 -07003383 } else {
3384 mTimestampRetrogradeTimeReported = false;
Phil Burk1b420972015-04-22 10:52:21 -07003385 }
3386
3387 // Looking at signed delta will work even when the timestamps
3388 // are wrapping around.
Andy Hung90e8a972015-11-09 16:42:40 -08003389 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition)
3390 - mPreviousTimestamp.mPosition).signedValue();
Phil Burk4c5a3672015-04-30 16:18:53 -07003391 if (deltaPosition < 0) {
3392 // Only report once per position instead of spamming the log.
Andy Hungcf3b7152019-04-19 18:29:21 -07003393 if (!mTimestampRetrogradePositionReported) {
Andy Hungfb8ede22018-09-12 19:03:24 -07003394 ALOGW("%s(%d): retrograde timestamp position corrected, %d = %u - %u",
Eric Laurent973db022018-11-20 14:54:31 -08003395 __func__, mPortId,
Phil Burk4c5a3672015-04-30 16:18:53 -07003396 deltaPosition,
3397 timestamp.mPosition,
3398 mPreviousTimestamp.mPosition);
Andy Hungcf3b7152019-04-19 18:29:21 -07003399 mTimestampRetrogradePositionReported = true;
Phil Burk4c5a3672015-04-30 16:18:53 -07003400 }
3401 } else {
Andy Hungcf3b7152019-04-19 18:29:21 -07003402 mTimestampRetrogradePositionReported = false;
Phil Burk4c5a3672015-04-30 16:18:53 -07003403 }
Andy Hung5d313802016-10-10 15:09:39 -07003404 if (deltaPosition < 0) {
3405 timestamp.mPosition = mPreviousTimestamp.mPosition;
3406 deltaPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07003407 }
Andy Hung5d313802016-10-10 15:09:39 -07003408#if 0
3409 // Uncomment this to verify audio timestamp rate.
3410 const int64_t deltaTime =
Andy Hung2b01f002017-07-05 12:01:36 -07003411 audio_utils_ns_from_timespec(&timestamp.mTime) - previousTimeNanos;
Andy Hung5d313802016-10-10 15:09:39 -07003412 if (deltaTime != 0) {
3413 const int64_t computedSampleRate =
3414 deltaPosition * (long long)NANOS_PER_SECOND / deltaTime;
Andy Hungfb8ede22018-09-12 19:03:24 -07003415 ALOGD("%s(%d): computedSampleRate:%u sampleRate:%u",
Eric Laurent973db022018-11-20 14:54:31 -08003416 __func__, mPortId,
Andy Hung5d313802016-10-10 15:09:39 -07003417 (unsigned)computedSampleRate, mSampleRate);
3418 }
3419#endif
Phil Burk1b420972015-04-22 10:52:21 -07003420 }
3421 mPreviousTimestamp = timestamp;
3422 mPreviousTimestampValid = true;
3423 }
3424
Glenn Kastenfe346c72013-08-30 13:28:22 -07003425 return status;
Glenn Kastence703742013-07-19 16:33:58 -07003426}
3427
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003428String8 AudioTrack::getParameters(const String8& keys)
3429{
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003430 audio_io_handle_t output = getOutput();
Glenn Kasten142f5192014-03-25 17:44:59 -07003431 if (output != AUDIO_IO_HANDLE_NONE) {
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003432 return AudioSystem::getParameters(output, keys);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01003433 } else {
3434 return String8::empty();
3435 }
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003436}
3437
Glenn Kasten23a75452014-01-13 10:37:17 -08003438bool AudioTrack::isOffloaded() const
3439{
3440 AutoMutex lock(mLock);
3441 return isOffloaded_l();
3442}
3443
Eric Laurentab5cdba2014-06-09 17:22:27 -07003444bool AudioTrack::isDirect() const
3445{
3446 AutoMutex lock(mLock);
3447 return isDirect_l();
3448}
3449
3450bool AudioTrack::isOffloadedOrDirect() const
3451{
3452 AutoMutex lock(mLock);
3453 return isOffloadedOrDirect_l();
3454}
3455
3456
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003457status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003458{
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003459 String8 result;
3460
3461 result.append(" AudioTrack::dump\n");
Andy Hungfb8ede22018-09-12 19:03:24 -07003462 result.appendFormat(" id(%d) status(%d), state(%d), session Id(%d), flags(%#x)\n",
Eric Laurent973db022018-11-20 14:54:31 -08003463 mPortId, mStatus, mState, mSessionId, mFlags);
Eric Laurentd114b622017-11-27 18:37:04 -08003464 result.appendFormat(" stream type(%d), left - right volume(%f, %f)\n",
François Gaffie58d4be52018-11-06 15:30:12 +01003465 mStreamType,
Eric Laurentd114b622017-11-27 18:37:04 -08003466 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
