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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*/));
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800178 media::AudioAttributesInternal attributesAidl = VALUE_OR_RETURN(
179 legacy2aidl_audio_attributes_t_AudioAttributesInternal(attributes));
180 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}
328
329AudioTrack::AudioTrack(
330 audio_stream_type_t streamType,
331 uint32_t sampleRate,
332 audio_format_t format,
333 audio_channel_mask_t channelMask,
334 size_t frameCount,
335 audio_output_flags_t flags,
336 legacy_callback_t callback,
337 void* user,
338 int32_t notificationFrames,
339 audio_session_t sessionId,
340 transfer_type transferType,
341 const audio_offload_info_t *offloadInfo,
342 const AttributionSourceState& attributionSource,
343 const audio_attributes_t* pAttributes,
344 bool doNotReconnect,
345 float maxRequiredSpeed,
346 audio_port_handle_t selectedDeviceId)
347 : mStatus(NO_INIT),
348 mState(STATE_STOPPED),
349 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
350 mPreviousSchedulingGroup(SP_DEFAULT),
351 mPausedPosition(0),
352 mAudioTrackCallback(new AudioTrackCallback())
353{
354 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
355 if (callback != nullptr) {
356 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
357 } else if (user) {
358 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
359 }
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500360 mSetParams = std::unique_ptr<SetParams>{new SetParams{
361 streamType, sampleRate, format, channelMask, frameCount, flags, mLegacyCallbackWrapper,
362 notificationFrames, 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId,
363 transferType, offloadInfo, attributionSource, pAttributes, doNotReconnect,
364 maxRequiredSpeed, selectedDeviceId}};
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800365}
366
Andreas Huberc8139852012-01-18 10:51:55 -0800367AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800368 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800369 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800370 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700371 audio_channel_mask_t channelMask,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800372 const sp<IMemory>& sharedBuffer,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700373 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400374 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700375 int32_t notificationFrames,
Glenn Kastend848eb42016-03-08 13:42:11 -0800376 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000377 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800378 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000379 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700380 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700381 bool doNotReconnect,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700382 float maxRequiredSpeed)
Glenn Kasten87913512011-06-22 16:15:25 -0700383 : mStatus(NO_INIT),
Haynes Mathew George9b3359f2016-03-23 19:09:10 -0700384 mState(STATE_STOPPED),
John Grossman4ff14ba2012-02-08 16:37:41 -0800385 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800386 mPreviousSchedulingGroup(SP_DEFAULT),
Paul McLeanaa981192015-03-21 09:55:15 -0700387 mPausedPosition(0),
jiabinf6eb4c32020-02-25 14:06:25 -0800388 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
389 mAudioTrackCallback(new AudioTrackCallback())
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800390{
François Gaffie393f0e02019-04-10 09:09:08 +0200391 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Ippei Murata5fa32ed2018-10-01 17:37:50 +0900392
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500393 mSetParams = std::unique_ptr<SetParams>{
394 new SetParams{streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags,
395 callback, notificationFrames, sharedBuffer, false /*threadCanCallJava*/,
396 sessionId, transferType, offloadInfo, attributionSource, pAttributes,
397 doNotReconnect, maxRequiredSpeed, AUDIO_PORT_HANDLE_NONE}};
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800398}
399
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400400AudioTrack::AudioTrack(
401 audio_stream_type_t streamType,
402 uint32_t sampleRate,
403 audio_format_t format,
404 audio_channel_mask_t channelMask,
405 const sp<IMemory>& sharedBuffer,
406 audio_output_flags_t flags,
407 legacy_callback_t callback,
408 void* user,
409 int32_t notificationFrames,
410 audio_session_t sessionId,
411 transfer_type transferType,
412 const audio_offload_info_t *offloadInfo,
413 const AttributionSourceState& attributionSource,
414 const audio_attributes_t* pAttributes,
415 bool doNotReconnect,
416 float maxRequiredSpeed)
417 : mStatus(NO_INIT),
418 mState(STATE_STOPPED),
419 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
420 mPreviousSchedulingGroup(SP_DEFAULT),
421 mPausedPosition(0),
422 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
423 mAudioTrackCallback(new AudioTrackCallback())
424{
425 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
426 if (callback) {
427 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
428 } else if (user) {
429 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
430 }
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500431 mSetParams = std::unique_ptr<SetParams>{new SetParams{
432 streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags,
433 mLegacyCallbackWrapper, notificationFrames, sharedBuffer, false /*threadCanCallJava*/,
434 sessionId, transferType, offloadInfo, attributionSource, pAttributes, doNotReconnect,
435 maxRequiredSpeed, AUDIO_PORT_HANDLE_NONE}};
436}
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400437
Atneya Nairaf12f3c2022-02-28 15:44:25 -0500438void AudioTrack::onFirstRef() {
439 if (mSetParams) {
440 set(*mSetParams);
441 mSetParams.reset();
442 }
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400443}
444
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800445AudioTrack::~AudioTrack()
446{
Ray Essicked304702017-12-12 14:00:57 -0800447 // pull together the numbers, before we clean up our structures
448 mMediaMetrics.gather(this);
449
Andy Hungb68f5eb2019-12-03 16:49:17 -0800450 mediametrics::LogItem(mMetricsId)
451 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DTOR)
Phil Burkd3813f32020-04-23 16:26:15 -0700452 .set(AMEDIAMETRICS_PROP_CALLERNAME,
453 mCallerName.empty()
Andy Hunga6b27032020-04-27 10:34:24 -0700454 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
Phil Burkd3813f32020-04-23 16:26:15 -0700455 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800456 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
457 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)mStatus)
458 .record();
459
Phil Burk7a9577c2021-03-12 20:12:11 +0000460 stopAndJoinCallbacks(); // checks mStatus
461
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800462 if (mStatus == NO_ERROR) {
Marco Nelissenf8880202014-11-14 07:58:25 -0800463 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten53cec222013-08-29 09:01:02 -0700464 mAudioTrack.clear();
Eric Laurent3bcffa12014-06-12 18:38:45 -0700465 mCblkMemory.clear();
466 mSharedBuffer.clear();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800467 IPCThreadState::self()->flushCommands();
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000468 pid_t clientPid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
Andy Hungfb8ede22018-09-12 19:03:24 -0700469 ALOGV("%s(%d), releasing session id %d from %d on behalf of %d",
Eric Laurent973db022018-11-20 14:54:31 -0800470 __func__, mPortId,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700471 mSessionId, IPCThreadState::self()->getCallingPid(), clientPid);
472 AudioSystem::releaseAudioSessionId(mSessionId, clientPid);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800473 }
474}
475
Phil Burk7a9577c2021-03-12 20:12:11 +0000476void AudioTrack::stopAndJoinCallbacks() {
477 // Prevent nullptr crash if it did not open properly.
478 if (mStatus != NO_ERROR) return;
479
480 // Make sure that callback function exits in the case where
481 // it is looping on buffer full condition in obtainBuffer().
482 // Otherwise the callback thread will never exit.
483 stop();
484 if (mAudioTrackThread != 0) { // not thread safe
Phil Burk7a9577c2021-03-12 20:12:11 +0000485 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
Kuowei Lifdf8e0f2021-11-26 17:38:34 +0800486 mProxy->interrupt();
Phil Burk7a9577c2021-03-12 20:12:11 +0000487 mAudioTrackThread->requestExitAndWait();
488 mAudioTrackThread.clear();
489 }
zhenjun.zhang116df6a2021-12-08 09:17:13 +0800490
491 AutoMutex lock(mLock);
Phil Burk7a9577c2021-03-12 20:12:11 +0000492 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) {
493 // This may not stop all of these device callbacks!
494 // TODO: Add some sort of protection.
495 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
496 mDeviceCallback.clear();
497 }
498}
499
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800500status_t AudioTrack::set(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800501 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800502 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800503 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700504 audio_channel_mask_t channelMask,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800505 size_t frameCount,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700506 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400507 legacy_callback_t callback,
508 void * user,
509 int32_t notificationFrames,
510 const sp<IMemory>& sharedBuffer,
511 bool threadCanCallJava,
512 audio_session_t sessionId,
513 transfer_type transferType,
514 const audio_offload_info_t *offloadInfo,
515 const AttributionSourceState& attributionSource,
516 const audio_attributes_t* pAttributes,
517 bool doNotReconnect,
518 float maxRequiredSpeed,
519 audio_port_handle_t selectedDeviceId)
520{
521 if (callback) {
522 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
523 } else if (user) {
524 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
525 }
526 return set(streamType, sampleRate,format, channelMask, frameCount, flags,
527 mLegacyCallbackWrapper, notificationFrames, sharedBuffer, threadCanCallJava,
528 sessionId, transferType, offloadInfo, attributionSource, pAttributes,
529 doNotReconnect, maxRequiredSpeed, selectedDeviceId);
530}
531status_t AudioTrack::set(
532 audio_stream_type_t streamType,
533 uint32_t sampleRate,
534 audio_format_t format,
535 audio_channel_mask_t channelMask,
536 size_t frameCount,
537 audio_output_flags_t flags,
538 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700539 int32_t notificationFrames,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800540 const sp<IMemory>& sharedBuffer,
Eric Laurentbe916aa2010-06-01 23:49:17 -0700541 bool threadCanCallJava,
Glenn Kastend848eb42016-03-08 13:42:11 -0800542 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000543 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800544 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000545 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700546 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700547 bool doNotReconnect,
jiabin156c6872017-10-06 09:47:15 -0700548 float maxRequiredSpeed,
549 audio_port_handle_t selectedDeviceId)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800550{
Atneya Nair14aabae2021-11-30 17:36:24 -0500551 LOG_ALWAYS_FATAL_IF(mInitialized, "%s: should not be called twice", __func__);
552 mInitialized = true;
Eric Laurentf32d7812017-11-30 14:44:07 -0800553 status_t status;
554 uint32_t channelCount;
555 pid_t callingPid;
556 pid_t myPid;
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000557 uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
558 pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
Andy Hung3acde2c2021-11-11 09:18:08 -0800559 std::string errorMessage;
Eric Laurent973db022018-11-20 14:54:31 -0800560 // Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
Andy Hungfb8ede22018-09-12 19:03:24 -0700561 ALOGV("%s(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
Glenn Kastenea38ee72016-04-18 11:08:01 -0700562 "flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
Andy Hungfb8ede22018-09-12 19:03:24 -0700563 __func__,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800564 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000565 sessionId, transferType, attributionSource.uid, attributionSource.pid);
Glenn Kasten86f04662014-02-24 15:13:05 -0800566
Phil Burk33ff89b2015-11-30 11:16:01 -0800567 mThreadCanCallJava = threadCanCallJava;
Andy Hungde602302021-12-07 21:35:49 -0800568
569 // These variables are pulled in an error report, so we initialize them early.
jiabin156c6872017-10-06 09:47:15 -0700570 mSelectedDeviceId = selectedDeviceId;
Eric Laurent21da6472017-11-09 16:29:26 -0800571 mSessionId = sessionId;
Andy Hungde602302021-12-07 21:35:49 -0800572 mChannelMask = channelMask;
Andy Hungde602302021-12-07 21:35:49 -0800573 mReqFrameCount = mFrameCount = frameCount;
574 mSampleRate = sampleRate;
575 mOriginalSampleRate = sampleRate;
576 mAttributes = pAttributes != nullptr ? *pAttributes : AUDIO_ATTRIBUTES_INITIALIZER;
577 mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
Phil Burk33ff89b2015-11-30 11:16:01 -0800578
Eric Laurentd7f33c52022-01-06 13:54:56 +0100579 // update format and flags before storing them in mFormat, mOrigFlags and mFlags
580 if (pAttributes != NULL) {
581 // stream type shouldn't be looked at, this track has audio attributes
582 ALOGV("%s(): Building AudioTrack with attributes:"
583 " usage=%d content=%d flags=0x%x tags=[%s]",
584 __func__,
585 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
586 audio_flags_to_audio_output_flags(mAttributes.flags, &flags);
587 }
588
589 // these below should probably come from the audioFlinger too...
590 if (format == AUDIO_FORMAT_DEFAULT) {
591 format = AUDIO_FORMAT_PCM_16_BIT;
592 } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
593 flags = static_cast<audio_output_flags_t>(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
594 }
595
596 // force direct flag if format is not linear PCM
597 // or offload was requested
598 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
599 || !audio_is_linear_pcm(format)) {
600 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
601 ? "%s(): Offload request, forcing to Direct Output"
602 : "%s(): Not linear PCM, forcing to Direct Output",
603 __func__);
604 flags = (audio_output_flags_t)
605 // FIXME why can't we allow direct AND fast?
606 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
607 }
608
609 // force direct flag if HW A/V sync requested
610 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
611 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
612 }
613
614 mFormat = format;
615 mOrigFlags = mFlags = flags;
616
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800617 switch (transferType) {
618 case TRANSFER_DEFAULT:
619 if (sharedBuffer != 0) {
620 transferType = TRANSFER_SHARED;
Atneya Nairba809b82022-03-04 18:11:10 -0500621 } else if (callback == nullptr|| threadCanCallJava) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800622 transferType = TRANSFER_SYNC;
623 } else {
624 transferType = TRANSFER_CALLBACK;
625 }
626 break;
627 case TRANSFER_CALLBACK:
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700628 case TRANSFER_SYNC_NOTIF_CALLBACK:
Atneya Nairba809b82022-03-04 18:11:10 -0500629 if (callback == nullptr || sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800630 errorMessage = StringPrintf(
631 "%s: Transfer type %s but callback == nullptr || sharedBuffer != 0",
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700632 convertTransferToText(transferType), __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800633 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800634 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800635 }
636 break;
637 case TRANSFER_OBTAIN:
638 case TRANSFER_SYNC:
639 if (sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800640 errorMessage = StringPrintf(
641 "%s: Transfer type TRANSFER_OBTAIN but sharedBuffer != 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800642 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800643 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800644 }
645 break;
646 case TRANSFER_SHARED:
647 if (sharedBuffer == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800648 errorMessage = StringPrintf(
649 "%s: Transfer type TRANSFER_SHARED but sharedBuffer == 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800650 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800651 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800652 }
653 break;
654 default:
Andy Hung3acde2c2021-11-11 09:18:08 -0800655 errorMessage = StringPrintf("%s: Invalid transfer type %d", __func__, transferType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800656 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800657 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800658 }
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800659 mSharedBuffer = sharedBuffer;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800660 mTransfer = transferType;
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700661 mDoNotReconnect = doNotReconnect;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800662
Andy Hungfb8ede22018-09-12 19:03:24 -0700663 ALOGV_IF(sharedBuffer != 0, "%s(): sharedBuffer: %p, size: %zu",
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -0700664 __func__, sharedBuffer->unsecurePointer(), sharedBuffer->size());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800665
Glenn Kasten53cec222013-08-29 09:01:02 -0700666 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Eric Laurent1dd70b92009-04-21 07:56:33 -0700667 if (mAudioTrack != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800668 errorMessage = StringPrintf("%s: Track already in use", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800669 status = INVALID_OPERATION;
Andy Hung3acde2c2021-11-11 09:18:08 -0800670 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800671 }
672
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800673 // handle default values first.
Eric Laurente83b55d2014-11-14 10:06:21 -0800674 if (streamType == AUDIO_STREAM_DEFAULT) {
Dima Zavinfce7a472011-04-19 22:30:36 -0700675 streamType = AUDIO_STREAM_MUSIC;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800676 }
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700677 if (pAttributes == NULL) {
Eric Laurent223fd5c2014-11-11 13:43:36 -0800678 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800679 errorMessage = StringPrintf("%s: Invalid stream type %d", __func__, streamType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800680 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800681 goto error;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700682 }
Andy Hunga2159aa2021-07-20 13:01:52 -0700683 mOriginalStreamType = streamType;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700684 } else {
Andy Hunga2159aa2021-07-20 13:01:52 -0700685 mOriginalStreamType = AUDIO_STREAM_DEFAULT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800686 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800687
688 // validate parameters
Dima Zavinfce7a472011-04-19 22:30:36 -0700689 if (!audio_is_valid_format(format)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800690 errorMessage = StringPrintf("%s: Invalid format %#x", __func__, format);
Eric Laurentf32d7812017-11-30 14:44:07 -0800691 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800692 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800693 }
Eric Laurentc2f1f072009-07-17 12:17:14 -0700694
Glenn Kasten8ba90322013-10-30 11:29:27 -0700695 if (!audio_is_output_channel(channelMask)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800696 errorMessage = StringPrintf("%s: Invalid channel mask %#x", __func__, channelMask);
Eric Laurentf32d7812017-11-30 14:44:07 -0800697 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800698 goto error;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700699 }
Eric Laurentf32d7812017-11-30 14:44:07 -0800700 channelCount = audio_channel_count_from_out_mask(channelMask);
Glenn Kastene3247bf2014-02-24 15:19:07 -0800701 mChannelCount = channelCount;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700702
Eric Laurentd7f33c52022-01-06 13:54:56 +0100703 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
Phil Burkfdb3c072016-02-09 10:47:02 -0800704 if (audio_has_proportional_frames(format)) {
Glenn Kastenb7730382014-04-30 15:50:31 -0700705 mFrameSize = channelCount * audio_bytes_per_sample(format);
706 } else {
707 mFrameSize = sizeof(uint8_t);
708 }
Glenn Kastene3aa6592012-12-04 12:22:46 -0800709 } else {
Phil Burkfdb3c072016-02-09 10:47:02 -0800710 ALOG_ASSERT(audio_has_proportional_frames(format));
Glenn Kastenb7730382014-04-30 15:50:31 -0700711 mFrameSize = channelCount * audio_bytes_per_sample(format);
Glenn Kastenb7730382014-04-30 15:50:31 -0700712 // createTrack will return an error if PCM format is not supported by server,
713 // so no need to check for specific PCM formats here
Glenn Kastene3aa6592012-12-04 12:22:46 -0800714 }
715
Eric Laurent0d6db582014-11-12 18:39:44 -0800716 // sampling rate must be specified for direct outputs
Eric Laurentd7f33c52022-01-06 13:54:56 +0100717 if (sampleRate == 0 && (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800718 errorMessage = StringPrintf(
719 "%s: sample rate must be specified for direct outputs", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800720 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800721 goto error;
Eric Laurent0d6db582014-11-12 18:39:44 -0800722 }
Andy Hungff874dc2016-04-11 16:49:09 -0700723 // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
724 mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
Eric Laurent0d6db582014-11-12 18:39:44 -0800725
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800726 // Make copy of input parameter offloadInfo so that in the future:
727 // (a) createTrack_l doesn't need it as an input parameter
728 // (b) we can support re-creation of offloaded tracks
729 if (offloadInfo != NULL) {
730 mOffloadInfoCopy = *offloadInfo;
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800731 } else {
Eric Laurent20b9ef02016-12-05 11:03:16 -0800732 memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
Ytai Ben-Tsviffa2fd92020-10-20 09:13:53 -0700733 mOffloadInfoCopy = AUDIO_INFO_INITIALIZER;
ziyang zhang6f11f9d2022-05-10 13:28:38 +0800734 mOffloadInfoCopy.format = format;
735 mOffloadInfoCopy.sample_rate = sampleRate;
736 mOffloadInfoCopy.channel_mask = channelMask;
737 mOffloadInfoCopy.stream_type = streamType;
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800738 }
739
Glenn Kasten66e46352014-01-16 17:44:23 -0800740 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
741 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
Glenn Kasten05632a52012-01-03 14:22:33 -0800742 mSendLevel = 0.0f;
Glenn Kasten396fabd2014-01-08 08:54:23 -0800743 // mFrameCount is initialized in createTrack_l
Glenn Kastenea38ee72016-04-18 11:08:01 -0700744 if (notificationFrames >= 0) {
745 mNotificationFramesReq = notificationFrames;
746 mNotificationsPerBufferReq = 0;
747 } else {
Eric Laurentd7f33c52022-01-06 13:54:56 +0100748 if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800749 errorMessage = StringPrintf(
750 "%s: notificationFrames=%d not permitted for non-fast track",
Andy Hungfb8ede22018-09-12 19:03:24 -0700751 __func__, notificationFrames);
Eric Laurentf32d7812017-11-30 14:44:07 -0800752 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800753 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700754 }
755 if (frameCount > 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700756 ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
757 __func__, notificationFrames, frameCount);
Eric Laurentf32d7812017-11-30 14:44:07 -0800758 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800759 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700760 }
761 mNotificationFramesReq = 0;
762 const uint32_t minNotificationsPerBuffer = 1;
763 const uint32_t maxNotificationsPerBuffer = 8;
764 mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
765 max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
766 ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
Andy Hungfb8ede22018-09-12 19:03:24 -0700767 "%s(): notificationFrames=%d clamped to the range -%u to -%u",
768 __func__,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700769 notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
770 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800771 mNotificationFramesAct = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700772 // TODO b/182392553: refactor or remove
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000773 mClientAttributionSource = AttributionSourceState(attributionSource);
Eric Laurentf32d7812017-11-30 14:44:07 -0800774 callingPid = IPCThreadState::self()->getCallingPid();
775 myPid = getpid();
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700776 if (uid == -1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000777 mClientAttributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700778 IPCThreadState::self()->getCallingUid()));
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800779 }
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700780 if (pid == (pid_t)-1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000781 mClientAttributionSource.pid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(callingPid));
Marco Nelissend457c972014-02-11 08:47:07 -0800782 }
Eric Laurent2beeb502010-07-16 07:43:46 -0700783 mAuxEffectId = 0;
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400784 mCallback = callback;
Eric Laurentbe916aa2010-06-01 23:49:17 -0700785
Atneya Nairba809b82022-03-04 18:11:10 -0500786 if (callback != nullptr) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400787 mAudioTrackThread = sp<AudioTrackThread>::make(*this);
Glenn Kastena997e7a2012-08-07 09:44:19 -0700788 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
Glenn Kastenbfd31842015-03-20 09:01:44 -0700789 // thread begins in paused state, and will not reference us until start()
Glenn Kastena997e7a2012-08-07 09:44:19 -0700790 }
791
Eric Laurent34f1d8e2009-11-04 08:27:26 -0800792 // create the IAudioTrack
Francois Gaffie24a9fb02019-01-18 17:51:34 +0100793 {
794 AutoMutex lock(mLock);
795 status = createTrack_l();
796 }
Glenn Kastena997e7a2012-08-07 09:44:19 -0700797 if (status != NO_ERROR) {
798 if (mAudioTrackThread != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100799 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
800 mAudioTrackThread->requestExitAndWait();
Glenn Kastena997e7a2012-08-07 09:44:19 -0700801 mAudioTrackThread.clear();
802 }
Andy Hung3acde2c2021-11-11 09:18:08 -0800803 // We do not goto error to prevent double-logging errors.
