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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>
32#include <media/AudioTrack.h>
33#include <utils/Log.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080034#include <private/media/AudioTrackShared.h>
Suren Baghdasaryan7435e7d2018-12-19 17:09:28 -080035#include <processgroup/sched_policy.h>
Glenn Kasten1ab85ec2013-05-31 09:18:43 -070036#include <media/IAudioFlinger.h>
Dean Wheatleya70eef72018-01-04 14:23:50 +110037#include <media/AudioParameter.h>
Andy Hungcd044842014-08-07 11:04:34 -070038#include <media/AudioResamplerPublic.h>
Michael Chana94fbb22018-04-24 14:31:19 +100039#include <media/AudioSystem.h>
Ray Essickf27e9872019-12-07 06:28:46 -080040#include <media/MediaMetricsItem.h>
Ray Essicked304702017-12-12 14:00:57 -080041#include <media/TypeConverter.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080042
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +010043#define WAIT_PERIOD_MS 10
44#define WAIT_STREAM_END_TIMEOUT_SEC 120
Andy Hung53c3b5f2014-12-15 16:42:05 -080045static const int kMaxLoopCountNotifications = 32;
Glenn Kasten511754b2012-01-11 09:52:19 -080046
Kuowei Lid4adbdb2020-08-13 14:44:25 +080047using ::android::aidl_utils::statusTFromBinderStatus;
Andy Hung3acde2c2021-11-11 09:18:08 -080048using ::android::base::StringPrintf;
Kuowei Lid4adbdb2020-08-13 14:44:25 +080049
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080050namespace android {
Chia-chi Yeh33005a92010-06-16 06:33:13 +080051// ---------------------------------------------------------------------------
52
Ivan Lozano8cf3a072017-08-09 09:01:33 -070053using media::VolumeShaper;
Svet Ganov3e5f14f2021-05-13 22:51:08 +000054using android::content::AttributionSourceState;
Ivan Lozano8cf3a072017-08-09 09:01:33 -070055
Andy Hunga7f03352015-05-31 21:54:49 -070056// TODO: Move to a separate .h
57
Andy Hung4ede21d2014-12-12 15:37:34 -080058template <typename T>
Andy Hunga7f03352015-05-31 21:54:49 -070059static inline const T &min(const T &x, const T &y) {
Andy Hung4ede21d2014-12-12 15:37:34 -080060 return x < y ? x : y;
61}
62
Andy Hunga7f03352015-05-31 21:54:49 -070063template <typename T>
64static inline const T &max(const T &x, const T &y) {
65 return x > y ? x : y;
66}
67
68static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed)
69{
70 return ((double)frames * 1000000000) / ((double)sampleRate * speed);
71}
72
Andy Hung7f1bc8a2014-09-12 14:43:11 -070073static int64_t convertTimespecToUs(const struct timespec &tv)
74{
Chih-Hung Hsieh50fa06c2018-12-11 13:51:36 -080075 return tv.tv_sec * 1000000LL + tv.tv_nsec / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -070076}
77
Andy Hungffa36952017-08-17 10:41:51 -070078// TODO move to audio_utils.
79static inline struct timespec convertNsToTimespec(int64_t ns) {
80 struct timespec tv;
81 tv.tv_sec = static_cast<time_t>(ns / NANOS_PER_SECOND);
Andy Hung06a730b2020-04-09 13:28:31 -070082 tv.tv_nsec = static_cast<int64_t>(ns % NANOS_PER_SECOND);
Andy Hungffa36952017-08-17 10:41:51 -070083 return tv;
84}
85
Andy Hung7f1bc8a2014-09-12 14:43:11 -070086// current monotonic time in microseconds.
87static int64_t getNowUs()
88{
89 struct timespec tv;
90 (void) clock_gettime(CLOCK_MONOTONIC, &tv);
91 return convertTimespecToUs(tv);
92}
93
Andy Hung26145642015-04-15 21:56:53 -070094// FIXME: we don't use the pitch setting in the time stretcher (not working);
95// instead we emulate it using our sample rate converter.
96static const bool kFixPitch = true; // enable pitch fix
97static inline uint32_t adjustSampleRate(uint32_t sampleRate, float pitch)
98{
99 return kFixPitch ? (sampleRate * pitch + 0.5) : sampleRate;
100}
101
102static inline float adjustSpeed(float speed, float pitch)
103{
Ricardo Garcia6c7f0622015-04-30 18:39:16 -0700104 return kFixPitch ? speed / max(pitch, AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) : speed;
Andy Hung26145642015-04-15 21:56:53 -0700105}
106
107static inline float adjustPitch(float pitch)
108{
109 return kFixPitch ? AUDIO_TIMESTRETCH_PITCH_NORMAL : pitch;
110}
111
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800112// static
113status_t AudioTrack::getMinFrameCount(
Glenn Kastene33054e2012-11-14 12:54:39 -0800114 size_t* frameCount,
Glenn Kastenfff6d712012-01-12 16:38:12 -0800115 audio_stream_type_t streamType,
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800116 uint32_t sampleRate)
117{
Glenn Kastend65d73c2012-06-22 17:21:07 -0700118 if (frameCount == NULL) {
119 return BAD_VALUE;
120 }
Glenn Kasten04cd0182012-06-25 11:49:27 -0700121
Andy Hung0e48d252015-01-26 11:43:15 -0800122 // FIXME handle in server, like createTrack_l(), possible missing info:
Glenn Kastene0fa4672012-04-24 14:35:14 -0700123 // audio_io_handle_t output
124 // audio_format_t format
125 // audio_channel_mask_t channelMask
Andy Hung0e48d252015-01-26 11:43:15 -0800126 // audio_output_flags_t flags (FAST)
Glenn Kasten3b16c762012-11-14 08:44:39 -0800127 uint32_t afSampleRate;
Glenn Kasten66a04672014-01-08 08:53:44 -0800128 status_t status;
129 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
130 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700131 ALOGE("%s(): Unable to query output sample rate for stream type %d; status %d",
132 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800133 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800134 }
Glenn Kastene33054e2012-11-14 12:54:39 -0800135 size_t afFrameCount;
Glenn Kasten66a04672014-01-08 08:53:44 -0800136 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
137 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700138 ALOGE("%s(): Unable to query output frame count for stream type %d; status %d",
139 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800140 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800141 }
142 uint32_t afLatency;
Glenn Kasten66a04672014-01-08 08:53:44 -0800143 status = AudioSystem::getOutputLatency(&afLatency, streamType);
144 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700145 ALOGE("%s(): Unable to query output latency for stream type %d; status %d",
146 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800147 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800148 }
149
Andy Hung8edb8dc2015-03-26 19:13:55 -0700150 // When called from createTrack, speed is 1.0f (normal speed).
151 // This is rechecked again on setting playback rate (TODO: on setting sample rate, too).
Eric Laurent21da6472017-11-09 16:29:26 -0800152 *frameCount = AudioSystem::calculateMinFrameCount(afLatency, afFrameCount, afSampleRate,
153 sampleRate, 1.0f /*, 0 notificationsPerBufferReq*/);
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800154
Andy Hung0e48d252015-01-26 11:43:15 -0800155 // The formula above should always produce a non-zero value under normal circumstances:
156 // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX.
157 // 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 -0800158 if (*frameCount == 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700159 ALOGE("%s(): failed for streamType %d, sampleRate %u",
160 __func__, streamType, sampleRate);
Glenn Kasten66a04672014-01-08 08:53:44 -0800161 return BAD_VALUE;
162 }
Andy Hungfb8ede22018-09-12 19:03:24 -0700163 ALOGV("%s(): getMinFrameCount=%zu: afFrameCount=%zu, afSampleRate=%u, afLatency=%u",
164 __func__, *frameCount, afFrameCount, afSampleRate, afLatency);
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800165 return NO_ERROR;
166}
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800167
Michael Chana94fbb22018-04-24 14:31:19 +1000168// static
169bool AudioTrack::isDirectOutputSupported(const audio_config_base_t& config,
170 const audio_attributes_t& attributes) {
171 ALOGV("%s()", __FUNCTION__);
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800172 const sp<media::IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
Michael Chana94fbb22018-04-24 14:31:19 +1000173 if (aps == 0) return false;
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800174
175 auto result = [&]() -> ConversionResult<bool> {
Mikhail Naganovdbf03642021-08-25 18:15:32 -0700176 media::audio::common::AudioConfigBase configAidl = VALUE_OR_RETURN(
Mikhail Naganov9dec7012021-07-21 10:30:57 -0700177 legacy2aidl_audio_config_base_t_AudioConfigBase(config, false /*isInput*/));
Mikhail Naganov2a1cf612023-05-17 11:48:43 -0700178 media::audio::common::AudioAttributes attributesAidl = VALUE_OR_RETURN(
179 legacy2aidl_audio_attributes_t_AudioAttributes(attributes));
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800180 bool retAidl;
181 RETURN_IF_ERROR(aidl_utils::statusTFromBinderStatus(
182 aps->isDirectOutputSupported(configAidl, attributesAidl, &retAidl)));
183 return retAidl;
184 }();
185 return result.value_or(false);
Michael Chana94fbb22018-04-24 14:31:19 +1000186}
187
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800188// ---------------------------------------------------------------------------
189
Ray Essicked304702017-12-12 14:00:57 -0800190void AudioTrack::MediaMetrics::gather(const AudioTrack *track)
191{
Ray Essick88394302018-01-24 14:52:05 -0800192 // only if we're in a good state...
193 // XXX: shall we gather alternative info if failing?
194 const status_t lstatus = track->initCheck();
195 if (lstatus != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700196 ALOGD("%s(): no metrics gathered, track status=%d", __func__, (int) lstatus);
Ray Essick88394302018-01-24 14:52:05 -0800197 return;
198 }
199
Andy Hungd0979812019-02-21 15:51:44 -0800200#define MM_PREFIX "android.media.audiotrack." // avoid cut-n-paste errors.
Ray Essicked304702017-12-12 14:00:57 -0800201
Andy Hungde602302021-12-07 21:35:49 -0800202 // Do not change this without changing the MediaMetricsService side.
Andy Hungd0979812019-02-21 15:51:44 -0800203 // Java API 28 entries, do not change.
Ray Essickf27e9872019-12-07 06:28:46 -0800204 mMetricsItem->setCString(MM_PREFIX "streamtype", toString(track->streamType()).c_str());
205 mMetricsItem->setCString(MM_PREFIX "type",
Andy Hungd0979812019-02-21 15:51:44 -0800206 toString(track->mAttributes.content_type).c_str());
Ray Essickf27e9872019-12-07 06:28:46 -0800207 mMetricsItem->setCString(MM_PREFIX "usage", toString(track->mAttributes.usage).c_str());
Ray Essicked304702017-12-12 14:00:57 -0800208
Andy Hungd0979812019-02-21 15:51:44 -0800209 // Non-API entries, these can change due to a Java string mistake.
Ray Essickf27e9872019-12-07 06:28:46 -0800210 mMetricsItem->setInt32(MM_PREFIX "sampleRate", (int32_t)track->mSampleRate);
211 mMetricsItem->setInt64(MM_PREFIX "channelMask", (int64_t)track->mChannelMask);
Andy Hungd0979812019-02-21 15:51:44 -0800212 // Non-API entries, these can change.
Ray Essickf27e9872019-12-07 06:28:46 -0800213 mMetricsItem->setInt32(MM_PREFIX "portId", (int32_t)track->mPortId);
214 mMetricsItem->setCString(MM_PREFIX "encoding", toString(track->mFormat).c_str());
215 mMetricsItem->setInt32(MM_PREFIX "frameCount", (int32_t)track->mFrameCount);
216 mMetricsItem->setCString(MM_PREFIX "attributes", toString(track->mAttributes).c_str());
Andy Hung6f451f02021-02-24 21:53:29 -0800217 mMetricsItem->setCString(MM_PREFIX "logSessionId", track->mLogSessionId.c_str());
Andy Hungde602302021-12-07 21:35:49 -0800218 mMetricsItem->setInt32(MM_PREFIX "underrunFrames", (int32_t)track->getUnderrunFrames());
Ray Essicked304702017-12-12 14:00:57 -0800219}
220
Ray Essick88394302018-01-24 14:52:05 -0800221// hand the user a snapshot of the metrics.
Ray Essickf27e9872019-12-07 06:28:46 -0800222status_t AudioTrack::getMetrics(mediametrics::Item * &item)
Ray Essick88394302018-01-24 14:52:05 -0800223{
224 mMediaMetrics.gather(this);
Ray Essickf27e9872019-12-07 06:28:46 -0800225 mediametrics::Item *tmp = mMediaMetrics.dup();
Ray Essick88394302018-01-24 14:52:05 -0800226 if (tmp == nullptr) {
227 return BAD_VALUE;
228 }
229 item = tmp;
230 return NO_ERROR;
231}
Ray Essicked304702017-12-12 14:00:57 -0800232
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000233AudioTrack::AudioTrack() : AudioTrack(AttributionSourceState())
Colin Crossb8a9dbb2020-08-27 04:12:26 +0000234{
235}
236
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000237AudioTrack::AudioTrack(const AttributionSourceState& attributionSource)
Glenn Kasten87913512011-06-22 16:15:25 -0700238 : mStatus(NO_INIT),
Haynes Mathew George9b3359f2016-03-23 19:09:10 -0700239 mState(STATE_STOPPED),
John Grossman4ff14ba2012-02-08 16:37:41 -0800240 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800241 mPreviousSchedulingGroup(SP_DEFAULT),
Paul McLeanaa981192015-03-21 09:55:15 -0700242 mPausedPosition(0),
Eric Laurent20b9ef02016-12-05 11:03:16 -0800243 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
jiabinf6eb4c32020-02-25 14:06:25 -0800244 mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000245 mClientAttributionSource(attributionSource),
jiabinf6eb4c32020-02-25 14:06:25 -0800246 mAudioTrackCallback(new AudioTrackCallback())
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800247{
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700248 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
249 mAttributes.usage = AUDIO_USAGE_UNKNOWN;
Mikhail Naganov55773032020-10-01 15:08:13 -0700250 mAttributes.flags = AUDIO_FLAG_NONE;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700251 strcpy(mAttributes.tags, "");
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800252}
253
254AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800255 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800256 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800257 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700258 audio_channel_mask_t channelMask,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800259 size_t frameCount,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700260 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400261 const wp<IAudioTrackCallback> & callback,
Atneyaf86d2692021-10-14 14:02:36 -0400262 int32_t notificationFrames,
263 audio_session_t sessionId,
264 transfer_type transferType,
265 const audio_offload_info_t *offloadInfo,
266 const AttributionSourceState& attributionSource,
267 const audio_attributes_t* pAttributes,
268 bool doNotReconnect,
269 float maxRequiredSpeed,
270 audio_port_handle_t selectedDeviceId)
271 : mStatus(NO_INIT),
272 mState(STATE_STOPPED),
273 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
274 mPreviousSchedulingGroup(SP_DEFAULT),
275 mPausedPosition(0),
276 mAudioTrackCallback(new AudioTrackCallback())
277{
278 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Daniel Chapinf13b98a2021-10-25 21:58:31 +0000279
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500280 // make_unique does not aggregate init until c++20
281 mSetParams = std::unique_ptr<SetParams>{
282 new SetParams{streamType, sampleRate, format, channelMask, frameCount, flags, callback,
283 notificationFrames, 0 /*sharedBuffer*/, false /*threadCanCallJava*/,
284 sessionId, transferType, offloadInfo, attributionSource, pAttributes,
285 doNotReconnect, maxRequiredSpeed, selectedDeviceId}};
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400286}
287
288namespace {
289 class LegacyCallbackWrapper : public AudioTrack::IAudioTrackCallback {
290 const AudioTrack::legacy_callback_t mCallback;
291 void * const mData;
292 public:
293 LegacyCallbackWrapper(AudioTrack::legacy_callback_t callback, void* user)
294 : mCallback(callback), mData(user) {}
295 size_t onMoreData(const AudioTrack::Buffer & buffer) override {
296 AudioTrack::Buffer copy = buffer;
297 mCallback(AudioTrack::EVENT_MORE_DATA, mData, static_cast<void*>(&copy));
Atneya Nair03079272022-01-18 17:03:14 -0500298 return copy.size();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400299 }
300 void onUnderrun() override {
301 mCallback(AudioTrack::EVENT_UNDERRUN, mData, nullptr);
302 }
303 void onLoopEnd(int32_t loopsRemaining) override {
304 mCallback(AudioTrack::EVENT_LOOP_END, mData, &loopsRemaining);
305 }
306 void onMarker(uint32_t markerPosition) override {
307 mCallback(AudioTrack::EVENT_MARKER, mData, &markerPosition);
308 }
309 void onNewPos(uint32_t newPos) override {
310 mCallback(AudioTrack::EVENT_NEW_POS, mData, &newPos);
311 }
312 void onBufferEnd() override {
313 mCallback(AudioTrack::EVENT_BUFFER_END, mData, nullptr);
314 }
315 void onNewIAudioTrack() override {
316 mCallback(AudioTrack::EVENT_NEW_IAUDIOTRACK, mData, nullptr);
317 }
318 void onStreamEnd() override {
319 mCallback(AudioTrack::EVENT_STREAM_END, mData, nullptr);
320 }
321 size_t onCanWriteMoreData(const AudioTrack::Buffer & buffer) override {
322 AudioTrack::Buffer copy = buffer;
323 mCallback(AudioTrack::EVENT_CAN_WRITE_MORE_DATA, mData, static_cast<void*>(&copy));
Atneya Nair03079272022-01-18 17:03:14 -0500324 return copy.size();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400325 }
326 };
327}
Andreas Huberc8139852012-01-18 10:51:55 -0800328AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800329 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800330 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800331 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700332 audio_channel_mask_t channelMask,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800333 const sp<IMemory>& sharedBuffer,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700334 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400335 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700336 int32_t notificationFrames,
Glenn Kastend848eb42016-03-08 13:42:11 -0800337 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000338 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800339 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000340 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700341 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700342 bool doNotReconnect,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700343 float maxRequiredSpeed)
Glenn Kasten87913512011-06-22 16:15:25 -0700344 : mStatus(NO_INIT),
Haynes Mathew George9b3359f2016-03-23 19:09:10 -0700345 mState(STATE_STOPPED),
John Grossman4ff14ba2012-02-08 16:37:41 -0800346 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800347 mPreviousSchedulingGroup(SP_DEFAULT),
Paul McLeanaa981192015-03-21 09:55:15 -0700348 mPausedPosition(0),
jiabinf6eb4c32020-02-25 14:06:25 -0800349 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
350 mAudioTrackCallback(new AudioTrackCallback())
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800351{
François Gaffie393f0e02019-04-10 09:09:08 +0200352 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Ippei Murata5fa32ed2018-10-01 17:37:50 +0900353
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500354 mSetParams = std::unique_ptr<SetParams>{
355 new SetParams{streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags,
356 callback, notificationFrames, sharedBuffer, false /*threadCanCallJava*/,
357 sessionId, transferType, offloadInfo, attributionSource, pAttributes,
358 doNotReconnect, maxRequiredSpeed, AUDIO_PORT_HANDLE_NONE}};
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800359}
360
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500361void AudioTrack::onFirstRef() {
362 if (mSetParams) {
363 set(*mSetParams);
364 mSetParams.reset();
365 }
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400366}
367
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800368AudioTrack::~AudioTrack()
369{
Ray Essicked304702017-12-12 14:00:57 -0800370 // pull together the numbers, before we clean up our structures
371 mMediaMetrics.gather(this);
372
Andy Hungb68f5eb2019-12-03 16:49:17 -0800373 mediametrics::LogItem(mMetricsId)
374 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DTOR)
Phil Burkd3813f32020-04-23 16:26:15 -0700375 .set(AMEDIAMETRICS_PROP_CALLERNAME,
376 mCallerName.empty()
Andy Hunga6b27032020-04-27 10:34:24 -0700377 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
Phil Burkd3813f32020-04-23 16:26:15 -0700378 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800379 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
380 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)mStatus)
381 .record();
382
Phil Burk7a9577c2021-03-12 20:12:11 +0000383 stopAndJoinCallbacks(); // checks mStatus
384
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800385 if (mStatus == NO_ERROR) {
Marco Nelissenf8880202014-11-14 07:58:25 -0800386 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten53cec222013-08-29 09:01:02 -0700387 mAudioTrack.clear();
Eric Laurent3bcffa12014-06-12 18:38:45 -0700388 mCblkMemory.clear();
389 mSharedBuffer.clear();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800390 IPCThreadState::self()->flushCommands();
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000391 pid_t clientPid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
Andy Hungfb8ede22018-09-12 19:03:24 -0700392 ALOGV("%s(%d), releasing session id %d from %d on behalf of %d",
Eric Laurent973db022018-11-20 14:54:31 -0800393 __func__, mPortId,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700394 mSessionId, IPCThreadState::self()->getCallingPid(), clientPid);
395 AudioSystem::releaseAudioSessionId(mSessionId, clientPid);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800396 }
397}
398
Phil Burk7a9577c2021-03-12 20:12:11 +0000399void AudioTrack::stopAndJoinCallbacks() {
400 // Prevent nullptr crash if it did not open properly.
