blob: 24fd625195845afb73a5939fab922d9b44fb5b11 [file] [log] [blame]
Glenn Kasten99e53b82012-01-19 08:59:58 -08001/*
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9** http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080018//#define LOG_NDEBUG 0
19#define LOG_TAG "AudioTrack"
20
Mark Salyzyn34fb2962014-06-18 16:30:56 -070021#include <inttypes.h>
Glenn Kastenc56f3422014-03-21 17:53:17 -070022#include <math.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080023#include <sys/resource.h>
Andy Hung959b5b82021-09-24 10:46:20 -070024#include <thread>
Mark Salyzyn34fb2962014-06-18 16:30:56 -070025
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -080026#include <android/media/IAudioPolicyService.h>
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -070027#include <android-base/macros.h>
Andy Hung3acde2c2021-11-11 09:18:08 -080028#include <android-base/stringprintf.h>
Andy Hung2b01f002017-07-05 12:01:36 -070029#include <audio_utils/clock.h>
Glenn Kasten9f80dd22012-12-18 15:57:32 -080030#include <audio_utils/primitives.h>
31#include <binder/IPCThreadState.h>
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
Atneyaf86d2692021-10-14 14:02:36 -0400280 (void)set(streamType, sampleRate, format, channelMask,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400281 frameCount, flags, callback, notificationFrames,
282 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
283 attributionSource, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
284}
285
286namespace {
287 class LegacyCallbackWrapper : public AudioTrack::IAudioTrackCallback {
288 const AudioTrack::legacy_callback_t mCallback;
289 void * const mData;
290 public:
291 LegacyCallbackWrapper(AudioTrack::legacy_callback_t callback, void* user)
292 : mCallback(callback), mData(user) {}
293 size_t onMoreData(const AudioTrack::Buffer & buffer) override {
294 AudioTrack::Buffer copy = buffer;
295 mCallback(AudioTrack::EVENT_MORE_DATA, mData, static_cast<void*>(&copy));
296 return copy.size;
297 }
298 void onUnderrun() override {
299 mCallback(AudioTrack::EVENT_UNDERRUN, mData, nullptr);
300 }
301 void onLoopEnd(int32_t loopsRemaining) override {
302 mCallback(AudioTrack::EVENT_LOOP_END, mData, &loopsRemaining);
303 }
304 void onMarker(uint32_t markerPosition) override {
305 mCallback(AudioTrack::EVENT_MARKER, mData, &markerPosition);
306 }
307 void onNewPos(uint32_t newPos) override {
308 mCallback(AudioTrack::EVENT_NEW_POS, mData, &newPos);
309 }
310 void onBufferEnd() override {
311 mCallback(AudioTrack::EVENT_BUFFER_END, mData, nullptr);
312 }
313 void onNewIAudioTrack() override {
314 mCallback(AudioTrack::EVENT_NEW_IAUDIOTRACK, mData, nullptr);
315 }
316 void onStreamEnd() override {
317 mCallback(AudioTrack::EVENT_STREAM_END, mData, nullptr);
318 }
319 size_t onCanWriteMoreData(const AudioTrack::Buffer & buffer) override {
320 AudioTrack::Buffer copy = buffer;
321 mCallback(AudioTrack::EVENT_CAN_WRITE_MORE_DATA, mData, static_cast<void*>(&copy));
322 return copy.size;
323 }
324 };
325}
326
327AudioTrack::AudioTrack(
328 audio_stream_type_t streamType,
329 uint32_t sampleRate,
330 audio_format_t format,
331 audio_channel_mask_t channelMask,
332 size_t frameCount,
333 audio_output_flags_t flags,
334 legacy_callback_t callback,
335 void* user,
336 int32_t notificationFrames,
337 audio_session_t sessionId,
338 transfer_type transferType,
339 const audio_offload_info_t *offloadInfo,
340 const AttributionSourceState& attributionSource,
341 const audio_attributes_t* pAttributes,
342 bool doNotReconnect,
343 float maxRequiredSpeed,
344 audio_port_handle_t selectedDeviceId)
345 : mStatus(NO_INIT),
346 mState(STATE_STOPPED),
347 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
348 mPreviousSchedulingGroup(SP_DEFAULT),
349 mPausedPosition(0),
350 mAudioTrackCallback(new AudioTrackCallback())
351{
352 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
353 if (callback != nullptr) {
354 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
355 } else if (user) {
356 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
357 }
358 (void)set(streamType, sampleRate, format, channelMask,
359 frameCount, flags, mLegacyCallbackWrapper, notificationFrames,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000360 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
361 attributionSource, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800362}
363
Andreas Huberc8139852012-01-18 10:51:55 -0800364AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800365 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800366 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800367 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700368 audio_channel_mask_t channelMask,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800369 const sp<IMemory>& sharedBuffer,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700370 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400371 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700372 int32_t notificationFrames,
Glenn Kastend848eb42016-03-08 13:42:11 -0800373 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000374 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800375 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000376 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700377 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700378 bool doNotReconnect,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700379 float maxRequiredSpeed)
Glenn Kasten87913512011-06-22 16:15:25 -0700380 : mStatus(NO_INIT),
Haynes Mathew George9b3359f2016-03-23 19:09:10 -0700381 mState(STATE_STOPPED),
John Grossman4ff14ba2012-02-08 16:37:41 -0800382 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800383 mPreviousSchedulingGroup(SP_DEFAULT),
Paul McLeanaa981192015-03-21 09:55:15 -0700384 mPausedPosition(0),
jiabinf6eb4c32020-02-25 14:06:25 -0800385 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
386 mAudioTrackCallback(new AudioTrackCallback())
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800387{
François Gaffie393f0e02019-04-10 09:09:08 +0200388 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Ippei Murata5fa32ed2018-10-01 17:37:50 +0900389
Eric Laurentf32d7812017-11-30 14:44:07 -0800390 (void)set(streamType, sampleRate, format, channelMask,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400391 0 /*frameCount*/, flags, callback, notificationFrames,
Marco Nelissend457c972014-02-11 08:47:07 -0800392 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000393 attributionSource, pAttributes, doNotReconnect, maxRequiredSpeed);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800394}
395
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400396AudioTrack::AudioTrack(
397 audio_stream_type_t streamType,
398 uint32_t sampleRate,
399 audio_format_t format,
400 audio_channel_mask_t channelMask,
401 const sp<IMemory>& sharedBuffer,
402 audio_output_flags_t flags,
403 legacy_callback_t callback,
404 void* user,
405 int32_t notificationFrames,
406 audio_session_t sessionId,
407 transfer_type transferType,
408 const audio_offload_info_t *offloadInfo,
409 const AttributionSourceState& attributionSource,
410 const audio_attributes_t* pAttributes,
411 bool doNotReconnect,
412 float maxRequiredSpeed)
413 : mStatus(NO_INIT),
414 mState(STATE_STOPPED),
415 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
416 mPreviousSchedulingGroup(SP_DEFAULT),
417 mPausedPosition(0),
418 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
419 mAudioTrackCallback(new AudioTrackCallback())
420{
421 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
422 if (callback) {
423 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
424 } else if (user) {
425 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
426 }
427
428 (void)set(streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags,
429 mLegacyCallbackWrapper, notificationFrames, sharedBuffer,
430 false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, attributionSource,
431 pAttributes, doNotReconnect, maxRequiredSpeed);
432}
433
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800434AudioTrack::~AudioTrack()
435{
Ray Essicked304702017-12-12 14:00:57 -0800436 // pull together the numbers, before we clean up our structures
437 mMediaMetrics.gather(this);
438
Andy Hungb68f5eb2019-12-03 16:49:17 -0800439 mediametrics::LogItem(mMetricsId)
440 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DTOR)
Phil Burkd3813f32020-04-23 16:26:15 -0700441 .set(AMEDIAMETRICS_PROP_CALLERNAME,
442 mCallerName.empty()
Andy Hunga6b27032020-04-27 10:34:24 -0700443 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
Phil Burkd3813f32020-04-23 16:26:15 -0700444 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800445 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
446 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)mStatus)
447 .record();
448
Phil Burk7a9577c2021-03-12 20:12:11 +0000449 stopAndJoinCallbacks(); // checks mStatus
450
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800451 if (mStatus == NO_ERROR) {
Marco Nelissenf8880202014-11-14 07:58:25 -0800452 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten53cec222013-08-29 09:01:02 -0700453 mAudioTrack.clear();
Eric Laurent3bcffa12014-06-12 18:38:45 -0700454 mCblkMemory.clear();
455 mSharedBuffer.clear();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800456 IPCThreadState::self()->flushCommands();
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000457 pid_t clientPid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
Andy Hungfb8ede22018-09-12 19:03:24 -0700458 ALOGV("%s(%d), releasing session id %d from %d on behalf of %d",
Eric Laurent973db022018-11-20 14:54:31 -0800459 __func__, mPortId,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700460 mSessionId, IPCThreadState::self()->getCallingPid(), clientPid);
461 AudioSystem::releaseAudioSessionId(mSessionId, clientPid);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800462 }
463}
464
Phil Burk7a9577c2021-03-12 20:12:11 +0000465void AudioTrack::stopAndJoinCallbacks() {
466 // Prevent nullptr crash if it did not open properly.