Eric Laurentf32d7812017-11-30 14:44:07 -0800804 goto exit;
Glenn Kasten5d464eb2012-06-22 17:19:53 -0700805 }
806
Andy Hung4ede21d2014-12-12 15:37:34 -0800807 mLoopCount = 0;
808 mLoopStart = 0;
809 mLoopEnd = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -0800810 mLoopCountNotified = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800811 mMarkerPosition = 0;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -0700812 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800813 mNewPosition = 0;
814 mUpdatePeriod = 0;
Glenn Kasten200092b2014-08-15 15:13:30 -0700815 mPosition = 0;
816 mReleased = 0;
Andy Hungffa36952017-08-17 10:41:51 -0700817 mStartNs = 0;
818 mStartFromZeroUs = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700819 AudioSystem::acquireAudioSessionId(mSessionId, pid, uid);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800820 mSequence = 1;
821 mObservedSequence = mSequence;
822 mInUnderrun = false;
Phil Burk1b420972015-04-22 10:52:21 -0700823 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700824 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700825 mTimestampRetrogradePositionReported = false;
826 mTimestampRetrogradeTimeReported = false;
827 mTimestampStallReported = false;
828 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700829 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Andy Hung65ffdfc2016-10-10 15:52:11 -0700830 mStartTs.mPosition = 0;
Phil Burk2812d9e2016-01-04 10:34:30 -0800831 mUnderrunCountOffset = 0;
Andy Hungea2b9c02016-02-12 17:06:53 -0800832 mFramesWritten = 0;
833 mFramesWrittenServerOffset = 0;
Andy Hungf20a4e92016-08-15 19:10:34 -0700834 mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
Ivan Lozano8cf3a072017-08-09 09:01:33 -0700835 mVolumeHandler = new media::VolumeHandler();
Eric Laurentf32d7812017-11-30 14:44:07 -0800836
Andy Hung3acde2c2021-11-11 09:18:08 -0800837error:
838 if (status != NO_ERROR) {
839 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
840 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
841 }
842 // fall through
Eric Laurentf32d7812017-11-30 14:44:07 -0800843exit:
844 mStatus = status;
845 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800846}
847
Mikhail Naganov55773032020-10-01 15:08:13 -0700848
849status_t AudioTrack::set(
850 audio_stream_type_t streamType,
851 uint32_t sampleRate,
852 audio_format_t format,
853 uint32_t channelMask,
854 size_t frameCount,
855 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400856 legacy_callback_t callback,
Mikhail Naganov55773032020-10-01 15:08:13 -0700857 void* user,
858 int32_t notificationFrames,
859 const sp<IMemory>& sharedBuffer,
860 bool threadCanCallJava,
861 audio_session_t sessionId,
862 transfer_type transferType,
863 const audio_offload_info_t *offloadInfo,
864 uid_t uid,
865 pid_t pid,
866 const audio_attributes_t* pAttributes,
867 bool doNotReconnect,
868 float maxRequiredSpeed,
869 audio_port_handle_t selectedDeviceId)
870{
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000871 AttributionSourceState attributionSource;
872 attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(uid));
873 attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(pid));
874 attributionSource.token = sp<BBinder>::make();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400875 if (callback) {
876 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
877 } else if (user) {
878 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
879 }
880 return set(streamType, sampleRate, format, static_cast<audio_channel_mask_t>(channelMask),
881 frameCount, flags, mLegacyCallbackWrapper, notificationFrames, sharedBuffer,
882 threadCanCallJava, sessionId, transferType, offloadInfo, attributionSource,
883 pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
Mikhail Naganov55773032020-10-01 15:08:13 -0700884}
885
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800886// -------------------------------------------------------------------------
887
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100888status_t AudioTrack::start()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800889{
Eric Laurentf5aafb22010-11-18 08:40:16 -0800890 AutoMutex lock(mLock);
Andy Hungb68f5eb2019-12-03 16:49:17 -0800891
Andy Hung10fb4be2020-05-27 22:22:22 -0700892 if (mState == STATE_ACTIVE) {
893 return INVALID_OPERATION;
894 }
895
896 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
897
898 // Defer logging here due to OpenSL ES repeated start calls.
899 // TODO(b/154868033) after fix, restore this logging back to the beginning of start().
900 const int64_t beginNs = systemTime();
Andy Hungb68f5eb2019-12-03 16:49:17 -0800901 status_t status = NO_ERROR; // logged: make sure to set this before returning.
Andy Hung06a730b2020-04-09 13:28:31 -0700902 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800903 mediametrics::LogItem(mMetricsId)
Andy Hunga6b27032020-04-27 10:34:24 -0700904 .set(AMEDIAMETRICS_PROP_CALLERNAME,
905 mCallerName.empty()
906 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
907 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800908 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_START)
Andy Hungea840382020-05-05 21:50:17 -0700909 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800910 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
911 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
912 .record(); });
913
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800914
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800915 mInUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800916
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800917 State previousState = mState;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100918 if (previousState == STATE_PAUSED_STOPPING) {
919 mState = STATE_STOPPING;
920 } else {
921 mState = STATE_ACTIVE;
922 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700923 (void) updateAndGetPosition_l();
Andy Hung65ffdfc2016-10-10 15:52:11 -0700924
925 // save start timestamp
926 if (isOffloadedOrDirect_l()) {
927 if (getTimestamp_l(mStartTs) != OK) {
928 mStartTs.mPosition = 0;
929 }
930 } else {
931 if (getTimestamp_l(&mStartEts) != OK) {
932 mStartEts.clear();
933 }
934 }
Andy Hungffa36952017-08-17 10:41:51 -0700935 mStartNs = systemTime(); // save this for timestamp adjustment after starting.
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800936 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
937 // reset current position as seen by client to 0
Glenn Kasten200092b2014-08-15 15:13:30 -0700938 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -0700939 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700940 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700941 mTimestampRetrogradePositionReported = false;
942 mTimestampRetrogradeTimeReported = false;
943 mTimestampStallReported = false;
944 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700945 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Phil Burk1b420972015-04-22 10:52:21 -0700946
Andy Hung65ffdfc2016-10-10 15:52:11 -0700947 if (!isOffloadedOrDirect_l()
948 && mStartEts.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) {
Andy Hunge1e98462016-04-12 10:18:51 -0700949 // Server side has consumed something, but is it finished consuming?
950 // It is possible since flush and stop are asynchronous that the server
951 // is still active at this point.
Andy Hungfb8ede22018-09-12 19:03:24 -0700952 ALOGV("%s(%d): server read:%lld cumulative flushed:%lld client written:%lld",
Eric Laurent973db022018-11-20 14:54:31 -0800953 __func__, mPortId,
Andy Hunge1e98462016-04-12 10:18:51 -0700954 (long long)(mFramesWrittenServerOffset
Andy Hung65ffdfc2016-10-10 15:52:11 -0700955 + mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]),
956 (long long)mStartEts.mFlushed,
Andy Hunge1e98462016-04-12 10:18:51 -0700957 (long long)mFramesWritten);
Andy Hungc4e60eb2017-09-06 11:14:57 -0700958 // mStartEts is already adjusted by mFramesWrittenServerOffset, so we delta adjust.
959 mFramesWrittenServerOffset -= mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER];
Andy Hung61be8412015-10-06 10:51:09 -0700960 }
Andy Hunge1e98462016-04-12 10:18:51 -0700961 mFramesWritten = 0;
962 mProxy->clearTimestamp(); // need new server push for valid timestamp
963 mMarkerReached = false;
Andy Hung61be8412015-10-06 10:51:09 -0700964
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700965 // For offloaded tracks, we don't know if the hardware counters are really zero here,
966 // since the flush is asynchronous and stop may not fully drain.
967 // We save the time when the track is started to later verify whether
968 // the counters are realistic (i.e. start from zero after this time).
Andy Hungffa36952017-08-17 10:41:51 -0700969 mStartFromZeroUs = mStartNs / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700970
Eric Laurentec9a0322013-08-28 10:23:01 -0700971 // force refresh of remaining frames by processAudioBuffer() as last
972 // write before stop could be partial.
973 mRefreshRemaining = true;
Tomoharu Kasahara6e8bf982017-06-09 16:17:33 +0900974
975 // for static track, clear the old flags when starting from stopped state
976 if (mSharedBuffer != 0) {
977 android_atomic_and(
978 ~(CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
979 &mCblk->mFlags);
980 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800981 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700982 mNewPosition = mPosition + mUpdatePeriod;
Andy Hung4be3b832016-10-13 17:51:43 -0700983 int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800984
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800985 if (!(flags & CBLK_INVALID)) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -0800986 mAudioTrack->start(&status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800987 if (status == DEAD_OBJECT) {
988 flags |= CBLK_INVALID;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800989 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800990 }
991 if (flags & CBLK_INVALID) {
992 status = restoreTrack_l("start");
993 }
994
Andy Hung79629f02016-03-24 13:57:40 -0700995 // resume or pause the callback thread as needed.
996 sp<AudioTrackThread> t = mAudioTrackThread;
997 if (status == NO_ERROR) {
998 if (t != 0) {
999 if (previousState == STATE_STOPPING) {
1000 mProxy->interrupt();
1001 } else {
1002 t->resume();
1003 }
1004 } else {
1005 mPreviousPriority = getpriority(PRIO_PROCESS, 0);
1006 get_sched_policy(0, &mPreviousSchedulingGroup);
1007 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
1008 }
Andy Hung39399b62017-04-21 15:07:45 -07001009
1010 // Start our local VolumeHandler for restoration purposes.
1011 mVolumeHandler->setStarted();
Andy Hung79629f02016-03-24 13:57:40 -07001012 } else {
Eric Laurent973db022018-11-20 14:54:31 -08001013 ALOGE("%s(%d): status %d", __func__, mPortId, status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001014 mState = previousState;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001015 if (t != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001016 if (previousState != STATE_STOPPING) {
1017 t->pause();
1018 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001019 } else {
Glenn Kasten87913512011-06-22 16:15:25 -07001020 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
Glenn Kastena6364332012-04-19 09:35:04 -07001021 set_sched_policy(0, mPreviousSchedulingGroup);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001022 }
1023 }
1024
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001025 return status;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001026}
1027
1028void AudioTrack::stop()
1029{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001030 const int64_t beginNs = systemTime();
1031
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001032 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -07001033 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001034 mediametrics::LogItem(mMetricsId)
1035 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_STOP)
Andy Hungea840382020-05-05 21:50:17 -07001036 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001037 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Phil Burk64e16a72020-06-01 13:25:51 -07001038 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1039 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getUnderrunCount_l())
Phil Burka9876702020-04-20 18:16:15 -07001040 .record();
Phil Burka9876702020-04-20 18:16:15 -07001041 });
Andy Hungb68f5eb2019-12-03 16:49:17 -08001042
Eric Laurent973db022018-11-20 14:54:31 -08001043 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001044
Glenn Kasten397edb32013-08-30 15:10:13 -07001045 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001046 return;
1047 }
1048
Glenn Kasten23a75452014-01-13 10:37:17 -08001049 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001050 mState = STATE_STOPPING;
1051 } else {
1052 mState = STATE_STOPPED;
Andy Hung2148bf02016-11-28 19:01:02 -08001053 ALOGD_IF(mSharedBuffer == nullptr,
Eric Laurent973db022018-11-20 14:54:31 -08001054 "%s(%d): called with %u frames delivered", __func__, mPortId, mReleased.value());
Andy Hungc2813e52014-10-16 17:54:34 -07001055 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001056 }
1057
Andy Hung1d3556d2018-03-29 16:30:14 -07001058 mProxy->stop(); // notify server not to read beyond current client position until start().
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001059 mProxy->interrupt();
1060 mAudioTrack->stop();
Andy Hung61be8412015-10-06 10:51:09 -07001061
1062 // Note: legacy handling - stop does not clear playback marker
1063 // and periodic update counter, but flush does for streaming tracks.
Andy Hung9b461582014-12-01 17:56:29 -08001064
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001065 if (mSharedBuffer != 0) {
Andy Hung9b461582014-12-01 17:56:29 -08001066 // clear buffer position and loop count.
Andy Hung9b461582014-12-01 17:56:29 -08001067 mStaticProxy->setBufferPositionAndLoop(0 /* position */,
1068 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001069 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001070
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001071 sp<AudioTrackThread> t = mAudioTrackThread;
1072 if (t != 0) {
Glenn Kasten23a75452014-01-13 10:37:17 -08001073 if (!isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001074 t->pause();
Jean-Michel Trivia60e9a22018-11-15 16:13:36 -08001075 } else if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
Jean-Michel Trivi0248a2a2018-11-21 10:05:57 -08001076 // causes wake up of the playback thread, that will callback the client for
1077 // EVENT_STREAM_END in processAudioBuffer()
1078 t->wake();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001079 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001080 } else {
1081 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
1082 set_sched_policy(0, mPreviousSchedulingGroup);
1083 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001084}
1085
1086bool AudioTrack::stopped() const
1087{
Glenn Kasten9a2aaf92012-01-03 09:42:47 -08001088 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001089 return mState != STATE_ACTIVE;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001090}
1091
1092void AudioTrack::flush()
1093{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001094 const int64_t beginNs = systemTime();
Andy Hungfb8ede22018-09-12 19:03:24 -07001095 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -07001096 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001097 mediametrics::LogItem(mMetricsId)
1098 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_FLUSH)
Andy Hungea840382020-05-05 21:50:17 -07001099 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001100 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
1101 .record(); });
1102
Eric Laurent973db022018-11-20 14:54:31 -08001103 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001104
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001105 if (mSharedBuffer != 0) {
1106 return;
Glenn Kasten4bae3642012-11-30 13:41:12 -08001107 }
Andy Hung4c5ed302018-05-09 11:16:21 -07001108 if (mState == STATE_ACTIVE) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001109 return;
1110 }
1111 flush_l();
Eric Laurent1703cdf2011-03-07 14:52:59 -08001112}
1113
Eric Laurent1703cdf2011-03-07 14:52:59 -08001114void AudioTrack::flush_l()
1115{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001116 ALOG_ASSERT(mState != STATE_ACTIVE);
Eric Laurentc2f1f072009-07-17 12:17:14 -07001117
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001118 // clear playback marker and periodic update counter
1119 mMarkerPosition = 0;
1120 mMarkerReached = false;
1121 mUpdatePeriod = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001122 mRefreshRemaining = true;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001123
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001124 mState = STATE_FLUSHED;
Andy Hungc2813e52014-10-16 17:54:34 -07001125 mReleased = 0;
Glenn Kasten23a75452014-01-13 10:37:17 -08001126 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001127 mProxy->interrupt();
1128 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001129 mProxy->flush();
Glenn Kasten4bae3642012-11-30 13:41:12 -08001130 mAudioTrack->flush();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001131}
1132
Andy Hung959b5b82021-09-24 10:46:20 -07001133bool AudioTrack::pauseAndWait(const std::chrono::milliseconds& timeout)
1134{
1135 using namespace std::chrono_literals;
1136
Andy Hungd87a53a2022-01-19 16:56:17 -08001137 // We use atomic access here for state variables - these are used as hints
1138 // to ensure we have ramped down audio.
1139 const int priorState = mProxy->getState();
1140 const uint32_t priorPosition = mProxy->getPosition().unsignedValue();
1141
Andy Hung959b5b82021-09-24 10:46:20 -07001142 pause();
1143
Andy Hungd87a53a2022-01-19 16:56:17 -08001144 // Only if we were previously active, do we wait to ramp down the audio.
1145 if (priorState != CBLK_STATE_ACTIVE) return true;
1146
Andy Hung959b5b82021-09-24 10:46:20 -07001147 AutoMutex lock(mLock);
1148 // offload and direct tracks do not wait because pause volume ramp is handled by hardware.
1149 if (isOffloadedOrDirect_l()) return true;
1150
1151 // Wait for the track state to be anything besides pausing.
1152 // This ensures that the volume has ramped down.
1153 constexpr auto SLEEP_INTERVAL_MS = 10ms;
Andy Hungd87a53a2022-01-19 16:56:17 -08001154 constexpr auto POSITION_TIMEOUT_MS = 40ms; // don't wait longer than this for position change.
Andy Hung959b5b82021-09-24 10:46:20 -07001155 auto begin = std::chrono::steady_clock::now();
1156 while (true) {
Andy Hungd87a53a2022-01-19 16:56:17 -08001157 // Wait for state and position to change.
1158 // After pause() the server state should be PAUSING, but that may immediately
1159 // convert to PAUSED by prepareTracks before data is read into the mixer.
1160 // Hence we check that the state is not PAUSING and that the server position
1161 // has advanced to be a more reliable estimate that the volume ramp has completed.
Andy Hung959b5b82021-09-24 10:46:20 -07001162 const int state = mProxy->getState();
Andy Hungd87a53a2022-01-19 16:56:17 -08001163 const uint32_t position = mProxy->getPosition().unsignedValue();
Andy Hung959b5b82021-09-24 10:46:20 -07001164
1165 mLock.unlock(); // only local variables accessed until lock.
1166 auto elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(
1167 std::chrono::steady_clock::now() - begin);
Andy Hungd87a53a2022-01-19 16:56:17 -08001168 if (state != CBLK_STATE_PAUSING &&
1169 (elapsed >= POSITION_TIMEOUT_MS || position != priorPosition)) {
1170 ALOGV("%s: success state:%d, position:%u after %lld ms"
1171 " (prior state:%d prior position:%u)",
1172 __func__, state, position, elapsed.count(), priorState, priorPosition);
Andy Hung959b5b82021-09-24 10:46:20 -07001173 return true;
1174 }
1175 std::chrono::milliseconds remaining = timeout - elapsed;
1176 if (remaining.count() <= 0) {
1177 ALOGW("%s: timeout expired state:%d still pausing:%d after %lld ms",
1178 __func__, state, CBLK_STATE_PAUSING, elapsed.count());
1179 return false;
1180 }
1181 // It is conceivable that the track is restored while sleeping;
1182 // as this logic is advisory, we allow that.
1183 std::this_thread::sleep_for(std::min(remaining, SLEEP_INTERVAL_MS));
1184 mLock.lock();
1185 }
1186}
1187
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001188void AudioTrack::pause()
1189{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001190 const int64_t beginNs = systemTime();
Eric Laurentf5aafb22010-11-18 08:40:16 -08001191 AutoMutex lock(mLock);
Andy Hunga6b27032020-04-27 10:34:24 -07001192 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001193 mediametrics::LogItem(mMetricsId)
1194 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_PAUSE)
Andy Hungea840382020-05-05 21:50:17 -07001195 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001196 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
1197 .record(); });
1198
Eric Laurent973db022018-11-20 14:54:31 -08001199 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001200
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001201 if (mState == STATE_ACTIVE) {
1202 mState = STATE_PAUSED;
1203 } else if (mState == STATE_STOPPING) {
1204 mState = STATE_PAUSED_STOPPING;
1205 } else {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001206 return;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001207 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001208 mProxy->interrupt();
1209 mAudioTrack->pause();
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001210
Marco Nelissen3a90f282014-03-10 11:21:43 -07001211 if (isOffloaded_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001212 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001213 // An offload output can be re-used between two audio tracks having
1214 // the same configuration. A timestamp query for a paused track
1215 // while the other is running would return an incorrect time.
1216 // To fix this, cache the playback position on a pause() and return
1217 // this time when requested until the track is resumed.
1218
1219 // OffloadThread sends HAL pause in its threadLoop. Time saved
1220 // here can be slightly off.
1221
1222 // TODO: check return code for getRenderPosition.