Glenn Kasten49f36ba2017-12-06 13:02:02 -08003467 result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
Eric Laurentd114b622017-11-27 18:37:04 -08003468 mFormat, mChannelMask, mChannelCount);
3469 result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
3470 mSampleRate, mOriginalSampleRate, mPlaybackRate.mSpeed);
3471 result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
3472 mFrameCount, mReqFrameCount);
3473 result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
3474 " req. notif. per buff(%u)\n",
3475 mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
3476 result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
3477 mLatency, mSelectedDeviceId, mRoutedDeviceId);
3478 result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
3479 mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003480 ::write(fd, result.string(), result.size());
3481 return NO_ERROR;
3482}
3483
Phil Burk2812d9e2016-01-04 10:34:30 -08003484uint32_t AudioTrack::getUnderrunCount() const
3485{
3486 AutoMutex lock(mLock);
3487 return getUnderrunCount_l();
3488}
3489
3490uint32_t AudioTrack::getUnderrunCount_l() const
3491{
3492 return mProxy->getUnderrunCount() + mUnderrunCountOffset;
3493}
3494
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003495uint32_t AudioTrack::getUnderrunFrames() const
3496{
3497 AutoMutex lock(mLock);
3498 return mProxy->getUnderrunFrames();
3499}
3500
Andy Hung3a5c2f32021-02-17 15:06:42 -08003501void AudioTrack::setLogSessionId(const char *logSessionId)
3502{
3503 AutoMutex lock(mLock);
Andy Hung1a9c21b2021-02-25 20:43:18 -08003504 if (logSessionId == nullptr) logSessionId = ""; // an empty string is an unset session id.
Andy Hung3a5c2f32021-02-17 15:06:42 -08003505 if (mLogSessionId == logSessionId) return;
3506
3507 mLogSessionId = logSessionId;
3508 mediametrics::LogItem(mMetricsId)
3509 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETLOGSESSIONID)
3510 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, logSessionId)
3511 .record();
3512}
3513
Andy Hung839a3062021-02-17 11:15:16 -08003514void AudioTrack::setPlayerIId(int playerIId)
3515{
3516 AutoMutex lock(mLock);
3517 if (mPlayerIId == playerIId) return;
3518
3519 mPlayerIId = playerIId;
Vlad Popaad0fe922022-06-10 00:43:14 +02003520 triggerPortIdUpdate_l();
Andy Hung839a3062021-02-17 11:15:16 -08003521 mediametrics::LogItem(mMetricsId)
3522 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYERIID)
3523 .set(AMEDIAMETRICS_PROP_PLAYERIID, playerIId)
3524 .record();
3525}
3526
Vlad Popaad0fe922022-06-10 00:43:14 +02003527void AudioTrack::triggerPortIdUpdate_l() {
3528 if (mAudioManager == nullptr) {
3529 // use checkService() to avoid blocking if audio service is not up yet
3530 sp<IBinder> binder =
3531 defaultServiceManager()->checkService(String16(kAudioServiceName));
3532 if (binder == nullptr) {
3533 ALOGE("%s(%d): binding to audio service failed.",
3534 __func__,
3535 mPlayerIId);
3536 return;
3537 }
3538
3539 mAudioManager = interface_cast<IAudioManager>(binder);
3540 }
3541
3542 // first time when the track is created we do not have a valid piid
3543 if (mPlayerIId != PLAYER_PIID_INVALID) {
3544 mAudioManager->playerEvent(mPlayerIId, PLAYER_UPDATE_PORT_ID, mPortId);
3545 }
3546}
3547
Eric Laurent296fb132015-05-01 11:38:42 -07003548status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback)
3549{
Eric Laurent09f1ed22019-04-24 17:45:17 -07003550
Eric Laurent296fb132015-05-01 11:38:42 -07003551 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003552 ALOGW("%s(%d): adding NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003553 return BAD_VALUE;
3554 }
3555 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07003556 if (mDeviceCallback.unsafe_get() == callback.get()) {
Eric Laurent973db022018-11-20 14:54:31 -08003557 ALOGW("%s(%d): adding same callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003558 return INVALID_OPERATION;
3559 }
3560 status_t status = NO_ERROR;
3561 if (mOutput != AUDIO_IO_HANDLE_NONE) {
3562 if (mDeviceCallback != 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003563 ALOGW("%s(%d): callback already present!", __func__, mPortId);