401 if (mStatus != NO_ERROR) return;
402
403 // Make sure that callback function exits in the case where
404 // it is looping on buffer full condition in obtainBuffer().
405 // Otherwise the callback thread will never exit.
406 stop();
407 if (mAudioTrackThread != 0) { // not thread safe
Phil Burk7a9577c2021-03-12 20:12:11 +0000408 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
Kuowei Lifdf8e0f2021-11-26 17:38:34 +0800409 mProxy->interrupt();
Phil Burk7a9577c2021-03-12 20:12:11 +0000410 mAudioTrackThread->requestExitAndWait();
411 mAudioTrackThread.clear();
412 }
zhenjun.zhang116df6a2021-12-08 09:17:13 +0800413
414 AutoMutex lock(mLock);
Phil Burk7a9577c2021-03-12 20:12:11 +0000415 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) {
416 // This may not stop all of these device callbacks!
417 // TODO: Add some sort of protection.
418 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
419 mDeviceCallback.clear();
420 }
421}
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400422status_t AudioTrack::set(
423 audio_stream_type_t streamType,
424 uint32_t sampleRate,
425 audio_format_t format,
426 audio_channel_mask_t channelMask,
427 size_t frameCount,
428 audio_output_flags_t flags,
429 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700430 int32_t notificationFrames,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800431 const sp<IMemory>& sharedBuffer,
Eric Laurentbe916aa2010-06-01 23:49:17 -0700432 bool threadCanCallJava,
Glenn Kastend848eb42016-03-08 13:42:11 -0800433 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000434 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800435 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000436 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700437 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700438 bool doNotReconnect,
jiabin156c6872017-10-06 09:47:15 -0700439 float maxRequiredSpeed,
440 audio_port_handle_t selectedDeviceId)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800441{
Atneya Nair14aabae2021-11-30 17:36:24 -0500442 LOG_ALWAYS_FATAL_IF(mInitialized, "%s: should not be called twice", __func__);
443 mInitialized = true;
Eric Laurentf32d7812017-11-30 14:44:07 -0800444 status_t status;
445 uint32_t channelCount;
446 pid_t callingPid;
447 pid_t myPid;
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000448 uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
449 pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
Andy Hung3acde2c2021-11-11 09:18:08 -0800450 std::string errorMessage;
Eric Laurent973db022018-11-20 14:54:31 -0800451 // Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
Andy Hungfb8ede22018-09-12 19:03:24 -0700452 ALOGV("%s(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
wanggang1471f644f2022-07-08 11:10:20 +0800453 "flags %#x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
Andy Hungfb8ede22018-09-12 19:03:24 -0700454 __func__,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800455 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000456 sessionId, transferType, attributionSource.uid, attributionSource.pid);
Glenn Kasten86f04662014-02-24 15:13:05 -0800457
Phil Burk33ff89b2015-11-30 11:16:01 -0800458 mThreadCanCallJava = threadCanCallJava;
Andy Hungde602302021-12-07 21:35:49 -0800459
460 // These variables are pulled in an error report, so we initialize them early.
jiabin156c6872017-10-06 09:47:15 -0700461 mSelectedDeviceId = selectedDeviceId;
Eric Laurent21da6472017-11-09 16:29:26 -0800462 mSessionId = sessionId;
Andy Hungde602302021-12-07 21:35:49 -0800463 mChannelMask = channelMask;
Andy Hungde602302021-12-07 21:35:49 -0800464 mReqFrameCount = mFrameCount = frameCount;
465 mSampleRate = sampleRate;
466 mOriginalSampleRate = sampleRate;
467 mAttributes = pAttributes != nullptr ? *pAttributes : AUDIO_ATTRIBUTES_INITIALIZER;
468 mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
Phil Burk33ff89b2015-11-30 11:16:01 -0800469
Eric Laurentd7f33c52022-01-06 13:54:56 +0100470 // update format and flags before storing them in mFormat, mOrigFlags and mFlags
471 if (pAttributes != NULL) {
472 // stream type shouldn't be looked at, this track has audio attributes
473 ALOGV("%s(): Building AudioTrack with attributes:"
474 " usage=%d content=%d flags=0x%x tags=[%s]",
475 __func__,
476 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
477 audio_flags_to_audio_output_flags(mAttributes.flags, &flags);
478 }
479
480 // these below should probably come from the audioFlinger too...
481 if (format == AUDIO_FORMAT_DEFAULT) {
482 format = AUDIO_FORMAT_PCM_16_BIT;
483 } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
484 flags = static_cast<audio_output_flags_t>(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
485 }
486
487 // force direct flag if format is not linear PCM
488 // or offload was requested
489 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
490 || !audio_is_linear_pcm(format)) {
491 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
492 ? "%s(): Offload request, forcing to Direct Output"
493 : "%s(): Not linear PCM, forcing to Direct Output",
494 __func__);
495 flags = (audio_output_flags_t)
496 // FIXME why can't we allow direct AND fast?
497 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
498 }
499
500 // force direct flag if HW A/V sync requested
501 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
502 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
503 }
504
505 mFormat = format;
506 mOrigFlags = mFlags = flags;
507
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800508 switch (transferType) {
509 case TRANSFER_DEFAULT:
510 if (sharedBuffer != 0) {
511 transferType = TRANSFER_SHARED;
Atneya Nairba809b82022-03-04 18:11:10 -0500512 } else if (callback == nullptr|| threadCanCallJava) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800513 transferType = TRANSFER_SYNC;
514 } else {
515 transferType = TRANSFER_CALLBACK;
516 }
517 break;
518 case TRANSFER_CALLBACK:
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700519 case TRANSFER_SYNC_NOTIF_CALLBACK:
Atneya Nairba809b82022-03-04 18:11:10 -0500520 if (callback == nullptr || sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800521 errorMessage = StringPrintf(
522 "%s: Transfer type %s but callback == nullptr || sharedBuffer != 0",
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700523 convertTransferToText(transferType), __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800524 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800525 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800526 }
527 break;
528 case TRANSFER_OBTAIN:
529 case TRANSFER_SYNC:
530 if (sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800531 errorMessage = StringPrintf(
532 "%s: Transfer type TRANSFER_OBTAIN but sharedBuffer != 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800533 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800534 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800535 }
536 break;
537 case TRANSFER_SHARED:
538 if (sharedBuffer == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800539 errorMessage = StringPrintf(
540 "%s: Transfer type TRANSFER_SHARED but sharedBuffer == 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800541 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800542 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800543 }
544 break;
545 default:
Andy Hung3acde2c2021-11-11 09:18:08 -0800546 errorMessage = StringPrintf("%s: Invalid transfer type %d", __func__, transferType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800547 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800548 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800549 }
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800550 mSharedBuffer = sharedBuffer;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800551 mTransfer = transferType;
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700552 mDoNotReconnect = doNotReconnect;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800553
Andy Hungfb8ede22018-09-12 19:03:24 -0700554 ALOGV_IF(sharedBuffer != 0, "%s(): sharedBuffer: %p, size: %zu",
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -0700555 __func__, sharedBuffer->unsecurePointer(), sharedBuffer->size());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800556
Glenn Kasten53cec222013-08-29 09:01:02 -0700557 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Eric Laurent1dd70b92009-04-21 07:56:33 -0700558 if (mAudioTrack != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800559 errorMessage = StringPrintf("%s: Track already in use", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800560 status = INVALID_OPERATION;
Andy Hung3acde2c2021-11-11 09:18:08 -0800561 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800562 }
563
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800564 // handle default values first.
Eric Laurente83b55d2014-11-14 10:06:21 -0800565 if (streamType == AUDIO_STREAM_DEFAULT) {
Dima Zavinfce7a472011-04-19 22:30:36 -0700566 streamType = AUDIO_STREAM_MUSIC;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800567 }
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700568 if (pAttributes == NULL) {
Eric Laurent223fd5c2014-11-11 13:43:36 -0800569 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800570 errorMessage = StringPrintf("%s: Invalid stream type %d", __func__, streamType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800571 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800572 goto error;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700573 }
Andy Hunga2159aa2021-07-20 13:01:52 -0700574 mOriginalStreamType = streamType;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700575 } else {
Andy Hunga2159aa2021-07-20 13:01:52 -0700576 mOriginalStreamType = AUDIO_STREAM_DEFAULT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800577 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800578
579 // validate parameters
Dima Zavinfce7a472011-04-19 22:30:36 -0700580 if (!audio_is_valid_format(format)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800581 errorMessage = StringPrintf("%s: Invalid format %#x", __func__, format);
Eric Laurentf32d7812017-11-30 14:44:07 -0800582 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800583 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800584 }
Eric Laurentc2f1f072009-07-17 12:17:14 -0700585
Glenn Kasten8ba90322013-10-30 11:29:27 -0700586 if (!audio_is_output_channel(channelMask)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800587 errorMessage = StringPrintf("%s: Invalid channel mask %#x", __func__, channelMask);
Eric Laurentf32d7812017-11-30 14:44:07 -0800588 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800589 goto error;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700590 }
Eric Laurentf32d7812017-11-30 14:44:07 -0800591 channelCount = audio_channel_count_from_out_mask(channelMask);
Glenn Kastene3247bf2014-02-24 15:19:07 -0800592 mChannelCount = channelCount;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700593
Eric Laurentd7f33c52022-01-06 13:54:56 +0100594 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
Phil Burkfdb3c072016-02-09 10:47:02 -0800595 if (audio_has_proportional_frames(format)) {
Glenn Kastenb7730382014-04-30 15:50:31 -0700596 mFrameSize = channelCount * audio_bytes_per_sample(format);
597 } else {
598 mFrameSize = sizeof(uint8_t);
599 }
Glenn Kastene3aa6592012-12-04 12:22:46 -0800600 } else {
Phil Burkfdb3c072016-02-09 10:47:02 -0800601 ALOG_ASSERT(audio_has_proportional_frames(format));
Glenn Kastenb7730382014-04-30 15:50:31 -0700602 mFrameSize = channelCount * audio_bytes_per_sample(format);
Glenn Kastenb7730382014-04-30 15:50:31 -0700603 // createTrack will return an error if PCM format is not supported by server,
604 // so no need to check for specific PCM formats here
Glenn Kastene3aa6592012-12-04 12:22:46 -0800605 }
606
Eric Laurent0d6db582014-11-12 18:39:44 -0800607 // sampling rate must be specified for direct outputs
Eric Laurentd7f33c52022-01-06 13:54:56 +0100608 if (sampleRate == 0 && (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800609 errorMessage = StringPrintf(
610 "%s: sample rate must be specified for direct outputs", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800611 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800612 goto error;
Eric Laurent0d6db582014-11-12 18:39:44 -0800613 }
Andy Hungff874dc2016-04-11 16:49:09 -0700614 // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
615 mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
Eric Laurent0d6db582014-11-12 18:39:44 -0800616
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800617 // Make copy of input parameter offloadInfo so that in the future:
618 // (a) createTrack_l doesn't need it as an input parameter
619 // (b) we can support re-creation of offloaded tracks
620 if (offloadInfo != NULL) {
621 mOffloadInfoCopy = *offloadInfo;
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800622 } else {
Eric Laurent20b9ef02016-12-05 11:03:16 -0800623 memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
Ytai Ben-Tsviffa2fd92020-10-20 09:13:53 -0700624 mOffloadInfoCopy = AUDIO_INFO_INITIALIZER;
ziyang zhangb3ec8aa2022-05-10 13:28:38 +0800625 mOffloadInfoCopy.format = format;
626 mOffloadInfoCopy.sample_rate = sampleRate;
627 mOffloadInfoCopy.channel_mask = channelMask;
628 mOffloadInfoCopy.stream_type = streamType;
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800629 }
630
Glenn Kasten66e46352014-01-16 17:44:23 -0800631 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
632 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
Glenn Kasten05632a52012-01-03 14:22:33 -0800633 mSendLevel = 0.0f;
Glenn Kasten396fabd2014-01-08 08:54:23 -0800634 // mFrameCount is initialized in createTrack_l
Glenn Kastenea38ee72016-04-18 11:08:01 -0700635 if (notificationFrames >= 0) {
636 mNotificationFramesReq = notificationFrames;
637 mNotificationsPerBufferReq = 0;
638 } else {
Eric Laurentd7f33c52022-01-06 13:54:56 +0100639 if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800640 errorMessage = StringPrintf(
641 "%s: notificationFrames=%d not permitted for non-fast track",
Andy Hungfb8ede22018-09-12 19:03:24 -0700642 __func__, notificationFrames);
Eric Laurentf32d7812017-11-30 14:44:07 -0800643 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800644 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700645 }
646 if (frameCount > 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700647 ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
648 __func__, notificationFrames, frameCount);
Eric Laurentf32d7812017-11-30 14:44:07 -0800649 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800650 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700651 }
652 mNotificationFramesReq = 0;
653 const uint32_t minNotificationsPerBuffer = 1;
654 const uint32_t maxNotificationsPerBuffer = 8;
655 mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
656 max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
657 ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
Andy Hungfb8ede22018-09-12 19:03:24 -0700658 "%s(): notificationFrames=%d clamped to the range -%u to -%u",
659 __func__,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700660 notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
661 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800662 mNotificationFramesAct = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700663 // TODO b/182392553: refactor or remove
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000664 mClientAttributionSource = AttributionSourceState(attributionSource);
Eric Laurentf32d7812017-11-30 14:44:07 -0800665 callingPid = IPCThreadState::self()->getCallingPid();
666 myPid = getpid();
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700667 if (uid == -1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000668 mClientAttributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700669 IPCThreadState::self()->getCallingUid()));
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800670 }
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700671 if (pid == (pid_t)-1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000672 mClientAttributionSource.pid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(callingPid));
Marco Nelissend457c972014-02-11 08:47:07 -0800673 }
Eric Laurent2beeb502010-07-16 07:43:46 -0700674 mAuxEffectId = 0;
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400675 mCallback = callback;
Eric Laurentbe916aa2010-06-01 23:49:17 -0700676
Atneya Nairba809b82022-03-04 18:11:10 -0500677 if (callback != nullptr) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400678 mAudioTrackThread = sp<AudioTrackThread>::make(*this);
Glenn Kastena997e7a2012-08-07 09:44:19 -0700679 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
Glenn Kastenbfd31842015-03-20 09:01:44 -0700680 // thread begins in paused state, and will not reference us until start()
Glenn Kastena997e7a2012-08-07 09:44:19 -0700681 }
682
Eric Laurent34f1d8e2009-11-04 08:27:26 -0800683 // create the IAudioTrack
Francois Gaffie24a9fb02019-01-18 17:51:34 +0100684 {
685 AutoMutex lock(mLock);
686 status = createTrack_l();
687 }
Glenn Kastena997e7a2012-08-07 09:44:19 -0700688 if (status != NO_ERROR) {
689 if (mAudioTrackThread != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100690 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
691 mAudioTrackThread->requestExitAndWait();
Glenn Kastena997e7a2012-08-07 09:44:19 -0700692 mAudioTrackThread.clear();
693 }
Andy Hung3acde2c2021-11-11 09:18:08 -0800694 // We do not goto error to prevent double-logging errors.
Eric Laurentf32d7812017-11-30 14:44:07 -0800695 goto exit;
Glenn Kasten5d464eb2012-06-22 17:19:53 -0700696 }
697
Andy Hung4ede21d2014-12-12 15:37:34 -0800698 mLoopCount = 0;
699 mLoopStart = 0;
700 mLoopEnd = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -0800701 mLoopCountNotified = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800702 mMarkerPosition = 0;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -0700703 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800704 mNewPosition = 0;
705 mUpdatePeriod = 0;
Glenn Kasten200092b2014-08-15 15:13:30 -0700706 mPosition = 0;
707 mReleased = 0;
Andy Hungffa36952017-08-17 10:41:51 -0700708 mStartNs = 0;
709 mStartFromZeroUs = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700710 AudioSystem::acquireAudioSessionId(mSessionId, pid, uid);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800711 mSequence = 1;
712 mObservedSequence = mSequence;
713 mInUnderrun = false;
Phil Burk1b420972015-04-22 10:52:21 -0700714 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700715 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700716 mTimestampRetrogradePositionReported = false;
717 mTimestampRetrogradeTimeReported = false;
718 mTimestampStallReported = false;
719 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700720 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Andy Hung65ffdfc2016-10-10 15:52:11 -0700721 mStartTs.mPosition = 0;
Phil Burk2812d9e2016-01-04 10:34:30 -0800722 mUnderrunCountOffset = 0;
Andy Hungea2b9c02016-02-12 17:06:53 -0800723 mFramesWritten = 0;
724 mFramesWrittenServerOffset = 0;
Andy Hungf20a4e92016-08-15 19:10:34 -0700725 mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
Ivan Lozano8cf3a072017-08-09 09:01:33 -0700726 mVolumeHandler = new media::VolumeHandler();
Eric Laurentf32d7812017-11-30 14:44:07 -0800727
Andy Hung3acde2c2021-11-11 09:18:08 -0800728error:
729 if (status != NO_ERROR) {
730 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
731 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
732 }
733 // fall through
Eric Laurentf32d7812017-11-30 14:44:07 -0800734exit:
735 mStatus = status;
736 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800737}
738
Mikhail Naganov55773032020-10-01 15:08:13 -0700739
740status_t AudioTrack::set(
741 audio_stream_type_t streamType,
742 uint32_t sampleRate,
743 audio_format_t format,
744 uint32_t channelMask,
745 size_t frameCount,
746 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400747 legacy_callback_t callback,
Mikhail Naganov55773032020-10-01 15:08:13 -0700748 void* user,
749 int32_t notificationFrames,
750 const sp<IMemory>& sharedBuffer,
751 bool threadCanCallJava,
752 audio_session_t sessionId,
753 transfer_type transferType,
754 const audio_offload_info_t *offloadInfo,
755 uid_t uid,
756 pid_t pid,
757 const audio_attributes_t* pAttributes,
758 bool doNotReconnect,
759 float maxRequiredSpeed,
760 audio_port_handle_t selectedDeviceId)
761{
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000762 AttributionSourceState attributionSource;
763 attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(uid));
764 attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(pid));
765 attributionSource.token = sp<BBinder>::make();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400766 if (callback) {
767 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
768 } else if (user) {
769 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
770 }
771 return set(streamType, sampleRate, format, static_cast<audio_channel_mask_t>(channelMask),
772 frameCount, flags, mLegacyCallbackWrapper, notificationFrames, sharedBuffer,
773 threadCanCallJava, sessionId, transferType, offloadInfo, attributionSource,
774 pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
Mikhail Naganov55773032020-10-01 15:08:13 -0700775}
776
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800777// -------------------------------------------------------------------------
778
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100779status_t AudioTrack::start()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800780{
Eric Laurentf5aafb22010-11-18 08:40:16 -0800781 AutoMutex lock(mLock);
Andy Hungb68f5eb2019-12-03 16:49:17 -0800782
Andy Hung10fb4be2020-05-27 22:22:22 -0700783 if (mState == STATE_ACTIVE) {
784 return INVALID_OPERATION;
785 }
786
787 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
788
789 // Defer logging here due to OpenSL ES repeated start calls.
790 // TODO(b/154868033) after fix, restore this logging back to the beginning of start().
791 const int64_t beginNs = systemTime();
Andy Hungb68f5eb2019-12-03 16:49:17 -0800792 status_t status = NO_ERROR; // logged: make sure to set this before returning.
Andy Hung06a730b2020-04-09 13:28:31 -0700793 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800794 mediametrics::LogItem(mMetricsId)
Andy Hunga6b27032020-04-27 10:34:24 -0700795 .set(AMEDIAMETRICS_PROP_CALLERNAME,
796 mCallerName.empty()
797 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
798 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800799 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_START)
Andy Hungea840382020-05-05 21:50:17 -0700800 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800801 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
802 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
803 .record(); });
804
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800805
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800806 mInUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800807
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800808 State previousState = mState;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100809 if (previousState == STATE_PAUSED_STOPPING) {
810 mState = STATE_STOPPING;
811 } else {
812 mState = STATE_ACTIVE;
813 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700814 (void) updateAndGetPosition_l();
Andy Hung65ffdfc2016-10-10 15:52:11 -0700815
816 // save start timestamp
817 if (isOffloadedOrDirect_l()) {
818 if (getTimestamp_l(mStartTs) != OK) {
819 mStartTs.mPosition = 0;
820 }
821 } else {
822 if (getTimestamp_l(&mStartEts) != OK) {
823 mStartEts.clear();
824 }
825 }
Andy Hungffa36952017-08-17 10:41:51 -0700826 mStartNs = systemTime(); // save this for timestamp adjustment after starting.
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800827 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
828 // reset current position as seen by client to 0
Glenn Kasten200092b2014-08-15 15:13:30 -0700829 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -0700830 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700831 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700832 mTimestampRetrogradePositionReported = false;
833 mTimestampRetrogradeTimeReported = false;
834 mTimestampStallReported = false;
835 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700836 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Phil Burk1b420972015-04-22 10:52:21 -0700837
Andy Hung65ffdfc2016-10-10 15:52:11 -0700838 if (!isOffloadedOrDirect_l()
839 && mStartEts.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) {
Andy Hunge1e98462016-04-12 10:18:51 -0700840 // Server side has consumed something, but is it finished consuming?
841 // It is possible since flush and stop are asynchronous that the server
842 // is still active at this point.