1223
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001224 uint32_t halFrames;
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001225 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition);
Andy Hungfb8ede22018-09-12 19:03:24 -07001226 ALOGV("%s(%d): for offload, cache current position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001227 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001228 }
1229 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001230}
1231
Eric Laurentbe916aa2010-06-01 23:49:17 -07001232status_t AudioTrack::setVolume(float left, float right)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001233{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001234 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1235 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY ||
1236 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001237 return BAD_VALUE;
1238 }
1239
Andy Hungb68f5eb2019-12-03 16:49:17 -08001240 mediametrics::LogItem(mMetricsId)
1241 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOLUME)
1242 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)left)
1243 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)right)
1244 .record();
1245
Eric Laurent1703cdf2011-03-07 14:52:59 -08001246 AutoMutex lock(mLock);
Glenn Kasten66e46352014-01-16 17:44:23 -08001247 mVolume[AUDIO_INTERLEAVE_LEFT] = left;
1248 mVolume[AUDIO_INTERLEAVE_RIGHT] = right;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001249
Glenn Kastenc56f3422014-03-21 17:53:17 -07001250 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right)));
Eric Laurentbe916aa2010-06-01 23:49:17 -07001251
Glenn Kasten23a75452014-01-13 10:37:17 -08001252 if (isOffloaded_l()) {
Eric Laurent59fe0102013-09-27 18:48:26 -07001253 mAudioTrack->signal();
1254 }
Eric Laurentbe916aa2010-06-01 23:49:17 -07001255 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001256}
1257
Glenn Kastenb1c09932012-02-27 16:21:04 -08001258status_t AudioTrack::setVolume(float volume)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001259{
Glenn Kastenb1c09932012-02-27 16:21:04 -08001260 return setVolume(volume, volume);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001261}
1262
Eric Laurent2beeb502010-07-16 07:43:46 -07001263status_t AudioTrack::setAuxEffectSendLevel(float level)
Eric Laurentbe916aa2010-06-01 23:49:17 -07001264{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001265 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1266 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001267 return BAD_VALUE;
1268 }
1269
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001270 AutoMutex lock(mLock);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001271 mSendLevel = level;
Glenn Kastene3aa6592012-12-04 12:22:46 -08001272 mProxy->setSendLevel(level);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001273
1274 return NO_ERROR;
1275}
1276
Glenn Kastena5224f32012-01-04 12:41:44 -08001277void AudioTrack::getAuxEffectSendLevel(float* level) const
Eric Laurentbe916aa2010-06-01 23:49:17 -07001278{
1279 if (level != NULL) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001280 *level = mSendLevel;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001281 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001282}
1283
Glenn Kasten3b16c762012-11-14 08:44:39 -08001284status_t AudioTrack::setSampleRate(uint32_t rate)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001285{
Andy Hung5cbb5782015-03-27 18:39:59 -07001286 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08001287 ALOGV("%s(%d): prior state:%s rate:%u", __func__, mPortId, stateToString(mState), rate);
Andy Hungfb8ede22018-09-12 19:03:24 -07001288
Andy Hung5cbb5782015-03-27 18:39:59 -07001289 if (rate == mSampleRate) {
1290 return NO_ERROR;
1291 }
jiabinf4de6112018-12-19 12:40:08 -08001292 if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)
1293 || (mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL)) {
John Grossman4ff14ba2012-02-08 16:37:41 -08001294 return INVALID_OPERATION;
1295 }
Eric Laurent0d6db582014-11-12 18:39:44 -08001296 if (mOutput == AUDIO_IO_HANDLE_NONE) {
1297 return NO_INIT;
1298 }
Andy Hung5cbb5782015-03-27 18:39:59 -07001299 // NOTE: it is theoretically possible, but highly unlikely, that a device change
1300 // could mean a previously allowed sampling rate is no longer allowed.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001301 uint32_t afSamplingRate;
Eric Laurent0d6db582014-11-12 18:39:44 -08001302 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
Eric Laurent57326622009-07-07 07:10:45 -07001303 return NO_INIT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001304 }
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 effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001307 if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001308 return BAD_VALUE;
1309 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001310 // TODO: Should we also check if the buffer size is compatible?
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001311
Glenn Kastene3aa6592012-12-04 12:22:46 -08001312 mSampleRate = rate;
Andy Hung26145642015-04-15 21:56:53 -07001313 mProxy->setSampleRate(effectiveSampleRate);
Glenn Kastene3aa6592012-12-04 12:22:46 -08001314
Eric Laurent57326622009-07-07 07:10:45 -07001315 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001316}
1317
Glenn Kastena5224f32012-01-04 12:41:44 -08001318uint32_t AudioTrack::getSampleRate() const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001319{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001320 AutoMutex lock(mLock);
Eric Laurent6f59db12013-07-26 17:16:50 -07001321
1322 // sample rate can be updated during playback by the offloaded decoder so we need to
1323 // query the HAL and update if needed.
1324// FIXME use Proxy return channel to update the rate from server and avoid polling here
Eric Laurentab5cdba2014-06-09 17:22:27 -07001325 if (isOffloadedOrDirect_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001326 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent6f59db12013-07-26 17:16:50 -07001327 uint32_t sampleRate = 0;
Jean-Michel Trivib7f24b12014-06-11 10:05:30 -07001328 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
Eric Laurent6f59db12013-07-26 17:16:50 -07001329 if (status == NO_ERROR) {
1330 mSampleRate = sampleRate;
1331 }
1332 }
1333 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001334 return mSampleRate;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001335}
1336
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001337uint32_t AudioTrack::getOriginalSampleRate() const
1338{
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001339 return mOriginalSampleRate;
1340}
1341
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001342status_t AudioTrack::setDualMonoMode(audio_dual_mono_mode_t mode)
1343{
1344 AutoMutex lock(mLock);
1345 return setDualMonoMode_l(mode);
1346}
1347
1348status_t AudioTrack::setDualMonoMode_l(audio_dual_mono_mode_t mode)
1349{
1350 const status_t status = statusTFromBinderStatus(
1351 mAudioTrack->setDualMonoMode(VALUE_OR_RETURN_STATUS(
1352 legacy2aidl_audio_dual_mono_mode_t_AudioDualMonoMode(mode))));
1353 if (status == NO_ERROR) mDualMonoMode = mode;
1354 return status;
1355}
1356
1357status_t AudioTrack::getDualMonoMode(audio_dual_mono_mode_t* mode) const
1358{
1359 AutoMutex lock(mLock);
1360 media::AudioDualMonoMode mediaMode;
1361 const status_t status = statusTFromBinderStatus(mAudioTrack->getDualMonoMode(&mediaMode));
1362 if (status == NO_ERROR) {
1363 *mode = VALUE_OR_RETURN_STATUS(
1364 aidl2legacy_AudioDualMonoMode_audio_dual_mono_mode_t(mediaMode));
1365 }
1366 return status;
1367}
1368
1369status_t AudioTrack::setAudioDescriptionMixLevel(float leveldB)
1370{
1371 AutoMutex lock(mLock);
1372 return setAudioDescriptionMixLevel_l(leveldB);
1373}
1374
1375status_t AudioTrack::setAudioDescriptionMixLevel_l(float leveldB)
1376{
1377 const status_t status = statusTFromBinderStatus(
1378 mAudioTrack->setAudioDescriptionMixLevel(leveldB));
1379 if (status == NO_ERROR) mAudioDescriptionMixLeveldB = leveldB;
1380 return status;
1381}
1382
1383status_t AudioTrack::getAudioDescriptionMixLevel(float* leveldB) const
1384{
1385 AutoMutex lock(mLock);
1386 return statusTFromBinderStatus(mAudioTrack->getAudioDescriptionMixLevel(leveldB));
1387}
1388
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001389status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
Andy Hung8edb8dc2015-03-26 19:13:55 -07001390{
Andy Hung8edb8dc2015-03-26 19:13:55 -07001391 AutoMutex lock(mLock);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001392 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
Andy Hung8edb8dc2015-03-26 19:13:55 -07001393 return NO_ERROR;
1394 }
Glenn Kastend79072e2016-01-06 08:41:20 -08001395 if (isOffloadedOrDirect_l()) {
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001396 const status_t status = statusTFromBinderStatus(mAudioTrack->setPlaybackRateParameters(
1397 VALUE_OR_RETURN_STATUS(
1398 legacy2aidl_audio_playback_rate_t_AudioPlaybackRate(playbackRate))));
1399 if (status == NO_ERROR) {
1400 mPlaybackRate = playbackRate;
1401 }
1402 return status;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001403 }
1404 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
1405 return INVALID_OPERATION;
1406 }
Andy Hungff874dc2016-04-11 16:49:09 -07001407
Andy Hungfb8ede22018-09-12 19:03:24 -07001408 ALOGV("%s(%d): mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001409 __func__, mPortId, mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001410 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001411 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
1412 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
1413 const float effectivePitch = adjustPitch(playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001414 AudioPlaybackRate playbackRateTemp = playbackRate;
1415 playbackRateTemp.mSpeed = effectiveSpeed;
1416 playbackRateTemp.mPitch = effectivePitch;
1417
Andy Hungfb8ede22018-09-12 19:03:24 -07001418 ALOGV("%s(%d) (effective) mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001419 __func__, mPortId, effectiveRate, effectiveSpeed, effectivePitch);
Andy Hungff874dc2016-04-11 16:49:09 -07001420
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001421 if (!isAudioPlaybackRateValid(playbackRateTemp)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001422 ALOGW("%s(%d) (%f, %f) failed (effective rate out of bounds)",
Eric Laurent973db022018-11-20 14:54:31 -08001423 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001424 return BAD_VALUE;
1425 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001426 // Check if the buffer size is compatible.
Andy Hung26145642015-04-15 21:56:53 -07001427 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001428 ALOGW("%s(%d) (%f, %f) failed (buffer size)",
Eric Laurent973db022018-11-20 14:54:31 -08001429 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung8edb8dc2015-03-26 19:13:55 -07001430 return BAD_VALUE;
1431 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001432
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001433 // Check resampler ratios are within bounds
Glenn Kastend3bb6452016-12-05 18:14:37 -08001434 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
1435 (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001436 ALOGW("%s(%d) (%f, %f) failed. Resample rate exceeds max accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001437 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001438 return BAD_VALUE;
1439 }
1440
Dan Austine34eae22015-10-27 16:14:52 -07001441 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001442 ALOGW("%s(%d) (%f, %f) failed. Resample rate below min accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001443 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001444 return BAD_VALUE;
1445 }
1446 mPlaybackRate = playbackRate;
1447 //set effective rates
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001448 mProxy->setPlaybackRate(playbackRateTemp);
Andy Hung26145642015-04-15 21:56:53 -07001449 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
Andy Hungb68f5eb2019-12-03 16:49:17 -08001450
1451 mediametrics::LogItem(mMetricsId)
1452 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYBACKPARAM)
1453 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
1454 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
1455 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
1456 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1457 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveRate)
1458 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1459 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)playbackRateTemp.mSpeed)
1460 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1461 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)playbackRateTemp.mPitch)
1462 .record();
1463
Andy Hung8edb8dc2015-03-26 19:13:55 -07001464 return NO_ERROR;
1465}
1466
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001467const AudioPlaybackRate& AudioTrack::getPlaybackRate()
Andy Hung8edb8dc2015-03-26 19:13:55 -07001468{
1469 AutoMutex lock(mLock);
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001470 if (isOffloadedOrDirect_l()) {
1471 media::AudioPlaybackRate playbackRateTemp;
1472 const status_t status = statusTFromBinderStatus(
1473 mAudioTrack->getPlaybackRateParameters(&playbackRateTemp));
1474 if (status == NO_ERROR) { // update local version if changed.
1475 mPlaybackRate =
1476 aidl2legacy_AudioPlaybackRate_audio_playback_rate_t(playbackRateTemp).value();
1477 }
1478 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001479 return mPlaybackRate;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001480}
1481
Phil Burkc0adecb2016-01-08 12:44:11 -08001482ssize_t AudioTrack::getBufferSizeInFrames()
1483{
1484 AutoMutex lock(mLock);
1485 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1486 return NO_INIT;
1487 }
Phil Burka9876702020-04-20 18:16:15 -07001488
Phil Burke8972b02016-03-04 11:29:57 -08001489 return (ssize_t) mProxy->getBufferSizeInFrames();
Phil Burkc0adecb2016-01-08 12:44:11 -08001490}
1491
Andy Hungf2c87b32016-04-07 19:49:29 -07001492status_t AudioTrack::getBufferDurationInUs(int64_t *duration)
1493{
1494 if (duration == nullptr) {
1495 return BAD_VALUE;
1496 }
1497 AutoMutex lock(mLock);
1498 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1499 return NO_INIT;
1500 }
1501 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames();
1502 if (bufferSizeInFrames < 0) {
1503 return (status_t)bufferSizeInFrames;
1504 }
1505 *duration = (int64_t)((double)bufferSizeInFrames * 1000000
1506 / ((double)mSampleRate * mPlaybackRate.mSpeed));
1507 return NO_ERROR;
1508}
1509
Phil Burkc0adecb2016-01-08 12:44:11 -08001510ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames)
1511{
1512 AutoMutex lock(mLock);
1513 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1514 return NO_INIT;
1515 }
Phil Burka9876702020-04-20 18:16:15 -07001516
1517 ssize_t originalBufferSize = mProxy->getBufferSizeInFrames();
1518 ssize_t finalBufferSize = mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
1519 if (originalBufferSize != finalBufferSize) {
Phil Burk64e16a72020-06-01 13:25:51 -07001520 android::mediametrics::LogItem(mMetricsId)
1521 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
1522 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1523 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t)getUnderrunCount_l())
1524 .record();
Phil Burka9876702020-04-20 18:16:15 -07001525 }
1526 return finalBufferSize;
Phil Burkc0adecb2016-01-08 12:44:11 -08001527}
1528
Andy Hung3c7f47a2021-03-16 17:30:09 -07001529ssize_t AudioTrack::getStartThresholdInFrames() const
1530{
1531 AutoMutex lock(mLock);
1532 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1533 return NO_INIT;
1534 }
1535 return (ssize_t) mProxy->getStartThresholdInFrames();
1536}
1537
1538ssize_t AudioTrack::setStartThresholdInFrames(size_t startThresholdInFrames)
1539{
1540 if (startThresholdInFrames > INT32_MAX || startThresholdInFrames == 0) {
1541 // contractually we could simply return the current threshold in frames
1542 // to indicate the request was ignored, but we return an error here.
1543 return BAD_VALUE;
1544 }
1545 AutoMutex lock(mLock);
1546 // We do not permit calling setStartThresholdInFrames() between the AudioTrack
1547 // default ctor AudioTrack() and set(...) but rather fail such an attempt.
1548 // (To do so would require a cached mOrigStartThresholdInFrames and we may
1549 // not have proper validation for the actual set value).
1550 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1551 return NO_INIT;
1552 }
1553 const uint32_t original = mProxy->getStartThresholdInFrames();
1554 const uint32_t final = mProxy->setStartThresholdInFrames(startThresholdInFrames);
1555 if (original != final) {
1556 android::mediametrics::LogItem(mMetricsId)
1557 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD)
1558 .set(AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES, (int32_t)final)
1559 .record();
1560 if (original > final) {
1561 // restart track if it was disabled by audioflinger due to previous underrun
1562 // and we reduced the number of frames for the threshold.
1563 restartIfDisabled();
1564 }
1565 }
1566 return final;
1567}
1568
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001569status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1570{
Glenn Kastend79072e2016-01-06 08:41:20 -08001571 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001572 return INVALID_OPERATION;
1573 }
1574
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001575 if (loopCount == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001576 ;
1577 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount &&
1578 loopEnd - loopStart >= MIN_LOOP) {
1579 ;
1580 } else {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001581 return BAD_VALUE;
1582 }
1583
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001584 AutoMutex lock(mLock);
1585 // See setPosition() regarding setting parameters such as loop points or position while active
1586 if (mState == STATE_ACTIVE) {
1587 return INVALID_OPERATION;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001588 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001589 setLoop_l(loopStart, loopEnd, loopCount);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001590 return NO_ERROR;
1591}
1592
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001593void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1594{
Andy Hung4ede21d2014-12-12 15:37:34 -08001595 // We do not update the periodic notification point.
1596 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
1597 mLoopCount = loopCount;
1598 mLoopEnd = loopEnd;
1599 mLoopStart = loopStart;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001600 mLoopCountNotified = loopCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001601 mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
Andy Hung3c09c782014-12-29 18:39:32 -08001602
1603 // Waking the AudioTrackThread is not needed as this cannot be called when active.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001604}
1605
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001606status_t AudioTrack::setMarkerPosition(uint32_t marker)
1607{
Atneya Nair14aabae2021-11-30 17:36:24 -05001608 AutoMutex lock(mLock);
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001609 // The only purpose of setting marker position is to get a callback
Atneya Nair14aabae2021-11-30 17:36:24 -05001610 if (!mCallback.promote() || isOffloadedOrDirect_l()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001611 return INVALID_OPERATION;
1612 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001613
1614 mMarkerPosition = marker;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001615 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001616
Andy Hung3c09c782014-12-29 18:39:32 -08001617 sp<AudioTrackThread> t = mAudioTrackThread;
1618 if (t != 0) {
1619 t->wake();
1620 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001621 return NO_ERROR;
1622}
1623
Glenn Kastena5224f32012-01-04 12:41:44 -08001624status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001625{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001626 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001627 return INVALID_OPERATION;
1628 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001629 if (marker == NULL) {
1630 return BAD_VALUE;
1631 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001632
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001633 AutoMutex lock(mLock);
Andy Hung90e8a972015-11-09 16:42:40 -08001634 mMarkerPosition.getValue(marker);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001635
1636 return NO_ERROR;
1637}
1638
1639status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
1640{
Atneya Nair14aabae2021-11-30 17:36:24 -05001641 AutoMutex lock(mLock);
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001642 // The only purpose of setting position update period is to get a callback
Atneya Nair14aabae2021-11-30 17:36:24 -05001643 if (!mCallback.promote() || isOffloadedOrDirect_l()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001644 return INVALID_OPERATION;
1645 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001646
Glenn Kasten200092b2014-08-15 15:13:30 -07001647 mNewPosition = updateAndGetPosition_l() + updatePeriod;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001648 mUpdatePeriod = updatePeriod;
Glenn Kasten2b2165c2014-01-13 08:53:36 -08001649
Andy Hung3c09c782014-12-29 18:39:32 -08001650 sp<AudioTrackThread> t = mAudioTrackThread;
1651 if (t != 0) {
1652 t->wake();
1653 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001654 return NO_ERROR;
1655}
1656
Glenn Kastena5224f32012-01-04 12:41:44 -08001657status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001658{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001659 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001660 return INVALID_OPERATION;
1661 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001662 if (updatePeriod == NULL) {
1663 return BAD_VALUE;
1664 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001665
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001666 AutoMutex lock(mLock);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001667 *updatePeriod = mUpdatePeriod;
1668
1669 return NO_ERROR;
1670}
1671
1672status_t AudioTrack::setPosition(uint32_t position)
1673{
Glenn Kastend79072e2016-01-06 08:41:20 -08001674 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001675 return INVALID_OPERATION;
1676 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001677 if (position > mFrameCount) {
1678 return BAD_VALUE;
1679 }
John Grossman4ff14ba2012-02-08 16:37:41 -08001680
Eric Laurent1703cdf2011-03-07 14:52:59 -08001681 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001682 // Currently we require that the player is inactive before setting parameters such as position
1683 // or loop points. Otherwise, there could be a race condition: the application could read the
1684 // current position, compute a new position or loop parameters, and then set that position or
1685 // loop parameters but it would do the "wrong" thing since the position has continued to advance
1686 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app
1687 // to specify how it wants to handle such scenarios.
1688 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001689 return INVALID_OPERATION;
1690 }
Andy Hung9b461582014-12-01 17:56:29 -08001691 // After setting the position, use full update period before notification.
Glenn Kasten200092b2014-08-15 15:13:30 -07001692 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001693 mStaticProxy->setBufferPosition(position);
Andy Hung3c09c782014-12-29 18:39:32 -08001694
1695 // Waking the AudioTrackThread is not needed as this cannot be called when active.
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001696 return NO_ERROR;
1697}
1698
Glenn Kasten200092b2014-08-15 15:13:30 -07001699status_t AudioTrack::getPosition(uint32_t *position)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001700{
Glenn Kastend65d73c2012-06-22 17:21:07 -07001701 if (position == NULL) {
1702 return BAD_VALUE;
1703 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001704
Eric Laurent1703cdf2011-03-07 14:52:59 -08001705 AutoMutex lock(mLock);
Andy Hung7a490e72016-03-23 15:58:10 -07001706 // FIXME: offloaded and direct tracks call into the HAL for render positions
1707 // for compressed/synced data; however, we use proxy position for pure linear pcm data
1708 // as we do not know the capability of the HAL for pcm position support and standby.
1709 // There may be some latency differences between the HAL position and the proxy position.
1710 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001711 uint32_t dspFrames = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001712
Eric Laurentab5cdba2014-06-09 17:22:27 -07001713 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001714 ALOGV("%s(%d): called in paused state, return cached position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001715 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001716 *position = mPausedPosition;
1717 return NO_ERROR;
1718 }
1719
Glenn Kasten142f5192014-03-25 17:44:59 -07001720 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung1f1db832015-06-08 13:26:10 -07001721 uint32_t halFrames; // actually unused
1722 (void) AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames);
1723 // FIXME: on getRenderPosition() error, we return OK with frame position 0.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001724 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001725 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED)
1726 // due to hardware latency. We leave this behavior for now.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001727 *position = dspFrames;
1728 } else {
Eric Laurent275e8e92014-11-30 15:14:47 -08001729 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung1f1db832015-06-08 13:26:10 -07001730 (void) restoreTrack_l("getPosition");
1731 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l()
1732 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position.
Eric Laurent275e8e92014-11-30 15:14:47 -08001733 }
1734
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001735 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
Glenn Kasten200092b2014-08-15 15:13:30 -07001736 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ?