467 if (mStatus != NO_ERROR) return;
468
469 // Make sure that callback function exits in the case where
470 // it is looping on buffer full condition in obtainBuffer().
471 // Otherwise the callback thread will never exit.
472 stop();
473 if (mAudioTrackThread != 0) { // not thread safe
Phil Burk7a9577c2021-03-12 20:12:11 +0000474 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
Kuowei Lifdf8e0f2021-11-26 17:38:34 +0800475 mProxy->interrupt();
Phil Burk7a9577c2021-03-12 20:12:11 +0000476 mAudioTrackThread->requestExitAndWait();
477 mAudioTrackThread.clear();
478 }
479 // No lock here: worst case we remove a NULL callback which will be a nop
480 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) {
481 // This may not stop all of these device callbacks!
482 // TODO: Add some sort of protection.
483 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
484 mDeviceCallback.clear();
485 }
486}
487
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800488status_t AudioTrack::set(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800489 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800490 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800491 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700492 audio_channel_mask_t channelMask,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800493 size_t frameCount,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700494 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400495 legacy_callback_t callback,
496 void * user,
497 int32_t notificationFrames,
498 const sp<IMemory>& sharedBuffer,
499 bool threadCanCallJava,
500 audio_session_t sessionId,
501 transfer_type transferType,
502 const audio_offload_info_t *offloadInfo,
503 const AttributionSourceState& attributionSource,
504 const audio_attributes_t* pAttributes,
505 bool doNotReconnect,
506 float maxRequiredSpeed,
507 audio_port_handle_t selectedDeviceId)
508{
509 if (callback) {
510 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
511 } else if (user) {
512 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
513 }
514 return set(streamType, sampleRate,format, channelMask, frameCount, flags,
515 mLegacyCallbackWrapper, notificationFrames, sharedBuffer, threadCanCallJava,
516 sessionId, transferType, offloadInfo, attributionSource, pAttributes,
517 doNotReconnect, maxRequiredSpeed, selectedDeviceId);
518}
519status_t AudioTrack::set(
520 audio_stream_type_t streamType,
521 uint32_t sampleRate,
522 audio_format_t format,
523 audio_channel_mask_t channelMask,
524 size_t frameCount,
525 audio_output_flags_t flags,
526 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700527 int32_t notificationFrames,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800528 const sp<IMemory>& sharedBuffer,
Eric Laurentbe916aa2010-06-01 23:49:17 -0700529 bool threadCanCallJava,
Glenn Kastend848eb42016-03-08 13:42:11 -0800530 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000531 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800532 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000533 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700534 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700535 bool doNotReconnect,
jiabin156c6872017-10-06 09:47:15 -0700536 float maxRequiredSpeed,
537 audio_port_handle_t selectedDeviceId)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800538{
Eric Laurentf32d7812017-11-30 14:44:07 -0800539 status_t status;
540 uint32_t channelCount;
541 pid_t callingPid;
542 pid_t myPid;
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000543 uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
544 pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400545 sp<IAudioTrackCallback> _callback = callback.promote();
Andy Hung3acde2c2021-11-11 09:18:08 -0800546 std::string errorMessage;
Eric Laurent973db022018-11-20 14:54:31 -0800547 // Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
Andy Hungfb8ede22018-09-12 19:03:24 -0700548 ALOGV("%s(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
Glenn Kastenea38ee72016-04-18 11:08:01 -0700549 "flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
Andy Hungfb8ede22018-09-12 19:03:24 -0700550 __func__,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800551 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000552 sessionId, transferType, attributionSource.uid, attributionSource.pid);
Glenn Kasten86f04662014-02-24 15:13:05 -0800553
Phil Burk33ff89b2015-11-30 11:16:01 -0800554 mThreadCanCallJava = threadCanCallJava;
Andy Hungde602302021-12-07 21:35:49 -0800555
556 // These variables are pulled in an error report, so we initialize them early.
jiabin156c6872017-10-06 09:47:15 -0700557 mSelectedDeviceId = selectedDeviceId;
Eric Laurent21da6472017-11-09 16:29:26 -0800558 mSessionId = sessionId;
Andy Hungde602302021-12-07 21:35:49 -0800559 mChannelMask = channelMask;
560 mFormat = format;
561 mOrigFlags = mFlags = flags;
562 mReqFrameCount = mFrameCount = frameCount;
563 mSampleRate = sampleRate;
564 mOriginalSampleRate = sampleRate;
565 mAttributes = pAttributes != nullptr ? *pAttributes : AUDIO_ATTRIBUTES_INITIALIZER;
566 mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
Phil Burk33ff89b2015-11-30 11:16:01 -0800567
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800568 switch (transferType) {
569 case TRANSFER_DEFAULT:
570 if (sharedBuffer != 0) {
571 transferType = TRANSFER_SHARED;
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400572 } else if (_callback == nullptr|| threadCanCallJava) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800573 transferType = TRANSFER_SYNC;
574 } else {
575 transferType = TRANSFER_CALLBACK;
576 }
577 break;
578 case TRANSFER_CALLBACK:
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700579 case TRANSFER_SYNC_NOTIF_CALLBACK:
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400580 if (_callback == nullptr || sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800581 errorMessage = StringPrintf(
582 "%s: Transfer type %s but callback == nullptr || sharedBuffer != 0",
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700583 convertTransferToText(transferType), __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800584 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800585 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800586 }
587 break;
588 case TRANSFER_OBTAIN:
589 case TRANSFER_SYNC:
590 if (sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800591 errorMessage = StringPrintf(
592 "%s: Transfer type TRANSFER_OBTAIN but sharedBuffer != 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800593 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800594 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800595 }
596 break;
597 case TRANSFER_SHARED:
598 if (sharedBuffer == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800599 errorMessage = StringPrintf(
600 "%s: Transfer type TRANSFER_SHARED but sharedBuffer == 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800601 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800602 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800603 }
604 break;
605 default:
Andy Hung3acde2c2021-11-11 09:18:08 -0800606 errorMessage = StringPrintf("%s: Invalid transfer type %d", __func__, transferType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800607 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800608 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800609 }
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800610 mSharedBuffer = sharedBuffer;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800611 mTransfer = transferType;
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700612 mDoNotReconnect = doNotReconnect;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800613
Andy Hungfb8ede22018-09-12 19:03:24 -0700614 ALOGV_IF(sharedBuffer != 0, "%s(): sharedBuffer: %p, size: %zu",
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -0700615 __func__, sharedBuffer->unsecurePointer(), sharedBuffer->size());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800616
Andy Hungfb8ede22018-09-12 19:03:24 -0700617 ALOGV("%s(): streamType %d frameCount %zu flags %04x",
618 __func__, streamType, frameCount, flags);
Eric Laurent1a9ed112012-03-20 18:36:01 -0700619
Glenn Kasten53cec222013-08-29 09:01:02 -0700620 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Eric Laurent1dd70b92009-04-21 07:56:33 -0700621 if (mAudioTrack != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800622 errorMessage = StringPrintf("%s: Track already in use", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800623 status = INVALID_OPERATION;
Andy Hung3acde2c2021-11-11 09:18:08 -0800624 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800625 }
626
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800627 // handle default values first.