Andy Hungfb8ede22018-09-12 19:03:24 -0700843 ALOGV("%s(%d): server read:%lld cumulative flushed:%lld client written:%lld",
Eric Laurent973db022018-11-20 14:54:31 -0800844 __func__, mPortId,
Andy Hunge1e98462016-04-12 10:18:51 -0700845 (long long)(mFramesWrittenServerOffset
Andy Hung65ffdfc2016-10-10 15:52:11 -0700846 + mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]),
847 (long long)mStartEts.mFlushed,
Andy Hunge1e98462016-04-12 10:18:51 -0700848 (long long)mFramesWritten);
Andy Hungc4e60eb2017-09-06 11:14:57 -0700849 // mStartEts is already adjusted by mFramesWrittenServerOffset, so we delta adjust.
850 mFramesWrittenServerOffset -= mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER];
Andy Hung61be8412015-10-06 10:51:09 -0700851 }
Andy Hunge1e98462016-04-12 10:18:51 -0700852 mFramesWritten = 0;
853 mProxy->clearTimestamp(); // need new server push for valid timestamp
854 mMarkerReached = false;
Andy Hung61be8412015-10-06 10:51:09 -0700855
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700856 // For offloaded tracks, we don't know if the hardware counters are really zero here,
857 // since the flush is asynchronous and stop may not fully drain.
858 // We save the time when the track is started to later verify whether
859 // the counters are realistic (i.e. start from zero after this time).
Andy Hungffa36952017-08-17 10:41:51 -0700860 mStartFromZeroUs = mStartNs / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700861
Eric Laurentec9a0322013-08-28 10:23:01 -0700862 // force refresh of remaining frames by processAudioBuffer() as last
863 // write before stop could be partial.
864 mRefreshRemaining = true;
Tomoharu Kasahara6e8bf982017-06-09 16:17:33 +0900865
866 // for static track, clear the old flags when starting from stopped state
867 if (mSharedBuffer != 0) {
868 android_atomic_and(
869 ~(CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
870 &mCblk->mFlags);
871 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800872 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700873 mNewPosition = mPosition + mUpdatePeriod;
Andy Hung4be3b832016-10-13 17:51:43 -0700874 int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800875
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800876 if (!(flags & CBLK_INVALID)) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -0800877 mAudioTrack->start(&status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800878 if (status == DEAD_OBJECT) {
879 flags |= CBLK_INVALID;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800880 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800881 }
882 if (flags & CBLK_INVALID) {
883 status = restoreTrack_l("start");
884 }
885
Andy Hung79629f02016-03-24 13:57:40 -0700886 // resume or pause the callback thread as needed.
887 sp<AudioTrackThread> t = mAudioTrackThread;
888 if (status == NO_ERROR) {
889 if (t != 0) {
890 if (previousState == STATE_STOPPING) {
891 mProxy->interrupt();
892 } else {
893 t->resume();
894 }
895 } else {
896 mPreviousPriority = getpriority(PRIO_PROCESS, 0);
897 get_sched_policy(0, &mPreviousSchedulingGroup);
898 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
899 }
Andy Hung39399b62017-04-21 15:07:45 -0700900
901 // Start our local VolumeHandler for restoration purposes.
902 mVolumeHandler->setStarted();
Andy Hung79629f02016-03-24 13:57:40 -0700903 } else {
Eric Laurent973db022018-11-20 14:54:31 -0800904 ALOGE("%s(%d): status %d", __func__, mPortId, status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800905 mState = previousState;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800906 if (t != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100907 if (previousState != STATE_STOPPING) {
908 t->pause();
909 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800910 } else {
Glenn Kasten87913512011-06-22 16:15:25 -0700911 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
Glenn Kastena6364332012-04-19 09:35:04 -0700912 set_sched_policy(0, mPreviousSchedulingGroup);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800913 }
914 }
915
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100916 return status;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800917}
918
919void AudioTrack::stop()
920{
Andy Hungb68f5eb2019-12-03 16:49:17 -0800921 const int64_t beginNs = systemTime();
922
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800923 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -0700924 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800925 mediametrics::LogItem(mMetricsId)
926 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_STOP)
Andy Hungea840382020-05-05 21:50:17 -0700927 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800928 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Phil Burk64e16a72020-06-01 13:25:51 -0700929 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
930 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getUnderrunCount_l())
Phil Burka9876702020-04-20 18:16:15 -0700931 .record();
Phil Burka9876702020-04-20 18:16:15 -0700932 });
Andy Hungb68f5eb2019-12-03 16:49:17 -0800933
Eric Laurent973db022018-11-20 14:54:31 -0800934 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -0700935
Glenn Kasten397edb32013-08-30 15:10:13 -0700936 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800937 return;
938 }
939
Glenn Kasten23a75452014-01-13 10:37:17 -0800940 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100941 mState = STATE_STOPPING;
942 } else {
943 mState = STATE_STOPPED;
Andy Hung2148bf02016-11-28 19:01:02 -0800944 ALOGD_IF(mSharedBuffer == nullptr,
Eric Laurent973db022018-11-20 14:54:31 -0800945 "%s(%d): called with %u frames delivered", __func__, mPortId, mReleased.value());
Andy Hungc2813e52014-10-16 17:54:34 -0700946 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100947 }
948
Andy Hung1d3556d2018-03-29 16:30:14 -0700949 mProxy->stop(); // notify server not to read beyond current client position until start().
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800950 mProxy->interrupt();
951 mAudioTrack->stop();
Andy Hung61be8412015-10-06 10:51:09 -0700952
953 // Note: legacy handling - stop does not clear playback marker
954 // and periodic update counter, but flush does for streaming tracks.
Andy Hung9b461582014-12-01 17:56:29 -0800955
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800956 if (mSharedBuffer != 0) {
Andy Hung9b461582014-12-01 17:56:29 -0800957 // clear buffer position and loop count.
Andy Hung9b461582014-12-01 17:56:29 -0800958 mStaticProxy->setBufferPositionAndLoop(0 /* position */,
959 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800960 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100961
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800962 sp<AudioTrackThread> t = mAudioTrackThread;
963 if (t != 0) {
Glenn Kasten23a75452014-01-13 10:37:17 -0800964 if (!isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100965 t->pause();
Jean-Michel Trivia60e9a22018-11-15 16:13:36 -0800966 } else if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
Jean-Michel Trivi0248a2a2018-11-21 10:05:57 -0800967 // causes wake up of the playback thread, that will callback the client for
968 // EVENT_STREAM_END in processAudioBuffer()
969 t->wake();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100970 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800971 } else {
972 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
973 set_sched_policy(0, mPreviousSchedulingGroup);
974 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800975}
976
977bool AudioTrack::stopped() const
978{
Glenn Kasten9a2aaf92012-01-03 09:42:47 -0800979 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800980 return mState != STATE_ACTIVE;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800981}
982
983void AudioTrack::flush()
984{
Andy Hungb68f5eb2019-12-03 16:49:17 -0800985 const int64_t beginNs = systemTime();
Andy Hungfb8ede22018-09-12 19:03:24 -0700986 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -0700987 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800988 mediametrics::LogItem(mMetricsId)
989 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_FLUSH)
Andy Hungea840382020-05-05 21:50:17 -0700990 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800991 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
992 .record(); });
993
Eric Laurent973db022018-11-20 14:54:31 -0800994 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -0700995
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800996 if (mSharedBuffer != 0) {
997 return;
Glenn Kasten4bae3642012-11-30 13:41:12 -0800998 }
Andy Hung4c5ed302018-05-09 11:16:21 -0700999 if (mState == STATE_ACTIVE) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001000 return;
1001 }
1002 flush_l();
Eric Laurent1703cdf2011-03-07 14:52:59 -08001003}
1004
Eric Laurent1703cdf2011-03-07 14:52:59 -08001005void AudioTrack::flush_l()
1006{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001007 ALOG_ASSERT(mState != STATE_ACTIVE);
Eric Laurentc2f1f072009-07-17 12:17:14 -07001008
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001009 // clear playback marker and periodic update counter
1010 mMarkerPosition = 0;
1011 mMarkerReached = false;
1012 mUpdatePeriod = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001013 mRefreshRemaining = true;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001014
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001015 mState = STATE_FLUSHED;
Andy Hungc2813e52014-10-16 17:54:34 -07001016 mReleased = 0;
Glenn Kasten23a75452014-01-13 10:37:17 -08001017 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001018 mProxy->interrupt();
1019 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001020 mProxy->flush();
Glenn Kasten4bae3642012-11-30 13:41:12 -08001021 mAudioTrack->flush();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001022}
1023
Andy Hung959b5b82021-09-24 10:46:20 -07001024bool AudioTrack::pauseAndWait(const std::chrono::milliseconds& timeout)
1025{
1026 using namespace std::chrono_literals;
1027
Andy Hungd87a53a2022-01-19 16:56:17 -08001028 // We use atomic access here for state variables - these are used as hints
1029 // to ensure we have ramped down audio.
1030 const int priorState = mProxy->getState();
1031 const uint32_t priorPosition = mProxy->getPosition().unsignedValue();
1032
Andy Hung959b5b82021-09-24 10:46:20 -07001033 pause();
1034
Andy Hungd87a53a2022-01-19 16:56:17 -08001035 // Only if we were previously active, do we wait to ramp down the audio.
1036 if (priorState != CBLK_STATE_ACTIVE) return true;
1037
Andy Hung959b5b82021-09-24 10:46:20 -07001038 AutoMutex lock(mLock);
1039 // offload and direct tracks do not wait because pause volume ramp is handled by hardware.
1040 if (isOffloadedOrDirect_l()) return true;
1041
1042 // Wait for the track state to be anything besides pausing.
1043 // This ensures that the volume has ramped down.
1044 constexpr auto SLEEP_INTERVAL_MS = 10ms;
Andy Hungd87a53a2022-01-19 16:56:17 -08001045 constexpr auto POSITION_TIMEOUT_MS = 40ms; // don't wait longer than this for position change.
Andy Hung959b5b82021-09-24 10:46:20 -07001046 auto begin = std::chrono::steady_clock::now();
1047 while (true) {
Andy Hungd87a53a2022-01-19 16:56:17 -08001048 // Wait for state and position to change.
1049 // After pause() the server state should be PAUSING, but that may immediately
1050 // convert to PAUSED by prepareTracks before data is read into the mixer.
1051 // Hence we check that the state is not PAUSING and that the server position
1052 // has advanced to be a more reliable estimate that the volume ramp has completed.
Andy Hung959b5b82021-09-24 10:46:20 -07001053 const int state = mProxy->getState();
Andy Hungd87a53a2022-01-19 16:56:17 -08001054 const uint32_t position = mProxy->getPosition().unsignedValue();
Andy Hung959b5b82021-09-24 10:46:20 -07001055
1056 mLock.unlock(); // only local variables accessed until lock.
1057 auto elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(
1058 std::chrono::steady_clock::now() - begin);
Andy Hungd87a53a2022-01-19 16:56:17 -08001059 if (state != CBLK_STATE_PAUSING &&
1060 (elapsed >= POSITION_TIMEOUT_MS || position != priorPosition)) {
1061 ALOGV("%s: success state:%d, position:%u after %lld ms"
1062 " (prior state:%d prior position:%u)",
1063 __func__, state, position, elapsed.count(), priorState, priorPosition);
Andy Hung959b5b82021-09-24 10:46:20 -07001064 return true;
1065 }
1066 std::chrono::milliseconds remaining = timeout - elapsed;
1067 if (remaining.count() <= 0) {
1068 ALOGW("%s: timeout expired state:%d still pausing:%d after %lld ms",
1069 __func__, state, CBLK_STATE_PAUSING, elapsed.count());
1070 return false;
1071 }
1072 // It is conceivable that the track is restored while sleeping;
1073 // as this logic is advisory, we allow that.
1074 std::this_thread::sleep_for(std::min(remaining, SLEEP_INTERVAL_MS));
1075 mLock.lock();
1076 }
1077}
1078
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001079void AudioTrack::pause()
1080{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001081 const int64_t beginNs = systemTime();
Eric Laurentf5aafb22010-11-18 08:40:16 -08001082 AutoMutex lock(mLock);
Andy Hunga6b27032020-04-27 10:34:24 -07001083 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001084 mediametrics::LogItem(mMetricsId)
1085 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_PAUSE)
Andy Hungea840382020-05-05 21:50:17 -07001086 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001087 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
1088 .record(); });
1089
Eric Laurent973db022018-11-20 14:54:31 -08001090 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001091
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001092 if (mState == STATE_ACTIVE) {
1093 mState = STATE_PAUSED;
1094 } else if (mState == STATE_STOPPING) {
1095 mState = STATE_PAUSED_STOPPING;
1096 } else {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001097 return;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001098 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001099 mProxy->interrupt();
1100 mAudioTrack->pause();
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001101
Marco Nelissen3a90f282014-03-10 11:21:43 -07001102 if (isOffloaded_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001103 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001104 // An offload output can be re-used between two audio tracks having
1105 // the same configuration. A timestamp query for a paused track
1106 // while the other is running would return an incorrect time.
1107 // To fix this, cache the playback position on a pause() and return
1108 // this time when requested until the track is resumed.
1109
1110 // OffloadThread sends HAL pause in its threadLoop. Time saved
1111 // here can be slightly off.
1112
1113 // TODO: check return code for getRenderPosition.
1114
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001115 uint32_t halFrames;
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001116 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition);
Andy Hungfb8ede22018-09-12 19:03:24 -07001117 ALOGV("%s(%d): for offload, cache current position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001118 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001119 }
1120 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001121}
1122
Eric Laurentbe916aa2010-06-01 23:49:17 -07001123status_t AudioTrack::setVolume(float left, float right)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001124{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001125 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1126 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY ||
1127 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001128 return BAD_VALUE;
1129 }
1130
Andy Hungb68f5eb2019-12-03 16:49:17 -08001131 mediametrics::LogItem(mMetricsId)
1132 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOLUME)
1133 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)left)
1134 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)right)
1135 .record();
1136
Eric Laurent1703cdf2011-03-07 14:52:59 -08001137 AutoMutex lock(mLock);
Glenn Kasten66e46352014-01-16 17:44:23 -08001138 mVolume[AUDIO_INTERLEAVE_LEFT] = left;
1139 mVolume[AUDIO_INTERLEAVE_RIGHT] = right;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001140
Glenn Kastenc56f3422014-03-21 17:53:17 -07001141 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right)));
Eric Laurentbe916aa2010-06-01 23:49:17 -07001142
Glenn Kasten23a75452014-01-13 10:37:17 -08001143 if (isOffloaded_l()) {
Eric Laurent59fe0102013-09-27 18:48:26 -07001144 mAudioTrack->signal();
1145 }
Eric Laurentbe916aa2010-06-01 23:49:17 -07001146 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001147}
1148
Glenn Kastenb1c09932012-02-27 16:21:04 -08001149status_t AudioTrack::setVolume(float volume)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001150{
Glenn Kastenb1c09932012-02-27 16:21:04 -08001151 return setVolume(volume, volume);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001152}
1153
Eric Laurent2beeb502010-07-16 07:43:46 -07001154status_t AudioTrack::setAuxEffectSendLevel(float level)
Eric Laurentbe916aa2010-06-01 23:49:17 -07001155{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001156 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1157 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001158 return BAD_VALUE;
1159 }
1160
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001161 AutoMutex lock(mLock);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001162 mSendLevel = level;
Glenn Kastene3aa6592012-12-04 12:22:46 -08001163 mProxy->setSendLevel(level);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001164
1165 return NO_ERROR;
1166}
1167
Glenn Kastena5224f32012-01-04 12:41:44 -08001168void AudioTrack::getAuxEffectSendLevel(float* level) const
Eric Laurentbe916aa2010-06-01 23:49:17 -07001169{
1170 if (level != NULL) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001171 *level = mSendLevel;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001172 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001173}
1174
Glenn Kasten3b16c762012-11-14 08:44:39 -08001175status_t AudioTrack::setSampleRate(uint32_t rate)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001176{
Andy Hung5cbb5782015-03-27 18:39:59 -07001177 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08001178 ALOGV("%s(%d): prior state:%s rate:%u", __func__, mPortId, stateToString(mState), rate);
Andy Hungfb8ede22018-09-12 19:03:24 -07001179
Andy Hung5cbb5782015-03-27 18:39:59 -07001180 if (rate == mSampleRate) {
1181 return NO_ERROR;
1182 }
jiabinf4de6112018-12-19 12:40:08 -08001183 if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)
1184 || (mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL)) {
John Grossman4ff14ba2012-02-08 16:37:41 -08001185 return INVALID_OPERATION;
1186 }
Eric Laurent0d6db582014-11-12 18:39:44 -08001187 if (mOutput == AUDIO_IO_HANDLE_NONE) {
1188 return NO_INIT;
1189 }
Andy Hung5cbb5782015-03-27 18:39:59 -07001190 // NOTE: it is theoretically possible, but highly unlikely, that a device change
1191 // could mean a previously allowed sampling rate is no longer allowed.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001192 uint32_t afSamplingRate;
Eric Laurent0d6db582014-11-12 18:39:44 -08001193 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
Eric Laurent57326622009-07-07 07:10:45 -07001194 return NO_INIT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001195 }
Andy Hung26145642015-04-15 21:56:53 -07001196 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001197 const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001198 if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001199 return BAD_VALUE;
1200 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001201 // TODO: Should we also check if the buffer size is compatible?
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001202
Glenn Kastene3aa6592012-12-04 12:22:46 -08001203 mSampleRate = rate;
Andy Hung26145642015-04-15 21:56:53 -07001204 mProxy->setSampleRate(effectiveSampleRate);
Glenn Kastene3aa6592012-12-04 12:22:46 -08001205
Eric Laurent57326622009-07-07 07:10:45 -07001206 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001207}
1208
Glenn Kastena5224f32012-01-04 12:41:44 -08001209uint32_t AudioTrack::getSampleRate() const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001210{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001211 AutoMutex lock(mLock);
Eric Laurent6f59db12013-07-26 17:16:50 -07001212
1213 // sample rate can be updated during playback by the offloaded decoder so we need to
1214 // query the HAL and update if needed.
1215// FIXME use Proxy return channel to update the rate from server and avoid polling here
Eric Laurentab5cdba2014-06-09 17:22:27 -07001216 if (isOffloadedOrDirect_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001217 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent6f59db12013-07-26 17:16:50 -07001218 uint32_t sampleRate = 0;
Jean-Michel Trivib7f24b12014-06-11 10:05:30 -07001219 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
Eric Laurent6f59db12013-07-26 17:16:50 -07001220 if (status == NO_ERROR) {
1221 mSampleRate = sampleRate;
1222 }
1223 }
1224 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001225 return mSampleRate;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001226}
1227
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001228uint32_t AudioTrack::getOriginalSampleRate() const
1229{
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001230 return mOriginalSampleRate;
1231}
1232
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001233status_t AudioTrack::setDualMonoMode(audio_dual_mono_mode_t mode)
1234{
1235 AutoMutex lock(mLock);
1236 return setDualMonoMode_l(mode);
1237}
1238
1239status_t AudioTrack::setDualMonoMode_l(audio_dual_mono_mode_t mode)
1240{
1241 const status_t status = statusTFromBinderStatus(
1242 mAudioTrack->setDualMonoMode(VALUE_OR_RETURN_STATUS(
1243 legacy2aidl_audio_dual_mono_mode_t_AudioDualMonoMode(mode))));
1244 if (status == NO_ERROR) mDualMonoMode = mode;
1245 return status;
1246}
1247
1248status_t AudioTrack::getDualMonoMode(audio_dual_mono_mode_t* mode) const
1249{
1250 AutoMutex lock(mLock);
Mikhail Naganovb1a075b2022-12-18 02:48:14 +00001251 media::audio::common::AudioDualMonoMode mediaMode;
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001252 const status_t status = statusTFromBinderStatus(mAudioTrack->getDualMonoMode(&mediaMode));
1253 if (status == NO_ERROR) {
1254 *mode = VALUE_OR_RETURN_STATUS(
1255 aidl2legacy_AudioDualMonoMode_audio_dual_mono_mode_t(mediaMode));
1256 }
1257 return status;
1258}
1259
1260status_t AudioTrack::setAudioDescriptionMixLevel(float leveldB)
1261{
1262 AutoMutex lock(mLock);
1263 return setAudioDescriptionMixLevel_l(leveldB);
1264}
1265
1266status_t AudioTrack::setAudioDescriptionMixLevel_l(float leveldB)
1267{
1268 const status_t status = statusTFromBinderStatus(
1269 mAudioTrack->setAudioDescriptionMixLevel(leveldB));
1270 if (status == NO_ERROR) mAudioDescriptionMixLeveldB = leveldB;
1271 return status;
1272}
1273
1274status_t AudioTrack::getAudioDescriptionMixLevel(float* leveldB) const
1275{
1276 AutoMutex lock(mLock);
1277 return statusTFromBinderStatus(mAudioTrack->getAudioDescriptionMixLevel(leveldB));
1278}
1279
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001280status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
Andy Hung8edb8dc2015-03-26 19:13:55 -07001281{
Andy Hung8edb8dc2015-03-26 19:13:55 -07001282 AutoMutex lock(mLock);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001283 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
Andy Hung8edb8dc2015-03-26 19:13:55 -07001284 return NO_ERROR;
1285 }
Glenn Kastend79072e2016-01-06 08:41:20 -08001286 if (isOffloadedOrDirect_l()) {
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001287 const status_t status = statusTFromBinderStatus(mAudioTrack->setPlaybackRateParameters(
1288 VALUE_OR_RETURN_STATUS(
1289 legacy2aidl_audio_playback_rate_t_AudioPlaybackRate(playbackRate))));
1290 if (status == NO_ERROR) {
1291 mPlaybackRate = playbackRate;
Bertil Ã…kesson081fb742022-07-14 16:39:36 +02001292 } else if (status == INVALID_OPERATION
1293 && playbackRate.mSpeed == 1.0f && mPlaybackRate.mPitch == 1.0f) {
1294 mPlaybackRate = playbackRate;
1295 return NO_ERROR;
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001296 }
1297 return status;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001298 }
1299 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
1300 return INVALID_OPERATION;
1301 }
Andy Hungff874dc2016-04-11 16:49:09 -07001302
Andy Hungfb8ede22018-09-12 19:03:24 -07001303 ALOGV("%s(%d): mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001304 __func__, mPortId, mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001305 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001306 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
1307 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
1308 const float effectivePitch = adjustPitch(playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001309 AudioPlaybackRate playbackRateTemp = playbackRate;
1310 playbackRateTemp.mSpeed = effectiveSpeed;
1311 playbackRateTemp.mPitch = effectivePitch;
1312
Andy Hungfb8ede22018-09-12 19:03:24 -07001313 ALOGV("%s(%d) (effective) mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001314 __func__, mPortId, effectiveRate, effectiveSpeed, effectivePitch);
Andy Hungff874dc2016-04-11 16:49:09 -07001315
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001316 if (!isAudioPlaybackRateValid(playbackRateTemp)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001317 ALOGW("%s(%d) (%f, %f) failed (effective rate out of bounds)",
Eric Laurent973db022018-11-20 14:54:31 -08001318 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001319 return BAD_VALUE;
1320 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001321 // Check if the buffer size is compatible.