Andy Hung90e8a972015-11-09 16:42:40 -08001737 0 : updateAndGetPosition_l().value();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001738 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001739 return NO_ERROR;
1740}
1741
Kévin PETIT377b2ec2014-02-03 12:35:36 +00001742status_t AudioTrack::getBufferPosition(uint32_t *position)
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001743{
Glenn Kastend79072e2016-01-06 08:41:20 -08001744 if (mSharedBuffer == 0) {
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001745 return INVALID_OPERATION;
1746 }
1747 if (position == NULL) {
1748 return BAD_VALUE;
1749 }
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001750
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001751 AutoMutex lock(mLock);
1752 *position = mStaticProxy->getBufferPosition();
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001753 return NO_ERROR;
1754}
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001755
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001756status_t AudioTrack::reload()
1757{
Glenn Kastend79072e2016-01-06 08:41:20 -08001758 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001759 return INVALID_OPERATION;
1760 }
1761
Eric Laurent1703cdf2011-03-07 14:52:59 -08001762 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001763 // See setPosition() regarding setting parameters such as loop points or position while active
1764 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001765 return INVALID_OPERATION;
1766 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001767 mNewPosition = mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001768 (void) updateAndGetPosition_l();
1769 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07001770 mPreviousTimestampValid = false;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001771#if 0
Andy Hung9b461582014-12-01 17:56:29 -08001772 // The documentation is not clear on the behavior of reload() and the restoration
Andy Hung53c3b5f2014-12-15 16:42:05 -08001773 // of loop count. Historically we have not restored loop count, start, end,
1774 // but it makes sense if one desires to repeat playing a particular sound.
1775 if (mLoopCount != 0) {
1776 mLoopCountNotified = mLoopCount;
1777 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
1778 }
1779#endif
Andy Hung9b461582014-12-01 17:56:29 -08001780 mStaticProxy->setBufferPosition(0);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001781 return NO_ERROR;
1782}
1783
Glenn Kasten38e905b2014-01-13 10:21:48 -08001784audio_io_handle_t AudioTrack::getOutput() const
Eric Laurentc2f1f072009-07-17 12:17:14 -07001785{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001786 AutoMutex lock(mLock);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001787 return mOutput;
Eric Laurent1703cdf2011-03-07 14:52:59 -08001788}
1789
Paul McLeanaa981192015-03-21 09:55:15 -07001790status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) {
1791 AutoMutex lock(mLock);
Eric Laurent2f2c1982021-06-02 14:03:11 +02001792 ALOGV("%s(%d): deviceId=%d mSelectedDeviceId=%d",
1793 __func__, mPortId, deviceId, mSelectedDeviceId);
Paul McLeanaa981192015-03-21 09:55:15 -07001794 if (mSelectedDeviceId != deviceId) {
1795 mSelectedDeviceId = deviceId;
Eric Laurentfb00fc72017-05-25 18:17:12 -07001796 if (mStatus == NO_ERROR) {
1797 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
jiabin156c6872017-10-06 09:47:15 -07001798 mProxy->interrupt();
Eric Laurentfb00fc72017-05-25 18:17:12 -07001799 }
Paul McLeanaa981192015-03-21 09:55:15 -07001800 }
Eric Laurent493404d2015-04-21 15:07:36 -07001801 return NO_ERROR;
Paul McLeanaa981192015-03-21 09:55:15 -07001802}
1803
1804audio_port_handle_t AudioTrack::getOutputDevice() {
1805 AutoMutex lock(mLock);
1806 return mSelectedDeviceId;
1807}
1808
Eric Laurentad2e7b92017-09-14 20:06:42 -07001809// must be called with mLock held
1810void AudioTrack::updateRoutedDeviceId_l()
1811{
1812 // if the track is inactive, do not update actual device as the output stream maybe routed
1813 // to a device not relevant to this client because of other active use cases.
1814 if (mState != STATE_ACTIVE) {
1815 return;
1816 }
1817 if (mOutput != AUDIO_IO_HANDLE_NONE) {
1818 audio_port_handle_t deviceId = AudioSystem::getDeviceIdForIo(mOutput);
1819 if (deviceId != AUDIO_PORT_HANDLE_NONE) {
1820 mRoutedDeviceId = deviceId;
1821 }
1822 }
1823}
1824
Eric Laurent296fb132015-05-01 11:38:42 -07001825audio_port_handle_t AudioTrack::getRoutedDeviceId() {
1826 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001827 updateRoutedDeviceId_l();
1828 return mRoutedDeviceId;
Eric Laurent296fb132015-05-01 11:38:42 -07001829}
1830
Eric Laurentbe916aa2010-06-01 23:49:17 -07001831status_t AudioTrack::attachAuxEffect(int effectId)
1832{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001833 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001834 status_t status;
1835 mAudioTrack->attachAuxEffect(effectId, &status);
Eric Laurent2beeb502010-07-16 07:43:46 -07001836 if (status == NO_ERROR) {
1837 mAuxEffectId = effectId;
1838 }
1839 return status;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001840}
1841
Eric Laurente83b55d2014-11-14 10:06:21 -08001842audio_stream_type_t AudioTrack::streamType() const
1843{
Eric Laurente83b55d2014-11-14 10:06:21 -08001844 return mStreamType;
1845}
1846
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001847uint32_t AudioTrack::latency()
1848{
1849 AutoMutex lock(mLock);
1850 updateLatency_l();
1851 return mLatency;
1852}
1853
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001854// -------------------------------------------------------------------------
1855
Eric Laurent1703cdf2011-03-07 14:52:59 -08001856// must be called with mLock held
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001857void AudioTrack::updateLatency_l()
1858{
1859 status_t status = AudioSystem::getLatency(mOutput, &mAfLatency);
1860 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08001861 ALOGW("%s(%d): getLatency(%d) failed status %d", __func__, mPortId, mOutput, status);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001862 } else {
1863 // FIXME don't believe this lie
Andy Hung13969262017-09-11 17:24:21 -07001864 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001865 }
1866}
1867
Phil Burkadbb75a2017-06-16 12:19:42 -07001868// TODO Move this macro to a common header file for enum to string conversion in audio framework.
1869#define MEDIA_CASE_ENUM(name) case name: return #name
1870const char * AudioTrack::convertTransferToText(transfer_type transferType) {
1871 switch (transferType) {
1872 MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
1873 MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
1874 MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
1875 MEDIA_CASE_ENUM(TRANSFER_SYNC);
1876 MEDIA_CASE_ENUM(TRANSFER_SHARED);
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001877 MEDIA_CASE_ENUM(TRANSFER_SYNC_NOTIF_CALLBACK);
Phil Burkadbb75a2017-06-16 12:19:42 -07001878 default:
1879 return "UNRECOGNIZED";
1880 }
1881}
1882
Glenn Kasten200092b2014-08-15 15:13:30 -07001883status_t AudioTrack::createTrack_l()
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001884{
Eric Laurentf32d7812017-11-30 14:44:07 -08001885 status_t status;
1886 bool callbackAdded = false;
Andy Hung3acde2c2021-11-11 09:18:08 -08001887 std::string errorMessage;
Eric Laurentf32d7812017-11-30 14:44:07 -08001888
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001889 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
1890 if (audioFlinger == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001891 errorMessage = StringPrintf("%s(%d): Could not get audioflinger",
Eric Laurent973db022018-11-20 14:54:31 -08001892 __func__, mPortId);
Andy Hung3acde2c2021-11-11 09:18:08 -08001893 status = DEAD_OBJECT;
Eric Laurentf32d7812017-11-30 14:44:07 -08001894 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001895 }
1896
Eric Laurent21da6472017-11-09 16:29:26 -08001897 {
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08001898 // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
1899 // After fast request is denied, we will request again if IAudioTrack is re-created.
Glenn Kastend79072e2016-01-06 08:41:20 -08001900 // Client can only express a preference for FAST. Server will perform additional tests.
Phil Burk33ff89b2015-11-30 11:16:01 -08001901 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Phil Burkadbb75a2017-06-16 12:19:42 -07001902 // either of these use cases:
1903 // use case 1: shared buffer
1904 bool sharedBuffer = mSharedBuffer != 0;
1905 bool transferAllowed =
Glenn Kastenc6ba8232014-02-27 13:34:29 -08001906 // use case 2: callback transfer mode
Glenn Kasten1dfe2f92015-03-09 12:03:14 -07001907 (mTransfer == TRANSFER_CALLBACK) ||
1908 // use case 3: obtain/release mode
Phil Burk33ff89b2015-11-30 11:16:01 -08001909 (mTransfer == TRANSFER_OBTAIN) ||
1910 // use case 4: synchronous write
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001911 ((mTransfer == TRANSFER_SYNC || mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK)
1912 && mThreadCanCallJava);
Phil Burkadbb75a2017-06-16 12:19:42 -07001913
Eric Laurent21da6472017-11-09 16:29:26 -08001914 bool fastAllowed = sharedBuffer || transferAllowed;
1915 if (!fastAllowed) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001916 ALOGW("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by client,"
1917 " not shared buffer and transfer = %s",
Eric Laurent973db022018-11-20 14:54:31 -08001918 __func__, mPortId,
Phil Burkadbb75a2017-06-16 12:19:42 -07001919 convertTransferToText(mTransfer));
Phil Burk33ff89b2015-11-30 11:16:01 -08001920 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
1921 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001922 }
1923
Eric Laurent21da6472017-11-09 16:29:26 -08001924 IAudioFlinger::CreateTrackInput input;
Andy Hunga2159aa2021-07-20 13:01:52 -07001925 if (mOriginalStreamType != AUDIO_STREAM_DEFAULT) {
1926 // Legacy: This is based on original parameters even if the track is recreated.
1927 input.attr = AudioSystem::streamTypeToAttributes(mOriginalStreamType);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001928 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08001929 input.attr = mAttributes;
Eric Laurentd1b449a2010-05-14 03:26:45 -07001930 }
Eric Laurent21da6472017-11-09 16:29:26 -08001931 input.config = AUDIO_CONFIG_INITIALIZER;
1932 input.config.sample_rate = mSampleRate;
1933 input.config.channel_mask = mChannelMask;
1934 input.config.format = mFormat;
1935 input.config.offload_info = mOffloadInfoCopy;
Svet Ganov3e5f14f2021-05-13 22:51:08 +00001936 input.clientInfo.attributionSource = mClientAttributionSource;
Eric Laurent21da6472017-11-09 16:29:26 -08001937 input.clientInfo.clientTid = -1;
Glenn Kasten363fb752014-01-15 12:27:31 -08001938 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Glenn Kasten47d55172017-05-23 11:19:30 -07001939 // It is currently meaningless to request SCHED_FIFO for a Java thread. Even if the
1940 // application-level code follows all non-blocking design rules, the language runtime
1941 // doesn't also follow those rules, so the thread will not benefit overall.
Phil Burk33ff89b2015-11-30 11:16:01 -08001942 if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
Eric Laurent21da6472017-11-09 16:29:26 -08001943 input.clientInfo.clientTid = mAudioTrackThread->getTid();
Glenn Kasten3acbd052012-02-28 10:39:56 -08001944 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001945 }
Eric Laurent21da6472017-11-09 16:29:26 -08001946 input.sharedBuffer = mSharedBuffer;
1947 input.notificationsPerBuffer = mNotificationsPerBufferReq;
1948 input.speed = 1.0;
1949 if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
1950 (mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
1951 input.speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
1952 max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
1953 }
1954 input.flags = mFlags;
1955 input.frameCount = mReqFrameCount;
1956 input.notificationFrameCount = mNotificationFramesReq;
1957 input.selectedDeviceId = mSelectedDeviceId;
1958 input.sessionId = mSessionId;
jiabinf6eb4c32020-02-25 14:06:25 -08001959 input.audioTrackCallback = mAudioTrackCallback;
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001960
Ytai Ben-Tsvi4dfeb622020-11-02 12:47:30 -08001961 media::CreateTrackResponse response;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001962 status = audioFlinger->createTrack(VALUE_OR_FATAL(input.toAidl()), response);
Ytai Ben-Tsvi357e26a2021-01-05 13:21:19 -08001963
1964 IAudioFlinger::CreateTrackOutput output{};
1965 if (status == NO_ERROR) {
1966 output = VALUE_OR_FATAL(IAudioFlinger::CreateTrackOutput::fromAidl(response));
1967 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001968
Eric Laurent21da6472017-11-09 16:29:26 -08001969 if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001970 errorMessage = StringPrintf(
1971 "%s(%d): AudioFlinger could not create track, status: %d output %d",
Eric Laurent973db022018-11-20 14:54:31 -08001972 __func__, mPortId, status, output.outputId);
Eric Laurentf32d7812017-11-30 14:44:07 -08001973 if (status == NO_ERROR) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001974 status = INVALID_OPERATION; // device not ready
Eric Laurentf32d7812017-11-30 14:44:07 -08001975 }
1976 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001977 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001978 ALOG_ASSERT(output.audioTrack != 0);
Glenn Kastenc08d20b2014-02-24 15:21:10 -08001979
Eric Laurent21da6472017-11-09 16:29:26 -08001980 mFrameCount = output.frameCount;
1981 mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
1982 mRoutedDeviceId = output.selectedDeviceId;
1983 mSessionId = output.sessionId;
Andy Hunga2159aa2021-07-20 13:01:52 -07001984 mStreamType = output.streamType;
Eric Laurent21da6472017-11-09 16:29:26 -08001985
1986 mSampleRate = output.sampleRate;
1987 if (mOriginalSampleRate == 0) {
1988 mOriginalSampleRate = mSampleRate;
1989 }
1990
1991 mAfFrameCount = output.afFrameCount;
1992 mAfSampleRate = output.afSampleRate;
1993 mAfLatency = output.afLatencyMs;
1994
1995 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
1996
Glenn Kasten38e905b2014-01-13 10:21:48 -08001997 // AudioFlinger now owns the reference to the I/O handle,
1998 // so we are no longer responsible for releasing it.
1999
Glenn Kasten7fd04222016-02-02 12:38:16 -08002000 // FIXME compare to AudioRecord
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002001 std::optional<media::SharedFileRegion> sfr;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08002002 output.audioTrack->getCblk(&sfr);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002003 sp<IMemory> iMem = VALUE_OR_FATAL(aidl2legacy_NullableSharedFileRegion_IMemory(sfr));
Glenn Kastend2c38fc2012-11-01 14:58:02 -07002004 if (iMem == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08002005 errorMessage = StringPrintf("%s(%d): Could not get control block", __func__, mPortId);
2006 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08002007 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002008 }
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07002009 // TODO: Using unsecurePointer() has some associated security pitfalls
2010 // (see declaration for details).
2011 // Either document why it is safe in this case or address the
2012 // issue (e.g. by copying).
2013 void *iMemPointer = iMem->unsecurePointer();
Glenn Kasten0cde0762014-01-16 15:06:36 -08002014 if (iMemPointer == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08002015 errorMessage = StringPrintf(
2016 "%s(%d): Could not get control block pointer", __func__, mPortId);
2017 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08002018 goto exit;
Glenn Kasten0cde0762014-01-16 15:06:36 -08002019 }
Glenn Kasten53cec222013-08-29 09:01:02 -07002020 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002021 if (mAudioTrack != 0) {
Marco Nelissenf8880202014-11-14 07:58:25 -08002022 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002023 mDeathNotifier.clear();
2024 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08002025 mAudioTrack = output.audioTrack;
Glenn Kastend2c38fc2012-11-01 14:58:02 -07002026 mCblkMemory = iMem;
Eric Laurent3bcffa12014-06-12 18:38:45 -07002027 IPCThreadState::self()->flushCommands();
2028
Glenn Kasten0cde0762014-01-16 15:06:36 -08002029 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
Glenn Kastend2c38fc2012-11-01 14:58:02 -07002030 mCblk = cblk;
Glenn Kasten5f631512014-02-24 15:16:07 -08002031
Glenn Kastena07f17c2013-04-23 12:39:37 -07002032 mAwaitBoost = false;
Glenn Kasten363fb752014-01-15 12:27:31 -08002033 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Eric Laurent21da6472017-11-09 16:29:26 -08002034 if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002035 ALOGI("%s(%d): AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08002036 __func__, mPortId, mReqFrameCount, mFrameCount);
Phil Burk33ff89b2015-11-30 11:16:01 -08002037 if (!mThreadCanCallJava) {
2038 mAwaitBoost = true;
2039 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08002040 } else {
Robert Wuc121cd12021-08-13 17:51:40 +00002041 ALOGV("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08002042 __func__, mPortId, mReqFrameCount, mFrameCount);
Glenn Kastene0fa4672012-04-24 14:35:14 -07002043 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08002044 }
Eric Laurent21da6472017-11-09 16:29:26 -08002045 mFlags = output.flags;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002046
Eric Laurentad2e7b92017-09-14 20:06:42 -07002047 //mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
Eric Laurent09f1ed22019-04-24 17:45:17 -07002048 if (mDeviceCallback != 0) {
Eric Laurentad2e7b92017-09-14 20:06:42 -07002049 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07002050 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07002051 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07002052 AudioSystem::addAudioDeviceCallback(this, output.outputId, output.portId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07002053 callbackAdded = true;
2054 }
2055
Eric Laurent09f1ed22019-04-24 17:45:17 -07002056 mPortId = output.portId;
Glenn Kasten38e905b2014-01-13 10:21:48 -08002057 // We retain a copy of the I/O handle, but don't own the reference
Eric Laurent21da6472017-11-09 16:29:26 -08002058 mOutput = output.outputId;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002059 mRefreshRemaining = true;
2060
2061 // Starting address of buffers in shared memory. If there is a shared buffer, buffers
2062 // is the value of pointer() for the shared buffer, otherwise buffers points
2063 // immediately after the control block. This address is for the mapping within client
2064 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
2065 void* buffers;
Glenn Kasten363fb752014-01-15 12:27:31 -08002066 if (mSharedBuffer == 0) {
Glenn Kasten138d6f92015-03-20 10:54:51 -07002067 buffers = cblk + 1;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002068 } else {
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07002069 // TODO: Using unsecurePointer() has some associated security pitfalls
2070 // (see declaration for details).
2071 // Either document why it is safe in this case or address the
2072 // issue (e.g. by copying).
2073 buffers = mSharedBuffer->unsecurePointer();
Glenn Kasten138d6f92015-03-20 10:54:51 -07002074 if (buffers == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08002075 errorMessage = StringPrintf(
2076 "%s(%d): Could not get buffer pointer", __func__, mPortId);
2077 ALOGE("%s", errorMessage.c_str());
2078 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08002079 goto exit;
Glenn Kasten138d6f92015-03-20 10:54:51 -07002080 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002081 }
2082
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002083 mAudioTrack->attachAuxEffect(mAuxEffectId, &status);
Glenn Kasten5f631512014-02-24 15:16:07 -08002084
Glenn Kasten093000f2012-05-03 09:35:36 -07002085 // If IAudioTrack is re-created, don't let the requested frameCount
2086 // decrease. This can confuse clients that cache frameCount().
Eric Laurent21da6472017-11-09 16:29:26 -08002087 if (mFrameCount > mReqFrameCount) {
2088 mReqFrameCount = mFrameCount;
Glenn Kasten093000f2012-05-03 09:35:36 -07002089 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08002090
Andy Hungd7bd69e2015-07-24 07:52:41 -07002091 // reset server position to 0 as we have new cblk.
2092 mServer = 0;
2093
Glenn Kastene3aa6592012-12-04 12:22:46 -08002094 // update proxy
Glenn Kasten363fb752014-01-15 12:27:31 -08002095 if (mSharedBuffer == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002096 mStaticProxy.clear();
Eric Laurent21da6472017-11-09 16:29:26 -08002097 mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002098 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08002099 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002100 mProxy = mStaticProxy;
2101 }
seunghak.hane6a9d6582014-11-22 15:22:35 +09002102
2103 mProxy->setVolumeLR(gain_minifloat_pack(
2104 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
2105 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
2106
Glenn Kastene3aa6592012-12-04 12:22:46 -08002107 mProxy->setSendLevel(mSendLevel);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002108 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
2109 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
2110 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07002111 mProxy->setSampleRate(effectiveSampleRate);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002112
2113 AudioPlaybackRate playbackRateTemp = mPlaybackRate;
2114 playbackRateTemp.mSpeed = effectiveSpeed;
2115 playbackRateTemp.mPitch = effectivePitch;
2116 mProxy->setPlaybackRate(playbackRateTemp);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002117 mProxy->setMinimum(mNotificationFramesAct);
2118
Kuowei Lid4adbdb2020-08-13 14:44:25 +08002119 if (mDualMonoMode != AUDIO_DUAL_MONO_MODE_OFF) {
2120 setDualMonoMode_l(mDualMonoMode);
2121 }
2122 if (mAudioDescriptionMixLeveldB != -std::numeric_limits<float>::infinity()) {
2123 setAudioDescriptionMixLevel_l(mAudioDescriptionMixLeveldB);
2124 }
2125
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002126 mDeathNotifier = new DeathNotifier(this);
Marco Nelissenf8880202014-11-14 07:58:25 -08002127 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
Glenn Kastene3aa6592012-12-04 12:22:46 -08002128
Andy Hungb68f5eb2019-12-03 16:49:17 -08002129 // This is the first log sent from the AudioTrack client.
2130 // The creation of the audio track by AudioFlinger (in the code above)
2131 // is the first log of the AudioTrack and must be present before
2132 // any AudioTrack client logs will be accepted.