Eric Laurente83b55d2014-11-14 10:06:21 -0800628 if (streamType == AUDIO_STREAM_DEFAULT) {
Dima Zavinfce7a472011-04-19 22:30:36 -0700629 streamType = AUDIO_STREAM_MUSIC;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800630 }
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700631 if (pAttributes == NULL) {
Eric Laurent223fd5c2014-11-11 13:43:36 -0800632 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800633 errorMessage = StringPrintf("%s: Invalid stream type %d", __func__, streamType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800634 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800635 goto error;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700636 }
Andy Hunga2159aa2021-07-20 13:01:52 -0700637 mOriginalStreamType = streamType;
Eric Laurente83b55d2014-11-14 10:06:21 -0800638
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700639 } else {
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700640 // stream type shouldn't be looked at, this track has audio attributes
Andy Hungfb8ede22018-09-12 19:03:24 -0700641 ALOGV("%s(): Building AudioTrack with attributes:"
642 " usage=%d content=%d flags=0x%x tags=[%s]",
643 __func__,
644 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
Andy Hunga2159aa2021-07-20 13:01:52 -0700645 mOriginalStreamType = AUDIO_STREAM_DEFAULT;
François Gaffie58d4be52018-11-06 15:30:12 +0100646 audio_flags_to_audio_output_flags(mAttributes.flags, &flags);
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800647 }
Glenn Kastenea7939a2012-03-14 12:56:26 -0700648
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800649 // these below should probably come from the audioFlinger too...
Glenn Kastene1c39622012-01-04 09:36:37 -0800650 if (format == AUDIO_FORMAT_DEFAULT) {
Dima Zavinfce7a472011-04-19 22:30:36 -0700651 format = AUDIO_FORMAT_PCM_16_BIT;
Phil Burkfdb3c072016-02-09 10:47:02 -0800652 } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
Mikhail Naganov55773032020-10-01 15:08:13 -0700653 flags = static_cast<audio_output_flags_t>(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800654 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800655
656 // validate parameters
Dima Zavinfce7a472011-04-19 22:30:36 -0700657 if (!audio_is_valid_format(format)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800658 errorMessage = StringPrintf("%s: Invalid format %#x", __func__, format);
Eric Laurentf32d7812017-11-30 14:44:07 -0800659 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800660 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800661 }
Eric Laurentc2f1f072009-07-17 12:17:14 -0700662
Glenn Kasten8ba90322013-10-30 11:29:27 -0700663 if (!audio_is_output_channel(channelMask)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800664 errorMessage = StringPrintf("%s: Invalid channel mask %#x", __func__, channelMask);
Eric Laurentf32d7812017-11-30 14:44:07 -0800665 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800666 goto error;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700667 }
Eric Laurentf32d7812017-11-30 14:44:07 -0800668 channelCount = audio_channel_count_from_out_mask(channelMask);
Glenn Kastene3247bf2014-02-24 15:19:07 -0800669 mChannelCount = channelCount;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700670
Eric Laurentc2f1f072009-07-17 12:17:14 -0700671 // force direct flag if format is not linear PCM
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100672 // or offload was requested
673 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
674 || !audio_is_linear_pcm(format)) {
675 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
Andy Hungfb8ede22018-09-12 19:03:24 -0700676 ? "%s(): Offload request, forcing to Direct Output"
677 : "%s(): Not linear PCM, forcing to Direct Output",
678 __func__);
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700679 flags = (audio_output_flags_t)
Glenn Kasten3acbd052012-02-28 10:39:56 -0800680 // FIXME why can't we allow direct AND fast?
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700681 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
Eric Laurentc2f1f072009-07-17 12:17:14 -0700682 }
683
Eric Laurentd1f69b02014-12-15 14:33:13 -0800684 // force direct flag if HW A/V sync requested
685 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
686 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
687 }
688
Glenn Kastenb7730382014-04-30 15:50:31 -0700689 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
Phil Burkfdb3c072016-02-09 10:47:02 -0800690 if (audio_has_proportional_frames(format)) {
Glenn Kastenb7730382014-04-30 15:50:31 -0700691 mFrameSize = channelCount * audio_bytes_per_sample(format);
692 } else {
693 mFrameSize = sizeof(uint8_t);
694 }
Glenn Kastene3aa6592012-12-04 12:22:46 -0800695 } else {
Phil Burkfdb3c072016-02-09 10:47:02 -0800696 ALOG_ASSERT(audio_has_proportional_frames(format));
Glenn Kastenb7730382014-04-30 15:50:31 -0700697 mFrameSize = channelCount * audio_bytes_per_sample(format);
Glenn Kastenb7730382014-04-30 15:50:31 -0700698 // createTrack will return an error if PCM format is not supported by server,
699 // so no need to check for specific PCM formats here
Glenn Kastene3aa6592012-12-04 12:22:46 -0800700 }
701
Eric Laurent0d6db582014-11-12 18:39:44 -0800702 // sampling rate must be specified for direct outputs
703 if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800704 errorMessage = StringPrintf(
705 "%s: sample rate must be specified for direct outputs", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800706 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800707 goto error;
Eric Laurent0d6db582014-11-12 18:39:44 -0800708 }
Andy Hungff874dc2016-04-11 16:49:09 -0700709 // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
710 mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
Eric Laurent0d6db582014-11-12 18:39:44 -0800711
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800712 // Make copy of input parameter offloadInfo so that in the future:
713 // (a) createTrack_l doesn't need it as an input parameter
714 // (b) we can support re-creation of offloaded tracks
715 if (offloadInfo != NULL) {
716 mOffloadInfoCopy = *offloadInfo;
717 mOffloadInfo = &mOffloadInfoCopy;
718 } else {
719 mOffloadInfo = NULL;
Eric Laurent20b9ef02016-12-05 11:03:16 -0800720 memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
Ytai Ben-Tsviffa2fd92020-10-20 09:13:53 -0700721 mOffloadInfoCopy = AUDIO_INFO_INITIALIZER;
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800722 }
723
Glenn Kasten66e46352014-01-16 17:44:23 -0800724 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
725 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
Glenn Kasten05632a52012-01-03 14:22:33 -0800726 mSendLevel = 0.0f;
Glenn Kasten396fabd2014-01-08 08:54:23 -0800727 // mFrameCount is initialized in createTrack_l
Glenn Kastenea38ee72016-04-18 11:08:01 -0700728 if (notificationFrames >= 0) {
729 mNotificationFramesReq = notificationFrames;
730 mNotificationsPerBufferReq = 0;
731 } else {
732 if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800733 errorMessage = StringPrintf(
734 "%s: notificationFrames=%d not permitted for non-fast track",
Andy Hungfb8ede22018-09-12 19:03:24 -0700735 __func__, notificationFrames);
Eric Laurentf32d7812017-11-30 14:44:07 -0800736 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800737 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700738 }
739 if (frameCount > 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700740 ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
741 __func__, notificationFrames, frameCount);