Andy Hung26145642015-04-15 21:56:53 -07001322 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001323 ALOGW("%s(%d) (%f, %f) failed (buffer size)",
Eric Laurent973db022018-11-20 14:54:31 -08001324 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung8edb8dc2015-03-26 19:13:55 -07001325 return BAD_VALUE;
1326 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001327
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001328 // Check resampler ratios are within bounds
Glenn Kastend3bb6452016-12-05 18:14:37 -08001329 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
1330 (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001331 ALOGW("%s(%d) (%f, %f) failed. Resample rate exceeds max accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001332 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001333 return BAD_VALUE;
1334 }
1335
Dan Austine34eae22015-10-27 16:14:52 -07001336 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001337 ALOGW("%s(%d) (%f, %f) failed. Resample rate below min accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001338 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001339 return BAD_VALUE;
1340 }
1341 mPlaybackRate = playbackRate;
1342 //set effective rates
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001343 mProxy->setPlaybackRate(playbackRateTemp);
Andy Hung26145642015-04-15 21:56:53 -07001344 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
Andy Hungb68f5eb2019-12-03 16:49:17 -08001345
1346 mediametrics::LogItem(mMetricsId)
1347 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYBACKPARAM)
1348 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
1349 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
1350 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
1351 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1352 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveRate)
1353 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1354 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)playbackRateTemp.mSpeed)
1355 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1356 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)playbackRateTemp.mPitch)
1357 .record();
1358
Andy Hung8edb8dc2015-03-26 19:13:55 -07001359 return NO_ERROR;
1360}
1361
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001362const AudioPlaybackRate& AudioTrack::getPlaybackRate()
Andy Hung8edb8dc2015-03-26 19:13:55 -07001363{
1364 AutoMutex lock(mLock);
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001365 if (isOffloadedOrDirect_l()) {
Mikhail Naganovb1a075b2022-12-18 02:48:14 +00001366 media::audio::common::AudioPlaybackRate playbackRateTemp;
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001367 const status_t status = statusTFromBinderStatus(
1368 mAudioTrack->getPlaybackRateParameters(&playbackRateTemp));
1369 if (status == NO_ERROR) { // update local version if changed.
1370 mPlaybackRate =
1371 aidl2legacy_AudioPlaybackRate_audio_playback_rate_t(playbackRateTemp).value();
1372 }
1373 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001374 return mPlaybackRate;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001375}
1376
Phil Burkc0adecb2016-01-08 12:44:11 -08001377ssize_t AudioTrack::getBufferSizeInFrames()
1378{
1379 AutoMutex lock(mLock);
1380 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1381 return NO_INIT;
1382 }
Phil Burka9876702020-04-20 18:16:15 -07001383
Phil Burke8972b02016-03-04 11:29:57 -08001384 return (ssize_t) mProxy->getBufferSizeInFrames();
Phil Burkc0adecb2016-01-08 12:44:11 -08001385}
1386
Andy Hungf2c87b32016-04-07 19:49:29 -07001387status_t AudioTrack::getBufferDurationInUs(int64_t *duration)
1388{
1389 if (duration == nullptr) {
1390 return BAD_VALUE;
1391 }
1392 AutoMutex lock(mLock);
1393 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1394 return NO_INIT;
1395 }
1396 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames();
1397 if (bufferSizeInFrames < 0) {
1398 return (status_t)bufferSizeInFrames;
1399 }
1400 *duration = (int64_t)((double)bufferSizeInFrames * 1000000
1401 / ((double)mSampleRate * mPlaybackRate.mSpeed));
1402 return NO_ERROR;
1403}
1404
Phil Burkc0adecb2016-01-08 12:44:11 -08001405ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames)
1406{
1407 AutoMutex lock(mLock);
1408 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1409 return NO_INIT;
1410 }
Phil Burka9876702020-04-20 18:16:15 -07001411
1412 ssize_t originalBufferSize = mProxy->getBufferSizeInFrames();
1413 ssize_t finalBufferSize = mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
1414 if (originalBufferSize != finalBufferSize) {
Phil Burk64e16a72020-06-01 13:25:51 -07001415 android::mediametrics::LogItem(mMetricsId)
1416 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
1417 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1418 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t)getUnderrunCount_l())
1419 .record();
Phil Burka9876702020-04-20 18:16:15 -07001420 }
1421 return finalBufferSize;
Phil Burkc0adecb2016-01-08 12:44:11 -08001422}
1423
Andy Hung3c7f47a2021-03-16 17:30:09 -07001424ssize_t AudioTrack::getStartThresholdInFrames() const
1425{
1426 AutoMutex lock(mLock);
1427 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1428 return NO_INIT;
1429 }
1430 return (ssize_t) mProxy->getStartThresholdInFrames();
1431}
1432
1433ssize_t AudioTrack::setStartThresholdInFrames(size_t startThresholdInFrames)
1434{
1435 if (startThresholdInFrames > INT32_MAX || startThresholdInFrames == 0) {
1436 // contractually we could simply return the current threshold in frames
1437 // to indicate the request was ignored, but we return an error here.
1438 return BAD_VALUE;
1439 }
1440 AutoMutex lock(mLock);
1441 // We do not permit calling setStartThresholdInFrames() between the AudioTrack
1442 // default ctor AudioTrack() and set(...) but rather fail such an attempt.
1443 // (To do so would require a cached mOrigStartThresholdInFrames and we may
1444 // not have proper validation for the actual set value).
1445 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1446 return NO_INIT;
1447 }
1448 const uint32_t original = mProxy->getStartThresholdInFrames();
1449 const uint32_t final = mProxy->setStartThresholdInFrames(startThresholdInFrames);
1450 if (original != final) {
1451 android::mediametrics::LogItem(mMetricsId)
1452 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD)
1453 .set(AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES, (int32_t)final)
1454 .record();
1455 if (original > final) {
1456 // restart track if it was disabled by audioflinger due to previous underrun
1457 // and we reduced the number of frames for the threshold.
1458 restartIfDisabled();
1459 }
1460 }
1461 return final;
1462}
1463
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001464status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1465{
Glenn Kastend79072e2016-01-06 08:41:20 -08001466 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001467 return INVALID_OPERATION;
1468 }
1469
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001470 if (loopCount == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001471 ;
1472 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount &&
1473 loopEnd - loopStart >= MIN_LOOP) {
1474 ;
1475 } else {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001476 return BAD_VALUE;
1477 }
1478
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001479 AutoMutex lock(mLock);
1480 // See setPosition() regarding setting parameters such as loop points or position while active
1481 if (mState == STATE_ACTIVE) {
1482 return INVALID_OPERATION;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001483 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001484 setLoop_l(loopStart, loopEnd, loopCount);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001485 return NO_ERROR;
1486}
1487
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001488void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1489{
Andy Hung4ede21d2014-12-12 15:37:34 -08001490 // We do not update the periodic notification point.
1491 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
1492 mLoopCount = loopCount;
1493 mLoopEnd = loopEnd;
1494 mLoopStart = loopStart;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001495 mLoopCountNotified = loopCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001496 mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
Andy Hung3c09c782014-12-29 18:39:32 -08001497
1498 // Waking the AudioTrackThread is not needed as this cannot be called when active.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001499}
1500
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001501status_t AudioTrack::setMarkerPosition(uint32_t marker)
1502{
Atneya Nair14aabae2021-11-30 17:36:24 -05001503 AutoMutex lock(mLock);
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001504 // The only purpose of setting marker position is to get a callback
Atneya Nair14aabae2021-11-30 17:36:24 -05001505 if (!mCallback.promote() || isOffloadedOrDirect_l()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001506 return INVALID_OPERATION;
1507 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001508
1509 mMarkerPosition = marker;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001510 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001511
Andy Hung3c09c782014-12-29 18:39:32 -08001512 sp<AudioTrackThread> t = mAudioTrackThread;
1513 if (t != 0) {
1514 t->wake();
1515 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001516 return NO_ERROR;
1517}
1518
Glenn Kastena5224f32012-01-04 12:41:44 -08001519status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001520{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001521 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001522 return INVALID_OPERATION;
1523 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001524 if (marker == NULL) {
1525 return BAD_VALUE;
1526 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001527
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001528 AutoMutex lock(mLock);
Andy Hung90e8a972015-11-09 16:42:40 -08001529 mMarkerPosition.getValue(marker);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001530
1531 return NO_ERROR;
1532}
1533
1534status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
1535{
Atneya Nair14aabae2021-11-30 17:36:24 -05001536 AutoMutex lock(mLock);
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001537 // The only purpose of setting position update period is to get a callback
Atneya Nair14aabae2021-11-30 17:36:24 -05001538 if (!mCallback.promote() || isOffloadedOrDirect_l()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001539 return INVALID_OPERATION;
1540 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001541
Glenn Kasten200092b2014-08-15 15:13:30 -07001542 mNewPosition = updateAndGetPosition_l() + updatePeriod;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001543 mUpdatePeriod = updatePeriod;
Glenn Kasten2b2165c2014-01-13 08:53:36 -08001544
Andy Hung3c09c782014-12-29 18:39:32 -08001545 sp<AudioTrackThread> t = mAudioTrackThread;
1546 if (t != 0) {
1547 t->wake();
1548 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001549 return NO_ERROR;
1550}
1551
Glenn Kastena5224f32012-01-04 12:41:44 -08001552status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001553{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001554 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001555 return INVALID_OPERATION;
1556 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001557 if (updatePeriod == NULL) {
1558 return BAD_VALUE;
1559 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001560
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001561 AutoMutex lock(mLock);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001562 *updatePeriod = mUpdatePeriod;
1563
1564 return NO_ERROR;
1565}
1566
1567status_t AudioTrack::setPosition(uint32_t position)
1568{
Glenn Kastend79072e2016-01-06 08:41:20 -08001569 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001570 return INVALID_OPERATION;
1571 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001572 if (position > mFrameCount) {
1573 return BAD_VALUE;
1574 }
John Grossman4ff14ba2012-02-08 16:37:41 -08001575
Eric Laurent1703cdf2011-03-07 14:52:59 -08001576 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001577 // Currently we require that the player is inactive before setting parameters such as position
1578 // or loop points. Otherwise, there could be a race condition: the application could read the
1579 // current position, compute a new position or loop parameters, and then set that position or
1580 // loop parameters but it would do the "wrong" thing since the position has continued to advance
1581 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app
1582 // to specify how it wants to handle such scenarios.
1583 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001584 return INVALID_OPERATION;
1585 }
Andy Hung9b461582014-12-01 17:56:29 -08001586 // After setting the position, use full update period before notification.
Glenn Kasten200092b2014-08-15 15:13:30 -07001587 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001588 mStaticProxy->setBufferPosition(position);
Andy Hung3c09c782014-12-29 18:39:32 -08001589
1590 // Waking the AudioTrackThread is not needed as this cannot be called when active.
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001591 return NO_ERROR;
1592}
1593
Glenn Kasten200092b2014-08-15 15:13:30 -07001594status_t AudioTrack::getPosition(uint32_t *position)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001595{
Glenn Kastend65d73c2012-06-22 17:21:07 -07001596 if (position == NULL) {
1597 return BAD_VALUE;
1598 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001599
Eric Laurent1703cdf2011-03-07 14:52:59 -08001600 AutoMutex lock(mLock);
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001601 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
1602 if (mState == STATE_STOPPED || mState == STATE_FLUSHED) {
1603 *position = 0;
1604 return NO_ERROR;
1605 }
Andy Hung7a490e72016-03-23 15:58:10 -07001606 // FIXME: offloaded and direct tracks call into the HAL for render positions
1607 // for compressed/synced data; however, we use proxy position for pure linear pcm data
1608 // as we do not know the capability of the HAL for pcm position support and standby.
1609 // There may be some latency differences between the HAL position and the proxy position.
1610 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) {
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
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001618 uint32_t dspFrames = 0;
Glenn Kasten142f5192014-03-25 17:44:59 -07001619 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung1f1db832015-06-08 13:26:10 -07001620 uint32_t halFrames; // actually unused
Andy Hung1f1db832015-06-08 13:26:10 -07001621 // FIXME: on getRenderPosition() error, we return OK with frame position 0.
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001622 if (AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames) != NO_ERROR) {
1623 *position = 0;
1624 return NO_ERROR;
1625 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001626 }
1627 *position = dspFrames;
1628 } else {
Eric Laurent275e8e92014-11-30 15:14:47 -08001629 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung1f1db832015-06-08 13:26:10 -07001630 (void) restoreTrack_l("getPosition");
1631 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l()
1632 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position.
Eric Laurent275e8e92014-11-30 15:14:47 -08001633 }
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001634 *position = updateAndGetPosition_l().value();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001635 }
Dean Wheatley0f51fd02022-08-31 16:41:40 +10001636
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;
Glenn Kasten38e905b2014-01-13 10:21:48 -08001955 // We retain a copy of the I/O handle, but don't own the reference
Eric Laurent21da6472017-11-09 16:29:26 -08001956 mOutput = output.outputId;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001957 mRefreshRemaining = true;
1958
1959 // Starting address of buffers in shared memory. If there is a shared buffer, buffers
1960 // is the value of pointer() for the shared buffer, otherwise buffers points
1961 // immediately after the control block. This address is for the mapping within client
1962 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
1963 void* buffers;
Glenn Kasten363fb752014-01-15 12:27:31 -08001964 if (mSharedBuffer == 0) {
Glenn Kasten138d6f92015-03-20 10:54:51 -07001965 buffers = cblk + 1;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001966 } else {
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001967 // TODO: Using unsecurePointer() has some associated security pitfalls
1968 // (see declaration for details).
1969 // Either document why it is safe in this case or address the
1970 // issue (e.g. by copying).
1971 buffers = mSharedBuffer->unsecurePointer();
Glenn Kasten138d6f92015-03-20 10:54:51 -07001972 if (buffers == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001973 errorMessage = StringPrintf(
1974 "%s(%d): Could not get buffer pointer", __func__, mPortId);
1975 ALOGE("%s", errorMessage.c_str());
1976 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001977 goto exit;
Glenn Kasten138d6f92015-03-20 10:54:51 -07001978 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001979 }
1980
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001981 mAudioTrack->attachAuxEffect(mAuxEffectId, &status);
Glenn Kasten5f631512014-02-24 15:16:07 -08001982
Glenn Kasten093000f2012-05-03 09:35:36 -07001983 // If IAudioTrack is re-created, don't let the requested frameCount
1984 // decrease. This can confuse clients that cache frameCount().
Eric Laurent21da6472017-11-09 16:29:26 -08001985 if (mFrameCount > mReqFrameCount) {
1986 mReqFrameCount = mFrameCount;
Glenn Kasten093000f2012-05-03 09:35:36 -07001987 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001988
Andy Hungd7bd69e2015-07-24 07:52:41 -07001989 // reset server position to 0 as we have new cblk.
1990 mServer = 0;
1991
Glenn Kastene3aa6592012-12-04 12:22:46 -08001992 // update proxy
Glenn Kasten363fb752014-01-15 12:27:31 -08001993 if (mSharedBuffer == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001994 mStaticProxy.clear();
Eric Laurent21da6472017-11-09 16:29:26 -08001995 mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001996 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08001997 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001998 mProxy = mStaticProxy;
1999 }
seunghak.hane6a9d6582014-11-22 15:22:35 +09002000
2001 mProxy->setVolumeLR(gain_minifloat_pack(
2002 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
2003 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
2004
Glenn Kastene3aa6592012-12-04 12:22:46 -08002005 mProxy->setSendLevel(mSendLevel);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002006 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
2007 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
2008 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07002009 mProxy->setSampleRate(effectiveSampleRate);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002010
2011 AudioPlaybackRate playbackRateTemp = mPlaybackRate;
2012 playbackRateTemp.mSpeed = effectiveSpeed;
2013 playbackRateTemp.mPitch = effectivePitch;
2014 mProxy->setPlaybackRate(playbackRateTemp);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002015 mProxy->setMinimum(mNotificationFramesAct);
2016
Kuowei Lid4adbdb2020-08-13 14:44:25 +08002017 if (mDualMonoMode != AUDIO_DUAL_MONO_MODE_OFF) {
2018 setDualMonoMode_l(mDualMonoMode);
2019 }
2020 if (mAudioDescriptionMixLeveldB != -std::numeric_limits<float>::infinity()) {
2021 setAudioDescriptionMixLevel_l(mAudioDescriptionMixLeveldB);
2022 }
2023
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002024 mDeathNotifier = new DeathNotifier(this);
Marco Nelissenf8880202014-11-14 07:58:25 -08002025 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
Glenn Kastene3aa6592012-12-04 12:22:46 -08002026
Andy Hungb68f5eb2019-12-03 16:49:17 -08002027 // This is the first log sent from the AudioTrack client.
2028 // The creation of the audio track by AudioFlinger (in the code above)
2029 // is the first log of the AudioTrack and must be present before
2030 // any AudioTrack client logs will be accepted.
Andy Hungea840382020-05-05 21:50:17 -07002031
Andy Hungb68f5eb2019-12-03 16:49:17 -08002032 mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(mPortId);
2033 mediametrics::LogItem(mMetricsId)
2034 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE)
2035 // the following are immutable
Andy Hunga629bd12020-06-05 16:03:53 -07002036 .set(AMEDIAMETRICS_PROP_FLAGS, toString(mFlags).c_str())
2037 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -08002038 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
Andy Hung3a5c2f32021-02-17 15:06:42 -08002039 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, mLogSessionId)
Andy Hung839a3062021-02-17 11:15:16 -08002040 .set(AMEDIAMETRICS_PROP_PLAYERIID, mPlayerIId)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002041 .set(AMEDIAMETRICS_PROP_TRACKID, mPortId) // dup from key
Andy Hungb68f5eb2019-12-03 16:49:17 -08002042 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2043 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2044 .set(AMEDIAMETRICS_PROP_THREADID, (int32_t)output.outputId)
2045 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2046 .set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)mRoutedDeviceId)
2047 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2048 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
2049 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
2050 // the following are NOT immutable
2051 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)mVolume[AUDIO_INTERLEAVE_LEFT])
2052 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)mVolume[AUDIO_INTERLEAVE_RIGHT])
2053 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Andy Hung73dc2f92021-12-07 21:50:04 -08002054 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)NO_ERROR)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002055 .set(AMEDIAMETRICS_PROP_AUXEFFECTID, (int32_t)mAuxEffectId)
2056 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2057 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2058 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2059 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2060 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveSampleRate)
2061 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2062 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)effectiveSpeed)
2063 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2064 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)effectivePitch)
2065 .record();
2066
2067 // mSendLevel
2068 // mReqFrameCount?
2069 // mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq
2070 // mLatency, mAfLatency, mAfFrameCount, mAfSampleRate
2071
Glenn Kasten38e905b2014-01-13 10:21:48 -08002072 }
2073
Eric Laurentf32d7812017-11-30 14:44:07 -08002074exit:
Andy Hung3acde2c2021-11-11 09:18:08 -08002075 if (status != NO_ERROR) {
2076 if (callbackAdded) {
2077 // note: mOutput is always valid is callbackAdded is true
2078 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
2079 }
2080 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
2081 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
Eric Laurentad2e7b92017-09-14 20:06:42 -07002082 }
Eric Laurentf32d7812017-11-30 14:44:07 -08002083 mStatus = status;
Eric Laurent21da6472017-11-09 16:29:26 -08002084
2085 // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
Glenn Kasten38e905b2014-01-13 10:21:48 -08002086 return status;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002087}
2088
Andy Hung3acde2c2021-11-11 09:18:08 -08002089void AudioTrack::reportError(status_t status, const char *event, const char *message) const
2090{
2091 if (status == NO_ERROR) return;
2092 // We report error on the native side because some callers do not come
2093 // from Java.