Andy Hungea840382020-05-05 21:50:17 -07002133
Andy Hungb68f5eb2019-12-03 16:49:17 -08002134 mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(mPortId);
2135 mediametrics::LogItem(mMetricsId)
2136 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE)
2137 // the following are immutable
Andy Hunga629bd12020-06-05 16:03:53 -07002138 .set(AMEDIAMETRICS_PROP_FLAGS, toString(mFlags).c_str())
2139 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -08002140 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
Andy Hung3a5c2f32021-02-17 15:06:42 -08002141 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, mLogSessionId)
Andy Hung839a3062021-02-17 11:15:16 -08002142 .set(AMEDIAMETRICS_PROP_PLAYERIID, mPlayerIId)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002143 .set(AMEDIAMETRICS_PROP_TRACKID, mPortId) // dup from key
Andy Hungb68f5eb2019-12-03 16:49:17 -08002144 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2145 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2146 .set(AMEDIAMETRICS_PROP_THREADID, (int32_t)output.outputId)
2147 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2148 .set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)mRoutedDeviceId)
2149 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2150 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
2151 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
2152 // the following are NOT immutable
2153 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)mVolume[AUDIO_INTERLEAVE_LEFT])
2154 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)mVolume[AUDIO_INTERLEAVE_RIGHT])
2155 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Andy Hung73dc2f92021-12-07 21:50:04 -08002156 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)NO_ERROR)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002157 .set(AMEDIAMETRICS_PROP_AUXEFFECTID, (int32_t)mAuxEffectId)
2158 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2159 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2160 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2161 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2162 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveSampleRate)
2163 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2164 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)effectiveSpeed)
2165 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2166 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)effectivePitch)
2167 .record();
2168
2169 // mSendLevel
2170 // mReqFrameCount?
2171 // mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq
2172 // mLatency, mAfLatency, mAfFrameCount, mAfSampleRate
2173
Glenn Kasten38e905b2014-01-13 10:21:48 -08002174 }
2175
Eric Laurentf32d7812017-11-30 14:44:07 -08002176exit:
Andy Hung3acde2c2021-11-11 09:18:08 -08002177 if (status != NO_ERROR) {
2178 if (callbackAdded) {
2179 // note: mOutput is always valid is callbackAdded is true
2180 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
2181 }
2182 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
2183 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
Eric Laurentad2e7b92017-09-14 20:06:42 -07002184 }
Eric Laurentf32d7812017-11-30 14:44:07 -08002185 mStatus = status;
Eric Laurent21da6472017-11-09 16:29:26 -08002186
2187 // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
Glenn Kasten38e905b2014-01-13 10:21:48 -08002188 return status;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002189}
2190
Andy Hung3acde2c2021-11-11 09:18:08 -08002191void AudioTrack::reportError(status_t status, const char *event, const char *message) const
2192{
2193 if (status == NO_ERROR) return;
2194 // We report error on the native side because some callers do not come
2195 // from Java.
Andy Hungde602302021-12-07 21:35:49 -08002196 // Ensure these variables are initialized in set().
2197 mediametrics::LogItem(AMEDIAMETRICS_KEY_AUDIO_TRACK_ERROR)
Andy Hung3acde2c2021-11-11 09:18:08 -08002198 .set(AMEDIAMETRICS_PROP_EVENT, event)
Andy Hung73dc2f92021-12-07 21:50:04 -08002199 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
2200 .set(AMEDIAMETRICS_PROP_STATUSMESSAGE, message)
Andy Hung3acde2c2021-11-11 09:18:08 -08002201 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
2202 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
2203 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2204 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2205 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2206 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2207 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
Andy Hung3acde2c2021-11-11 09:18:08 -08002208 // the following are NOT immutable
Andy Hungde602302021-12-07 21:35:49 -08002209 // frame count is initially the requested frame count, but may be adjusted
2210 // by AudioFlinger after creation.
2211 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
Andy Hung3acde2c2021-11-11 09:18:08 -08002212 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2213 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2214 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2215 .record();
2216}
2217
Glenn Kastenb46f3942015-03-09 12:00:30 -07002218status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002219{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002220 if (audioBuffer == NULL) {
Glenn Kasten551b5352015-03-20 11:30:28 -07002221 if (nonContig != NULL) {
2222 *nonContig = 0;
2223 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002224 return BAD_VALUE;
Eric Laurent9b7d9502011-03-21 11:49:00 -07002225 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002226 if (mTransfer != TRANSFER_OBTAIN) {
2227 audioBuffer->frameCount = 0;
Atneya Nair03079272022-01-18 17:03:14 -05002228 audioBuffer->mSize = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002229 audioBuffer->raw = NULL;
Glenn Kasten551b5352015-03-20 11:30:28 -07002230 if (nonContig != NULL) {
2231 *nonContig = 0;
2232 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002233 return INVALID_OPERATION;
2234 }
Eric Laurent9b7d9502011-03-21 11:49:00 -07002235
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002236 const struct timespec *requested;
Eric Laurentdf576992014-01-27 18:13:39 -08002237 struct timespec timeout;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002238 if (waitCount == -1) {
2239 requested = &ClientProxy::kForever;
2240 } else if (waitCount == 0) {
2241 requested = &ClientProxy::kNonBlocking;
2242 } else if (waitCount > 0) {
Chih-Hung Hsiehbca74292018-08-10 16:06:07 -07002243 time_t ms = WAIT_PERIOD_MS * (time_t) waitCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002244 timeout.tv_sec = ms / 1000;
Andy Hung06a730b2020-04-09 13:28:31 -07002245 timeout.tv_nsec = (ms % 1000) * 1000000;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002246 requested = &timeout;
2247 } else {
Eric Laurent973db022018-11-20 14:54:31 -08002248 ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002249 requested = NULL;
2250 }
Glenn Kastenb46f3942015-03-09 12:00:30 -07002251 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002252}
Eric Laurent1703cdf2011-03-07 14:52:59 -08002253
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002254status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
2255 struct timespec *elapsed, size_t *nonContig)
2256{
2257 // previous and new IAudioTrack sequence numbers are used to detect track re-creation
2258 uint32_t oldSequence = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002259
2260 Proxy::Buffer buffer;
2261 status_t status = NO_ERROR;
2262
2263 static const int32_t kMaxTries = 5;
2264 int32_t tryCounter = kMaxTries;
2265
2266 do {
2267 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
2268 // keep them from going away if another thread re-creates the track during obtainBuffer()
2269 sp<AudioTrackClientProxy> proxy;
2270 sp<IMemory> iMem;
2271
2272 { // start of lock scope
2273 AutoMutex lock(mLock);
2274
Glenn Kasten305996c2020-01-27 08:03:37 -08002275 uint32_t newSequence = mSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002276 // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
2277 if (status == DEAD_OBJECT) {
2278 // re-create track, unless someone else has already done so
2279 if (newSequence == oldSequence) {
2280 status = restoreTrack_l("obtainBuffer");
2281 if (status != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002282 buffer.mFrameCount = 0;
2283 buffer.mRaw = NULL;
2284 buffer.mNonContig = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002285 break;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002286 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002287 }
2288 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002289 oldSequence = newSequence;
2290
Eric Laurent4d231dc2016-03-11 18:38:23 -08002291 if (status == NOT_ENOUGH_DATA) {
2292 restartIfDisabled();
2293 }
2294
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002295 // Keep the extra references
2296 proxy = mProxy;
2297 iMem = mCblkMemory;
2298
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002299 if (mState == STATE_STOPPING) {
2300 status = -EINTR;
2301 buffer.mFrameCount = 0;
2302 buffer.mRaw = NULL;
2303 buffer.mNonContig = 0;
2304 break;
2305 }
2306
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002307 // Non-blocking if track is stopped or paused
2308 if (mState != STATE_ACTIVE) {
2309 requested = &ClientProxy::kNonBlocking;
2310 }
2311
2312 } // end of lock scope
2313
2314 buffer.mFrameCount = audioBuffer->frameCount;
2315 // FIXME starts the requested timeout and elapsed over from scratch
2316 status = proxy->obtainBuffer(&buffer, requested, elapsed);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002317 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002318
2319 audioBuffer->frameCount = buffer.mFrameCount;
Atneya Nair03079272022-01-18 17:03:14 -05002320 audioBuffer->mSize = buffer.mFrameCount * mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002321 audioBuffer->raw = buffer.mRaw;
Glenn Kasten305996c2020-01-27 08:03:37 -08002322 audioBuffer->sequence = oldSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002323 if (nonContig != NULL) {
2324 *nonContig = buffer.mNonContig;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002325 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002326 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002327}
2328
Glenn Kasten54a8a452015-03-09 12:03:00 -07002329void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002330{
Glenn Kasten3f02be22015-03-09 11:59:04 -07002331 // FIXME add error checking on mode, by adding an internal version
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002332 if (mTransfer == TRANSFER_SHARED) {
2333 return;
2334 }
2335
Atneya Nair03079272022-01-18 17:03:14 -05002336 size_t stepCount = audioBuffer->mSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002337 if (stepCount == 0) {
2338 return;
2339 }
2340
2341 Proxy::Buffer buffer;
2342 buffer.mFrameCount = stepCount;
2343 buffer.mRaw = audioBuffer->raw;
Glenn Kastene3aa6592012-12-04 12:22:46 -08002344
Eric Laurent1703cdf2011-03-07 14:52:59 -08002345 AutoMutex lock(mLock);
Glenn Kasten305996c2020-01-27 08:03:37 -08002346 if (audioBuffer->sequence != mSequence) {
2347 // This Buffer came from a different IAudioTrack instance, so ignore the releaseBuffer
2348 ALOGD("%s is no-op due to IAudioTrack sequence mismatch %u != %u",
2349 __func__, audioBuffer->sequence, mSequence);
2350 return;
2351 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002352 mReleased += stepCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002353 mInUnderrun = false;
2354 mProxy->releaseBuffer(&buffer);
2355
2356 // restart track if it was disabled by audioflinger due to previous underrun
Eric Laurent4d231dc2016-03-11 18:38:23 -08002357 restartIfDisabled();
2358}
2359
2360void AudioTrack::restartIfDisabled()
2361{
2362 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
2363 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002364 ALOGW("%s(%d): releaseBuffer() track %p disabled due to previous underrun, restarting",
Eric Laurent973db022018-11-20 14:54:31 -08002365 __func__, mPortId, this);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002366 // FIXME ignoring status
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002367 status_t status;
2368 mAudioTrack->start(&status);
Eric Laurentdf839842012-05-31 14:27:14 -07002369 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002370}
2371
2372// -------------------------------------------------------------------------
2373
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002374ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002375{
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002376 if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07002377 return INVALID_OPERATION;
2378 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002379
Eric Laurentab5cdba2014-06-09 17:22:27 -07002380 if (isDirect()) {
2381 AutoMutex lock(mLock);
2382 int32_t flags = android_atomic_and(
2383 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
2384 &mCblk->mFlags);
2385 if (flags & CBLK_INVALID) {
2386 return DEAD_OBJECT;
2387 }
2388 }
2389
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002390 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
Jiabin Huang447cea72020-07-28 22:35:18 +00002391 // Validation: user is most-likely passing an error code, and it would
Glenn Kasten99e53b82012-01-19 08:59:58 -08002392 // make the return value ambiguous (actualSize vs error).
Andy Hungfb8ede22018-09-12 19:03:24 -07002393 ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
Eric Laurent973db022018-11-20 14:54:31 -08002394 __func__, mPortId, buffer, userSize, userSize);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002395 return BAD_VALUE;
2396 }
2397
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002398 size_t written = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002399 Buffer audioBuffer;
2400
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002401 while (userSize >= mFrameSize) {
2402 audioBuffer.frameCount = userSize / mFrameSize;
Eric Laurentc2f1f072009-07-17 12:17:14 -07002403
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002404 status_t err = obtainBuffer(&audioBuffer,
2405 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002406 if (err < 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002407 if (written > 0) {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002408 break;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002409 }
Glenn Kasten0a2f1512016-07-22 08:06:37 -07002410 if (err == TIMED_OUT || err == -EINTR) {
2411 err = WOULD_BLOCK;
2412 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002413 return ssize_t(err);
2414 }
2415
Atneya Nair03079272022-01-18 17:03:14 -05002416 size_t toWrite = audioBuffer.size();
2417 memcpy(audioBuffer.raw, buffer, toWrite);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002418 buffer = ((const char *) buffer) + toWrite;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002419 userSize -= toWrite;
2420 written += toWrite;
2421
2422 releaseBuffer(&audioBuffer);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002423 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002424
Andy Hungea2b9c02016-02-12 17:06:53 -08002425 if (written > 0) {
2426 mFramesWritten += written / mFrameSize;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002427
2428 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2429 const sp<AudioTrackThread> t = mAudioTrackThread;
2430 if (t != 0) {
2431 // causes wake up of the playback thread, that will callback the client for
2432 // more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
2433 t->wake();
2434 }
2435 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002436 }
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002437
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002438 return written;
2439}
2440
2441// -------------------------------------------------------------------------
2442
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08002443nsecs_t AudioTrack::processAudioBuffer()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002444{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002445 // Currently the AudioTrack thread is not created if there are no callbacks.
2446 // Would it ever make sense to run the thread, even without callbacks?
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002447 // If so, then replace this by checks at each use for mCallback != NULL.
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002448 LOG_ALWAYS_FATAL_IF(mCblk == NULL);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002449 mLock.lock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002450 sp<IAudioTrackCallback> callback = mCallback.promote();
2451 if (!callback) {
2452 mCallback = nullptr;
Atneya Naire260f5a2022-05-03 17:02:20 -04002453 mLock.unlock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002454 return NS_NEVER;
2455 }
Glenn Kastena07f17c2013-04-23 12:39:37 -07002456 if (mAwaitBoost) {
2457 mAwaitBoost = false;
2458 mLock.unlock();
2459 static const int32_t kMaxTries = 5;
2460 int32_t tryCounter = kMaxTries;
2461 uint32_t pollUs = 10000;
2462 do {
Glenn Kasten8255ba72016-08-23 13:54:23 -07002463 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002464 if (policy == SCHED_FIFO || policy == SCHED_RR) {
2465 break;
2466 }
2467 usleep(pollUs);
2468 pollUs <<= 1;
2469 } while (tryCounter-- > 0);
2470 if (tryCounter < 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002471 ALOGE("%s(%d): did not receive expected priority boost on time",
Eric Laurent973db022018-11-20 14:54:31 -08002472 __func__, mPortId);
Glenn Kastena07f17c2013-04-23 12:39:37 -07002473 }
Glenn Kastenb0dfd462013-07-10 16:52:47 -07002474 // Run again immediately
2475 return 0;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002476 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002477
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002478 // Can only reference mCblk while locked
2479 int32_t flags = android_atomic_and(
Glenn Kasten96f60d82013-07-12 10:21:18 -07002480 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);
Glenn Kastena47f3162012-11-07 10:13:08 -08002481
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002482 // Check for track invalidation
2483 if (flags & CBLK_INVALID) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002484 // for offloaded tracks restoreTrack_l() will just update the sequence and clear
2485 // AudioSystem cache. We should not exit here but after calling the callback so
2486 // that the upper layers can recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002487 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
Lajos Molnarf1063e22015-04-17 15:19:42 -07002488 status_t status __unused = restoreTrack_l("processAudioBuffer");
2489 // FIXME unused status
Andy Hung53c3b5f2014-12-15 16:42:05 -08002490 // after restoration, continue below to make sure that the loop and buffer events
2491 // are notified because they have been cleared from mCblk->mFlags above.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002492 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002493 }
2494
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002495 bool waitStreamEnd = mState == STATE_STOPPING;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002496 bool active = mState == STATE_ACTIVE;
2497
2498 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
2499 bool newUnderrun = false;
2500 if (flags & CBLK_UNDERRUN) {
2501#if 0
2502 // Currently in shared buffer mode, when the server reaches the end of buffer,
2503 // the track stays active in continuous underrun state. It's up to the application
2504 // to pause or stop the track, or set the position to a new offset within buffer.
2505 // This was some experimental code to auto-pause on underrun. Keeping it here
2506 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
2507 if (mTransfer == TRANSFER_SHARED) {
2508 mState = STATE_PAUSED;
2509 active = false;
2510 }
2511#endif
2512 if (!mInUnderrun) {
2513 mInUnderrun = true;
2514 newUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002515 }
2516 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002517
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002518 // Get current position of server
Andy Hung90e8a972015-11-09 16:42:40 -08002519 Modulo<uint32_t> position(updateAndGetPosition_l());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002520
2521 // Manage marker callback
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002522 bool markerReached = false;
Andy Hung90e8a972015-11-09 16:42:40 -08002523 Modulo<uint32_t> markerPosition(mMarkerPosition);
2524 // uses 32 bit wraparound for comparison with position.
2525 if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002526 mMarkerReached = markerReached = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002527 }
2528
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002529 // Determine number of new position callback(s) that will be needed, while locked
2530 size_t newPosCount = 0;
Andy Hung90e8a972015-11-09 16:42:40 -08002531 Modulo<uint32_t> newPosition(mNewPosition);
2532 uint32_t updatePeriod = mUpdatePeriod;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002533 // FIXME fails for wraparound, need 64 bits
2534 if (updatePeriod > 0 && position >= newPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002535 newPosCount = ((position - newPosition).value() / updatePeriod) + 1;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002536 mNewPosition += updatePeriod * newPosCount;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002537 }
2538
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002539 // Cache other fields that will be needed soon
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002540 uint32_t sampleRate = mSampleRate;
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002541 float speed = mPlaybackRate.mSpeed;
Andy Hunga7f03352015-05-31 21:54:49 -07002542 const uint32_t notificationFrames = mNotificationFramesAct;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002543 if (mRefreshRemaining) {
2544 mRefreshRemaining = false;
2545 mRemainingFrames = notificationFrames;
2546 mRetryOnPartialBuffer = false;
2547 }
2548 size_t misalignment = mProxy->getMisalignment();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002549 uint32_t sequence = mSequence;
Glenn Kasten96f04882013-09-20 09:28:56 -07002550 sp<AudioTrackClientProxy> proxy = mProxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002551
Andy Hung53c3b5f2014-12-15 16:42:05 -08002552 // Determine the number of new loop callback(s) that will be needed, while locked.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002553 uint32_t loopCountNotifications = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -08002554 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
2555
2556 if (mLoopCount > 0) {
2557 int loopCount;
2558 size_t bufferPosition;
2559 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
2560 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
2561 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
2562 mLoopCountNotified = loopCount; // discard any excess notifications
2563 } else if (mLoopCount < 0) {
2564 // FIXME: We're not accurate with notification count and position with infinite looping
2565 // since loopCount from server side will always return -1 (we could decrement it).
2566 size_t bufferPosition = mStaticProxy->getBufferPosition();
2567 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
2568 loopPeriod = mLoopEnd - bufferPosition;
2569 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
2570 size_t bufferPosition = mStaticProxy->getBufferPosition();
2571 loopPeriod = mFrameCount - bufferPosition;
2572 }
2573
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002574 // These fields don't need to be cached, because they are assigned only by set():
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002575 // mTransfer, mCallback, mUserData, mFormat, mFrameSize, mFlags
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002576 // mFlags is also assigned by createTrack_l(), but not the bit we care about.
2577
2578 mLock.unlock();
2579
Andy Hunga7f03352015-05-31 21:54:49 -07002580 // get anchor time to account for callbacks.
2581 const nsecs_t timeBeforeCallbacks = systemTime();
2582
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002583 if (waitStreamEnd) {
Andy Hunga7f03352015-05-31 21:54:49 -07002584 // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread
2585 // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
2586 // (and make sure we don't callback for more data while we're stopping).
2587 // This helps with position, marker notifications, and track invalidation.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002588 struct timespec timeout;
2589 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
2590 timeout.tv_nsec = 0;
2591
Glenn Kasten96f04882013-09-20 09:28:56 -07002592 status_t status = proxy->waitStreamEndDone(&timeout);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002593 switch (status) {
2594 case NO_ERROR:
2595 case DEAD_OBJECT:
2596 case TIMED_OUT:
Andy Hung39609a02015-09-03 16:38:38 -07002597 if (status != DEAD_OBJECT) {
2598 // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop();
2599 // instead, the application should handle the EVENT_NEW_IAUDIOTRACK.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002600 callback->onStreamEnd();
Andy Hung39609a02015-09-03 16:38:38 -07002601 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002602 {
2603 AutoMutex lock(mLock);
2604 // The previously assigned value of waitStreamEnd is no longer valid,
2605 // since the mutex has been unlocked and either the callback handler
2606 // or another thread could have re-started the AudioTrack during that time.
2607 waitStreamEnd = mState == STATE_STOPPING;
2608 if (waitStreamEnd) {
2609 mState = STATE_STOPPED;
Andy Hungc2813e52014-10-16 17:54:34 -07002610 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002611 }
2612 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002613 if (waitStreamEnd && status != DEAD_OBJECT) {
2614 return NS_INACTIVE;
2615 }
2616 break;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002617 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002618 return 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002619 }
2620
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002621 // perform callbacks while unlocked
2622 if (newUnderrun) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002623 callback->onUnderrun();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002624 }
Andy Hung53c3b5f2014-12-15 16:42:05 -08002625 while (loopCountNotifications > 0) {
Andy Hung53c3b5f2014-12-15 16:42:05 -08002626 --loopCountNotifications;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002627 callback->onLoopEnd(mLoopCount > 0 ? loopCountNotifications + mLoopCountNotified : -1);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002628 }
2629 if (flags & CBLK_BUFFER_END) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002630 callback->onBufferEnd();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002631 }
2632 if (markerReached) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002633 callback->onMarker(markerPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002634 }
2635 while (newPosCount > 0) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002636 callback->onNewPos(newPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002637 newPosition += updatePeriod;
2638 newPosCount--;
2639 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002640
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002641 if (mObservedSequence != sequence) {
2642 mObservedSequence = sequence;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002643 callback->onNewIAudioTrack();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002644 // for offloaded tracks, just wait for the upper layers to recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002645 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002646 return NS_INACTIVE;
2647 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002648 }
2649
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002650 // if inactive, then don't run me again until re-started
2651 if (!active) {
2652 return NS_INACTIVE;
Eric Laurent2267ba12011-09-07 11:13:23 -07002653 }
2654
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002655 // Compute the estimated time until the next timed event (position, markers, loops)
2656 // FIXME only for non-compressed audio
2657 uint32_t minFrames = ~0;
2658 if (!markerReached && position < markerPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002659 minFrames = (markerPosition - position).value();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002660 }
2661 if (loopPeriod > 0 && loopPeriod < minFrames) {
Andy Hung2d85f092015-01-07 12:45:13 -08002662 // loopPeriod is already adjusted for actual position.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002663 minFrames = loopPeriod;
2664 }
Andy Hung2d85f092015-01-07 12:45:13 -08002665 if (updatePeriod > 0) {
Andy Hung90e8a972015-11-09 16:42:40 -08002666 minFrames = min(minFrames, (newPosition - position).value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002667 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002668
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002669 // If > 0, poll periodically to recover from a stuck server. A good value is 2.