Eric Laurent09f1ed22019-04-24 17:45:17 -07003564 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003565 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003566 status = AudioSystem::addAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003567 }
3568 mDeviceCallback = callback;
3569 return status;
3570}
3571
3572status_t AudioTrack::removeAudioDeviceCallback(
3573 const sp<AudioSystem::AudioDeviceCallback>& callback)
3574{
3575 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003576 ALOGW("%s(%d): removing NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003577 return BAD_VALUE;
3578 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003579 AutoMutex lock(mLock);
3580 if (mDeviceCallback.unsafe_get() != callback.get()) {
3581 ALOGW("%s removing different callback!", __FUNCTION__);
3582 return INVALID_OPERATION;
Eric Laurent296fb132015-05-01 11:38:42 -07003583 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003584 mDeviceCallback.clear();
Eric Laurent296fb132015-05-01 11:38:42 -07003585 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07003586 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003587 }
Eric Laurent296fb132015-05-01 11:38:42 -07003588 return NO_ERROR;
3589}
3590
Eric Laurentad2e7b92017-09-14 20:06:42 -07003591
3592void AudioTrack::onAudioDeviceUpdate(audio_io_handle_t audioIo,
3593 audio_port_handle_t deviceId)
3594{
3595 sp<AudioSystem::AudioDeviceCallback> callback;
3596 {
3597 AutoMutex lock(mLock);
3598 if (audioIo != mOutput) {
3599 return;
3600 }
3601 callback = mDeviceCallback.promote();
3602 // only update device if the track is active as route changes due to other use cases are
3603 // irrelevant for this client
3604 if (mState == STATE_ACTIVE) {
3605 mRoutedDeviceId = deviceId;
3606 }
3607 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003608
Eric Laurentad2e7b92017-09-14 20:06:42 -07003609 if (callback.get() != nullptr) {
3610 callback->onAudioDeviceUpdate(mOutput, mRoutedDeviceId);
3611 }
3612}
3613
Andy Hunge13f8a62016-03-30 14:20:42 -07003614status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location)
3615{
3616 if (msec == nullptr ||
3617 (location != ExtendedTimestamp::LOCATION_SERVER
3618 && location != ExtendedTimestamp::LOCATION_KERNEL)) {
3619 return BAD_VALUE;
3620 }
3621 AutoMutex lock(mLock);
3622 // inclusive of offloaded and direct tracks.
3623 //
3624 // It is possible, but not enabled, to allow duration computation for non-pcm
3625 // audio_has_proportional_frames() formats because currently they have
3626 // the drain rate equivalent to the pcm sample rate * framesize.
3627 if (!isPurePcmData_l()) {
3628 return INVALID_OPERATION;
3629 }
3630 ExtendedTimestamp ets;
3631 if (getTimestamp_l(&ets) == OK
3632 && ets.mTimeNs[location] > 0) {
3633 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
3634 - ets.mPosition[location];
3635 if (diff < 0) {
3636 *msec = 0;
3637 } else {
3638 // ms is the playback time by frames
3639 int64_t ms = (int64_t)((double)diff * 1000 /
3640 ((double)mSampleRate * mPlaybackRate.mSpeed));
3641 // clockdiff is the timestamp age (negative)
3642 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
3643 ets.mTimeNs[location]
3644 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
3645 - systemTime(SYSTEM_TIME_MONOTONIC);
3646
3647 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
3648 static const int NANOS_PER_MILLIS = 1000000;
3649 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
3650 }
3651 return NO_ERROR;
3652 }
3653 if (location != ExtendedTimestamp::LOCATION_SERVER) {
3654 return INVALID_OPERATION; // LOCATION_KERNEL is not available
3655 }
3656 // use server position directly (offloaded and direct arrive here)
3657 updateAndGetPosition_l();
3658 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue();
3659 *msec = (diff <= 0) ? 0
3660 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed));
3661 return NO_ERROR;
3662}
3663
Andy Hung65ffdfc2016-10-10 15:52:11 -07003664bool AudioTrack::hasStarted()
3665{
3666 AutoMutex lock(mLock);
3667 switch (mState) {
3668 case STATE_STOPPED:
3669 if (isOffloadedOrDirect_l()) {
3670 // check if we have started in the past to return true.