Andy Hungde602302021-12-07 21:35:49 -08002094 // Ensure these variables are initialized in set().
2095 mediametrics::LogItem(AMEDIAMETRICS_KEY_AUDIO_TRACK_ERROR)
Andy Hung3acde2c2021-11-11 09:18:08 -08002096 .set(AMEDIAMETRICS_PROP_EVENT, event)
Andy Hung73dc2f92021-12-07 21:50:04 -08002097 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
2098 .set(AMEDIAMETRICS_PROP_STATUSMESSAGE, message)
Andy Hung3acde2c2021-11-11 09:18:08 -08002099 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
2100 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
2101 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2102 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2103 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2104 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2105 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
Andy Hung3acde2c2021-11-11 09:18:08 -08002106 // the following are NOT immutable
Andy Hungde602302021-12-07 21:35:49 -08002107 // frame count is initially the requested frame count, but may be adjusted
2108 // by AudioFlinger after creation.
2109 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
Andy Hung3acde2c2021-11-11 09:18:08 -08002110 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2111 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2112 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2113 .record();
2114}
2115
Glenn Kastenb46f3942015-03-09 12:00:30 -07002116status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002117{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002118 if (audioBuffer == NULL) {
Glenn Kasten551b5352015-03-20 11:30:28 -07002119 if (nonContig != NULL) {
2120 *nonContig = 0;
2121 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002122 return BAD_VALUE;
Eric Laurent9b7d9502011-03-21 11:49:00 -07002123 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002124 if (mTransfer != TRANSFER_OBTAIN) {
2125 audioBuffer->frameCount = 0;
Atneya Nair03079272022-01-18 17:03:14 -05002126 audioBuffer->mSize = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002127 audioBuffer->raw = NULL;
Glenn Kasten551b5352015-03-20 11:30:28 -07002128 if (nonContig != NULL) {
2129 *nonContig = 0;
2130 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002131 return INVALID_OPERATION;
2132 }
Eric Laurent9b7d9502011-03-21 11:49:00 -07002133
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002134 const struct timespec *requested;
Eric Laurentdf576992014-01-27 18:13:39 -08002135 struct timespec timeout;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002136 if (waitCount == -1) {
2137 requested = &ClientProxy::kForever;
2138 } else if (waitCount == 0) {
2139 requested = &ClientProxy::kNonBlocking;
2140 } else if (waitCount > 0) {
Chih-Hung Hsiehbca74292018-08-10 16:06:07 -07002141 time_t ms = WAIT_PERIOD_MS * (time_t) waitCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002142 timeout.tv_sec = ms / 1000;
Andy Hung06a730b2020-04-09 13:28:31 -07002143 timeout.tv_nsec = (ms % 1000) * 1000000;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002144 requested = &timeout;
2145 } else {
Eric Laurent973db022018-11-20 14:54:31 -08002146 ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002147 requested = NULL;
2148 }
Glenn Kastenb46f3942015-03-09 12:00:30 -07002149 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002150}
Eric Laurent1703cdf2011-03-07 14:52:59 -08002151
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002152status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
2153 struct timespec *elapsed, size_t *nonContig)
2154{
2155 // previous and new IAudioTrack sequence numbers are used to detect track re-creation
2156 uint32_t oldSequence = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002157
2158 Proxy::Buffer buffer;
2159 status_t status = NO_ERROR;
2160
2161 static const int32_t kMaxTries = 5;
2162 int32_t tryCounter = kMaxTries;
2163
2164 do {
2165 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
2166 // keep them from going away if another thread re-creates the track during obtainBuffer()
2167 sp<AudioTrackClientProxy> proxy;
2168 sp<IMemory> iMem;
2169
2170 { // start of lock scope
2171 AutoMutex lock(mLock);
2172
Glenn Kasten305996c2020-01-27 08:03:37 -08002173 uint32_t newSequence = mSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002174 // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
2175 if (status == DEAD_OBJECT) {
2176 // re-create track, unless someone else has already done so
2177 if (newSequence == oldSequence) {
2178 status = restoreTrack_l("obtainBuffer");
2179 if (status != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002180 buffer.mFrameCount = 0;
2181 buffer.mRaw = NULL;
2182 buffer.mNonContig = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002183 break;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002184 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002185 }
2186 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002187 oldSequence = newSequence;
2188
Eric Laurent4d231dc2016-03-11 18:38:23 -08002189 if (status == NOT_ENOUGH_DATA) {
2190 restartIfDisabled();
2191 }
2192
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002193 // Keep the extra references
2194 proxy = mProxy;
2195 iMem = mCblkMemory;
2196
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002197 if (mState == STATE_STOPPING) {
2198 status = -EINTR;
2199 buffer.mFrameCount = 0;
2200 buffer.mRaw = NULL;
2201 buffer.mNonContig = 0;
2202 break;
2203 }
2204
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002205 // Non-blocking if track is stopped or paused
2206 if (mState != STATE_ACTIVE) {
2207 requested = &ClientProxy::kNonBlocking;
2208 }
2209
2210 } // end of lock scope
2211
2212 buffer.mFrameCount = audioBuffer->frameCount;
2213 // FIXME starts the requested timeout and elapsed over from scratch
2214 status = proxy->obtainBuffer(&buffer, requested, elapsed);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002215 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002216
2217 audioBuffer->frameCount = buffer.mFrameCount;
Atneya Nair03079272022-01-18 17:03:14 -05002218 audioBuffer->mSize = buffer.mFrameCount * mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002219 audioBuffer->raw = buffer.mRaw;
Glenn Kasten305996c2020-01-27 08:03:37 -08002220 audioBuffer->sequence = oldSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002221 if (nonContig != NULL) {
2222 *nonContig = buffer.mNonContig;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002223 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002224 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002225}
2226
Glenn Kasten54a8a452015-03-09 12:03:00 -07002227void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002228{
Glenn Kasten3f02be22015-03-09 11:59:04 -07002229 // FIXME add error checking on mode, by adding an internal version
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002230 if (mTransfer == TRANSFER_SHARED) {
2231 return;
2232 }
2233
Atneya Nair03079272022-01-18 17:03:14 -05002234 size_t stepCount = audioBuffer->mSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002235 if (stepCount == 0) {
2236 return;
2237 }
2238
2239 Proxy::Buffer buffer;
2240 buffer.mFrameCount = stepCount;
2241 buffer.mRaw = audioBuffer->raw;
Glenn Kastene3aa6592012-12-04 12:22:46 -08002242
Eric Laurent1703cdf2011-03-07 14:52:59 -08002243 AutoMutex lock(mLock);
Glenn Kasten305996c2020-01-27 08:03:37 -08002244 if (audioBuffer->sequence != mSequence) {
2245 // This Buffer came from a different IAudioTrack instance, so ignore the releaseBuffer
2246 ALOGD("%s is no-op due to IAudioTrack sequence mismatch %u != %u",
2247 __func__, audioBuffer->sequence, mSequence);
2248 return;
2249 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002250 mReleased += stepCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002251 mInUnderrun = false;
2252 mProxy->releaseBuffer(&buffer);
2253
2254 // restart track if it was disabled by audioflinger due to previous underrun
Eric Laurent4d231dc2016-03-11 18:38:23 -08002255 restartIfDisabled();
2256}
2257
2258void AudioTrack::restartIfDisabled()
2259{
2260 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
2261 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002262 ALOGW("%s(%d): releaseBuffer() track %p disabled due to previous underrun, restarting",
Eric Laurent973db022018-11-20 14:54:31 -08002263 __func__, mPortId, this);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002264 // FIXME ignoring status
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002265 status_t status;
2266 mAudioTrack->start(&status);
Eric Laurentdf839842012-05-31 14:27:14 -07002267 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002268}
2269
2270// -------------------------------------------------------------------------
2271
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002272ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002273{
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002274 if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07002275 return INVALID_OPERATION;
2276 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002277
Eric Laurentab5cdba2014-06-09 17:22:27 -07002278 if (isDirect()) {
2279 AutoMutex lock(mLock);
2280 int32_t flags = android_atomic_and(
2281 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
2282 &mCblk->mFlags);
2283 if (flags & CBLK_INVALID) {
2284 return DEAD_OBJECT;
2285 }
2286 }
2287
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002288 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
Jiabin Huang447cea72020-07-28 22:35:18 +00002289 // Validation: user is most-likely passing an error code, and it would
Glenn Kasten99e53b82012-01-19 08:59:58 -08002290 // make the return value ambiguous (actualSize vs error).
Andy Hungfb8ede22018-09-12 19:03:24 -07002291 ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
Eric Laurent973db022018-11-20 14:54:31 -08002292 __func__, mPortId, buffer, userSize, userSize);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002293 return BAD_VALUE;
2294 }
2295
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002296 size_t written = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002297 Buffer audioBuffer;
2298
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002299 while (userSize >= mFrameSize) {
2300 audioBuffer.frameCount = userSize / mFrameSize;
Eric Laurentc2f1f072009-07-17 12:17:14 -07002301
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002302 status_t err = obtainBuffer(&audioBuffer,
2303 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002304 if (err < 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002305 if (written > 0) {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002306 break;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002307 }
Glenn Kasten0a2f1512016-07-22 08:06:37 -07002308 if (err == TIMED_OUT || err == -EINTR) {
2309 err = WOULD_BLOCK;
2310 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002311 return ssize_t(err);
2312 }
2313
Atneya Nair03079272022-01-18 17:03:14 -05002314 size_t toWrite = audioBuffer.size();
2315 memcpy(audioBuffer.raw, buffer, toWrite);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002316 buffer = ((const char *) buffer) + toWrite;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002317 userSize -= toWrite;
2318 written += toWrite;
2319
2320 releaseBuffer(&audioBuffer);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002321 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002322
Andy Hungea2b9c02016-02-12 17:06:53 -08002323 if (written > 0) {
2324 mFramesWritten += written / mFrameSize;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002325
2326 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2327 const sp<AudioTrackThread> t = mAudioTrackThread;
2328 if (t != 0) {
2329 // causes wake up of the playback thread, that will callback the client for
2330 // more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
2331 t->wake();
2332 }
2333 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002334 }
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002335
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002336 return written;
2337}
2338
2339// -------------------------------------------------------------------------
2340
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08002341nsecs_t AudioTrack::processAudioBuffer()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002342{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002343 // Currently the AudioTrack thread is not created if there are no callbacks.
2344 // Would it ever make sense to run the thread, even without callbacks?
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002345 // If so, then replace this by checks at each use for mCallback != NULL.
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002346 LOG_ALWAYS_FATAL_IF(mCblk == NULL);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002347 mLock.lock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002348 sp<IAudioTrackCallback> callback = mCallback.promote();
2349 if (!callback) {
2350 mCallback = nullptr;
Atneya Naire260f5a2022-05-03 17:02:20 -04002351 mLock.unlock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002352 return NS_NEVER;
2353 }
Glenn Kastena07f17c2013-04-23 12:39:37 -07002354 if (mAwaitBoost) {
2355 mAwaitBoost = false;
2356 mLock.unlock();
2357 static const int32_t kMaxTries = 5;
2358 int32_t tryCounter = kMaxTries;
2359 uint32_t pollUs = 10000;
2360 do {
Glenn Kasten8255ba72016-08-23 13:54:23 -07002361 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002362 if (policy == SCHED_FIFO || policy == SCHED_RR) {
2363 break;
2364 }
2365 usleep(pollUs);
2366 pollUs <<= 1;
2367 } while (tryCounter-- > 0);
2368 if (tryCounter < 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002369 ALOGE("%s(%d): did not receive expected priority boost on time",
Eric Laurent973db022018-11-20 14:54:31 -08002370 __func__, mPortId);
Glenn Kastena07f17c2013-04-23 12:39:37 -07002371 }
Glenn Kastenb0dfd462013-07-10 16:52:47 -07002372 // Run again immediately
2373 return 0;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002374 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002375
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002376 // Can only reference mCblk while locked
2377 int32_t flags = android_atomic_and(
Glenn Kasten96f60d82013-07-12 10:21:18 -07002378 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);
Glenn Kastena47f3162012-11-07 10:13:08 -08002379
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002380 // Check for track invalidation
2381 if (flags & CBLK_INVALID) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002382 // for offloaded tracks restoreTrack_l() will just update the sequence and clear
2383 // AudioSystem cache. We should not exit here but after calling the callback so
2384 // that the upper layers can recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002385 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
Lajos Molnarf1063e22015-04-17 15:19:42 -07002386 status_t status __unused = restoreTrack_l("processAudioBuffer");
2387 // FIXME unused status
Andy Hung53c3b5f2014-12-15 16:42:05 -08002388 // after restoration, continue below to make sure that the loop and buffer events
2389 // are notified because they have been cleared from mCblk->mFlags above.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002390 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002391 }
2392
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002393 bool waitStreamEnd = mState == STATE_STOPPING;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002394 bool active = mState == STATE_ACTIVE;
2395
2396 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
2397 bool newUnderrun = false;
2398 if (flags & CBLK_UNDERRUN) {
2399#if 0
2400 // Currently in shared buffer mode, when the server reaches the end of buffer,
2401 // the track stays active in continuous underrun state. It's up to the application
2402 // to pause or stop the track, or set the position to a new offset within buffer.
2403 // This was some experimental code to auto-pause on underrun. Keeping it here
2404 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
2405 if (mTransfer == TRANSFER_SHARED) {
2406 mState = STATE_PAUSED;
2407 active = false;
2408 }
2409#endif
2410 if (!mInUnderrun) {
2411 mInUnderrun = true;
2412 newUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002413 }
2414 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002415
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002416 // Get current position of server
Andy Hung90e8a972015-11-09 16:42:40 -08002417 Modulo<uint32_t> position(updateAndGetPosition_l());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002418
2419 // Manage marker callback
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002420 bool markerReached = false;
Andy Hung90e8a972015-11-09 16:42:40 -08002421 Modulo<uint32_t> markerPosition(mMarkerPosition);
2422 // uses 32 bit wraparound for comparison with position.
2423 if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002424 mMarkerReached = markerReached = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002425 }
2426
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002427 // Determine number of new position callback(s) that will be needed, while locked
2428 size_t newPosCount = 0;
Andy Hung90e8a972015-11-09 16:42:40 -08002429 Modulo<uint32_t> newPosition(mNewPosition);
2430 uint32_t updatePeriod = mUpdatePeriod;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002431 // FIXME fails for wraparound, need 64 bits
2432 if (updatePeriod > 0 && position >= newPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002433 newPosCount = ((position - newPosition).value() / updatePeriod) + 1;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002434 mNewPosition += updatePeriod * newPosCount;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002435 }
2436
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002437 // Cache other fields that will be needed soon
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002438 uint32_t sampleRate = mSampleRate;
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002439 float speed = mPlaybackRate.mSpeed;
Andy Hunga7f03352015-05-31 21:54:49 -07002440 const uint32_t notificationFrames = mNotificationFramesAct;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002441 if (mRefreshRemaining) {
2442 mRefreshRemaining = false;
2443 mRemainingFrames = notificationFrames;
2444 mRetryOnPartialBuffer = false;
2445 }
2446 size_t misalignment = mProxy->getMisalignment();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002447 uint32_t sequence = mSequence;
Glenn Kasten96f04882013-09-20 09:28:56 -07002448 sp<AudioTrackClientProxy> proxy = mProxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002449
Andy Hung53c3b5f2014-12-15 16:42:05 -08002450 // Determine the number of new loop callback(s) that will be needed, while locked.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002451 uint32_t loopCountNotifications = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -08002452 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
2453
2454 if (mLoopCount > 0) {
2455 int loopCount;
2456 size_t bufferPosition;
2457 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
2458 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
2459 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
2460 mLoopCountNotified = loopCount; // discard any excess notifications
2461 } else if (mLoopCount < 0) {
2462 // FIXME: We're not accurate with notification count and position with infinite looping
2463 // since loopCount from server side will always return -1 (we could decrement it).
2464 size_t bufferPosition = mStaticProxy->getBufferPosition();
2465 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
2466 loopPeriod = mLoopEnd - bufferPosition;
2467 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
2468 size_t bufferPosition = mStaticProxy->getBufferPosition();
2469 loopPeriod = mFrameCount - bufferPosition;
2470 }
2471
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002472 // These fields don't need to be cached, because they are assigned only by set():
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002473 // mTransfer, mCallback, mUserData, mFormat, mFrameSize, mFlags
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002474 // mFlags is also assigned by createTrack_l(), but not the bit we care about.
2475
2476 mLock.unlock();
2477
Andy Hunga7f03352015-05-31 21:54:49 -07002478 // get anchor time to account for callbacks.
2479 const nsecs_t timeBeforeCallbacks = systemTime();
2480
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002481 if (waitStreamEnd) {
Andy Hunga7f03352015-05-31 21:54:49 -07002482 // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread
2483 // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
2484 // (and make sure we don't callback for more data while we're stopping).
2485 // This helps with position, marker notifications, and track invalidation.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002486 struct timespec timeout;
2487 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
2488 timeout.tv_nsec = 0;
2489
Andy Hungeb0732d2023-03-29 20:31:47 -07002490 // Use timestamp progress to safeguard we don't falsely time out.
2491 AudioTimestamp timestamp{};
2492 const bool isTimestampValid = getTimestamp(timestamp) == OK;
2493 const auto frameCount = isTimestampValid ? timestamp.mPosition : 0;
2494
Glenn Kasten96f04882013-09-20 09:28:56 -07002495 status_t status = proxy->waitStreamEndDone(&timeout);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002496 switch (status) {
Andy Hungeb0732d2023-03-29 20:31:47 -07002497 case TIMED_OUT:
2498 if (isTimestampValid
2499 && getTimestamp(timestamp) == OK && frameCount != timestamp.mPosition) {
2500 ALOGD("%s: waitStreamEndDone retrying", __func__);
2501 break; // we retry again (and recheck possible state change).
2502 }
2503 [[fallthrough]];
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002504 case NO_ERROR:
2505 case DEAD_OBJECT:
Andy Hung39609a02015-09-03 16:38:38 -07002506 if (status != DEAD_OBJECT) {
2507 // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop();
2508 // instead, the application should handle the EVENT_NEW_IAUDIOTRACK.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002509 callback->onStreamEnd();
Andy Hung39609a02015-09-03 16:38:38 -07002510 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002511 {
2512 AutoMutex lock(mLock);
2513 // The previously assigned value of waitStreamEnd is no longer valid,
2514 // since the mutex has been unlocked and either the callback handler
2515 // or another thread could have re-started the AudioTrack during that time.
2516 waitStreamEnd = mState == STATE_STOPPING;
2517 if (waitStreamEnd) {
2518 mState = STATE_STOPPED;
Andy Hungc2813e52014-10-16 17:54:34 -07002519 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002520 }
2521 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002522 if (waitStreamEnd && status != DEAD_OBJECT) {
Andy Hungeb0732d2023-03-29 20:31:47 -07002523 ALOGV("%s: waitStreamEndDone complete", __func__);
Glenn Kasten96f04882013-09-20 09:28:56 -07002524 return NS_INACTIVE;
2525 }
2526 break;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002527 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002528 return 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002529 }
2530
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002531 // perform callbacks while unlocked
2532 if (newUnderrun) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002533 callback->onUnderrun();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002534 }
Andy Hung53c3b5f2014-12-15 16:42:05 -08002535 while (loopCountNotifications > 0) {
Andy Hung53c3b5f2014-12-15 16:42:05 -08002536 --loopCountNotifications;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002537 callback->onLoopEnd(mLoopCount > 0 ? loopCountNotifications + mLoopCountNotified : -1);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002538 }
2539 if (flags & CBLK_BUFFER_END) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002540 callback->onBufferEnd();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002541 }
2542 if (markerReached) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002543 callback->onMarker(markerPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002544 }
2545 while (newPosCount > 0) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002546 callback->onNewPos(newPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002547 newPosition += updatePeriod;
2548 newPosCount--;
2549 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002550
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002551 if (mObservedSequence != sequence) {
2552 mObservedSequence = sequence;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002553 callback->onNewIAudioTrack();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002554 // for offloaded tracks, just wait for the upper layers to recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002555 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002556 return NS_INACTIVE;
2557 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002558 }
2559
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002560 // if inactive, then don't run me again until re-started
2561 if (!active) {
2562 return NS_INACTIVE;
Eric Laurent2267ba12011-09-07 11:13:23 -07002563 }
2564
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002565 // Compute the estimated time until the next timed event (position, markers, loops)
2566 // FIXME only for non-compressed audio
2567 uint32_t minFrames = ~0;
2568 if (!markerReached && position < markerPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002569 minFrames = (markerPosition - position).value();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002570 }
2571 if (loopPeriod > 0 && loopPeriod < minFrames) {
Andy Hung2d85f092015-01-07 12:45:13 -08002572 // loopPeriod is already adjusted for actual position.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002573 minFrames = loopPeriod;
2574 }
Andy Hung2d85f092015-01-07 12:45:13 -08002575 if (updatePeriod > 0) {
Andy Hung90e8a972015-11-09 16:42:40 -08002576 minFrames = min(minFrames, (newPosition - position).value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002577 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002578
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002579 // If > 0, poll periodically to recover from a stuck server. A good value is 2.