2670 static const uint32_t kPoll = 0;
2671 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
2672 minFrames = kPoll * notificationFrames;
2673 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002674
Andy Hunga7f03352015-05-31 21:54:49 -07002675 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
2676 static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL;
2677 const nsecs_t timeAfterCallbacks = systemTime();
2678
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002679 // Convert frame units to time units
2680 nsecs_t ns = NS_WHENEVER;
2681 if (minFrames != (uint32_t) ~0) {
jiabinc7bb8322017-09-06 18:20:11 -07002682 // AudioFlinger consumption of client data may be irregular when coming out of device
2683 // standby since the kernel buffers require filling. This is throttled to no more than 2x
2684 // the expected rate in the MixerThread. Hence, we reduce the estimated time to wait by one
2685 // half (but no more than half a second) to improve callback accuracy during these temporary
2686 // data surges.
2687 const nsecs_t estimatedNs = framesToNanoseconds(minFrames, sampleRate, speed);
2688 constexpr nsecs_t maxThrottleCompensationNs = 500000000LL;
2689 ns = estimatedNs - min(estimatedNs / 2, maxThrottleCompensationNs) + kWaitPeriodNs;
Andy Hunga7f03352015-05-31 21:54:49 -07002690 ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time
2691 // TODO: Should we warn if the callback time is too long?
2692 if (ns < 0) ns = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002693 }
2694
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002695 // If not supplying data by EVENT_MORE_DATA or EVENT_CAN_WRITE_MORE_DATA, then we're done
2696 if (mTransfer != TRANSFER_CALLBACK && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002697 return ns;
2698 }
2699
Andy Hunga7f03352015-05-31 21:54:49 -07002700 // EVENT_MORE_DATA callback handling.
2701 // Timing for linear pcm audio data formats can be derived directly from the
2702 // buffer fill level.
2703 // Timing for compressed data is not directly available from the buffer fill level,
2704 // rather indirectly from waiting for blocking mode callbacks or waiting for obtain()
2705 // to return a certain fill level.
2706
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002707 struct timespec timeout;
2708 const struct timespec *requested = &ClientProxy::kForever;
2709 if (ns != NS_WHENEVER) {
2710 timeout.tv_sec = ns / 1000000000LL;
2711 timeout.tv_nsec = ns % 1000000000LL;
Andy Hungfb8ede22018-09-12 19:03:24 -07002712 ALOGV("%s(%d): timeout %ld.%03d",
Eric Laurent973db022018-11-20 14:54:31 -08002713 __func__, mPortId, timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002714 requested = &timeout;
2715 }
2716
Andy Hungea2b9c02016-02-12 17:06:53 -08002717 size_t writtenFrames = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002718 while (mRemainingFrames > 0) {
2719
2720 Buffer audioBuffer;
2721 audioBuffer.frameCount = mRemainingFrames;
2722 size_t nonContig;
2723 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
2724 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
Andy Hungfb8ede22018-09-12 19:03:24 -07002725 "%s(%d): obtainBuffer() err=%d frameCount=%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002726 __func__, mPortId, err, audioBuffer.frameCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002727 requested = &ClientProxy::kNonBlocking;
2728 size_t avail = audioBuffer.frameCount + nonContig;
Andy Hungfb8ede22018-09-12 19:03:24 -07002729 ALOGV("%s(%d): obtainBuffer(%u) returned %zu = %zu + %zu err %d",
Eric Laurent973db022018-11-20 14:54:31 -08002730 __func__, mPortId, mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002731 if (err != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002732 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
2733 (isOffloaded() && (err == DEAD_OBJECT))) {
Glenn Kasten606fbc12015-10-22 15:28:15 -07002734 // FIXME bug 25195759
2735 return 1000000;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002736 }
Andy Hungfb8ede22018-09-12 19:03:24 -07002737 ALOGE("%s(%d): Error %d obtaining an audio buffer, giving up.",
Eric Laurent973db022018-11-20 14:54:31 -08002738 __func__, mPortId, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002739 return NS_NEVER;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002740 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002741
Phil Burkfdb3c072016-02-09 10:47:02 -08002742 if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002743 mRetryOnPartialBuffer = false;
2744 if (avail < mRemainingFrames) {
Andy Hunga7f03352015-05-31 21:54:49 -07002745 if (ns > 0) { // account for obtain time
2746 const nsecs_t timeNow = systemTime();
2747 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2748 }
Andy Hungeb6e6502019-04-29 13:47:40 -07002749
2750 // delayNs is first computed by the additional frames required in the buffer.
2751 nsecs_t delayNs = framesToNanoseconds(
2752 mRemainingFrames - avail, sampleRate, speed);
2753
2754 // afNs is the AudioFlinger mixer period in ns.
2755 const nsecs_t afNs = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2756
2757 // If the AudioTrack is double buffered based on the AudioFlinger mixer period,
2758 // we may have a race if we wait based on the number of frames desired.
2759 // This is a possible issue with resampling and AAudio.
2760 //
2761 // The granularity of audioflinger processing is one mixer period; if
2762 // our wait time is less than one mixer period, wait at most half the period.
2763 if (delayNs < afNs) {
2764 delayNs = std::min(delayNs, afNs / 2);
2765 }
2766
2767 // adjust our ns wait by delayNs.
2768 if (ns < 0 /* NS_WHENEVER */ || delayNs < ns) {
2769 ns = delayNs;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002770 }
2771 return ns;
2772 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07002773 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002774
Atneya Nair03079272022-01-18 17:03:14 -05002775 size_t reqSize = audioBuffer.size();
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002776 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2777 // when notifying client it can write more data, pass the total size that can be
2778 // written in the next write() call, since it's not passed through the callback
Atneya Nair03079272022-01-18 17:03:14 -05002779 audioBuffer.mSize += nonContig;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002780 }
Atneya Nairc2dd1272021-10-26 19:39:51 -04002781 const size_t writtenSize = (mTransfer == TRANSFER_CALLBACK)
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002782 ? callback->onMoreData(audioBuffer)
2783 : callback->onCanWriteMoreData(audioBuffer);
Jiabin Huang447cea72020-07-28 22:35:18 +00002784 // Validate on returned size
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002785 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002786 ALOGE("%s(%d): EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
Eric Laurent973db022018-11-20 14:54:31 -08002787 __func__, mPortId, reqSize, ssize_t(writtenSize));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002788 return NS_NEVER;
2789 }
2790
2791 if (writtenSize == 0) {
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002792 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2793 // The callback EVENT_CAN_WRITE_MORE_DATA was processed in the JNI of
2794 // android.media.AudioTrack. The JNI is not using the callback to provide data,
2795 // it only signals to the Java client that it can provide more data, which
2796 // this track is read to accept now.
2797 // The playback thread will be awaken at the next ::write()
2798 return NS_WHENEVER;
2799 }
The Android Open Source Project8555d082009-03-05 14:34:35 -08002800 // The callback is done filling buffers
2801 // Keep this thread going to handle timed events and
2802 // still try to get more data in intervals of WAIT_PERIOD_MS
2803 // but don't just loop and block the CPU, so wait
Andy Hunga7f03352015-05-31 21:54:49 -07002804
2805 // mCbf(EVENT_MORE_DATA, ...) might either
2806 // (1) Block until it can fill the buffer, returning 0 size on EOS.
2807 // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS.
2808 // (3) Return 0 size when no data is available, does not wait for more data.
2809 //
2810 // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer.
2811 // We try to compute the wait time to avoid a tight sleep-wait cycle,
2812 // especially for case (3).
2813 //
2814 // The decision to support (1) and (2) affect the sizing of mRemainingFrames
2815 // and this loop; whereas for case (3) we could simply check once with the full
2816 // buffer size and skip the loop entirely.
2817
2818 nsecs_t myns;
Phil Burkfdb3c072016-02-09 10:47:02 -08002819 if (audio_has_proportional_frames(mFormat)) {
Andy Hunga7f03352015-05-31 21:54:49 -07002820 // time to wait based on buffer occupancy
2821 const nsecs_t datans = mRemainingFrames <= avail ? 0 :
2822 framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
2823 // audio flinger thread buffer size (TODO: adjust for fast tracks)
Glenn Kastenea38ee72016-04-18 11:08:01 -07002824 // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks.
Andy Hunga7f03352015-05-31 21:54:49 -07002825 const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2826 // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0.
2827 myns = datans + (afns / 2);
2828 } else {
2829 // FIXME: This could ping quite a bit if the buffer isn't full.
2830 // Note that when mState is stopping we waitStreamEnd, so it never gets here.
2831 myns = kWaitPeriodNs;
2832 }
2833 if (ns > 0) { // account for obtain and callback time
2834 const nsecs_t timeNow = systemTime();
2835 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2836 }
2837 if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
2838 ns = myns;
2839 }
2840 return ns;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002841 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002842
Atneya Nairc2dd1272021-10-26 19:39:51 -04002843 // releaseBuffer reads from audioBuffer.size
Atneya Nair03079272022-01-18 17:03:14 -05002844 audioBuffer.mSize = writtenSize;
Atneya Nairc2dd1272021-10-26 19:39:51 -04002845
Glenn Kasten138d6f92015-03-20 10:54:51 -07002846 size_t releasedFrames = writtenSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002847 audioBuffer.frameCount = releasedFrames;
2848 mRemainingFrames -= releasedFrames;
2849 if (misalignment >= releasedFrames) {
2850 misalignment -= releasedFrames;
2851 } else {
2852 misalignment = 0;
2853 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002854
2855 releaseBuffer(&audioBuffer);
Andy Hungea2b9c02016-02-12 17:06:53 -08002856 writtenFrames += releasedFrames;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002857
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002858 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
2859 // if callback doesn't like to accept the full chunk
2860 if (writtenSize < reqSize) {
2861 continue;
2862 }
2863
2864 // There could be enough non-contiguous frames available to satisfy the remaining request
2865 if (mRemainingFrames <= nonContig) {
2866 continue;
2867 }
2868
2869#if 0
2870 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
2871 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
2872 // that total to a sum == notificationFrames.
2873 if (0 < misalignment && misalignment <= mRemainingFrames) {
2874 mRemainingFrames = misalignment;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002875 return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002876 }
2877#endif
2878
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002879 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002880 if (writtenFrames > 0) {
2881 AutoMutex lock(mLock);
2882 mFramesWritten += writtenFrames;
2883 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002884 mRemainingFrames = notificationFrames;
2885 mRetryOnPartialBuffer = true;
2886
2887 // A lot has transpired since ns was calculated, so run again immediately and re-calculate
2888 return 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002889}
2890
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002891status_t AudioTrack::restoreTrack_l(const char *from)
Eric Laurent1703cdf2011-03-07 14:52:59 -08002892{
Andy Hungb68f5eb2019-12-03 16:49:17 -08002893 status_t result = NO_ERROR; // logged: make sure to set this before returning.
2894 const int64_t beginNs = systemTime();
Andy Hunga6b27032020-04-27 10:34:24 -07002895 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -08002896 mediametrics::LogItem(mMetricsId)
2897 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_RESTORE)
Andy Hungea840382020-05-05 21:50:17 -07002898 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08002899 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
2900 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
2901 .set(AMEDIAMETRICS_PROP_WHERE, from)
2902 .record(); });
2903
Andy Hungfb8ede22018-09-12 19:03:24 -07002904 ALOGW("%s(%d): dead IAudioTrack, %s, creating a new one from %s()",
Eric Laurent973db022018-11-20 14:54:31 -08002905 __func__, mPortId, isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002906 ++mSequence;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002907
Glenn Kastena47f3162012-11-07 10:13:08 -08002908 // refresh the audio configuration cache in this process to make sure we get new
Glenn Kastend2d089f2014-11-05 11:48:12 -08002909 // output parameters and new IAudioFlinger in createTrack_l()
Glenn Kastena47f3162012-11-07 10:13:08 -08002910 AudioSystem::clearAudioConfigCache();
Eric Laurent9f6530f2011-08-30 10:18:54 -07002911
Ronghua Wufaeb0f22015-05-21 12:20:21 -07002912 if (isOffloadedOrDirect_l() || mDoNotReconnect) {
Andy Hung1f1db832015-06-08 13:26:10 -07002913 // FIXME re-creation of offloaded and direct tracks is not yet implemented;
2914 // reconsider enabling for linear PCM encodings when position can be preserved.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002915 result = DEAD_OBJECT;
2916 return result;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002917 }
2918
Phil Burk2812d9e2016-01-04 10:34:30 -08002919 // Save so we can return count since creation.
2920 mUnderrunCountOffset = getUnderrunCount_l();
2921
Glenn Kasten200092b2014-08-15 15:13:30 -07002922 // save the old static buffer position
Andy Hungf20a4e92016-08-15 19:10:34 -07002923 uint32_t staticPosition = 0;
Andy Hung4ede21d2014-12-12 15:37:34 -08002924 size_t bufferPosition = 0;
2925 int loopCount = 0;
2926 if (mStaticProxy != 0) {
2927 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
Andy Hungf20a4e92016-08-15 19:10:34 -07002928 staticPosition = mStaticProxy->getPosition().unsignedValue();
Andy Hung4ede21d2014-12-12 15:37:34 -08002929 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002930
Andy Hung3c7f47a2021-03-16 17:30:09 -07002931 // save the old startThreshold and framecount
2932 const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
2933 const uint32_t originalFrameCount = mProxy->frameCount();
2934
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002935 // See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
2936 // causes a lot of churn on the service side, and it can reject starting
2937 // playback of a previously created track. May also apply to other cases.
2938 const int INITIAL_RETRIES = 3;
2939 int retries = INITIAL_RETRIES;
2940retry:
2941 if (retries < INITIAL_RETRIES) {
2942 // See the comment for clearAudioConfigCache at the start of the function.
2943 AudioSystem::clearAudioConfigCache();
2944 }
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08002945 mFlags = mOrigFlags;
2946
Glenn Kasten200092b2014-08-15 15:13:30 -07002947 // If a new IAudioTrack is successfully created, createTrack_l() will modify the
Glenn Kastena47f3162012-11-07 10:13:08 -08002948 // following member variables: mAudioTrack, mCblkMemory and mCblk.
Glenn Kasten200092b2014-08-15 15:13:30 -07002949 // It will also delete the strong references on previous IAudioTrack and IMemory.
2950 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002951 result = createTrack_l();
Eric Laurentcc21e4f2013-10-16 15:12:32 -07002952
Eric Laurent6ec546d2018-10-10 16:52:14 -07002953 if (result == NO_ERROR) {
Andy Hungd7bd69e2015-07-24 07:52:41 -07002954 // take the frames that will be lost by track recreation into account in saved position
2955 // For streaming tracks, this is the amount we obtained from the user/client
2956 // (not the number actually consumed at the server - those are already lost).
2957 if (mStaticProxy == 0) {
2958 mPosition = mReleased;
2959 }
Andy Hung4ede21d2014-12-12 15:37:34 -08002960 // Continue playback from last known position and restore loop.
2961 if (mStaticProxy != 0) {
2962 if (loopCount != 0) {
2963 mStaticProxy->setBufferPositionAndLoop(bufferPosition,
2964 mLoopStart, mLoopEnd, loopCount);
2965 } else {
2966 mStaticProxy->setBufferPosition(bufferPosition);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002967 if (bufferPosition == mFrameCount) {
Eric Laurent973db022018-11-20 14:54:31 -08002968 ALOGD("%s(%d): restoring track at end of static buffer", __func__, mPortId);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002969 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002970 }
2971 }
Andy Hung4ef88d72017-02-21 19:47:53 -08002972 // restore volume handler
Andy Hung39399b62017-04-21 15:07:45 -07002973 mVolumeHandler->forall([this](const VolumeShaper &shaper) -> VolumeShaper::Status {
2974 sp<VolumeShaper::Operation> operationToEnd =
2975 new VolumeShaper::Operation(shaper.mOperation);
Andy Hung4ef88d72017-02-21 19:47:53 -08002976 // TODO: Ideally we would restore to the exact xOffset position
2977 // as returned by getVolumeShaperState(), but we don't have that
2978 // information when restoring at the client unless we periodically poll
2979 // the server or create shared memory state.
2980 //
Andy Hung39399b62017-04-21 15:07:45 -07002981 // For now, we simply advance to the end of the VolumeShaper effect
2982 // if it has been started.
2983 if (shaper.isStarted()) {
Andy Hungf3702642017-05-05 17:33:32 -07002984 operationToEnd->setNormalizedTime(1.f);
Andy Hung39399b62017-04-21 15:07:45 -07002985 }
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002986 media::VolumeShaperConfiguration config;
2987 shaper.mConfiguration->writeToParcelable(&config);
2988 media::VolumeShaperOperation operation;
2989 operationToEnd->writeToParcelable(&operation);
2990 status_t status;
2991 mAudioTrack->applyVolumeShaper(config, operation, &status);
2992 return status;
Andy Hung4ef88d72017-02-21 19:47:53 -08002993 });
2994
Andy Hung3c7f47a2021-03-16 17:30:09 -07002995 // restore the original start threshold if different than frameCount.
2996 if (originalStartThresholdInFrames != originalFrameCount) {
2997 // Note: mProxy->setStartThresholdInFrames() call is in the Proxy
2998 // and does not trigger a restart.
2999 // (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
3000 // Any start would be triggered on the mState == ACTIVE check below.
3001 const uint32_t currentThreshold =
3002 mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
3003 ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
3004 "%s(%d) startThresholdInFrames changing from %u to %u",
3005 __func__, mPortId, originalStartThresholdInFrames, currentThreshold);
3006 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003007 if (mState == STATE_ACTIVE) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003008 mAudioTrack->start(&result);
Eric Laurent1703cdf2011-03-07 14:52:59 -08003009 }
Andy Hungf20a4e92016-08-15 19:10:34 -07003010 // server resets to zero so we offset
3011 mFramesWrittenServerOffset =
3012 mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten;
3013 mFramesWrittenAtRestore = mFramesWrittenServerOffset;
Eric Laurent1703cdf2011-03-07 14:52:59 -08003014 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003015 if (result != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08003016 ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07003017 if (--retries > 0) {
Eric Laurent6ec546d2018-10-10 16:52:14 -07003018 // leave time for an eventual race condition to clear before retrying
3019 usleep(500000);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07003020 goto retry;
3021 }
Eric Laurent6ec546d2018-10-10 16:52:14 -07003022 // if no retries left, set invalid bit to force restoring at next occasion
3023 // and avoid inconsistent active state on client and server sides
3024 if (mCblk != nullptr) {
3025 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
3026 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08003027 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08003028 return result;
3029}
3030
Andy Hung90e8a972015-11-09 16:42:40 -08003031Modulo<uint32_t> AudioTrack::updateAndGetPosition_l()
Glenn Kasten200092b2014-08-15 15:13:30 -07003032{
3033 // This is the sole place to read server consumed frames
Andy Hung90e8a972015-11-09 16:42:40 -08003034 Modulo<uint32_t> newServer(mProxy->getPosition());
3035 const int32_t delta = (newServer - mServer).signedValue();
Glenn Kasten200092b2014-08-15 15:13:30 -07003036 // TODO There is controversy about whether there can be "negative jitter" in server position.
3037 // This should be investigated further, and if possible, it should be addressed.
3038 // A more definite failure mode is infrequent polling by client.
3039 // One could call (void)getPosition_l() in releaseBuffer(),
3040 // so mReleased and mPosition are always lock-step as best possible.
3041 // That should ensure delta never goes negative for infrequent polling
3042 // unless the server has more than 2^31 frames in its buffer,
3043 // in which case the use of uint32_t for these counters has bigger issues.
Andy Hung90e8a972015-11-09 16:42:40 -08003044 ALOGE_IF(delta < 0,
Andy Hungfb8ede22018-09-12 19:03:24 -07003045 "%s(%d): detected illegal retrograde motion by the server: mServer advanced by %d",
Eric Laurent973db022018-11-20 14:54:31 -08003046 __func__, mPortId, delta);
Chad Brubaker039c27a2015-09-23 15:17:29 -07003047 mServer = newServer;
Andy Hung90e8a972015-11-09 16:42:40 -08003048 if (delta > 0) { // avoid retrograde
3049 mPosition += delta;
3050 }
3051 return mPosition;
Glenn Kasten200092b2014-08-15 15:13:30 -07003052}
3053
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003054bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed)
Andy Hung8edb8dc2015-03-26 19:13:55 -07003055{
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003056 updateLatency_l();
Andy Hung8edb8dc2015-03-26 19:13:55 -07003057 // applicable for mixing tracks only (not offloaded or direct)
3058 if (mStaticProxy != 0) {
3059 return true; // static tracks do not have issues with buffer sizing.