Andy Hungffa36952017-08-17 10:41:51 -07003671 return mStartFromZeroUs > 0;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003672 }
3673 // A normal audio track may still be draining, so
3674 // check if stream has ended. This covers fasttrack position
3675 // instability and start/stop without any data written.
3676 if (mProxy->getStreamEndDone()) {
3677 return true;
3678 }
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003679 FALLTHROUGH_INTENDED;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003680 case STATE_ACTIVE:
3681 case STATE_STOPPING:
3682 break;
3683 case STATE_PAUSED:
3684 case STATE_PAUSED_STOPPING:
3685 case STATE_FLUSHED:
3686 return false; // we're not active
3687 default:
Eric Laurent973db022018-11-20 14:54:31 -08003688 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in hasStarted(): %d", __func__, mPortId, mState);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003689 break;
3690 }
3691
3692 // wait indicates whether we need to wait for a timestamp.
3693 // This is conservatively figured - if we encounter an unexpected error
3694 // then we will not wait.
3695 bool wait = false;
3696 if (isOffloadedOrDirect_l()) {
3697 AudioTimestamp ts;
3698 status_t status = getTimestamp_l(ts);
3699 if (status == WOULD_BLOCK) {
3700 wait = true;
3701 } else if (status == OK) {
3702 wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition);
3703 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003704 ALOGV("%s(%d): hasStarted wait:%d ts:%u start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003705 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003706 (int)wait,
3707 ts.mPosition,
3708 (long long)mStartTs.mPosition);
3709 } else {
3710 int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG
3711 ExtendedTimestamp ets;
3712 status_t status = getTimestamp_l(&ets);
3713 if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets
3714 wait = true;
3715 } else if (status == OK) {
3716 for (location = ExtendedTimestamp::LOCATION_KERNEL;
3717 location >= ExtendedTimestamp::LOCATION_SERVER; --location) {
3718 if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) {
3719 continue;
3720 }
3721 wait = ets.mPosition[location] == 0
3722 || ets.mPosition[location] == mStartEts.mPosition[location];
3723 break;
3724 }
3725 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003726 ALOGV("%s(%d): hasStarted wait:%d ets:%lld start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003727 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003728 (int)wait,
3729 (long long)ets.mPosition[location],
3730 (long long)mStartEts.mPosition[location]);
3731 }
3732 return !wait;
3733}
3734
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003735// =========================================================================
3736
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003737void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003738{
3739 sp<AudioTrack> audioTrack = mAudioTrack.promote();
3740 if (audioTrack != 0) {
3741 AutoMutex lock(audioTrack->mLock);
3742 audioTrack->mProxy->binderDied();
3743 }
3744}
3745
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003746// =========================================================================
3747
Andy Hungca353672019-03-06 11:54:38 -08003748AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver)
Andy Hungeb46cf92019-03-06 10:13:38 -08003749 : Thread(true /* bCanCallJava */) // binder recursion on restoreTrack_l() may call Java.
3750 , mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
Glenn Kasten598de6c2013-10-16 17:02:13 -07003751 mIgnoreNextPausedInt(false)
Glenn Kasten3acbd052012-02-28 10:39:56 -08003752{
3753}
3754
3755AudioTrack::AudioTrackThread::~AudioTrackThread()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003756{
3757}
3758
3759bool AudioTrack::AudioTrackThread::threadLoop()
3760{
Glenn Kasten3acbd052012-02-28 10:39:56 -08003761 {
3762 AutoMutex _l(mMyLock);
3763 if (mPaused) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003764 // TODO check return value and handle or log
Glenn Kasten3acbd052012-02-28 10:39:56 -08003765 mMyCond.wait(mMyLock);
3766 // caller will check for exitPending()
3767 return true;
3768 }
Glenn Kasten598de6c2013-10-16 17:02:13 -07003769 if (mIgnoreNextPausedInt) {
3770 mIgnoreNextPausedInt = false;
3771 mPausedInt = false;
3772 }
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003773 if (mPausedInt) {
Glenn Kasten47d55172017-05-23 11:19:30 -07003774 // TODO use futex instead of condition, for event flag "or"
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003775 if (mPausedNs > 0) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003776 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003777 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
3778 } else {
Glenn Kasten75f79032017-05-23 11:22:44 -07003779 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003780 mMyCond.wait(mMyLock);
3781 }
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003782 mPausedInt = false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003783 return true;
3784 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08003785 }
Eric Laurent7985dcb2014-10-07 15:45:14 -07003786 if (exitPending()) {
3787 return false;
3788 }
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003789 nsecs_t ns = mReceiver.processAudioBuffer();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003790 switch (ns) {
3791 case 0:
3792 return true;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003793 case NS_INACTIVE:
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003794 pauseInternal();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003795 return true;
3796 case NS_NEVER:
3797 return false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003798 case NS_WHENEVER:
Andy Hung3c09c782014-12-29 18:39:32 -08003799 // Event driven: call wake() when callback notifications conditions change.