2580 static const uint32_t kPoll = 0;
2581 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
2582 minFrames = kPoll * notificationFrames;
2583 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002584
Andy Hunga7f03352015-05-31 21:54:49 -07002585 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
2586 static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL;
2587 const nsecs_t timeAfterCallbacks = systemTime();
2588
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002589 // Convert frame units to time units
2590 nsecs_t ns = NS_WHENEVER;
2591 if (minFrames != (uint32_t) ~0) {
jiabinc7bb8322017-09-06 18:20:11 -07002592 // AudioFlinger consumption of client data may be irregular when coming out of device
2593 // standby since the kernel buffers require filling. This is throttled to no more than 2x
2594 // the expected rate in the MixerThread. Hence, we reduce the estimated time to wait by one
2595 // half (but no more than half a second) to improve callback accuracy during these temporary
2596 // data surges.
2597 const nsecs_t estimatedNs = framesToNanoseconds(minFrames, sampleRate, speed);
2598 constexpr nsecs_t maxThrottleCompensationNs = 500000000LL;
2599 ns = estimatedNs - min(estimatedNs / 2, maxThrottleCompensationNs) + kWaitPeriodNs;
Andy Hunga7f03352015-05-31 21:54:49 -07002600 ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time
2601 // TODO: Should we warn if the callback time is too long?
2602 if (ns < 0) ns = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002603 }
2604
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002605 // If not supplying data by EVENT_MORE_DATA or EVENT_CAN_WRITE_MORE_DATA, then we're done
2606 if (mTransfer != TRANSFER_CALLBACK && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002607 return ns;
2608 }
2609
Andy Hunga7f03352015-05-31 21:54:49 -07002610 // EVENT_MORE_DATA callback handling.
2611 // Timing for linear pcm audio data formats can be derived directly from the
2612 // buffer fill level.
2613 // Timing for compressed data is not directly available from the buffer fill level,
2614 // rather indirectly from waiting for blocking mode callbacks or waiting for obtain()
2615 // to return a certain fill level.
2616
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002617 struct timespec timeout;
2618 const struct timespec *requested = &ClientProxy::kForever;
2619 if (ns != NS_WHENEVER) {
2620 timeout.tv_sec = ns / 1000000000LL;
2621 timeout.tv_nsec = ns % 1000000000LL;
Andy Hungfb8ede22018-09-12 19:03:24 -07002622 ALOGV("%s(%d): timeout %ld.%03d",
Eric Laurent973db022018-11-20 14:54:31 -08002623 __func__, mPortId, timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002624 requested = &timeout;
2625 }
2626
Andy Hungea2b9c02016-02-12 17:06:53 -08002627 size_t writtenFrames = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002628 while (mRemainingFrames > 0) {
2629
2630 Buffer audioBuffer;
2631 audioBuffer.frameCount = mRemainingFrames;
2632 size_t nonContig;
2633 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
2634 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
Andy Hungfb8ede22018-09-12 19:03:24 -07002635 "%s(%d): obtainBuffer() err=%d frameCount=%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002636 __func__, mPortId, err, audioBuffer.frameCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002637 requested = &ClientProxy::kNonBlocking;
2638 size_t avail = audioBuffer.frameCount + nonContig;
Andy Hungfb8ede22018-09-12 19:03:24 -07002639 ALOGV("%s(%d): obtainBuffer(%u) returned %zu = %zu + %zu err %d",
Eric Laurent973db022018-11-20 14:54:31 -08002640 __func__, mPortId, mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002641 if (err != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002642 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
2643 (isOffloaded() && (err == DEAD_OBJECT))) {
Glenn Kasten606fbc12015-10-22 15:28:15 -07002644 // FIXME bug 25195759
2645 return 1000000;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002646 }
Andy Hungfb8ede22018-09-12 19:03:24 -07002647 ALOGE("%s(%d): Error %d obtaining an audio buffer, giving up.",
Eric Laurent973db022018-11-20 14:54:31 -08002648 __func__, mPortId, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002649 return NS_NEVER;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002650 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002651
Phil Burkfdb3c072016-02-09 10:47:02 -08002652 if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002653 mRetryOnPartialBuffer = false;
2654 if (avail < mRemainingFrames) {
Andy Hunga7f03352015-05-31 21:54:49 -07002655 if (ns > 0) { // account for obtain time
2656 const nsecs_t timeNow = systemTime();
2657 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2658 }
Andy Hungeb6e6502019-04-29 13:47:40 -07002659
2660 // delayNs is first computed by the additional frames required in the buffer.
2661 nsecs_t delayNs = framesToNanoseconds(
2662 mRemainingFrames - avail, sampleRate, speed);
2663
2664 // afNs is the AudioFlinger mixer period in ns.
2665 const nsecs_t afNs = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2666
2667 // If the AudioTrack is double buffered based on the AudioFlinger mixer period,
2668 // we may have a race if we wait based on the number of frames desired.
2669 // This is a possible issue with resampling and AAudio.
2670 //
2671 // The granularity of audioflinger processing is one mixer period; if
2672 // our wait time is less than one mixer period, wait at most half the period.
2673 if (delayNs < afNs) {
2674 delayNs = std::min(delayNs, afNs / 2);
2675 }
2676
2677 // adjust our ns wait by delayNs.
2678 if (ns < 0 /* NS_WHENEVER */ || delayNs < ns) {
2679 ns = delayNs;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002680 }
2681 return ns;
2682 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07002683 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002684
Atneya Nair03079272022-01-18 17:03:14 -05002685 size_t reqSize = audioBuffer.size();
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002686 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2687 // when notifying client it can write more data, pass the total size that can be
2688 // written in the next write() call, since it's not passed through the callback
Atneya Nair03079272022-01-18 17:03:14 -05002689 audioBuffer.mSize += nonContig;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002690 }
Atneya Nairc2dd1272021-10-26 19:39:51 -04002691 const size_t writtenSize = (mTransfer == TRANSFER_CALLBACK)
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002692 ? callback->onMoreData(audioBuffer)
2693 : callback->onCanWriteMoreData(audioBuffer);
Jiabin Huang447cea72020-07-28 22:35:18 +00002694 // Validate on returned size
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002695 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002696 ALOGE("%s(%d): EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
Eric Laurent973db022018-11-20 14:54:31 -08002697 __func__, mPortId, reqSize, ssize_t(writtenSize));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002698 return NS_NEVER;
2699 }
2700
2701 if (writtenSize == 0) {
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002702 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2703 // The callback EVENT_CAN_WRITE_MORE_DATA was processed in the JNI of
2704 // android.media.AudioTrack. The JNI is not using the callback to provide data,
2705 // it only signals to the Java client that it can provide more data, which
2706 // this track is read to accept now.
2707 // The playback thread will be awaken at the next ::write()
2708 return NS_WHENEVER;
2709 }
The Android Open Source Project8555d082009-03-05 14:34:35 -08002710 // The callback is done filling buffers
2711 // Keep this thread going to handle timed events and
2712 // still try to get more data in intervals of WAIT_PERIOD_MS
2713 // but don't just loop and block the CPU, so wait
Andy Hunga7f03352015-05-31 21:54:49 -07002714
2715 // mCbf(EVENT_MORE_DATA, ...) might either
2716 // (1) Block until it can fill the buffer, returning 0 size on EOS.
2717 // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS.
2718 // (3) Return 0 size when no data is available, does not wait for more data.
2719 //
2720 // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer.
2721 // We try to compute the wait time to avoid a tight sleep-wait cycle,
2722 // especially for case (3).
2723 //
2724 // The decision to support (1) and (2) affect the sizing of mRemainingFrames
2725 // and this loop; whereas for case (3) we could simply check once with the full
2726 // buffer size and skip the loop entirely.
2727
2728 nsecs_t myns;
Phil Burkfdb3c072016-02-09 10:47:02 -08002729 if (audio_has_proportional_frames(mFormat)) {
Andy Hunga7f03352015-05-31 21:54:49 -07002730 // time to wait based on buffer occupancy
2731 const nsecs_t datans = mRemainingFrames <= avail ? 0 :
2732 framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
2733 // audio flinger thread buffer size (TODO: adjust for fast tracks)
Glenn Kastenea38ee72016-04-18 11:08:01 -07002734 // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks.
Andy Hunga7f03352015-05-31 21:54:49 -07002735 const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2736 // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0.
2737 myns = datans + (afns / 2);
2738 } else {
2739 // FIXME: This could ping quite a bit if the buffer isn't full.
2740 // Note that when mState is stopping we waitStreamEnd, so it never gets here.
2741 myns = kWaitPeriodNs;
2742 }
2743 if (ns > 0) { // account for obtain and callback time
2744 const nsecs_t timeNow = systemTime();
2745 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2746 }
2747 if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
2748 ns = myns;
2749 }
2750 return ns;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002751 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002752
Atneya Nairc2dd1272021-10-26 19:39:51 -04002753 // releaseBuffer reads from audioBuffer.size
Atneya Nair03079272022-01-18 17:03:14 -05002754 audioBuffer.mSize = writtenSize;
Atneya Nairc2dd1272021-10-26 19:39:51 -04002755
Glenn Kasten138d6f92015-03-20 10:54:51 -07002756 size_t releasedFrames = writtenSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002757 audioBuffer.frameCount = releasedFrames;
2758 mRemainingFrames -= releasedFrames;
2759 if (misalignment >= releasedFrames) {
2760 misalignment -= releasedFrames;
2761 } else {
2762 misalignment = 0;
2763 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002764
2765 releaseBuffer(&audioBuffer);
Andy Hungea2b9c02016-02-12 17:06:53 -08002766 writtenFrames += releasedFrames;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002767
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002768 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
2769 // if callback doesn't like to accept the full chunk
2770 if (writtenSize < reqSize) {
2771 continue;
2772 }
2773
2774 // There could be enough non-contiguous frames available to satisfy the remaining request
2775 if (mRemainingFrames <= nonContig) {
2776 continue;
2777 }
2778
2779#if 0
2780 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
2781 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
2782 // that total to a sum == notificationFrames.
2783 if (0 < misalignment && misalignment <= mRemainingFrames) {
2784 mRemainingFrames = misalignment;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002785 return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002786 }
2787#endif
2788
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002789 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002790 if (writtenFrames > 0) {
2791 AutoMutex lock(mLock);
2792 mFramesWritten += writtenFrames;
2793 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002794 mRemainingFrames = notificationFrames;
2795 mRetryOnPartialBuffer = true;
2796
2797 // A lot has transpired since ns was calculated, so run again immediately and re-calculate
2798 return 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002799}
2800
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002801status_t AudioTrack::restoreTrack_l(const char *from)
Eric Laurent1703cdf2011-03-07 14:52:59 -08002802{
Andy Hungb68f5eb2019-12-03 16:49:17 -08002803 status_t result = NO_ERROR; // logged: make sure to set this before returning.
2804 const int64_t beginNs = systemTime();
Andy Hunga6b27032020-04-27 10:34:24 -07002805 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -08002806 mediametrics::LogItem(mMetricsId)
2807 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_RESTORE)
Andy Hungea840382020-05-05 21:50:17 -07002808 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08002809 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
2810 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
2811 .set(AMEDIAMETRICS_PROP_WHERE, from)
2812 .record(); });
2813
Andy Hungfb8ede22018-09-12 19:03:24 -07002814 ALOGW("%s(%d): dead IAudioTrack, %s, creating a new one from %s()",
Eric Laurent973db022018-11-20 14:54:31 -08002815 __func__, mPortId, isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002816 ++mSequence;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002817
Glenn Kastena47f3162012-11-07 10:13:08 -08002818 // refresh the audio configuration cache in this process to make sure we get new
Glenn Kastend2d089f2014-11-05 11:48:12 -08002819 // output parameters and new IAudioFlinger in createTrack_l()
Glenn Kastena47f3162012-11-07 10:13:08 -08002820 AudioSystem::clearAudioConfigCache();
Eric Laurent9f6530f2011-08-30 10:18:54 -07002821
Ronghua Wufaeb0f22015-05-21 12:20:21 -07002822 if (isOffloadedOrDirect_l() || mDoNotReconnect) {
Andy Hung1f1db832015-06-08 13:26:10 -07002823 // FIXME re-creation of offloaded and direct tracks is not yet implemented;
2824 // reconsider enabling for linear PCM encodings when position can be preserved.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002825 result = DEAD_OBJECT;
2826 return result;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002827 }
2828
Phil Burk2812d9e2016-01-04 10:34:30 -08002829 // Save so we can return count since creation.
2830 mUnderrunCountOffset = getUnderrunCount_l();
2831
Glenn Kasten200092b2014-08-15 15:13:30 -07002832 // save the old static buffer position
Andy Hungf20a4e92016-08-15 19:10:34 -07002833 uint32_t staticPosition = 0;
Andy Hung4ede21d2014-12-12 15:37:34 -08002834 size_t bufferPosition = 0;
2835 int loopCount = 0;
2836 if (mStaticProxy != 0) {
2837 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
Andy Hungf20a4e92016-08-15 19:10:34 -07002838 staticPosition = mStaticProxy->getPosition().unsignedValue();
Andy Hung4ede21d2014-12-12 15:37:34 -08002839 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002840
Andy Hung3c7f47a2021-03-16 17:30:09 -07002841 // save the old startThreshold and framecount
2842 const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
2843 const uint32_t originalFrameCount = mProxy->frameCount();
2844
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002845 // See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
2846 // causes a lot of churn on the service side, and it can reject starting
2847 // playback of a previously created track. May also apply to other cases.
2848 const int INITIAL_RETRIES = 3;
2849 int retries = INITIAL_RETRIES;
2850retry:
2851 if (retries < INITIAL_RETRIES) {
2852 // See the comment for clearAudioConfigCache at the start of the function.
2853 AudioSystem::clearAudioConfigCache();
2854 }
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08002855 mFlags = mOrigFlags;
2856
Glenn Kasten200092b2014-08-15 15:13:30 -07002857 // If a new IAudioTrack is successfully created, createTrack_l() will modify the
Glenn Kastena47f3162012-11-07 10:13:08 -08002858 // following member variables: mAudioTrack, mCblkMemory and mCblk.
Glenn Kasten200092b2014-08-15 15:13:30 -07002859 // It will also delete the strong references on previous IAudioTrack and IMemory.
2860 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002861 result = createTrack_l();
Eric Laurentcc21e4f2013-10-16 15:12:32 -07002862
Eric Laurent6ec546d2018-10-10 16:52:14 -07002863 if (result == NO_ERROR) {
Andy Hungd7bd69e2015-07-24 07:52:41 -07002864 // take the frames that will be lost by track recreation into account in saved position
2865 // For streaming tracks, this is the amount we obtained from the user/client
2866 // (not the number actually consumed at the server - those are already lost).
2867 if (mStaticProxy == 0) {
2868 mPosition = mReleased;
2869 }
Andy Hung4ede21d2014-12-12 15:37:34 -08002870 // Continue playback from last known position and restore loop.
2871 if (mStaticProxy != 0) {
2872 if (loopCount != 0) {
2873 mStaticProxy->setBufferPositionAndLoop(bufferPosition,
2874 mLoopStart, mLoopEnd, loopCount);
2875 } else {
2876 mStaticProxy->setBufferPosition(bufferPosition);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002877 if (bufferPosition == mFrameCount) {
Eric Laurent973db022018-11-20 14:54:31 -08002878 ALOGD("%s(%d): restoring track at end of static buffer", __func__, mPortId);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002879 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002880 }
2881 }
Andy Hung4ef88d72017-02-21 19:47:53 -08002882 // restore volume handler
Andy Hung39399b62017-04-21 15:07:45 -07002883 mVolumeHandler->forall([this](const VolumeShaper &shaper) -> VolumeShaper::Status {
2884 sp<VolumeShaper::Operation> operationToEnd =
2885 new VolumeShaper::Operation(shaper.mOperation);
Andy Hung4ef88d72017-02-21 19:47:53 -08002886 // TODO: Ideally we would restore to the exact xOffset position
2887 // as returned by getVolumeShaperState(), but we don't have that
2888 // information when restoring at the client unless we periodically poll
2889 // the server or create shared memory state.
2890 //
Andy Hung39399b62017-04-21 15:07:45 -07002891 // For now, we simply advance to the end of the VolumeShaper effect
2892 // if it has been started.
2893 if (shaper.isStarted()) {
Andy Hungf3702642017-05-05 17:33:32 -07002894 operationToEnd->setNormalizedTime(1.f);
Andy Hung39399b62017-04-21 15:07:45 -07002895 }
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002896 media::VolumeShaperConfiguration config;
2897 shaper.mConfiguration->writeToParcelable(&config);
2898 media::VolumeShaperOperation operation;
2899 operationToEnd->writeToParcelable(&operation);
2900 status_t status;
2901 mAudioTrack->applyVolumeShaper(config, operation, &status);
2902 return status;
Andy Hung4ef88d72017-02-21 19:47:53 -08002903 });
2904
Andy Hung3c7f47a2021-03-16 17:30:09 -07002905 // restore the original start threshold if different than frameCount.
2906 if (originalStartThresholdInFrames != originalFrameCount) {
2907 // Note: mProxy->setStartThresholdInFrames() call is in the Proxy
2908 // and does not trigger a restart.
2909 // (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
2910 // Any start would be triggered on the mState == ACTIVE check below.
2911 const uint32_t currentThreshold =
2912 mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
2913 ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
2914 "%s(%d) startThresholdInFrames changing from %u to %u",
2915 __func__, mPortId, originalStartThresholdInFrames, currentThreshold);
2916 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002917 if (mState == STATE_ACTIVE) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002918 mAudioTrack->start(&result);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002919 }
Andy Hungf20a4e92016-08-15 19:10:34 -07002920 // server resets to zero so we offset
2921 mFramesWrittenServerOffset =
2922 mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten;
2923 mFramesWrittenAtRestore = mFramesWrittenServerOffset;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002924 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002925 if (result != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08002926 ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002927 if (--retries > 0) {
Eric Laurent6ec546d2018-10-10 16:52:14 -07002928 // leave time for an eventual race condition to clear before retrying
2929 usleep(500000);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002930 goto retry;
2931 }
Eric Laurent6ec546d2018-10-10 16:52:14 -07002932 // if no retries left, set invalid bit to force restoring at next occasion
2933 // and avoid inconsistent active state on client and server sides
2934 if (mCblk != nullptr) {
2935 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
2936 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002937 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002938 return result;
2939}
2940
Andy Hung90e8a972015-11-09 16:42:40 -08002941Modulo<uint32_t> AudioTrack::updateAndGetPosition_l()
Glenn Kasten200092b2014-08-15 15:13:30 -07002942{
2943 // This is the sole place to read server consumed frames
Andy Hung90e8a972015-11-09 16:42:40 -08002944 Modulo<uint32_t> newServer(mProxy->getPosition());
2945 const int32_t delta = (newServer - mServer).signedValue();
Glenn Kasten200092b2014-08-15 15:13:30 -07002946 // TODO There is controversy about whether there can be "negative jitter" in server position.
2947 // This should be investigated further, and if possible, it should be addressed.
2948 // A more definite failure mode is infrequent polling by client.
2949 // One could call (void)getPosition_l() in releaseBuffer(),
2950 // so mReleased and mPosition are always lock-step as best possible.
2951 // That should ensure delta never goes negative for infrequent polling
2952 // unless the server has more than 2^31 frames in its buffer,
2953 // in which case the use of uint32_t for these counters has bigger issues.
Andy Hung90e8a972015-11-09 16:42:40 -08002954 ALOGE_IF(delta < 0,
Andy Hungfb8ede22018-09-12 19:03:24 -07002955 "%s(%d): detected illegal retrograde motion by the server: mServer advanced by %d",
Eric Laurent973db022018-11-20 14:54:31 -08002956 __func__, mPortId, delta);
Chad Brubaker039c27a2015-09-23 15:17:29 -07002957 mServer = newServer;
Andy Hung90e8a972015-11-09 16:42:40 -08002958 if (delta > 0) { // avoid retrograde
2959 mPosition += delta;
2960 }
2961 return mPosition;
Glenn Kasten200092b2014-08-15 15:13:30 -07002962}
2963
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002964bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed)
Andy Hung8edb8dc2015-03-26 19:13:55 -07002965{
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002966 updateLatency_l();
Andy Hung8edb8dc2015-03-26 19:13:55 -07002967 // applicable for mixing tracks only (not offloaded or direct)
2968 if (mStaticProxy != 0) {
2969 return true; // static tracks do not have issues with buffer sizing.