Eric Laurentf32d7812017-11-30 14:44:07 -0800742 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800743 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700744 }
745 mNotificationFramesReq = 0;
746 const uint32_t minNotificationsPerBuffer = 1;
747 const uint32_t maxNotificationsPerBuffer = 8;
748 mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
749 max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
750 ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
Andy Hungfb8ede22018-09-12 19:03:24 -0700751 "%s(): notificationFrames=%d clamped to the range -%u to -%u",
752 __func__,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700753 notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
754 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800755 mNotificationFramesAct = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700756 // TODO b/182392553: refactor or remove
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000757 mClientAttributionSource = AttributionSourceState(attributionSource);
Eric Laurentf32d7812017-11-30 14:44:07 -0800758 callingPid = IPCThreadState::self()->getCallingPid();
759 myPid = getpid();
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700760 if (uid == -1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000761 mClientAttributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700762 IPCThreadState::self()->getCallingUid()));
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800763 }
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700764 if (pid == (pid_t)-1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000765 mClientAttributionSource.pid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(callingPid));
Marco Nelissend457c972014-02-11 08:47:07 -0800766 }
Eric Laurent2beeb502010-07-16 07:43:46 -0700767 mAuxEffectId = 0;
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400768 mCallback = callback;
Eric Laurentbe916aa2010-06-01 23:49:17 -0700769
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400770 if (_callback != nullptr) {
771 mAudioTrackThread = sp<AudioTrackThread>::make(*this);
Glenn Kastena997e7a2012-08-07 09:44:19 -0700772 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
Glenn Kastenbfd31842015-03-20 09:01:44 -0700773 // thread begins in paused state, and will not reference us until start()
Glenn Kastena997e7a2012-08-07 09:44:19 -0700774 }
775
Eric Laurent34f1d8e2009-11-04 08:27:26 -0800776 // create the IAudioTrack
Francois Gaffie24a9fb02019-01-18 17:51:34 +0100777 {
778 AutoMutex lock(mLock);
779 status = createTrack_l();
780 }
Glenn Kastena997e7a2012-08-07 09:44:19 -0700781 if (status != NO_ERROR) {
782 if (mAudioTrackThread != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100783 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
784 mAudioTrackThread->requestExitAndWait();
Glenn Kastena997e7a2012-08-07 09:44:19 -0700785 mAudioTrackThread.clear();
786 }
Andy Hung3acde2c2021-11-11 09:18:08 -0800787 // We do not goto error to prevent double-logging errors.
Eric Laurentf32d7812017-11-30 14:44:07 -0800788 goto exit;
Glenn Kasten5d464eb2012-06-22 17:19:53 -0700789 }
790
Andy Hung4ede21d2014-12-12 15:37:34 -0800791 mLoopCount = 0;
792 mLoopStart = 0;
793 mLoopEnd = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -0800794 mLoopCountNotified = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800795 mMarkerPosition = 0;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -0700796 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800797 mNewPosition = 0;
798 mUpdatePeriod = 0;
Glenn Kasten200092b2014-08-15 15:13:30 -0700799 mPosition = 0;
800 mReleased = 0;
Andy Hungffa36952017-08-17 10:41:51 -0700801 mStartNs = 0;
802 mStartFromZeroUs = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700803 AudioSystem::acquireAudioSessionId(mSessionId, pid, uid);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800804 mSequence = 1;
805 mObservedSequence = mSequence;
806 mInUnderrun = false;
Phil Burk1b420972015-04-22 10:52:21 -0700807 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700808 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700809 mTimestampRetrogradePositionReported = false;
810 mTimestampRetrogradeTimeReported = false;
811 mTimestampStallReported = false;
812 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700813 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Andy Hung65ffdfc2016-10-10 15:52:11 -0700814 mStartTs.mPosition = 0;
Phil Burk2812d9e2016-01-04 10:34:30 -0800815 mUnderrunCountOffset = 0;
Andy Hungea2b9c02016-02-12 17:06:53 -0800816 mFramesWritten = 0;
817 mFramesWrittenServerOffset = 0;
Andy Hungf20a4e92016-08-15 19:10:34 -0700818 mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
Ivan Lozano8cf3a072017-08-09 09:01:33 -0700819 mVolumeHandler = new media::VolumeHandler();
Eric Laurentf32d7812017-11-30 14:44:07 -0800820
Andy Hung3acde2c2021-11-11 09:18:08 -0800821error:
822 if (status != NO_ERROR) {
823 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
824 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
825 }
826 // fall through
Eric Laurentf32d7812017-11-30 14:44:07 -0800827exit:
828 mStatus = status;
829 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800830}
831
Mikhail Naganov55773032020-10-01 15:08:13 -0700832
833status_t AudioTrack::set(
834 audio_stream_type_t streamType,
835 uint32_t sampleRate,
836 audio_format_t format,
837 uint32_t channelMask,
838 size_t frameCount,
839 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400840 legacy_callback_t callback,
Mikhail Naganov55773032020-10-01 15:08:13 -0700841 void* user,
842 int32_t notificationFrames,
843 const sp<IMemory>& sharedBuffer,
844 bool threadCanCallJava,
845 audio_session_t sessionId,
846 transfer_type transferType,
847 const audio_offload_info_t *offloadInfo,
848 uid_t uid,
849 pid_t pid,
850 const audio_attributes_t* pAttributes,
851 bool doNotReconnect,
852 float maxRequiredSpeed,
853 audio_port_handle_t selectedDeviceId)
854{
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000855 AttributionSourceState attributionSource;
856 attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(uid));
857 attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(pid));
858 attributionSource.token = sp<BBinder>::make();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400859 if (callback) {
860 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
861 } else if (user) {
862 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
863 }
864 return set(streamType, sampleRate, format, static_cast<audio_channel_mask_t>(channelMask),
865 frameCount, flags, mLegacyCallbackWrapper, notificationFrames, sharedBuffer,
866 threadCanCallJava, sessionId, transferType, offloadInfo, attributionSource,
867 pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
Mikhail Naganov55773032020-10-01 15:08:13 -0700868}
869
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800870// -------------------------------------------------------------------------
871
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100872status_t AudioTrack::start()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800873{
Eric Laurentf5aafb22010-11-18 08:40:16 -0800874 AutoMutex lock(mLock);
Andy Hungb68f5eb2019-12-03 16:49:17 -0800875
Andy Hung10fb4be2020-05-27 22:22:22 -0700876 if (mState == STATE_ACTIVE) {
877 return INVALID_OPERATION;
878 }
879
880 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
881
882 // Defer logging here due to OpenSL ES repeated start calls.
883 // TODO(b/154868033) after fix, restore this logging back to the beginning of start().
884 const int64_t beginNs = systemTime();
Andy Hungb68f5eb2019-12-03 16:49:17 -0800885 status_t status = NO_ERROR; // logged: make sure to set this before returning.