3060 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07003061 const size_t minFrameCount =
Eric Laurent21da6472017-11-09 16:29:26 -08003062 AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
3063 sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003064 const bool allowed = mFrameCount >= minFrameCount;
3065 ALOGD_IF(!allowed,
Andy Hungfb8ede22018-09-12 19:03:24 -07003066 "%s(%d): denied "
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003067 "mAfLatency:%u mAfFrameCount:%zu mAfSampleRate:%u sampleRate:%u speed:%f "
3068 "mFrameCount:%zu < minFrameCount:%zu",
Eric Laurent973db022018-11-20 14:54:31 -08003069 __func__, mPortId,
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003070 mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed,
Andy Hung8edb8dc2015-03-26 19:13:55 -07003071 mFrameCount, minFrameCount);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003072 return allowed;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003073}
3074
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003075status_t AudioTrack::setParameters(const String8& keyValuePairs)
3076{
3077 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003078 status_t status;
3079 mAudioTrack->setParameters(keyValuePairs.c_str(), &status);
3080 return status;
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003081}
3082
Dean Wheatleya70eef72018-01-04 14:23:50 +11003083status_t AudioTrack::selectPresentation(int presentationId, int programId)
3084{
3085 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08003086 AudioParameter param = AudioParameter();
3087 param.addInt(String8(AudioParameter::keyPresentationId), presentationId);
3088 param.addInt(String8(AudioParameter::keyProgramId), programId);
3089 ALOGV("%s(%d): PresentationId/ProgramId[%s]",
3090 __func__, mPortId, param.toString().string());
3091
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003092 status_t status;
3093 mAudioTrack->setParameters(param.toString().c_str(), &status);
3094 return status;
Dean Wheatleya70eef72018-01-04 14:23:50 +11003095}
3096
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003097VolumeShaper::Status AudioTrack::applyVolumeShaper(
3098 const sp<VolumeShaper::Configuration>& configuration,
3099 const sp<VolumeShaper::Operation>& operation)
3100{
3101 AutoMutex lock(mLock);
Andy Hung4ef88d72017-02-21 19:47:53 -08003102 mVolumeHandler->setIdIfNecessary(configuration);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003103 media::VolumeShaperConfiguration config;
3104 configuration->writeToParcelable(&config);
3105 media::VolumeShaperOperation op;
3106 operation->writeToParcelable(&op);
3107 VolumeShaper::Status status;
3108 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003109
3110 if (status == DEAD_OBJECT) {
3111 if (restoreTrack_l("applyVolumeShaper") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003112 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003113 }
3114 }
Andy Hung4ef88d72017-02-21 19:47:53 -08003115 if (status >= 0) {
3116 // save VolumeShaper for restore
3117 mVolumeHandler->applyVolumeShaper(configuration, operation);
Andy Hung39399b62017-04-21 15:07:45 -07003118 if (mState == STATE_ACTIVE || mState == STATE_STOPPING) {
3119 mVolumeHandler->setStarted();
3120 }
3121 } else {
3122 // warn only if not an expected restore failure.
3123 ALOGW_IF(!((isOffloadedOrDirect_l() || mDoNotReconnect) && status == DEAD_OBJECT),
Eric Laurent973db022018-11-20 14:54:31 -08003124 "%s(%d): applyVolumeShaper failed: %d", __func__, mPortId, status);
Andy Hung4ef88d72017-02-21 19:47:53 -08003125 }
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003126 return status;
3127}
3128
3129sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id)
3130{
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003131 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003132 std::optional<media::VolumeShaperState> vss;
3133 mAudioTrack->getVolumeShaperState(id, &vss);
3134 sp<VolumeShaper::State> state;
3135 if (vss.has_value()) {
3136 state = new VolumeShaper::State();
3137 state->readFromParcelable(vss.value());
3138 }
Andy Hung39399b62017-04-21 15:07:45 -07003139 if (state.get() == nullptr && (mCblk->mFlags & CBLK_INVALID) != 0) {
3140 if (restoreTrack_l("getVolumeShaperState") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003141 mAudioTrack->getVolumeShaperState(id, &vss);
3142 if (vss.has_value()) {
3143 state = new VolumeShaper::State();
3144 state->readFromParcelable(vss.value());
3145 }
Andy Hung39399b62017-04-21 15:07:45 -07003146 }
3147 }
3148 return state;
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003149}
3150
Andy Hungea2b9c02016-02-12 17:06:53 -08003151status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp)
3152{
3153 if (timestamp == nullptr) {
3154 return BAD_VALUE;
3155 }
3156 AutoMutex lock(mLock);
Andy Hunge13f8a62016-03-30 14:20:42 -07003157 return getTimestamp_l(timestamp);
3158}
3159
3160status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp)
3161{
Andy Hungea2b9c02016-02-12 17:06:53 -08003162 if (mCblk->mFlags & CBLK_INVALID) {
3163 const status_t status = restoreTrack_l("getTimestampExtended");
3164 if (status != OK) {
3165 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3166 // recommending that the track be recreated.
3167 return DEAD_OBJECT;
3168 }
3169 }
3170 // check for offloaded/direct here in case restoring somehow changed those flags.
3171 if (isOffloadedOrDirect_l()) {
3172 return INVALID_OPERATION; // not supported
3173 }
3174 status_t status = mProxy->getTimestamp(timestamp);
Andy Hungfb8ede22018-09-12 19:03:24 -07003175 LOG_ALWAYS_FATAL_IF(status != OK, "%s(%d): status %d not allowed from proxy getTimestamp",
Eric Laurent973db022018-11-20 14:54:31 -08003176 __func__, mPortId, status);
Andy Hungea2b9c02016-02-12 17:06:53 -08003177 bool found = false;
Andy Hunge1e98462016-04-12 10:18:51 -07003178 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten;
3179 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
3180 // server side frame offset in case AudioTrack has been restored.
3181 for (int i = ExtendedTimestamp::LOCATION_SERVER;
3182 i < ExtendedTimestamp::LOCATION_MAX; ++i) {
3183 if (timestamp->mTimeNs[i] >= 0) {
3184 // apply server offset (frames flushed is ignored
3185 // so we don't report the jump when the flush occurs).
3186 timestamp->mPosition[i] += mFramesWrittenServerOffset;
3187 found = true;
Andy Hungea2b9c02016-02-12 17:06:53 -08003188 }
3189 }
3190 return found ? OK : WOULD_BLOCK;
3191}
3192
Glenn Kastence703742013-07-19 16:33:58 -07003193status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
3194{
Glenn Kasten53cec222013-08-29 09:01:02 -07003195 AutoMutex lock(mLock);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003196 return getTimestamp_l(timestamp);
3197}
Phil Burk1b420972015-04-22 10:52:21 -07003198
Andy Hung65ffdfc2016-10-10 15:52:11 -07003199status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp)
3200{
Phil Burk1b420972015-04-22 10:52:21 -07003201 bool previousTimestampValid = mPreviousTimestampValid;
3202 // Set false here to cover all the error return cases.
3203 mPreviousTimestampValid = false;
3204
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003205 switch (mState) {
3206 case STATE_ACTIVE:
3207 case STATE_PAUSED:
3208 break; // handle below
3209 case STATE_FLUSHED:
3210 case STATE_STOPPED:
3211 return WOULD_BLOCK;
3212 case STATE_STOPPING:
3213 case STATE_PAUSED_STOPPING:
3214 if (!isOffloaded_l()) {
3215 return INVALID_OPERATION;
3216 }
3217 break; // offloaded tracks handled below
3218 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003219 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in getTimestamp(): %d",
Eric Laurent973db022018-11-20 14:54:31 -08003220 __func__, mPortId, mState);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003221 break;
Glenn Kastenfe346c72013-08-30 13:28:22 -07003222 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003223
Eric Laurent275e8e92014-11-30 15:14:47 -08003224 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung6653c932015-06-08 13:27:48 -07003225 const status_t status = restoreTrack_l("getTimestamp");
3226 if (status != OK) {
3227 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3228 // recommending that the track be recreated.
3229 return DEAD_OBJECT;
3230 }
Eric Laurent275e8e92014-11-30 15:14:47 -08003231 }
3232
Glenn Kasten200092b2014-08-15 15:13:30 -07003233 // The presented frame count must always lag behind the consumed frame count.
3234 // To avoid a race, read the presented frames first. This ensures that presented <= consumed.
Andy Hung6ae58432016-02-16 18:32:24 -08003235
3236 status_t status;
Andy Hung818e7a32016-02-16 18:08:07 -08003237 if (isOffloadedOrDirect_l()) {
Andy Hung6ae58432016-02-16 18:32:24 -08003238 // use Binder to get timestamp
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003239 media::AudioTimestampInternal ts;
3240 mAudioTrack->getTimestamp(&ts, &status);
3241 if (status == OK) {
Andy Hung973638a2020-12-08 20:47:45 -08003242 timestamp = VALUE_OR_FATAL(aidl2legacy_AudioTimestampInternal_AudioTimestamp(ts));
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003243 }
Andy Hung6ae58432016-02-16 18:32:24 -08003244 } else {
3245 // read timestamp from shared memory
3246 ExtendedTimestamp ets;
3247 status = mProxy->getTimestamp(&ets);
3248 if (status == OK) {
Andy Hungb01faa32016-04-27 12:51:32 -07003249 ExtendedTimestamp::Location location;
3250 status = ets.getBestTimestamp(&timestamp, &location);
3251
3252 if (status == OK) {
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003253 updateLatency_l();
Andy Hungb01faa32016-04-27 12:51:32 -07003254 // It is possible that the best location has moved from the kernel to the server.
3255 // In this case we adjust the position from the previous computed latency.
3256 if (location == ExtendedTimestamp::LOCATION_SERVER) {
3257 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL,
Andy Hungfb8ede22018-09-12 19:03:24 -07003258 "%s(%d): location moved from kernel to server",
Eric Laurent973db022018-11-20 14:54:31 -08003259 __func__, mPortId);
Andy Hung07eee802016-06-21 16:47:16 -07003260 // check that the last kernel OK time info exists and the positions
3261 // are valid (if they predate the current track, the positions may
3262 // be zero or negative).
Andy Hungb01faa32016-04-27 12:51:32 -07003263 const int64_t frames =
Andy Hung6d7b1192016-05-07 22:59:48 -07003264 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
Andy Hung07eee802016-06-21 16:47:16 -07003265 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 ||
3266 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 ||
3267 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0)
Andy Hung6d7b1192016-05-07 22:59:48 -07003268 ?
3269 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed
3270 / 1000)
3271 :
3272 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3273 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]);
Andy Hungfb8ede22018-09-12 19:03:24 -07003274 ALOGV("%s(%d): frame adjustment:%lld timestamp:%s",
Eric Laurent973db022018-11-20 14:54:31 -08003275 __func__, mPortId, (long long)frames, ets.toString().c_str());
Andy Hungb01faa32016-04-27 12:51:32 -07003276 if (frames >= ets.mPosition[location]) {
3277 timestamp.mPosition = 0;
3278 } else {
3279 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames);
3280 }
Andy Hung69488c42016-05-16 18:43:33 -07003281 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) {
3282 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER,
Andy Hungfb8ede22018-09-12 19:03:24 -07003283 "%s(%d): location moved from server to kernel",
Eric Laurent973db022018-11-20 14:54:31 -08003284 __func__, mPortId);
Andy Hung98731a22019-04-08 19:19:07 -07003285
3286 if (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER] ==
3287 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL]) {
3288 // In Q, we don't return errors as an invalid time
3289 // but instead we leave the last kernel good timestamp alone.
3290 //
3291 // If server is identical to kernel, the device data pipeline is idle.
3292 // A better start time is now. The retrograde check ensures
3293 // timestamp monotonicity.
3294 const int64_t nowNs = systemTime();
Andy Hungcf3b7152019-04-19 18:29:21 -07003295 if (!mTimestampStallReported) {
3296 ALOGD("%s(%d): device stall time corrected using current time %lld",
3297 __func__, mPortId, (long long)nowNs);
3298 mTimestampStallReported = true;
3299 }
Andy Hung98731a22019-04-08 19:19:07 -07003300 timestamp.mTime = convertNsToTimespec(nowNs);
Andy Hungcf3b7152019-04-19 18:29:21 -07003301 } else {
3302 mTimestampStallReported = false;
Andy Hung98731a22019-04-08 19:19:07 -07003303 }
Andy Hungb01faa32016-04-27 12:51:32 -07003304 }
Andy Hung5d313802016-10-10 15:09:39 -07003305
3306 // We update the timestamp time even when paused.
3307 if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) {
3308 const int64_t now = systemTime();
Andy Hung2b01f002017-07-05 12:01:36 -07003309 const int64_t at = audio_utils_ns_from_timespec(&timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003310 const int64_t lag =
3311 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
3312 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0)
3313 ? int64_t(mAfLatency * 1000000LL)
3314 : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3315 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK])
3316 * NANOS_PER_SECOND / mSampleRate;
3317 const int64_t limit = now - lag; // no earlier than this limit
3318 if (at < limit) {
3319 ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld",
3320 (long long)lag, (long long)at, (long long)limit);
Andy Hungffa36952017-08-17 10:41:51 -07003321 timestamp.mTime = convertNsToTimespec(limit);
Andy Hung5d313802016-10-10 15:09:39 -07003322 }
3323 }
Andy Hungb01faa32016-04-27 12:51:32 -07003324 mPreviousLocation = location;
3325 } else {
3326 // right after AudioTrack is started, one may not find a timestamp
Eric Laurent973db022018-11-20 14:54:31 -08003327 ALOGV("%s(%d): getBestTimestamp did not find timestamp", __func__, mPortId);
Andy Hungb01faa32016-04-27 12:51:32 -07003328 }
Andy Hung6ae58432016-02-16 18:32:24 -08003329 }
3330 if (status == INVALID_OPERATION) {
Andy Hungf20a4e92016-08-15 19:10:34 -07003331 // INVALID_OPERATION occurs when no timestamp has been issued by the server;
3332 // other failures are signaled by a negative time.
3333 // If we come out of FLUSHED or STOPPED where the position is known
3334 // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of
3335 // "zero" for NuPlayer). We don't convert for track restoration as position
3336 // does not reset.
Andy Hungfb8ede22018-09-12 19:03:24 -07003337 ALOGV("%s(%d): timestamp server offset:%lld restore frames:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003338 __func__, mPortId,
Andy Hungf20a4e92016-08-15 19:10:34 -07003339 (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore);
3340 if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) {
3341 status = WOULD_BLOCK;
3342 }
Andy Hung6ae58432016-02-16 18:32:24 -08003343 }
3344 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003345 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08003346 ALOGV_IF(status != WOULD_BLOCK, "%s(%d): getTimestamp error:%#x", __func__, mPortId, status);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003347 return status;
3348 }
3349 if (isOffloadedOrDirect_l()) {
3350 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) {
3351 // use cached paused position in case another offloaded track is running.
3352 timestamp.mPosition = mPausedPosition;
3353 clock_gettime(CLOCK_MONOTONIC, &timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003354 // TODO: adjust for delay
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003355 return NO_ERROR;
3356 }
3357
3358 // Check whether a pending flush or stop has completed, as those commands may
Andy Hungc8e09c62015-06-03 23:43:36 -07003359 // be asynchronous or return near finish or exhibit glitchy behavior.
3360 //
3361 // Originally this showed up as the first timestamp being a continuation of
3362 // the previous song under gapless playback.
3363 // However, we sometimes see zero timestamps, then a glitch of
3364 // the previous song's position, and then correct timestamps afterwards.
Andy Hungffa36952017-08-17 10:41:51 -07003365 if (mStartFromZeroUs != 0 && mSampleRate != 0) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003366 static const int kTimeJitterUs = 100000; // 100 ms
3367 static const int k1SecUs = 1000000;
3368
3369 const int64_t timeNow = getNowUs();
3370
Andy Hungffa36952017-08-17 10:41:51 -07003371 if (timeNow < mStartFromZeroUs + k1SecUs) { // within first second of starting
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003372 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003373 if (timestampTimeUs < mStartFromZeroUs) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003374 return WOULD_BLOCK; // stale timestamp time, occurs before start.
3375 }
Andy Hungffa36952017-08-17 10:41:51 -07003376 const int64_t deltaTimeUs = timestampTimeUs - mStartFromZeroUs;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003377 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07003378 / ((double)mSampleRate * mPlaybackRate.mSpeed);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003379
3380 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
3381 // Verify that the counter can't count faster than the sample rate
Andy Hungc8e09c62015-06-03 23:43:36 -07003382 // since the start time. If greater, then that means we may have failed
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003383 // to completely flush or stop the previous playing track.
Andy Hungc8e09c62015-06-03 23:43:36 -07003384 ALOGW_IF(!mTimestampStartupGlitchReported,
Andy Hungfb8ede22018-09-12 19:03:24 -07003385 "%s(%d): startup glitch detected"
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003386 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)",
Eric Laurent973db022018-11-20 14:54:31 -08003387 __func__, mPortId,
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003388 (long long)deltaTimeUs, (long long)deltaPositionByUs,
3389 timestamp.mPosition);
Andy Hungc8e09c62015-06-03 23:43:36 -07003390 mTimestampStartupGlitchReported = true;
3391 if (previousTimestampValid
3392 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) {
3393 timestamp = mPreviousTimestamp;
3394 mPreviousTimestampValid = true;
3395 return NO_ERROR;
3396 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003397 return WOULD_BLOCK;
3398 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003399 if (deltaPositionByUs != 0) {
Andy Hungffa36952017-08-17 10:41:51 -07003400 mStartFromZeroUs = 0; // don't check again, we got valid nonzero position.
Andy Hungc8e09c62015-06-03 23:43:36 -07003401 }
3402 } else {
Andy Hungffa36952017-08-17 10:41:51 -07003403 mStartFromZeroUs = 0; // don't check again, start time expired.
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003404 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003405 mTimestampStartupGlitchReported = false;
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003406 }
3407 } else {
Glenn Kasten200092b2014-08-15 15:13:30 -07003408 // Update the mapping between local consumed (mPosition) and server consumed (mServer)
3409 (void) updateAndGetPosition_l();
3410 // Server consumed (mServer) and presented both use the same server time base,
3411 // and server consumed is always >= presented.
3412 // The delta between these represents the number of frames in the buffer pipeline.
3413 // If this delta between these is greater than the client position, it means that
3414 // actually presented is still stuck at the starting line (figuratively speaking),
3415 // waiting for the first frame to go by. So we can't report a valid timestamp yet.
Andy Hung90e8a972015-11-09 16:42:40 -08003416 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when
3417 // mPosition exceeds 32 bits.
3418 // TODO Remove when timestamp is updated to contain pipeline status info.
3419 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue();
3420 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */
3421 && (uint32_t)pipelineDepthInFrames > mPosition.value()) {
Glenn Kasten200092b2014-08-15 15:13:30 -07003422 return INVALID_OPERATION;
3423 }
3424 // Convert timestamp position from server time base to client time base.
3425 // TODO The following code should work OK now because timestamp.mPosition is 32-bit.
3426 // But if we change it to 64-bit then this could fail.
Andy Hung90e8a972015-11-09 16:42:40 -08003427 // Use Modulo computation here.
3428 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value();
Glenn Kasten200092b2014-08-15 15:13:30 -07003429 // Immediately after a call to getPosition_l(), mPosition and
3430 // mServer both represent the same frame position. mPosition is
3431 // in client's point of view, and mServer is in server's point of
3432 // view. So the difference between them is the "fudge factor"
3433 // between client and server views due to stop() and/or new
3434 // IAudioTrack. And timestamp.mPosition is initially in server's
3435 // point of view, so we need to apply the same fudge factor to it.
Glenn Kastenfe346c72013-08-30 13:28:22 -07003436 }
Phil Burk1b420972015-04-22 10:52:21 -07003437
3438 // Prevent retrograde motion in timestamp.
3439 // This is sometimes caused by erratic reports of the available space in the ALSA drivers.
3440 if (status == NO_ERROR) {
Andy Hung3b8c6332019-04-03 19:29:36 -07003441 // Fix stale time when checking timestamp right after start().
3442 // The position is at the last reported location but the time can be stale
3443 // due to pause or standby or cold start latency.
3444 //
3445 // We keep advancing the time (but not the position) to ensure that the
3446 // stale value does not confuse the application.
3447 //
3448 // For offload compatibility, use a default lag value here.
3449 // Any time discrepancy between this update and the pause timestamp is handled
3450 // by the retrograde check afterwards.
3451 int64_t currentTimeNanos = audio_utils_ns_from_timespec(&timestamp.mTime);
3452 const int64_t lagNs = int64_t(mAfLatency * 1000000LL);
3453 const int64_t limitNs = mStartNs - lagNs;
3454 if (currentTimeNanos < limitNs) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003455 if (!mTimestampStaleTimeReported) {
3456 ALOGD("%s(%d): stale timestamp time corrected, "
3457 "currentTimeNanos: %lld < limitNs: %lld < mStartNs: %lld",
3458 __func__, mPortId,
3459 (long long)currentTimeNanos, (long long)limitNs, (long long)mStartNs);
3460 mTimestampStaleTimeReported = true;
3461 }
Andy Hung3b8c6332019-04-03 19:29:36 -07003462 timestamp.mTime = convertNsToTimespec(limitNs);
3463 currentTimeNanos = limitNs;
Andy Hungcf3b7152019-04-19 18:29:21 -07003464 } else {
3465 mTimestampStaleTimeReported = false;
Andy Hung3b8c6332019-04-03 19:29:36 -07003466 }
3467
Andy Hungffa36952017-08-17 10:41:51 -07003468 // previousTimestampValid is set to false when starting after a stop or flush.