2970 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07002971 const size_t minFrameCount =
Eric Laurent21da6472017-11-09 16:29:26 -08002972 AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
2973 sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002974 const bool allowed = mFrameCount >= minFrameCount;
2975 ALOGD_IF(!allowed,
Andy Hungfb8ede22018-09-12 19:03:24 -07002976 "%s(%d): denied "
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002977 "mAfLatency:%u mAfFrameCount:%zu mAfSampleRate:%u sampleRate:%u speed:%f "
2978 "mFrameCount:%zu < minFrameCount:%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002979 __func__, mPortId,
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002980 mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed,
Andy Hung8edb8dc2015-03-26 19:13:55 -07002981 mFrameCount, minFrameCount);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002982 return allowed;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002983}
2984
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00002985status_t AudioTrack::setParameters(const String8& keyValuePairs)
2986{
2987 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002988 status_t status;
2989 mAudioTrack->setParameters(keyValuePairs.c_str(), &status);
2990 return status;
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00002991}
2992
Dean Wheatleya70eef72018-01-04 14:23:50 +11002993status_t AudioTrack::selectPresentation(int presentationId, int programId)
2994{
2995 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08002996 AudioParameter param = AudioParameter();
2997 param.addInt(String8(AudioParameter::keyPresentationId), presentationId);
2998 param.addInt(String8(AudioParameter::keyProgramId), programId);
2999 ALOGV("%s(%d): PresentationId/ProgramId[%s]",
3000 __func__, mPortId, param.toString().string());
3001
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003002 status_t status;
3003 mAudioTrack->setParameters(param.toString().c_str(), &status);
3004 return status;
Dean Wheatleya70eef72018-01-04 14:23:50 +11003005}
3006
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003007VolumeShaper::Status AudioTrack::applyVolumeShaper(
3008 const sp<VolumeShaper::Configuration>& configuration,
3009 const sp<VolumeShaper::Operation>& operation)
3010{
3011 AutoMutex lock(mLock);
Andy Hung4ef88d72017-02-21 19:47:53 -08003012 mVolumeHandler->setIdIfNecessary(configuration);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003013 media::VolumeShaperConfiguration config;
3014 configuration->writeToParcelable(&config);
3015 media::VolumeShaperOperation op;
3016 operation->writeToParcelable(&op);
3017 VolumeShaper::Status status;
3018 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003019
3020 if (status == DEAD_OBJECT) {
3021 if (restoreTrack_l("applyVolumeShaper") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003022 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003023 }
3024 }
Andy Hung4ef88d72017-02-21 19:47:53 -08003025 if (status >= 0) {
3026 // save VolumeShaper for restore
3027 mVolumeHandler->applyVolumeShaper(configuration, operation);
Andy Hung39399b62017-04-21 15:07:45 -07003028 if (mState == STATE_ACTIVE || mState == STATE_STOPPING) {
3029 mVolumeHandler->setStarted();
3030 }
3031 } else {
3032 // warn only if not an expected restore failure.
3033 ALOGW_IF(!((isOffloadedOrDirect_l() || mDoNotReconnect) && status == DEAD_OBJECT),
Eric Laurent973db022018-11-20 14:54:31 -08003034 "%s(%d): applyVolumeShaper failed: %d", __func__, mPortId, status);
Andy Hung4ef88d72017-02-21 19:47:53 -08003035 }
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003036 return status;
3037}
3038
3039sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id)
3040{
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003041 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003042 std::optional<media::VolumeShaperState> vss;
3043 mAudioTrack->getVolumeShaperState(id, &vss);
3044 sp<VolumeShaper::State> state;
3045 if (vss.has_value()) {
3046 state = new VolumeShaper::State();
3047 state->readFromParcelable(vss.value());
3048 }
Andy Hung39399b62017-04-21 15:07:45 -07003049 if (state.get() == nullptr && (mCblk->mFlags & CBLK_INVALID) != 0) {
3050 if (restoreTrack_l("getVolumeShaperState") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003051 mAudioTrack->getVolumeShaperState(id, &vss);
3052 if (vss.has_value()) {
3053 state = new VolumeShaper::State();
3054 state->readFromParcelable(vss.value());
3055 }
Andy Hung39399b62017-04-21 15:07:45 -07003056 }
3057 }
3058 return state;
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003059}
3060
Andy Hungea2b9c02016-02-12 17:06:53 -08003061status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp)
3062{
3063 if (timestamp == nullptr) {
3064 return BAD_VALUE;
3065 }
3066 AutoMutex lock(mLock);
Andy Hunge13f8a62016-03-30 14:20:42 -07003067 return getTimestamp_l(timestamp);
3068}
3069
3070status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp)
3071{
Andy Hungea2b9c02016-02-12 17:06:53 -08003072 if (mCblk->mFlags & CBLK_INVALID) {
3073 const status_t status = restoreTrack_l("getTimestampExtended");
3074 if (status != OK) {
3075 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3076 // recommending that the track be recreated.
3077 return DEAD_OBJECT;
3078 }
3079 }
3080 // check for offloaded/direct here in case restoring somehow changed those flags.
3081 if (isOffloadedOrDirect_l()) {
3082 return INVALID_OPERATION; // not supported
3083 }
3084 status_t status = mProxy->getTimestamp(timestamp);
Andy Hungfb8ede22018-09-12 19:03:24 -07003085 LOG_ALWAYS_FATAL_IF(status != OK, "%s(%d): status %d not allowed from proxy getTimestamp",
Eric Laurent973db022018-11-20 14:54:31 -08003086 __func__, mPortId, status);
Andy Hungea2b9c02016-02-12 17:06:53 -08003087 bool found = false;
Andy Hunge1e98462016-04-12 10:18:51 -07003088 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten;
3089 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
3090 // server side frame offset in case AudioTrack has been restored.
3091 for (int i = ExtendedTimestamp::LOCATION_SERVER;
3092 i < ExtendedTimestamp::LOCATION_MAX; ++i) {
3093 if (timestamp->mTimeNs[i] >= 0) {
3094 // apply server offset (frames flushed is ignored
3095 // so we don't report the jump when the flush occurs).
3096 timestamp->mPosition[i] += mFramesWrittenServerOffset;
3097 found = true;
Andy Hungea2b9c02016-02-12 17:06:53 -08003098 }
3099 }
3100 return found ? OK : WOULD_BLOCK;
3101}
3102
Glenn Kastence703742013-07-19 16:33:58 -07003103status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
3104{
Glenn Kasten53cec222013-08-29 09:01:02 -07003105 AutoMutex lock(mLock);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003106 return getTimestamp_l(timestamp);
3107}
Phil Burk1b420972015-04-22 10:52:21 -07003108
Andy Hung65ffdfc2016-10-10 15:52:11 -07003109status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp)
3110{
Phil Burk1b420972015-04-22 10:52:21 -07003111 bool previousTimestampValid = mPreviousTimestampValid;
3112 // Set false here to cover all the error return cases.
3113 mPreviousTimestampValid = false;
3114
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003115 switch (mState) {
3116 case STATE_ACTIVE:
3117 case STATE_PAUSED:
3118 break; // handle below
3119 case STATE_FLUSHED:
3120 case STATE_STOPPED:
3121 return WOULD_BLOCK;
3122 case STATE_STOPPING:
3123 case STATE_PAUSED_STOPPING:
3124 if (!isOffloaded_l()) {
3125 return INVALID_OPERATION;
3126 }
3127 break; // offloaded tracks handled below
3128 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003129 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in getTimestamp(): %d",
Eric Laurent973db022018-11-20 14:54:31 -08003130 __func__, mPortId, mState);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003131 break;
Glenn Kastenfe346c72013-08-30 13:28:22 -07003132 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003133
Eric Laurent275e8e92014-11-30 15:14:47 -08003134 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung6653c932015-06-08 13:27:48 -07003135 const status_t status = restoreTrack_l("getTimestamp");
3136 if (status != OK) {
3137 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3138 // recommending that the track be recreated.
3139 return DEAD_OBJECT;
3140 }
Eric Laurent275e8e92014-11-30 15:14:47 -08003141 }
3142
Glenn Kasten200092b2014-08-15 15:13:30 -07003143 // The presented frame count must always lag behind the consumed frame count.
3144 // To avoid a race, read the presented frames first. This ensures that presented <= consumed.
Andy Hung6ae58432016-02-16 18:32:24 -08003145
3146 status_t status;
Andy Hung818e7a32016-02-16 18:08:07 -08003147 if (isOffloadedOrDirect_l()) {
Andy Hung6ae58432016-02-16 18:32:24 -08003148 // use Binder to get timestamp
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003149 media::AudioTimestampInternal ts;
3150 mAudioTrack->getTimestamp(&ts, &status);
3151 if (status == OK) {
Andy Hung973638a2020-12-08 20:47:45 -08003152 timestamp = VALUE_OR_FATAL(aidl2legacy_AudioTimestampInternal_AudioTimestamp(ts));
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003153 }
Andy Hung6ae58432016-02-16 18:32:24 -08003154 } else {
3155 // read timestamp from shared memory
3156 ExtendedTimestamp ets;
3157 status = mProxy->getTimestamp(&ets);
3158 if (status == OK) {
Andy Hungb01faa32016-04-27 12:51:32 -07003159 ExtendedTimestamp::Location location;
3160 status = ets.getBestTimestamp(&timestamp, &location);
3161
3162 if (status == OK) {
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003163 updateLatency_l();
Andy Hungb01faa32016-04-27 12:51:32 -07003164 // It is possible that the best location has moved from the kernel to the server.
3165 // In this case we adjust the position from the previous computed latency.
3166 if (location == ExtendedTimestamp::LOCATION_SERVER) {
3167 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL,
Andy Hungfb8ede22018-09-12 19:03:24 -07003168 "%s(%d): location moved from kernel to server",
Eric Laurent973db022018-11-20 14:54:31 -08003169 __func__, mPortId);
Andy Hung07eee802016-06-21 16:47:16 -07003170 // check that the last kernel OK time info exists and the positions
3171 // are valid (if they predate the current track, the positions may
3172 // be zero or negative).
Andy Hungb01faa32016-04-27 12:51:32 -07003173 const int64_t frames =
Andy Hung6d7b1192016-05-07 22:59:48 -07003174 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
Andy Hung07eee802016-06-21 16:47:16 -07003175 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 ||
3176 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 ||
3177 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0)
Andy Hung6d7b1192016-05-07 22:59:48 -07003178 ?
3179 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed
3180 / 1000)
3181 :
3182 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3183 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]);
Andy Hungfb8ede22018-09-12 19:03:24 -07003184 ALOGV("%s(%d): frame adjustment:%lld timestamp:%s",
Eric Laurent973db022018-11-20 14:54:31 -08003185 __func__, mPortId, (long long)frames, ets.toString().c_str());
Andy Hungb01faa32016-04-27 12:51:32 -07003186 if (frames >= ets.mPosition[location]) {
3187 timestamp.mPosition = 0;
3188 } else {
3189 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames);
3190 }
Andy Hung69488c42016-05-16 18:43:33 -07003191 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) {
3192 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER,
Andy Hungfb8ede22018-09-12 19:03:24 -07003193 "%s(%d): location moved from server to kernel",
Eric Laurent973db022018-11-20 14:54:31 -08003194 __func__, mPortId);
Andy Hung98731a22019-04-08 19:19:07 -07003195
3196 if (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER] ==
3197 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL]) {
3198 // In Q, we don't return errors as an invalid time
3199 // but instead we leave the last kernel good timestamp alone.
3200 //
3201 // If server is identical to kernel, the device data pipeline is idle.
3202 // A better start time is now. The retrograde check ensures
3203 // timestamp monotonicity.
3204 const int64_t nowNs = systemTime();
Andy Hungcf3b7152019-04-19 18:29:21 -07003205 if (!mTimestampStallReported) {
3206 ALOGD("%s(%d): device stall time corrected using current time %lld",
3207 __func__, mPortId, (long long)nowNs);
3208 mTimestampStallReported = true;
3209 }
Andy Hung98731a22019-04-08 19:19:07 -07003210 timestamp.mTime = convertNsToTimespec(nowNs);
Andy Hungcf3b7152019-04-19 18:29:21 -07003211 } else {
3212 mTimestampStallReported = false;
Andy Hung98731a22019-04-08 19:19:07 -07003213 }
Andy Hungb01faa32016-04-27 12:51:32 -07003214 }
Andy Hung5d313802016-10-10 15:09:39 -07003215
3216 // We update the timestamp time even when paused.
3217 if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) {
3218 const int64_t now = systemTime();
Andy Hung2b01f002017-07-05 12:01:36 -07003219 const int64_t at = audio_utils_ns_from_timespec(&timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003220 const int64_t lag =
3221 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
3222 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0)
3223 ? int64_t(mAfLatency * 1000000LL)
3224 : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3225 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK])
3226 * NANOS_PER_SECOND / mSampleRate;
3227 const int64_t limit = now - lag; // no earlier than this limit
3228 if (at < limit) {
3229 ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld",
3230 (long long)lag, (long long)at, (long long)limit);
Andy Hungffa36952017-08-17 10:41:51 -07003231 timestamp.mTime = convertNsToTimespec(limit);
Andy Hung5d313802016-10-10 15:09:39 -07003232 }
3233 }
Andy Hungb01faa32016-04-27 12:51:32 -07003234 mPreviousLocation = location;
3235 } else {
3236 // right after AudioTrack is started, one may not find a timestamp
Eric Laurent973db022018-11-20 14:54:31 -08003237 ALOGV("%s(%d): getBestTimestamp did not find timestamp", __func__, mPortId);
Andy Hungb01faa32016-04-27 12:51:32 -07003238 }
Andy Hung6ae58432016-02-16 18:32:24 -08003239 }
3240 if (status == INVALID_OPERATION) {
Andy Hungf20a4e92016-08-15 19:10:34 -07003241 // INVALID_OPERATION occurs when no timestamp has been issued by the server;
3242 // other failures are signaled by a negative time.
3243 // If we come out of FLUSHED or STOPPED where the position is known
3244 // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of
3245 // "zero" for NuPlayer). We don't convert for track restoration as position
3246 // does not reset.
Andy Hungfb8ede22018-09-12 19:03:24 -07003247 ALOGV("%s(%d): timestamp server offset:%lld restore frames:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003248 __func__, mPortId,
Andy Hungf20a4e92016-08-15 19:10:34 -07003249 (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore);
3250 if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) {
3251 status = WOULD_BLOCK;
3252 }
Andy Hung6ae58432016-02-16 18:32:24 -08003253 }
3254 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003255 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08003256 ALOGV_IF(status != WOULD_BLOCK, "%s(%d): getTimestamp error:%#x", __func__, mPortId, status);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003257 return status;
3258 }
3259 if (isOffloadedOrDirect_l()) {
3260 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) {
3261 // use cached paused position in case another offloaded track is running.
3262 timestamp.mPosition = mPausedPosition;
3263 clock_gettime(CLOCK_MONOTONIC, &timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003264 // TODO: adjust for delay
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003265 return NO_ERROR;
3266 }
3267
3268 // Check whether a pending flush or stop has completed, as those commands may
Andy Hungc8e09c62015-06-03 23:43:36 -07003269 // be asynchronous or return near finish or exhibit glitchy behavior.
3270 //
3271 // Originally this showed up as the first timestamp being a continuation of
3272 // the previous song under gapless playback.
3273 // However, we sometimes see zero timestamps, then a glitch of
3274 // the previous song's position, and then correct timestamps afterwards.
Andy Hungffa36952017-08-17 10:41:51 -07003275 if (mStartFromZeroUs != 0 && mSampleRate != 0) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003276 static const int kTimeJitterUs = 100000; // 100 ms
3277 static const int k1SecUs = 1000000;
3278
3279 const int64_t timeNow = getNowUs();
3280
Andy Hungffa36952017-08-17 10:41:51 -07003281 if (timeNow < mStartFromZeroUs + k1SecUs) { // within first second of starting
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003282 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003283 if (timestampTimeUs < mStartFromZeroUs) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003284 return WOULD_BLOCK; // stale timestamp time, occurs before start.
3285 }
Andy Hungffa36952017-08-17 10:41:51 -07003286 const int64_t deltaTimeUs = timestampTimeUs - mStartFromZeroUs;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003287 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07003288 / ((double)mSampleRate * mPlaybackRate.mSpeed);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003289
3290 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
3291 // Verify that the counter can't count faster than the sample rate
Andy Hungc8e09c62015-06-03 23:43:36 -07003292 // since the start time. If greater, then that means we may have failed
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003293 // to completely flush or stop the previous playing track.
Andy Hungc8e09c62015-06-03 23:43:36 -07003294 ALOGW_IF(!mTimestampStartupGlitchReported,
Andy Hungfb8ede22018-09-12 19:03:24 -07003295 "%s(%d): startup glitch detected"
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003296 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)",
Eric Laurent973db022018-11-20 14:54:31 -08003297 __func__, mPortId,
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003298 (long long)deltaTimeUs, (long long)deltaPositionByUs,
3299 timestamp.mPosition);
Andy Hungc8e09c62015-06-03 23:43:36 -07003300 mTimestampStartupGlitchReported = true;
3301 if (previousTimestampValid
3302 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) {
3303 timestamp = mPreviousTimestamp;
3304 mPreviousTimestampValid = true;
3305 return NO_ERROR;
3306 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003307 return WOULD_BLOCK;
3308 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003309 if (deltaPositionByUs != 0) {
Andy Hungffa36952017-08-17 10:41:51 -07003310 mStartFromZeroUs = 0; // don't check again, we got valid nonzero position.
Andy Hungc8e09c62015-06-03 23:43:36 -07003311 }
3312 } else {
Andy Hungffa36952017-08-17 10:41:51 -07003313 mStartFromZeroUs = 0; // don't check again, start time expired.
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003314 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003315 mTimestampStartupGlitchReported = false;
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003316 }
3317 } else {
Glenn Kasten200092b2014-08-15 15:13:30 -07003318 // Update the mapping between local consumed (mPosition) and server consumed (mServer)
3319 (void) updateAndGetPosition_l();
3320 // Server consumed (mServer) and presented both use the same server time base,
3321 // and server consumed is always >= presented.
3322 // The delta between these represents the number of frames in the buffer pipeline.
3323 // If this delta between these is greater than the client position, it means that
3324 // actually presented is still stuck at the starting line (figuratively speaking),
3325 // waiting for the first frame to go by. So we can't report a valid timestamp yet.
Andy Hung90e8a972015-11-09 16:42:40 -08003326 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when
3327 // mPosition exceeds 32 bits.
3328 // TODO Remove when timestamp is updated to contain pipeline status info.
3329 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue();
3330 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */
3331 && (uint32_t)pipelineDepthInFrames > mPosition.value()) {
Glenn Kasten200092b2014-08-15 15:13:30 -07003332 return INVALID_OPERATION;
3333 }
3334 // Convert timestamp position from server time base to client time base.
3335 // TODO The following code should work OK now because timestamp.mPosition is 32-bit.
3336 // But if we change it to 64-bit then this could fail.
Andy Hung90e8a972015-11-09 16:42:40 -08003337 // Use Modulo computation here.
3338 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value();
Glenn Kasten200092b2014-08-15 15:13:30 -07003339 // Immediately after a call to getPosition_l(), mPosition and
3340 // mServer both represent the same frame position. mPosition is
3341 // in client's point of view, and mServer is in server's point of
3342 // view. So the difference between them is the "fudge factor"
3343 // between client and server views due to stop() and/or new
3344 // IAudioTrack. And timestamp.mPosition is initially in server's
3345 // point of view, so we need to apply the same fudge factor to it.
Glenn Kastenfe346c72013-08-30 13:28:22 -07003346 }
Phil Burk1b420972015-04-22 10:52:21 -07003347
3348 // Prevent retrograde motion in timestamp.
3349 // This is sometimes caused by erratic reports of the available space in the ALSA drivers.
3350 if (status == NO_ERROR) {
Andy Hung3b8c6332019-04-03 19:29:36 -07003351 // Fix stale time when checking timestamp right after start().
3352 // The position is at the last reported location but the time can be stale
3353 // due to pause or standby or cold start latency.
3354 //
3355 // We keep advancing the time (but not the position) to ensure that the
3356 // stale value does not confuse the application.
3357 //
3358 // For offload compatibility, use a default lag value here.
3359 // Any time discrepancy between this update and the pause timestamp is handled
3360 // by the retrograde check afterwards.
3361 int64_t currentTimeNanos = audio_utils_ns_from_timespec(&timestamp.mTime);
3362 const int64_t lagNs = int64_t(mAfLatency * 1000000LL);
3363 const int64_t limitNs = mStartNs - lagNs;
3364 if (currentTimeNanos < limitNs) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003365 if (!mTimestampStaleTimeReported) {
3366 ALOGD("%s(%d): stale timestamp time corrected, "
3367 "currentTimeNanos: %lld < limitNs: %lld < mStartNs: %lld",
3368 __func__, mPortId,
3369 (long long)currentTimeNanos, (long long)limitNs, (long long)mStartNs);
3370 mTimestampStaleTimeReported = true;
3371 }
Andy Hung3b8c6332019-04-03 19:29:36 -07003372 timestamp.mTime = convertNsToTimespec(limitNs);
3373 currentTimeNanos = limitNs;
Andy Hungcf3b7152019-04-19 18:29:21 -07003374 } else {
3375 mTimestampStaleTimeReported = false;
Andy Hung3b8c6332019-04-03 19:29:36 -07003376 }
3377
Andy Hungffa36952017-08-17 10:41:51 -07003378 // previousTimestampValid is set to false when starting after a stop or flush.