3800 ns = INT64_MAX;
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003801 FALLTHROUGH_INTENDED;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003802 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003803 LOG_ALWAYS_FATAL_IF(ns < 0, "%s(%d): processAudioBuffer() returned %lld",
Eric Laurent973db022018-11-20 14:54:31 -08003804 __func__, mReceiver.mPortId, (long long)ns);
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003805 pauseInternal(ns);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003806 return true;
Glenn Kastenca8b2802012-04-23 13:58:16 -07003807 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003808}
3809
Glenn Kasten3acbd052012-02-28 10:39:56 -08003810void AudioTrack::AudioTrackThread::requestExit()
3811{
3812 // must be in this order to avoid a race condition
3813 Thread::requestExit();
Glenn Kasten598de6c2013-10-16 17:02:13 -07003814 resume();
Glenn Kasten3acbd052012-02-28 10:39:56 -08003815}
3816
3817void AudioTrack::AudioTrackThread::pause()
3818{
3819 AutoMutex _l(mMyLock);
3820 mPaused = true;
3821}
3822
3823void AudioTrack::AudioTrackThread::resume()
3824{
3825 AutoMutex _l(mMyLock);
Glenn Kasten598de6c2013-10-16 17:02:13 -07003826 mIgnoreNextPausedInt = true;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003827 if (mPaused || mPausedInt) {
Glenn Kasten3acbd052012-02-28 10:39:56 -08003828 mPaused = false;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003829 mPausedInt = false;
Glenn Kasten3acbd052012-02-28 10:39:56 -08003830 mMyCond.signal();
3831 }
3832}
3833
Andy Hung3c09c782014-12-29 18:39:32 -08003834void AudioTrack::AudioTrackThread::wake()
3835{
3836 AutoMutex _l(mMyLock);
Andy Hunga8d08902015-07-22 11:52:13 -07003837 if (!mPaused) {
3838 // wake() might be called while servicing a callback - ignore the next
3839 // pause time and call processAudioBuffer.
Andy Hung3c09c782014-12-29 18:39:32 -08003840 mIgnoreNextPausedInt = true;
Andy Hunga8d08902015-07-22 11:52:13 -07003841 if (mPausedInt && mPausedNs > 0) {
3842 // audio track is active and internally paused with timeout.
3843 mPausedInt = false;
3844 mMyCond.signal();
3845 }
Andy Hung3c09c782014-12-29 18:39:32 -08003846 }
3847}
3848
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003849void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
3850{
3851 AutoMutex _l(mMyLock);
3852 mPausedInt = true;
3853 mPausedNs = ns;
3854}
3855
jiabinf6eb4c32020-02-25 14:06:25 -08003856binder::Status AudioTrack::AudioTrackCallback::onCodecFormatChanged(
3857 const std::vector<uint8_t>& audioMetadata)
3858{
3859 AutoMutex _l(mAudioTrackCbLock);
3860 sp<media::IAudioTrackCallback> callback = mCallback.promote();
3861 if (callback.get() != nullptr) {
3862 callback->onCodecFormatChanged(audioMetadata);
3863 } else {
3864 mCallback.clear();
3865 }
3866 return binder::Status::ok();
3867}
3868
3869void AudioTrack::AudioTrackCallback::setAudioTrackCallback(
3870 const sp<media::IAudioTrackCallback> &callback) {
3871 AutoMutex lock(mAudioTrackCbLock);
3872 mCallback = callback;
3873}
3874
Glenn Kasten40bc9062015-03-20 09:09:33 -07003875} // namespace android