Andy Hung06a730b2020-04-09 13:28:31 -0700886 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800887 mediametrics::LogItem(mMetricsId)
Andy Hunga6b27032020-04-27 10:34:24 -0700888 .set(AMEDIAMETRICS_PROP_CALLERNAME,
889 mCallerName.empty()
890 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
891 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800892 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_START)
Andy Hungea840382020-05-05 21:50:17 -0700893 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800894 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
895 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
896 .record(); });
897
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800898
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800899 mInUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800900
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800901 State previousState = mState;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100902 if (previousState == STATE_PAUSED_STOPPING) {
903 mState = STATE_STOPPING;
904 } else {
905 mState = STATE_ACTIVE;
906 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700907 (void) updateAndGetPosition_l();
Andy Hung65ffdfc2016-10-10 15:52:11 -0700908
909 // save start timestamp
910 if (isOffloadedOrDirect_l()) {
911 if (getTimestamp_l(mStartTs) != OK) {
912 mStartTs.mPosition = 0;
913 }
914 } else {
915 if (getTimestamp_l(&mStartEts) != OK) {
916 mStartEts.clear();
917 }
918 }
Andy Hungffa36952017-08-17 10:41:51 -0700919 mStartNs = systemTime(); // save this for timestamp adjustment after starting.
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800920 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
921 // reset current position as seen by client to 0
Glenn Kasten200092b2014-08-15 15:13:30 -0700922 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -0700923 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700924 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700925 mTimestampRetrogradePositionReported = false;
926 mTimestampRetrogradeTimeReported = false;
927 mTimestampStallReported = false;
928 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700929 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Phil Burk1b420972015-04-22 10:52:21 -0700930
Andy Hung65ffdfc2016-10-10 15:52:11 -0700931 if (!isOffloadedOrDirect_l()
932 && mStartEts.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) {
Andy Hunge1e98462016-04-12 10:18:51 -0700933 // Server side has consumed something, but is it finished consuming?
934 // It is possible since flush and stop are asynchronous that the server
935 // is still active at this point.
Andy Hungfb8ede22018-09-12 19:03:24 -0700936 ALOGV("%s(%d): server read:%lld cumulative flushed:%lld client written:%lld",
Eric Laurent973db022018-11-20 14:54:31 -0800937 __func__, mPortId,
Andy Hunge1e98462016-04-12 10:18:51 -0700938 (long long)(mFramesWrittenServerOffset
Andy Hung65ffdfc2016-10-10 15:52:11 -0700939 + mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]),
940 (long long)mStartEts.mFlushed,
Andy Hunge1e98462016-04-12 10:18:51 -0700941 (long long)mFramesWritten);
Andy Hungc4e60eb2017-09-06 11:14:57 -0700942 // mStartEts is already adjusted by mFramesWrittenServerOffset, so we delta adjust.
943 mFramesWrittenServerOffset -= mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER];
Andy Hung61be8412015-10-06 10:51:09 -0700944 }
Andy Hunge1e98462016-04-12 10:18:51 -0700945 mFramesWritten = 0;
946 mProxy->clearTimestamp(); // need new server push for valid timestamp
947 mMarkerReached = false;
Andy Hung61be8412015-10-06 10:51:09 -0700948
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700949 // For offloaded tracks, we don't know if the hardware counters are really zero here,
950 // since the flush is asynchronous and stop may not fully drain.
951 // We save the time when the track is started to later verify whether
952 // the counters are realistic (i.e. start from zero after this time).
Andy Hungffa36952017-08-17 10:41:51 -0700953 mStartFromZeroUs = mStartNs / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700954
Eric Laurentec9a0322013-08-28 10:23:01 -0700955 // force refresh of remaining frames by processAudioBuffer() as last
956 // write before stop could be partial.
957 mRefreshRemaining = true;
Tomoharu Kasahara6e8bf982017-06-09 16:17:33 +0900958
959 // for static track, clear the old flags when starting from stopped state
960 if (mSharedBuffer != 0) {
961 android_atomic_and(
962 ~(CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
963 &mCblk->mFlags);
964 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800965 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700966 mNewPosition = mPosition + mUpdatePeriod;
Andy Hung4be3b832016-10-13 17:51:43 -0700967 int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800968
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800969 if (!(flags & CBLK_INVALID)) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -0800970 mAudioTrack->start(&status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800971 if (status == DEAD_OBJECT) {
972 flags |= CBLK_INVALID;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800973 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800974 }
975 if (flags & CBLK_INVALID) {
976 status = restoreTrack_l("start");
977 }
978
Andy Hung79629f02016-03-24 13:57:40 -0700979 // resume or pause the callback thread as needed.
980 sp<AudioTrackThread> t = mAudioTrackThread;
981 if (status == NO_ERROR) {
982 if (t != 0) {
983 if (previousState == STATE_STOPPING) {
984 mProxy->interrupt();
985 } else {
986 t->resume();
987 }
988 } else {
989 mPreviousPriority = getpriority(PRIO_PROCESS, 0);
990 get_sched_policy(0, &mPreviousSchedulingGroup);
991 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
992 }
Andy Hung39399b62017-04-21 15:07:45 -0700993
994 // Start our local VolumeHandler for restoration purposes.
995 mVolumeHandler->setStarted();
Andy Hung79629f02016-03-24 13:57:40 -0700996 } else {
Eric Laurent973db022018-11-20 14:54:31 -0800997 ALOGE("%s(%d): status %d", __func__, mPortId, status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800998 mState = previousState;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800999 if (t != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001000 if (previousState != STATE_STOPPING) {
1001 t->pause();
1002 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001003 } else {
Glenn Kasten87913512011-06-22 16:15:25 -07001004 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
Glenn Kastena6364332012-04-19 09:35:04 -07001005 set_sched_policy(0, mPreviousSchedulingGroup);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001006 }
1007 }
1008
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001009 return status;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001010}
1011
1012void AudioTrack::stop()
1013{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001014 const int64_t beginNs = systemTime();
1015
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001016 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -07001017 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001018 mediametrics::LogItem(mMetricsId)
1019 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_STOP)
Andy Hungea840382020-05-05 21:50:17 -07001020 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001021 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Phil Burk64e16a72020-06-01 13:25:51 -07001022 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1023 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getUnderrunCount_l())
Phil Burka9876702020-04-20 18:16:15 -07001024 .record();
Phil Burka9876702020-04-20 18:16:15 -07001025 });
Andy Hungb68f5eb2019-12-03 16:49:17 -08001026
Eric Laurent973db022018-11-20 14:54:31 -08001027 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001028
Glenn Kasten397edb32013-08-30 15:10:13 -07001029 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001030 return;
1031 }
1032
Glenn Kasten23a75452014-01-13 10:37:17 -08001033 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001034 mState = STATE_STOPPING;
1035 } else {
1036 mState = STATE_STOPPED;
Andy Hung2148bf02016-11-28 19:01:02 -08001037 ALOGD_IF(mSharedBuffer == nullptr,
Eric Laurent973db022018-11-20 14:54:31 -08001038 "%s(%d): called with %u frames delivered", __func__, mPortId, mReleased.value());
Andy Hungc2813e52014-10-16 17:54:34 -07001039 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001040 }
1041
Andy Hung1d3556d2018-03-29 16:30:14 -07001042 mProxy->stop(); // notify server not to read beyond current client position until start().
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001043 mProxy->interrupt();
1044 mAudioTrack->stop();
Andy Hung61be8412015-10-06 10:51:09 -07001045
1046 // Note: legacy handling - stop does not clear playback marker
1047 // and periodic update counter, but flush does for streaming tracks.