Phil Burk1b420972015-04-22 10:52:21 -07003379 if (previousTimestampValid) {
Andy Hung2b01f002017-07-05 12:01:36 -07003380 const int64_t previousTimeNanos =
3381 audio_utils_ns_from_timespec(&mPreviousTimestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003382
3383 // retrograde check
Phil Burk1b420972015-04-22 10:52:21 -07003384 if (currentTimeNanos < previousTimeNanos) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003385 if (!mTimestampRetrogradeTimeReported) {
3386 ALOGW("%s(%d): retrograde timestamp time corrected, %lld < %lld",
3387 __func__, mPortId,
3388 (long long)currentTimeNanos, (long long)previousTimeNanos);
3389 mTimestampRetrogradeTimeReported = true;
3390 }
Andy Hung5d313802016-10-10 15:09:39 -07003391 timestamp.mTime = mPreviousTimestamp.mTime;
Andy Hungcf3b7152019-04-19 18:29:21 -07003392 } else {
3393 mTimestampRetrogradeTimeReported = false;
Phil Burk1b420972015-04-22 10:52:21 -07003394 }
3395
3396 // Looking at signed delta will work even when the timestamps
3397 // are wrapping around.
Andy Hung90e8a972015-11-09 16:42:40 -08003398 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition)
3399 - mPreviousTimestamp.mPosition).signedValue();
Phil Burk4c5a3672015-04-30 16:18:53 -07003400 if (deltaPosition < 0) {
3401 // Only report once per position instead of spamming the log.
Andy Hungcf3b7152019-04-19 18:29:21 -07003402 if (!mTimestampRetrogradePositionReported) {
Andy Hungfb8ede22018-09-12 19:03:24 -07003403 ALOGW("%s(%d): retrograde timestamp position corrected, %d = %u - %u",
Eric Laurent973db022018-11-20 14:54:31 -08003404 __func__, mPortId,
Phil Burk4c5a3672015-04-30 16:18:53 -07003405 deltaPosition,
3406 timestamp.mPosition,
3407 mPreviousTimestamp.mPosition);
Andy Hungcf3b7152019-04-19 18:29:21 -07003408 mTimestampRetrogradePositionReported = true;
Phil Burk4c5a3672015-04-30 16:18:53 -07003409 }
3410 } else {
Andy Hungcf3b7152019-04-19 18:29:21 -07003411 mTimestampRetrogradePositionReported = false;
Phil Burk4c5a3672015-04-30 16:18:53 -07003412 }
Andy Hung5d313802016-10-10 15:09:39 -07003413 if (deltaPosition < 0) {
3414 timestamp.mPosition = mPreviousTimestamp.mPosition;
3415 deltaPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07003416 }
Andy Hung5d313802016-10-10 15:09:39 -07003417#if 0
3418 // Uncomment this to verify audio timestamp rate.
3419 const int64_t deltaTime =
Andy Hung2b01f002017-07-05 12:01:36 -07003420 audio_utils_ns_from_timespec(&timestamp.mTime) - previousTimeNanos;
Andy Hung5d313802016-10-10 15:09:39 -07003421 if (deltaTime != 0) {
3422 const int64_t computedSampleRate =
3423 deltaPosition * (long long)NANOS_PER_SECOND / deltaTime;
Andy Hungfb8ede22018-09-12 19:03:24 -07003424 ALOGD("%s(%d): computedSampleRate:%u sampleRate:%u",
Eric Laurent973db022018-11-20 14:54:31 -08003425 __func__, mPortId,
Andy Hung5d313802016-10-10 15:09:39 -07003426 (unsigned)computedSampleRate, mSampleRate);
3427 }
3428#endif
Phil Burk1b420972015-04-22 10:52:21 -07003429 }
3430 mPreviousTimestamp = timestamp;
3431 mPreviousTimestampValid = true;
3432 }
3433
Glenn Kastenfe346c72013-08-30 13:28:22 -07003434 return status;
Glenn Kastence703742013-07-19 16:33:58 -07003435}
3436
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003437String8 AudioTrack::getParameters(const String8& keys)
3438{
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003439 audio_io_handle_t output = getOutput();
Glenn Kasten142f5192014-03-25 17:44:59 -07003440 if (output != AUDIO_IO_HANDLE_NONE) {
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003441 return AudioSystem::getParameters(output, keys);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01003442 } else {
3443 return String8::empty();
3444 }
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003445}
3446
Glenn Kasten23a75452014-01-13 10:37:17 -08003447bool AudioTrack::isOffloaded() const
3448{
3449 AutoMutex lock(mLock);
3450 return isOffloaded_l();
3451}
3452
Eric Laurentab5cdba2014-06-09 17:22:27 -07003453bool AudioTrack::isDirect() const
3454{
3455 AutoMutex lock(mLock);
3456 return isDirect_l();
3457}
3458
3459bool AudioTrack::isOffloadedOrDirect() const
3460{
3461 AutoMutex lock(mLock);
3462 return isOffloadedOrDirect_l();
3463}
3464
3465
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003466status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003467{
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003468 String8 result;
3469
3470 result.append(" AudioTrack::dump\n");
Andy Hungfb8ede22018-09-12 19:03:24 -07003471 result.appendFormat(" id(%d) status(%d), state(%d), session Id(%d), flags(%#x)\n",
Eric Laurent973db022018-11-20 14:54:31 -08003472 mPortId, mStatus, mState, mSessionId, mFlags);
Eric Laurentd114b622017-11-27 18:37:04 -08003473 result.appendFormat(" stream type(%d), left - right volume(%f, %f)\n",
François Gaffie58d4be52018-11-06 15:30:12 +01003474 mStreamType,
Eric Laurentd114b622017-11-27 18:37:04 -08003475 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
Glenn Kasten49f36ba2017-12-06 13:02:02 -08003476 result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
Eric Laurentd114b622017-11-27 18:37:04 -08003477 mFormat, mChannelMask, mChannelCount);
3478 result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
3479 mSampleRate, mOriginalSampleRate, mPlaybackRate.mSpeed);
3480 result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
3481 mFrameCount, mReqFrameCount);
3482 result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
3483 " req. notif. per buff(%u)\n",
3484 mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
3485 result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
3486 mLatency, mSelectedDeviceId, mRoutedDeviceId);
3487 result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
3488 mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003489 ::write(fd, result.string(), result.size());
3490 return NO_ERROR;
3491}
3492
Phil Burk2812d9e2016-01-04 10:34:30 -08003493uint32_t AudioTrack::getUnderrunCount() const
3494{
3495 AutoMutex lock(mLock);
3496 return getUnderrunCount_l();
3497}
3498
3499uint32_t AudioTrack::getUnderrunCount_l() const
3500{
3501 return mProxy->getUnderrunCount() + mUnderrunCountOffset;
3502}
3503
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003504uint32_t AudioTrack::getUnderrunFrames() const
3505{
3506 AutoMutex lock(mLock);
3507 return mProxy->getUnderrunFrames();
3508}
3509
Andy Hung3a5c2f32021-02-17 15:06:42 -08003510void AudioTrack::setLogSessionId(const char *logSessionId)
3511{
3512 AutoMutex lock(mLock);
Andy Hung1a9c21b2021-02-25 20:43:18 -08003513 if (logSessionId == nullptr) logSessionId = ""; // an empty string is an unset session id.
Andy Hung3a5c2f32021-02-17 15:06:42 -08003514 if (mLogSessionId == logSessionId) return;
3515
3516 mLogSessionId = logSessionId;
3517 mediametrics::LogItem(mMetricsId)
3518 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETLOGSESSIONID)
3519 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, logSessionId)
3520 .record();
3521}
3522
Andy Hung839a3062021-02-17 11:15:16 -08003523void AudioTrack::setPlayerIId(int playerIId)
3524{
3525 AutoMutex lock(mLock);
3526 if (mPlayerIId == playerIId) return;
3527
3528 mPlayerIId = playerIId;
3529 mediametrics::LogItem(mMetricsId)
3530 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYERIID)
3531 .set(AMEDIAMETRICS_PROP_PLAYERIID, playerIId)
3532 .record();
3533}
3534
Eric Laurent296fb132015-05-01 11:38:42 -07003535status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback)
3536{
Eric Laurent09f1ed22019-04-24 17:45:17 -07003537
Eric Laurent296fb132015-05-01 11:38:42 -07003538 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003539 ALOGW("%s(%d): adding NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003540 return BAD_VALUE;
3541 }
3542 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07003543 if (mDeviceCallback.unsafe_get() == callback.get()) {
Eric Laurent973db022018-11-20 14:54:31 -08003544 ALOGW("%s(%d): adding same callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003545 return INVALID_OPERATION;
3546 }
3547 status_t status = NO_ERROR;
3548 if (mOutput != AUDIO_IO_HANDLE_NONE) {
3549 if (mDeviceCallback != 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003550 ALOGW("%s(%d): callback already present!", __func__, mPortId);
Eric Laurent09f1ed22019-04-24 17:45:17 -07003551 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003552 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003553 status = AudioSystem::addAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003554 }
3555 mDeviceCallback = callback;
3556 return status;
3557}
3558
3559status_t AudioTrack::removeAudioDeviceCallback(
3560 const sp<AudioSystem::AudioDeviceCallback>& callback)
3561{
3562 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003563 ALOGW("%s(%d): removing NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003564 return BAD_VALUE;
3565 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003566 AutoMutex lock(mLock);
3567 if (mDeviceCallback.unsafe_get() != callback.get()) {
3568 ALOGW("%s removing different callback!", __FUNCTION__);
3569 return INVALID_OPERATION;
Eric Laurent296fb132015-05-01 11:38:42 -07003570 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003571 mDeviceCallback.clear();
Eric Laurent296fb132015-05-01 11:38:42 -07003572 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07003573 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003574 }
Eric Laurent296fb132015-05-01 11:38:42 -07003575 return NO_ERROR;
3576}
3577
Eric Laurentad2e7b92017-09-14 20:06:42 -07003578
3579void AudioTrack::onAudioDeviceUpdate(audio_io_handle_t audioIo,
3580 audio_port_handle_t deviceId)
3581{
3582 sp<AudioSystem::AudioDeviceCallback> callback;
3583 {
3584 AutoMutex lock(mLock);
3585 if (audioIo != mOutput) {
3586 return;
3587 }
3588 callback = mDeviceCallback.promote();
3589 // only update device if the track is active as route changes due to other use cases are
3590 // irrelevant for this client
3591 if (mState == STATE_ACTIVE) {
3592 mRoutedDeviceId = deviceId;
3593 }
3594 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003595
Eric Laurentad2e7b92017-09-14 20:06:42 -07003596 if (callback.get() != nullptr) {
3597 callback->onAudioDeviceUpdate(mOutput, mRoutedDeviceId);
3598 }
3599}
3600
Andy Hunge13f8a62016-03-30 14:20:42 -07003601status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location)
3602{
3603 if (msec == nullptr ||
3604 (location != ExtendedTimestamp::LOCATION_SERVER
3605 && location != ExtendedTimestamp::LOCATION_KERNEL)) {
3606 return BAD_VALUE;
3607 }
3608 AutoMutex lock(mLock);
3609 // inclusive of offloaded and direct tracks.
3610 //
3611 // It is possible, but not enabled, to allow duration computation for non-pcm
3612 // audio_has_proportional_frames() formats because currently they have
3613 // the drain rate equivalent to the pcm sample rate * framesize.
3614 if (!isPurePcmData_l()) {
3615 return INVALID_OPERATION;
3616 }
3617 ExtendedTimestamp ets;
3618 if (getTimestamp_l(&ets) == OK
3619 && ets.mTimeNs[location] > 0) {
3620 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
3621 - ets.mPosition[location];
3622 if (diff < 0) {
3623 *msec = 0;
3624 } else {
3625 // ms is the playback time by frames
3626 int64_t ms = (int64_t)((double)diff * 1000 /
3627 ((double)mSampleRate * mPlaybackRate.mSpeed));
3628 // clockdiff is the timestamp age (negative)
3629 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
3630 ets.mTimeNs[location]
3631 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
3632 - systemTime(SYSTEM_TIME_MONOTONIC);
3633
3634 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
3635 static const int NANOS_PER_MILLIS = 1000000;
3636 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
3637 }
3638 return NO_ERROR;
3639 }
3640 if (location != ExtendedTimestamp::LOCATION_SERVER) {
3641 return INVALID_OPERATION; // LOCATION_KERNEL is not available
3642 }
3643 // use server position directly (offloaded and direct arrive here)
3644 updateAndGetPosition_l();
3645 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue();
3646 *msec = (diff <= 0) ? 0
3647 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed));
3648 return NO_ERROR;
3649}
3650
Andy Hung65ffdfc2016-10-10 15:52:11 -07003651bool AudioTrack::hasStarted()
3652{
3653 AutoMutex lock(mLock);
3654 switch (mState) {
3655 case STATE_STOPPED:
3656 if (isOffloadedOrDirect_l()) {
3657 // check if we have started in the past to return true.
Andy Hungffa36952017-08-17 10:41:51 -07003658 return mStartFromZeroUs > 0;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003659 }
3660 // A normal audio track may still be draining, so
3661 // check if stream has ended. This covers fasttrack position
3662 // instability and start/stop without any data written.
3663 if (mProxy->getStreamEndDone()) {
3664 return true;
3665 }
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003666 FALLTHROUGH_INTENDED;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003667 case STATE_ACTIVE:
3668 case STATE_STOPPING:
3669 break;
3670 case STATE_PAUSED:
3671 case STATE_PAUSED_STOPPING:
3672 case STATE_FLUSHED:
3673 return false; // we're not active
3674 default:
Eric Laurent973db022018-11-20 14:54:31 -08003675 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in hasStarted(): %d", __func__, mPortId, mState);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003676 break;
3677 }
3678
3679 // wait indicates whether we need to wait for a timestamp.
3680 // This is conservatively figured - if we encounter an unexpected error
3681 // then we will not wait.
3682 bool wait = false;
3683 if (isOffloadedOrDirect_l()) {
3684 AudioTimestamp ts;
3685 status_t status = getTimestamp_l(ts);
3686 if (status == WOULD_BLOCK) {
3687 wait = true;
3688 } else if (status == OK) {
3689 wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition);
3690 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003691 ALOGV("%s(%d): hasStarted wait:%d ts:%u start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003692 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003693 (int)wait,
3694 ts.mPosition,
3695 (long long)mStartTs.mPosition);
3696 } else {
3697 int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG
3698 ExtendedTimestamp ets;
3699 status_t status = getTimestamp_l(&ets);
3700 if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets
3701 wait = true;
3702 } else if (status == OK) {
3703 for (location = ExtendedTimestamp::LOCATION_KERNEL;
3704 location >= ExtendedTimestamp::LOCATION_SERVER; --location) {
3705 if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) {
3706 continue;
3707 }
3708 wait = ets.mPosition[location] == 0
3709 || ets.mPosition[location] == mStartEts.mPosition[location];
3710 break;
3711 }
3712 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003713 ALOGV("%s(%d): hasStarted wait:%d ets:%lld start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003714 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003715 (int)wait,
3716 (long long)ets.mPosition[location],
3717 (long long)mStartEts.mPosition[location]);
3718 }
3719 return !wait;
3720}
3721
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003722// =========================================================================
3723
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003724void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003725{
3726 sp<AudioTrack> audioTrack = mAudioTrack.promote();
3727 if (audioTrack != 0) {
3728 AutoMutex lock(audioTrack->mLock);
3729 audioTrack->mProxy->binderDied();
3730 }
3731}
3732
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003733// =========================================================================
3734
Andy Hungca353672019-03-06 11:54:38 -08003735AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver)
Andy Hungeb46cf92019-03-06 10:13:38 -08003736 : Thread(true /* bCanCallJava */) // binder recursion on restoreTrack_l() may call Java.
3737 , mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
Glenn Kasten598de6c2013-10-16 17:02:13 -07003738 mIgnoreNextPausedInt(false)
Glenn Kasten3acbd052012-02-28 10:39:56 -08003739{
3740}
3741
3742AudioTrack::AudioTrackThread::~AudioTrackThread()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003743{
3744}
3745
3746bool AudioTrack::AudioTrackThread::threadLoop()
3747{
Glenn Kasten3acbd052012-02-28 10:39:56 -08003748 {
3749 AutoMutex _l(mMyLock);
3750 if (mPaused) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003751 // TODO check return value and handle or log
Glenn Kasten3acbd052012-02-28 10:39:56 -08003752 mMyCond.wait(mMyLock);
3753 // caller will check for exitPending()
3754 return true;
3755 }
Glenn Kasten598de6c2013-10-16 17:02:13 -07003756 if (mIgnoreNextPausedInt) {
3757 mIgnoreNextPausedInt = false;
3758 mPausedInt = false;
3759 }
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003760 if (mPausedInt) {
Glenn Kasten47d55172017-05-23 11:19:30 -07003761 // TODO use futex instead of condition, for event flag "or"
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003762 if (mPausedNs > 0) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003763 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003764 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
3765 } else {
Glenn Kasten75f79032017-05-23 11:22:44 -07003766 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003767 mMyCond.wait(mMyLock);
3768 }
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003769 mPausedInt = false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003770 return true;
3771 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08003772 }
Eric Laurent7985dcb2014-10-07 15:45:14 -07003773 if (exitPending()) {
3774 return false;
3775 }
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003776 nsecs_t ns = mReceiver.processAudioBuffer();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003777 switch (ns) {
3778 case 0:
3779 return true;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003780 case NS_INACTIVE:
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003781 pauseInternal();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003782 return true;
3783 case NS_NEVER:
3784 return false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003785 case NS_WHENEVER:
Andy Hung3c09c782014-12-29 18:39:32 -08003786 // Event driven: call wake() when callback notifications conditions change.
3787 ns = INT64_MAX;
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003788 FALLTHROUGH_INTENDED;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003789 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003790 LOG_ALWAYS_FATAL_IF(ns < 0, "%s(%d): processAudioBuffer() returned %lld",
Eric Laurent973db022018-11-20 14:54:31 -08003791 __func__, mReceiver.mPortId, (long long)ns);
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003792 pauseInternal(ns);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003793 return true;
Glenn Kastenca8b2802012-04-23 13:58:16 -07003794 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003795}
3796
Glenn Kasten3acbd052012-02-28 10:39:56 -08003797void AudioTrack::AudioTrackThread::requestExit()
3798{
3799 // must be in this order to avoid a race condition
3800 Thread::requestExit();
Glenn Kasten598de6c2013-10-16 17:02:13 -07003801 resume();
Glenn Kasten3acbd052012-02-28 10:39:56 -08003802}
3803
3804void AudioTrack::AudioTrackThread::pause()
3805{
3806 AutoMutex _l(mMyLock);
3807 mPaused = true;
3808}
3809
3810void AudioTrack::AudioTrackThread::resume()
3811{
3812 AutoMutex _l(mMyLock);
Glenn Kasten598de6c2013-10-16 17:02:13 -07003813 mIgnoreNextPausedInt = true;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003814 if (mPaused || mPausedInt) {
Glenn Kasten3acbd052012-02-28 10:39:56 -08003815 mPaused = false;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003816 mPausedInt = false;
Glenn Kasten3acbd052012-02-28 10:39:56 -08003817 mMyCond.signal();
3818 }
3819}
3820
Andy Hung3c09c782014-12-29 18:39:32 -08003821void AudioTrack::AudioTrackThread::wake()
3822{
3823 AutoMutex _l(mMyLock);
Andy Hunga8d08902015-07-22 11:52:13 -07003824 if (!mPaused) {
3825 // wake() might be called while servicing a callback - ignore the next
3826 // pause time and call processAudioBuffer.
Andy Hung3c09c782014-12-29 18:39:32 -08003827 mIgnoreNextPausedInt = true;
Andy Hunga8d08902015-07-22 11:52:13 -07003828 if (mPausedInt && mPausedNs > 0) {
3829 // audio track is active and internally paused with timeout.
3830 mPausedInt = false;
3831 mMyCond.signal();
3832 }
Andy Hung3c09c782014-12-29 18:39:32 -08003833 }
3834}
3835
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003836void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
3837{
3838 AutoMutex _l(mMyLock);
3839 mPausedInt = true;
3840 mPausedNs = ns;
3841}
3842
jiabinf6eb4c32020-02-25 14:06:25 -08003843binder::Status AudioTrack::AudioTrackCallback::onCodecFormatChanged(
3844 const std::vector<uint8_t>& audioMetadata)
3845{
3846 AutoMutex _l(mAudioTrackCbLock);
3847 sp<media::IAudioTrackCallback> callback = mCallback.promote();
3848 if (callback.get() != nullptr) {
3849 callback->onCodecFormatChanged(audioMetadata);
3850 } else {
3851 mCallback.clear();
3852 }
3853 return binder::Status::ok();
3854}
3855
3856void AudioTrack::AudioTrackCallback::setAudioTrackCallback(
3857 const sp<media::IAudioTrackCallback> &callback) {
3858 AutoMutex lock(mAudioTrackCbLock);
3859 mCallback = callback;
3860}
3861
Glenn Kasten40bc9062015-03-20 09:09:33 -07003862} // namespace android