Phil Burk1b420972015-04-22 10:52:21 -07003469 if (previousTimestampValid) {
Andy Hung2b01f002017-07-05 12:01:36 -07003470 const int64_t previousTimeNanos =
3471 audio_utils_ns_from_timespec(&mPreviousTimestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003472
3473 // retrograde check
Phil Burk1b420972015-04-22 10:52:21 -07003474 if (currentTimeNanos < previousTimeNanos) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003475 if (!mTimestampRetrogradeTimeReported) {
3476 ALOGW("%s(%d): retrograde timestamp time corrected, %lld < %lld",
3477 __func__, mPortId,
3478 (long long)currentTimeNanos, (long long)previousTimeNanos);
3479 mTimestampRetrogradeTimeReported = true;
3480 }
Andy Hung5d313802016-10-10 15:09:39 -07003481 timestamp.mTime = mPreviousTimestamp.mTime;
Andy Hungcf3b7152019-04-19 18:29:21 -07003482 } else {
3483 mTimestampRetrogradeTimeReported = false;
Phil Burk1b420972015-04-22 10:52:21 -07003484 }
3485
3486 // Looking at signed delta will work even when the timestamps
3487 // are wrapping around.
Andy Hung90e8a972015-11-09 16:42:40 -08003488 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition)
3489 - mPreviousTimestamp.mPosition).signedValue();
Phil Burk4c5a3672015-04-30 16:18:53 -07003490 if (deltaPosition < 0) {
3491 // Only report once per position instead of spamming the log.
Andy Hungcf3b7152019-04-19 18:29:21 -07003492 if (!mTimestampRetrogradePositionReported) {
Andy Hungfb8ede22018-09-12 19:03:24 -07003493 ALOGW("%s(%d): retrograde timestamp position corrected, %d = %u - %u",
Eric Laurent973db022018-11-20 14:54:31 -08003494 __func__, mPortId,
Phil Burk4c5a3672015-04-30 16:18:53 -07003495 deltaPosition,
3496 timestamp.mPosition,
3497 mPreviousTimestamp.mPosition);
Andy Hungcf3b7152019-04-19 18:29:21 -07003498 mTimestampRetrogradePositionReported = true;
Phil Burk4c5a3672015-04-30 16:18:53 -07003499 }
3500 } else {
Andy Hungcf3b7152019-04-19 18:29:21 -07003501 mTimestampRetrogradePositionReported = false;
Phil Burk4c5a3672015-04-30 16:18:53 -07003502 }
Andy Hung5d313802016-10-10 15:09:39 -07003503 if (deltaPosition < 0) {
3504 timestamp.mPosition = mPreviousTimestamp.mPosition;
3505 deltaPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07003506 }
Andy Hung5d313802016-10-10 15:09:39 -07003507#if 0
3508 // Uncomment this to verify audio timestamp rate.
3509 const int64_t deltaTime =
Andy Hung2b01f002017-07-05 12:01:36 -07003510 audio_utils_ns_from_timespec(&timestamp.mTime) - previousTimeNanos;
Andy Hung5d313802016-10-10 15:09:39 -07003511 if (deltaTime != 0) {
3512 const int64_t computedSampleRate =
3513 deltaPosition * (long long)NANOS_PER_SECOND / deltaTime;
Andy Hungfb8ede22018-09-12 19:03:24 -07003514 ALOGD("%s(%d): computedSampleRate:%u sampleRate:%u",
Eric Laurent973db022018-11-20 14:54:31 -08003515 __func__, mPortId,
Andy Hung5d313802016-10-10 15:09:39 -07003516 (unsigned)computedSampleRate, mSampleRate);
3517 }
3518#endif
Phil Burk1b420972015-04-22 10:52:21 -07003519 }
3520 mPreviousTimestamp = timestamp;
3521 mPreviousTimestampValid = true;
3522 }
3523
Glenn Kastenfe346c72013-08-30 13:28:22 -07003524 return status;
Glenn Kastence703742013-07-19 16:33:58 -07003525}
3526
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003527String8 AudioTrack::getParameters(const String8& keys)
3528{
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003529 audio_io_handle_t output = getOutput();
Glenn Kasten142f5192014-03-25 17:44:59 -07003530 if (output != AUDIO_IO_HANDLE_NONE) {
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003531 return AudioSystem::getParameters(output, keys);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01003532 } else {
3533 return String8::empty();
3534 }
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003535}
3536
Glenn Kasten23a75452014-01-13 10:37:17 -08003537bool AudioTrack::isOffloaded() const
3538{
3539 AutoMutex lock(mLock);
3540 return isOffloaded_l();
3541}
3542
Eric Laurentab5cdba2014-06-09 17:22:27 -07003543bool AudioTrack::isDirect() const
3544{
3545 AutoMutex lock(mLock);
3546 return isDirect_l();
3547}
3548
3549bool AudioTrack::isOffloadedOrDirect() const
3550{
3551 AutoMutex lock(mLock);
3552 return isOffloadedOrDirect_l();
3553}
3554
3555
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003556status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003557{
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003558 String8 result;
3559
3560 result.append(" AudioTrack::dump\n");
Andy Hungfb8ede22018-09-12 19:03:24 -07003561 result.appendFormat(" id(%d) status(%d), state(%d), session Id(%d), flags(%#x)\n",
Eric Laurent973db022018-11-20 14:54:31 -08003562 mPortId, mStatus, mState, mSessionId, mFlags);
Eric Laurentd114b622017-11-27 18:37:04 -08003563 result.appendFormat(" stream type(%d), left - right volume(%f, %f)\n",
François Gaffie58d4be52018-11-06 15:30:12 +01003564 mStreamType,
Eric Laurentd114b622017-11-27 18:37:04 -08003565 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
Glenn Kasten49f36ba2017-12-06 13:02:02 -08003566 result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
Eric Laurentd114b622017-11-27 18:37:04 -08003567 mFormat, mChannelMask, mChannelCount);
3568 result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
3569 mSampleRate, mOriginalSampleRate, mPlaybackRate.mSpeed);
3570 result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
3571 mFrameCount, mReqFrameCount);
3572 result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
3573 " req. notif. per buff(%u)\n",
3574 mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
3575 result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
3576 mLatency, mSelectedDeviceId, mRoutedDeviceId);
3577 result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
3578 mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003579 ::write(fd, result.string(), result.size());
3580 return NO_ERROR;
3581}
3582
Phil Burk2812d9e2016-01-04 10:34:30 -08003583uint32_t AudioTrack::getUnderrunCount() const
3584{
3585 AutoMutex lock(mLock);
3586 return getUnderrunCount_l();
3587}
3588
3589uint32_t AudioTrack::getUnderrunCount_l() const
3590{
3591 return mProxy->getUnderrunCount() + mUnderrunCountOffset;
3592}
3593
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003594uint32_t AudioTrack::getUnderrunFrames() const
3595{
3596 AutoMutex lock(mLock);
3597 return mProxy->getUnderrunFrames();
3598}
3599
Andy Hung3a5c2f32021-02-17 15:06:42 -08003600void AudioTrack::setLogSessionId(const char *logSessionId)
3601{
3602 AutoMutex lock(mLock);
Andy Hung1a9c21b2021-02-25 20:43:18 -08003603 if (logSessionId == nullptr) logSessionId = ""; // an empty string is an unset session id.
Andy Hung3a5c2f32021-02-17 15:06:42 -08003604 if (mLogSessionId == logSessionId) return;
3605
3606 mLogSessionId = logSessionId;
3607 mediametrics::LogItem(mMetricsId)
3608 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETLOGSESSIONID)
3609 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, logSessionId)
3610 .record();
3611}
3612
Andy Hung839a3062021-02-17 11:15:16 -08003613void AudioTrack::setPlayerIId(int playerIId)
3614{
3615 AutoMutex lock(mLock);
3616 if (mPlayerIId == playerIId) return;
3617
3618 mPlayerIId = playerIId;
3619 mediametrics::LogItem(mMetricsId)
3620 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYERIID)
3621 .set(AMEDIAMETRICS_PROP_PLAYERIID, playerIId)
3622 .record();
3623}
3624
Eric Laurent296fb132015-05-01 11:38:42 -07003625status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback)
3626{
Eric Laurent09f1ed22019-04-24 17:45:17 -07003627
Eric Laurent296fb132015-05-01 11:38:42 -07003628 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003629 ALOGW("%s(%d): adding NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003630 return BAD_VALUE;
3631 }
3632 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07003633 if (mDeviceCallback.unsafe_get() == callback.get()) {
Eric Laurent973db022018-11-20 14:54:31 -08003634 ALOGW("%s(%d): adding same callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003635 return INVALID_OPERATION;
3636 }
3637 status_t status = NO_ERROR;
3638 if (mOutput != AUDIO_IO_HANDLE_NONE) {
3639 if (mDeviceCallback != 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003640 ALOGW("%s(%d): callback already present!", __func__, mPortId);
Eric Laurent09f1ed22019-04-24 17:45:17 -07003641 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003642 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003643 status = AudioSystem::addAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003644 }
3645 mDeviceCallback = callback;
3646 return status;
3647}
3648
3649status_t AudioTrack::removeAudioDeviceCallback(
3650 const sp<AudioSystem::AudioDeviceCallback>& callback)
3651{
3652 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003653 ALOGW("%s(%d): removing NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003654 return BAD_VALUE;
3655 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003656 AutoMutex lock(mLock);
3657 if (mDeviceCallback.unsafe_get() != callback.get()) {
3658 ALOGW("%s removing different callback!", __FUNCTION__);
3659 return INVALID_OPERATION;
Eric Laurent296fb132015-05-01 11:38:42 -07003660 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003661 mDeviceCallback.clear();
Eric Laurent296fb132015-05-01 11:38:42 -07003662 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07003663 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003664 }
Eric Laurent296fb132015-05-01 11:38:42 -07003665 return NO_ERROR;
3666}
3667
Eric Laurentad2e7b92017-09-14 20:06:42 -07003668
3669void AudioTrack::onAudioDeviceUpdate(audio_io_handle_t audioIo,
3670 audio_port_handle_t deviceId)
3671{
3672 sp<AudioSystem::AudioDeviceCallback> callback;
3673 {
3674 AutoMutex lock(mLock);
3675 if (audioIo != mOutput) {
3676 return;
3677 }
3678 callback = mDeviceCallback.promote();
3679 // only update device if the track is active as route changes due to other use cases are
3680 // irrelevant for this client
3681 if (mState == STATE_ACTIVE) {
3682 mRoutedDeviceId = deviceId;
3683 }
3684 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003685
Eric Laurentad2e7b92017-09-14 20:06:42 -07003686 if (callback.get() != nullptr) {
3687 callback->onAudioDeviceUpdate(mOutput, mRoutedDeviceId);
3688 }
3689}
3690
Andy Hunge13f8a62016-03-30 14:20:42 -07003691status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location)
3692{
3693 if (msec == nullptr ||
3694 (location != ExtendedTimestamp::LOCATION_SERVER
3695 && location != ExtendedTimestamp::LOCATION_KERNEL)) {
3696 return BAD_VALUE;
3697 }
3698 AutoMutex lock(mLock);
3699 // inclusive of offloaded and direct tracks.
3700 //
3701 // It is possible, but not enabled, to allow duration computation for non-pcm
3702 // audio_has_proportional_frames() formats because currently they have
3703 // the drain rate equivalent to the pcm sample rate * framesize.
3704 if (!isPurePcmData_l()) {
3705 return INVALID_OPERATION;
3706 }
3707 ExtendedTimestamp ets;
3708 if (getTimestamp_l(&ets) == OK
3709 && ets.mTimeNs[location] > 0) {
3710 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
3711 - ets.mPosition[location];
3712 if (diff < 0) {
3713 *msec = 0;
3714 } else {
3715 // ms is the playback time by frames
3716 int64_t ms = (int64_t)((double)diff * 1000 /
3717 ((double)mSampleRate * mPlaybackRate.mSpeed));
3718 // clockdiff is the timestamp age (negative)
3719 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
3720 ets.mTimeNs[location]
3721 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
3722 - systemTime(SYSTEM_TIME_MONOTONIC);
3723
3724 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
3725 static const int NANOS_PER_MILLIS = 1000000;
3726 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
3727 }
3728 return NO_ERROR;
3729 }
3730 if (location != ExtendedTimestamp::LOCATION_SERVER) {
3731 return INVALID_OPERATION; // LOCATION_KERNEL is not available
3732 }
3733 // use server position directly (offloaded and direct arrive here)
3734 updateAndGetPosition_l();
3735 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue();
3736 *msec = (diff <= 0) ? 0
3737 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed));
3738 return NO_ERROR;
3739}
3740
Andy Hung65ffdfc2016-10-10 15:52:11 -07003741bool AudioTrack::hasStarted()
3742{
3743 AutoMutex lock(mLock);
3744 switch (mState) {
3745 case STATE_STOPPED:
3746 if (isOffloadedOrDirect_l()) {
3747 // check if we have started in the past to return true.
Andy Hungffa36952017-08-17 10:41:51 -07003748 return mStartFromZeroUs > 0;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003749 }
3750 // A normal audio track may still be draining, so
3751 // check if stream has ended. This covers fasttrack position
3752 // instability and start/stop without any data written.
3753 if (mProxy->getStreamEndDone()) {
3754 return true;
3755 }
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003756 FALLTHROUGH_INTENDED;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003757 case STATE_ACTIVE:
3758 case STATE_STOPPING:
3759 break;
3760 case STATE_PAUSED:
3761 case STATE_PAUSED_STOPPING:
3762 case STATE_FLUSHED:
3763 return false; // we're not active
3764 default:
Eric Laurent973db022018-11-20 14:54:31 -08003765 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in hasStarted(): %d", __func__, mPortId, mState);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003766 break;
3767 }
3768
3769 // wait indicates whether we need to wait for a timestamp.
3770 // This is conservatively figured - if we encounter an unexpected error
3771 // then we will not wait.
3772 bool wait = false;
3773 if (isOffloadedOrDirect_l()) {
3774 AudioTimestamp ts;
3775 status_t status = getTimestamp_l(ts);
3776 if (status == WOULD_BLOCK) {
3777 wait = true;
3778 } else if (status == OK) {
3779 wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition);
3780 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003781 ALOGV("%s(%d): hasStarted wait:%d ts:%u start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003782 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003783 (int)wait,
3784 ts.mPosition,
3785 (long long)mStartTs.mPosition);
3786 } else {
3787 int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG
3788 ExtendedTimestamp ets;
3789 status_t status = getTimestamp_l(&ets);
3790 if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets
3791 wait = true;
3792 } else if (status == OK) {
3793 for (location = ExtendedTimestamp::LOCATION_KERNEL;
3794 location >= ExtendedTimestamp::LOCATION_SERVER; --location) {
3795 if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) {
3796 continue;
3797 }
3798 wait = ets.mPosition[location] == 0
3799 || ets.mPosition[location] == mStartEts.mPosition[location];
3800 break;
3801 }
3802 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003803 ALOGV("%s(%d): hasStarted wait:%d ets:%lld start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003804 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003805 (int)wait,
3806 (long long)ets.mPosition[location],
3807 (long long)mStartEts.mPosition[location]);
3808 }
3809 return !wait;
3810}
3811
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003812// =========================================================================
3813
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003814void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003815{
3816 sp<AudioTrack> audioTrack = mAudioTrack.promote();
3817 if (audioTrack != 0) {
3818 AutoMutex lock(audioTrack->mLock);
3819 audioTrack->mProxy->binderDied();
3820 }
3821}
3822
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003823// =========================================================================
3824
Andy Hungca353672019-03-06 11:54:38 -08003825AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver)
Andy Hungeb46cf92019-03-06 10:13:38 -08003826 : Thread(true /* bCanCallJava */) // binder recursion on restoreTrack_l() may call Java.
3827 , mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
Glenn Kasten598de6c2013-10-16 17:02:13 -07003828 mIgnoreNextPausedInt(false)
Glenn Kasten3acbd052012-02-28 10:39:56 -08003829{
3830}
3831
3832AudioTrack::AudioTrackThread::~AudioTrackThread()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003833{
3834}
3835
3836bool AudioTrack::AudioTrackThread::threadLoop()
3837{
Glenn Kasten3acbd052012-02-28 10:39:56 -08003838 {
3839 AutoMutex _l(mMyLock);
3840 if (mPaused) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003841 // TODO check return value and handle or log
Glenn Kasten3acbd052012-02-28 10:39:56 -08003842 mMyCond.wait(mMyLock);
3843 // caller will check for exitPending()
3844 return true;
3845 }
Glenn Kasten598de6c2013-10-16 17:02:13 -07003846 if (mIgnoreNextPausedInt) {
3847 mIgnoreNextPausedInt = false;
3848 mPausedInt = false;
3849 }
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003850 if (mPausedInt) {
Glenn Kasten47d55172017-05-23 11:19:30 -07003851 // TODO use futex instead of condition, for event flag "or"
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003852 if (mPausedNs > 0) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003853 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003854 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
3855 } else {
Glenn Kasten75f79032017-05-23 11:22:44 -07003856 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003857 mMyCond.wait(mMyLock);
3858 }
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003859 mPausedInt = false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003860 return true;
3861 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08003862 }
Eric Laurent7985dcb2014-10-07 15:45:14 -07003863 if (exitPending()) {
3864 return false;
3865 }
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003866 nsecs_t ns = mReceiver.processAudioBuffer();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003867 switch (ns) {
3868 case 0:
3869 return true;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003870 case NS_INACTIVE:
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003871 pauseInternal();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003872 return true;
3873 case NS_NEVER:
3874 return false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003875 case NS_WHENEVER:
Andy Hung3c09c782014-12-29 18:39:32 -08003876 // Event driven: call wake() when callback notifications conditions change.
3877 ns = INT64_MAX;
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003878 FALLTHROUGH_INTENDED;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003879 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003880 LOG_ALWAYS_FATAL_IF(ns < 0, "%s(%d): processAudioBuffer() returned %lld",
Eric Laurent973db022018-11-20 14:54:31 -08003881 __func__, mReceiver.mPortId, (long long)ns);
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003882 pauseInternal(ns);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003883 return true;
Glenn Kastenca8b2802012-04-23 13:58:16 -07003884 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003885}
3886
Glenn Kasten3acbd052012-02-28 10:39:56 -08003887void AudioTrack::AudioTrackThread::requestExit()
3888{
3889 // must be in this order to avoid a race condition
3890 Thread::requestExit();
Glenn Kasten598de6c2013-10-16 17:02:13 -07003891 resume();
Glenn Kasten3acbd052012-02-28 10:39:56 -08003892}
3893
3894void AudioTrack::AudioTrackThread::pause()
3895{
3896 AutoMutex _l(mMyLock);
3897 mPaused = true;
3898}
3899
3900void AudioTrack::AudioTrackThread::resume()
3901{
3902 AutoMutex _l(mMyLock);
Glenn Kasten598de6c2013-10-16 17:02:13 -07003903 mIgnoreNextPausedInt = true;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003904 if (mPaused || mPausedInt) {
Glenn Kasten3acbd052012-02-28 10:39:56 -08003905 mPaused = false;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003906 mPausedInt = false;
Glenn Kasten3acbd052012-02-28 10:39:56 -08003907 mMyCond.signal();
3908 }
3909}
3910
Andy Hung3c09c782014-12-29 18:39:32 -08003911void AudioTrack::AudioTrackThread::wake()
3912{
3913 AutoMutex _l(mMyLock);
Andy Hunga8d08902015-07-22 11:52:13 -07003914 if (!mPaused) {
3915 // wake() might be called while servicing a callback - ignore the next
3916 // pause time and call processAudioBuffer.
Andy Hung3c09c782014-12-29 18:39:32 -08003917 mIgnoreNextPausedInt = true;
Andy Hunga8d08902015-07-22 11:52:13 -07003918 if (mPausedInt && mPausedNs > 0) {
3919 // audio track is active and internally paused with timeout.
3920 mPausedInt = false;
3921 mMyCond.signal();
3922 }
Andy Hung3c09c782014-12-29 18:39:32 -08003923 }
3924}
3925
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003926void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
3927{
3928 AutoMutex _l(mMyLock);
3929 mPausedInt = true;
3930 mPausedNs = ns;
3931}
3932
jiabinf6eb4c32020-02-25 14:06:25 -08003933binder::Status AudioTrack::AudioTrackCallback::onCodecFormatChanged(
3934 const std::vector<uint8_t>& audioMetadata)
3935{
3936 AutoMutex _l(mAudioTrackCbLock);
3937 sp<media::IAudioTrackCallback> callback = mCallback.promote();
3938 if (callback.get() != nullptr) {
3939 callback->onCodecFormatChanged(audioMetadata);
3940 } else {
3941 mCallback.clear();
3942 }
3943 return binder::Status::ok();
3944}
3945
3946void AudioTrack::AudioTrackCallback::setAudioTrackCallback(
3947 const sp<media::IAudioTrackCallback> &callback) {
3948 AutoMutex lock(mAudioTrackCbLock);
3949 mCallback = callback;
3950}
3951
Glenn Kasten40bc9062015-03-20 09:09:33 -07003952} // namespace android