Andy Hung9b461582014-12-01 17:56:29 -08001048
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001049 if (mSharedBuffer != 0) {
Andy Hung9b461582014-12-01 17:56:29 -08001050 // clear buffer position and loop count.
Andy Hung9b461582014-12-01 17:56:29 -08001051 mStaticProxy->setBufferPositionAndLoop(0 /* position */,
1052 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001053 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001054
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001055 sp<AudioTrackThread> t = mAudioTrackThread;
1056 if (t != 0) {
Glenn Kasten23a75452014-01-13 10:37:17 -08001057 if (!isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001058 t->pause();
Jean-Michel Trivia60e9a22018-11-15 16:13:36 -08001059 } else if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
Jean-Michel Trivi0248a2a2018-11-21 10:05:57 -08001060 // causes wake up of the playback thread, that will callback the client for
1061 // EVENT_STREAM_END in processAudioBuffer()
1062 t->wake();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001063 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001064 } else {
1065 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
1066 set_sched_policy(0, mPreviousSchedulingGroup);
1067 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001068}
1069
1070bool AudioTrack::stopped() const
1071{
Glenn Kasten9a2aaf92012-01-03 09:42:47 -08001072 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001073 return mState != STATE_ACTIVE;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001074}
1075
1076void AudioTrack::flush()
1077{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001078 const int64_t beginNs = systemTime();
Andy Hungfb8ede22018-09-12 19:03:24 -07001079 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -07001080 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001081 mediametrics::LogItem(mMetricsId)
1082 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_FLUSH)
Andy Hungea840382020-05-05 21:50:17 -07001083 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001084 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
1085 .record(); });
1086
Eric Laurent973db022018-11-20 14:54:31 -08001087 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001088
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001089 if (mSharedBuffer != 0) {
1090 return;
Glenn Kasten4bae3642012-11-30 13:41:12 -08001091 }
Andy Hung4c5ed302018-05-09 11:16:21 -07001092 if (mState == STATE_ACTIVE) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001093 return;
1094 }
1095 flush_l();
Eric Laurent1703cdf2011-03-07 14:52:59 -08001096}
1097
Eric Laurent1703cdf2011-03-07 14:52:59 -08001098void AudioTrack::flush_l()
1099{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001100 ALOG_ASSERT(mState != STATE_ACTIVE);
Eric Laurentc2f1f072009-07-17 12:17:14 -07001101
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001102 // clear playback marker and periodic update counter
1103 mMarkerPosition = 0;
1104 mMarkerReached = false;
1105 mUpdatePeriod = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001106 mRefreshRemaining = true;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001107
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001108 mState = STATE_FLUSHED;
Andy Hungc2813e52014-10-16 17:54:34 -07001109 mReleased = 0;
Glenn Kasten23a75452014-01-13 10:37:17 -08001110 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001111 mProxy->interrupt();
1112 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001113 mProxy->flush();
Glenn Kasten4bae3642012-11-30 13:41:12 -08001114 mAudioTrack->flush();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001115}
1116
Andy Hung959b5b82021-09-24 10:46:20 -07001117bool AudioTrack::pauseAndWait(const std::chrono::milliseconds& timeout)
1118{
1119 using namespace std::chrono_literals;
1120
1121 pause();
1122
1123 AutoMutex lock(mLock);
1124 // offload and direct tracks do not wait because pause volume ramp is handled by hardware.
1125 if (isOffloadedOrDirect_l()) return true;
1126
1127 // Wait for the track state to be anything besides pausing.
1128 // This ensures that the volume has ramped down.
1129 constexpr auto SLEEP_INTERVAL_MS = 10ms;
1130 auto begin = std::chrono::steady_clock::now();
1131 while (true) {
1132 // wait for state to change
1133 const int state = mProxy->getState();
1134
1135 mLock.unlock(); // only local variables accessed until lock.
1136 auto elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(
1137 std::chrono::steady_clock::now() - begin);
1138 if (state != CBLK_STATE_PAUSING) {
1139 ALOGV("%s: success state:%d after %lld ms", __func__, state, elapsed.count());
1140 return true;
1141 }
1142 std::chrono::milliseconds remaining = timeout - elapsed;
1143 if (remaining.count() <= 0) {
1144 ALOGW("%s: timeout expired state:%d still pausing:%d after %lld ms",
1145 __func__, state, CBLK_STATE_PAUSING, elapsed.count());
1146 return false;
1147 }
1148 // It is conceivable that the track is restored while sleeping;
1149 // as this logic is advisory, we allow that.
1150 std::this_thread::sleep_for(std::min(remaining, SLEEP_INTERVAL_MS));
1151 mLock.lock();
1152 }
1153}
1154
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001155void AudioTrack::pause()
1156{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001157 const int64_t beginNs = systemTime();
Eric Laurentf5aafb22010-11-18 08:40:16 -08001158 AutoMutex lock(mLock);
Andy Hunga6b27032020-04-27 10:34:24 -07001159 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001160 mediametrics::LogItem(mMetricsId)
1161 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_PAUSE)
Andy Hungea840382020-05-05 21:50:17 -07001162 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001163 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
1164 .record(); });
1165
Eric Laurent973db022018-11-20 14:54:31 -08001166 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001167
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001168 if (mState == STATE_ACTIVE) {
1169 mState = STATE_PAUSED;
1170 } else if (mState == STATE_STOPPING) {
1171 mState = STATE_PAUSED_STOPPING;
1172 } else {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001173 return;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001174 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001175 mProxy->interrupt();
1176 mAudioTrack->pause();
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001177
Marco Nelissen3a90f282014-03-10 11:21:43 -07001178 if (isOffloaded_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001179 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001180 // An offload output can be re-used between two audio tracks having
1181 // the same configuration. A timestamp query for a paused track
1182 // while the other is running would return an incorrect time.
1183 // To fix this, cache the playback position on a pause() and return
1184 // this time when requested until the track is resumed.
1185
1186 // OffloadThread sends HAL pause in its threadLoop. Time saved
1187 // here can be slightly off.
1188
1189 // TODO: check return code for getRenderPosition.
1190
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001191 uint32_t halFrames;
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001192 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition);
Andy Hungfb8ede22018-09-12 19:03:24 -07001193 ALOGV("%s(%d): for offload, cache current position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001194 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001195 }
1196 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001197}
1198
Eric Laurentbe916aa2010-06-01 23:49:17 -07001199status_t AudioTrack::setVolume(float left, float right)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001200{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001201 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1202 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY ||
1203 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001204 return BAD_VALUE;
1205 }
1206
Andy Hungb68f5eb2019-12-03 16:49:17 -08001207 mediametrics::LogItem(mMetricsId)
1208 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOLUME)
1209 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)left)
1210 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)right)
1211 .record();
1212
Eric Laurent1703cdf2011-03-07 14:52:59 -08001213 AutoMutex lock(mLock);
Glenn Kasten66e46352014-01-16 17:44:23 -08001214 mVolume[AUDIO_INTERLEAVE_LEFT] = left;
1215 mVolume[AUDIO_INTERLEAVE_RIGHT] = right;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001216
Glenn Kastenc56f3422014-03-21 17:53:17 -07001217 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right)));
Eric Laurentbe916aa2010-06-01 23:49:17 -07001218
Glenn Kasten23a75452014-01-13 10:37:17 -08001219 if (isOffloaded_l()) {
Eric Laurent59fe0102013-09-27 18:48:26 -07001220 mAudioTrack->signal();
1221 }
Eric Laurentbe916aa2010-06-01 23:49:17 -07001222 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001223}
1224
Glenn Kastenb1c09932012-02-27 16:21:04 -08001225status_t AudioTrack::setVolume(float volume)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001226{
Glenn Kastenb1c09932012-02-27 16:21:04 -08001227 return setVolume(volume, volume);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001228}
1229
Eric Laurent2beeb502010-07-16 07:43:46 -07001230status_t AudioTrack::setAuxEffectSendLevel(float level)
Eric Laurentbe916aa2010-06-01 23:49:17 -07001231{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001232 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1233 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001234 return BAD_VALUE;
1235 }
1236
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001237 AutoMutex lock(mLock);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001238 mSendLevel = level;
Glenn Kastene3aa6592012-12-04 12:22:46 -08001239 mProxy->setSendLevel(level);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001240
1241 return NO_ERROR;
1242}
1243
Glenn Kastena5224f32012-01-04 12:41:44 -08001244void AudioTrack::getAuxEffectSendLevel(float* level) const
Eric Laurentbe916aa2010-06-01 23:49:17 -07001245{
1246 if (level != NULL) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001247 *level = mSendLevel;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001248 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001249}
1250
Glenn Kasten3b16c762012-11-14 08:44:39 -08001251status_t AudioTrack::setSampleRate(uint32_t rate)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001252{
Andy Hung5cbb5782015-03-27 18:39:59 -07001253 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08001254 ALOGV("%s(%d): prior state:%s rate:%u", __func__, mPortId, stateToString(mState), rate);
Andy Hungfb8ede22018-09-12 19:03:24 -07001255
Andy Hung5cbb5782015-03-27 18:39:59 -07001256 if (rate == mSampleRate) {
1257 return NO_ERROR;
1258 }
jiabinf4de6112018-12-19 12:40:08 -08001259 if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)
1260 || (mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL)) {
John Grossman4ff14ba2012-02-08 16:37:41 -08001261 return INVALID_OPERATION;
1262 }
Eric Laurent0d6db582014-11-12 18:39:44 -08001263 if (mOutput == AUDIO_IO_HANDLE_NONE) {
1264 return NO_INIT;
1265 }
Andy Hung5cbb5782015-03-27 18:39:59 -07001266 // NOTE: it is theoretically possible, but highly unlikely, that a device change
1267 // could mean a previously allowed sampling rate is no longer allowed.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001268 uint32_t afSamplingRate;
Eric Laurent0d6db582014-11-12 18:39:44 -08001269 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
Eric Laurent57326622009-07-07 07:10:45 -07001270 return NO_INIT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001271 }
Andy Hung26145642015-04-15 21:56:53 -07001272 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001273 const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001274 if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001275 return BAD_VALUE;
1276 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001277 // TODO: Should we also check if the buffer size is compatible?
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001278
Glenn Kastene3aa6592012-12-04 12:22:46 -08001279 mSampleRate = rate;
Andy Hung26145642015-04-15 21:56:53 -07001280 mProxy->setSampleRate(effectiveSampleRate);
Glenn Kastene3aa6592012-12-04 12:22:46 -08001281
Eric Laurent57326622009-07-07 07:10:45 -07001282 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001283}
1284
Glenn Kastena5224f32012-01-04 12:41:44 -08001285uint32_t AudioTrack::getSampleRate() const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001286{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001287 AutoMutex lock(mLock);
Eric Laurent6f59db12013-07-26 17:16:50 -07001288
1289 // sample rate can be updated during playback by the offloaded decoder so we need to
1290 // query the HAL and update if needed.
1291// FIXME use Proxy return channel to update the rate from server and avoid polling here
Eric Laurentab5cdba2014-06-09 17:22:27 -07001292 if (isOffloadedOrDirect_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001293 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent6f59db12013-07-26 17:16:50 -07001294 uint32_t sampleRate = 0;
Jean-Michel Trivib7f24b12014-06-11 10:05:30 -07001295 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
Eric Laurent6f59db12013-07-26 17:16:50 -07001296 if (status == NO_ERROR) {
1297 mSampleRate = sampleRate;
1298 }
1299 }
1300 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001301 return mSampleRate;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001302}
1303
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001304uint32_t AudioTrack::getOriginalSampleRate() const
1305{
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001306 return mOriginalSampleRate;
1307}
1308
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001309status_t AudioTrack::setDualMonoMode(audio_dual_mono_mode_t mode)
1310{
1311 AutoMutex lock(mLock);
1312 return setDualMonoMode_l(mode);
1313}
1314
1315status_t AudioTrack::setDualMonoMode_l(audio_dual_mono_mode_t mode)
1316{
1317 const status_t status = statusTFromBinderStatus(
1318 mAudioTrack->setDualMonoMode(VALUE_OR_RETURN_STATUS(
1319 legacy2aidl_audio_dual_mono_mode_t_AudioDualMonoMode(mode))));
1320 if (status == NO_ERROR) mDualMonoMode = mode;
1321 return status;
1322}
1323
1324status_t AudioTrack::getDualMonoMode(audio_dual_mono_mode_t* mode) const
1325{
1326 AutoMutex lock(mLock);
1327 media::AudioDualMonoMode mediaMode;
1328 const status_t status = statusTFromBinderStatus(mAudioTrack->getDualMonoMode(&mediaMode));
1329 if (status == NO_ERROR) {
1330 *mode = VALUE_OR_RETURN_STATUS(
1331 aidl2legacy_AudioDualMonoMode_audio_dual_mono_mode_t(mediaMode));
1332 }
1333 return status;
1334}
1335
1336status_t AudioTrack::setAudioDescriptionMixLevel(float leveldB)
1337{
1338 AutoMutex lock(mLock);
1339 return setAudioDescriptionMixLevel_l(leveldB);
1340}
1341
1342status_t AudioTrack::setAudioDescriptionMixLevel_l(float leveldB)
1343{
1344 const status_t status = statusTFromBinderStatus(
1345 mAudioTrack->setAudioDescriptionMixLevel(leveldB));
1346 if (status == NO_ERROR) mAudioDescriptionMixLeveldB = leveldB;
1347 return status;
1348}
1349
1350status_t AudioTrack::getAudioDescriptionMixLevel(float* leveldB) const
1351{
1352 AutoMutex lock(mLock);
1353 return statusTFromBinderStatus(mAudioTrack->getAudioDescriptionMixLevel(leveldB));
1354}
1355
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001356status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
Andy Hung8edb8dc2015-03-26 19:13:55 -07001357{
Andy Hung8edb8dc2015-03-26 19:13:55 -07001358 AutoMutex lock(mLock);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001359 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
Andy Hung8edb8dc2015-03-26 19:13:55 -07001360 return NO_ERROR;
1361 }
Glenn Kastend79072e2016-01-06 08:41:20 -08001362 if (isOffloadedOrDirect_l()) {
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001363 const status_t status = statusTFromBinderStatus(mAudioTrack->setPlaybackRateParameters(
1364 VALUE_OR_RETURN_STATUS(
1365 legacy2aidl_audio_playback_rate_t_AudioPlaybackRate(playbackRate))));
1366 if (status == NO_ERROR) {
1367 mPlaybackRate = playbackRate;
1368 }
1369 return status;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001370 }
1371 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
1372 return INVALID_OPERATION;
1373 }
Andy Hungff874dc2016-04-11 16:49:09 -07001374
Andy Hungfb8ede22018-09-12 19:03:24 -07001375 ALOGV("%s(%d): mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001376 __func__, mPortId, mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001377 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001378 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
1379 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
1380 const float effectivePitch = adjustPitch(playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001381 AudioPlaybackRate playbackRateTemp = playbackRate;
1382 playbackRateTemp.mSpeed = effectiveSpeed;
1383 playbackRateTemp.mPitch = effectivePitch;
1384
Andy Hungfb8ede22018-09-12 19:03:24 -07001385 ALOGV("%s(%d) (effective) mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001386 __func__, mPortId, effectiveRate, effectiveSpeed, effectivePitch);
Andy Hungff874dc2016-04-11 16:49:09 -07001387
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001388 if (!isAudioPlaybackRateValid(playbackRateTemp)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001389 ALOGW("%s(%d) (%f, %f) failed (effective rate out of bounds)",
Eric Laurent973db022018-11-20 14:54:31 -08001390 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001391 return BAD_VALUE;
1392 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001393 // Check if the buffer size is compatible.
Andy Hung26145642015-04-15 21:56:53 -07001394 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001395 ALOGW("%s(%d) (%f, %f) failed (buffer size)",
Eric Laurent973db022018-11-20 14:54:31 -08001396 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung8edb8dc2015-03-26 19:13:55 -07001397 return BAD_VALUE;
1398 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001399
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001400 // Check resampler ratios are within bounds
Glenn Kastend3bb6452016-12-05 18:14:37 -08001401 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
1402 (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001403 ALOGW("%s(%d) (%f, %f) failed. Resample rate exceeds max accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001404 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001405 return BAD_VALUE;
1406 }
1407
Dan Austine34eae22015-10-27 16:14:52 -07001408 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001409 ALOGW("%s(%d) (%f, %f) failed. Resample rate below min accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001410 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001411 return BAD_VALUE;
1412 }
1413 mPlaybackRate = playbackRate;
1414 //set effective rates
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001415 mProxy->setPlaybackRate(playbackRateTemp);
Andy Hung26145642015-04-15 21:56:53 -07001416 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
Andy Hungb68f5eb2019-12-03 16:49:17 -08001417
1418 mediametrics::LogItem(mMetricsId)
1419 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYBACKPARAM)
1420 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
1421 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
1422 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
1423 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1424 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveRate)
1425 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1426 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)playbackRateTemp.mSpeed)
1427 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1428 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)playbackRateTemp.mPitch)
1429 .record();
1430
Andy Hung8edb8dc2015-03-26 19:13:55 -07001431 return NO_ERROR;
1432}
1433
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001434const AudioPlaybackRate& AudioTrack::getPlaybackRate()
Andy Hung8edb8dc2015-03-26 19:13:55 -07001435{
1436 AutoMutex lock(mLock);
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001437 if (isOffloadedOrDirect_l()) {
1438 media::AudioPlaybackRate playbackRateTemp;
1439 const status_t status = statusTFromBinderStatus(
1440 mAudioTrack->getPlaybackRateParameters(&playbackRateTemp));
1441 if (status == NO_ERROR) { // update local version if changed.
1442 mPlaybackRate =
1443 aidl2legacy_AudioPlaybackRate_audio_playback_rate_t(playbackRateTemp).value();
1444 }
1445 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001446 return mPlaybackRate;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001447}
1448
Phil Burkc0adecb2016-01-08 12:44:11 -08001449ssize_t AudioTrack::getBufferSizeInFrames()
1450{
1451 AutoMutex lock(mLock);
1452 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1453 return NO_INIT;
1454 }
Phil Burka9876702020-04-20 18:16:15 -07001455
Phil Burke8972b02016-03-04 11:29:57 -08001456 return (ssize_t) mProxy->getBufferSizeInFrames();
Phil Burkc0adecb2016-01-08 12:44:11 -08001457}
1458
Andy Hungf2c87b32016-04-07 19:49:29 -07001459status_t AudioTrack::getBufferDurationInUs(int64_t *duration)
1460{
1461 if (duration == nullptr) {
1462 return BAD_VALUE;
1463 }
1464 AutoMutex lock(mLock);
1465 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1466 return NO_INIT;
1467 }
1468 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames();
1469 if (bufferSizeInFrames < 0) {
1470 return (status_t)bufferSizeInFrames;
1471 }
1472 *duration = (int64_t)((double)bufferSizeInFrames * 1000000
1473 / ((double)mSampleRate * mPlaybackRate.mSpeed));
1474 return NO_ERROR;
1475}
1476
Phil Burkc0adecb2016-01-08 12:44:11 -08001477ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames)
1478{
1479 AutoMutex lock(mLock);
1480 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1481 return NO_INIT;
1482 }
Phil Burka9876702020-04-20 18:16:15 -07001483
1484 ssize_t originalBufferSize = mProxy->getBufferSizeInFrames();
1485 ssize_t finalBufferSize = mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
1486 if (originalBufferSize != finalBufferSize) {
Phil Burk64e16a72020-06-01 13:25:51 -07001487 android::mediametrics::LogItem(mMetricsId)
1488 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
1489 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1490 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t)getUnderrunCount_l())
1491 .record();
Phil Burka9876702020-04-20 18:16:15 -07001492 }
1493 return finalBufferSize;
Phil Burkc0adecb2016-01-08 12:44:11 -08001494}
1495
Andy Hung3c7f47a2021-03-16 17:30:09 -07001496ssize_t AudioTrack::getStartThresholdInFrames() const
1497{
1498 AutoMutex lock(mLock);
1499 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1500 return NO_INIT;
1501 }
1502 return (ssize_t) mProxy->getStartThresholdInFrames();
1503}
1504
1505ssize_t AudioTrack::setStartThresholdInFrames(size_t startThresholdInFrames)
1506{
1507 if (startThresholdInFrames > INT32_MAX || startThresholdInFrames == 0) {
1508 // contractually we could simply return the current threshold in frames
1509 // to indicate the request was ignored, but we return an error here.
1510 return BAD_VALUE;
1511 }
1512 AutoMutex lock(mLock);
1513 // We do not permit calling setStartThresholdInFrames() between the AudioTrack
1514 // default ctor AudioTrack() and set(...) but rather fail such an attempt.
1515 // (To do so would require a cached mOrigStartThresholdInFrames and we may
1516 // not have proper validation for the actual set value).
1517 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1518 return NO_INIT;
1519 }
1520 const uint32_t original = mProxy->getStartThresholdInFrames();
1521 const uint32_t final = mProxy->setStartThresholdInFrames(startThresholdInFrames);
1522 if (original != final) {
1523 android::mediametrics::LogItem(mMetricsId)
1524 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD)
1525 .set(AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES, (int32_t)final)
1526 .record();
1527 if (original > final) {
1528 // restart track if it was disabled by audioflinger due to previous underrun
1529 // and we reduced the number of frames for the threshold.
1530 restartIfDisabled();
1531 }
1532 }
1533 return final;
1534}
1535
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001536status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1537{
Glenn Kastend79072e2016-01-06 08:41:20 -08001538 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001539 return INVALID_OPERATION;
1540 }
1541
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001542 if (loopCount == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001543 ;
1544 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount &&
1545 loopEnd - loopStart >= MIN_LOOP) {
1546 ;
1547 } else {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001548 return BAD_VALUE;
1549 }
1550
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001551 AutoMutex lock(mLock);
1552 // See setPosition() regarding setting parameters such as loop points or position while active
1553 if (mState == STATE_ACTIVE) {
1554 return INVALID_OPERATION;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001555 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001556 setLoop_l(loopStart, loopEnd, loopCount);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001557 return NO_ERROR;
1558}
1559
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001560void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1561{
Andy Hung4ede21d2014-12-12 15:37:34 -08001562 // We do not update the periodic notification point.
1563 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
1564 mLoopCount = loopCount;
1565 mLoopEnd = loopEnd;
1566 mLoopStart = loopStart;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001567 mLoopCountNotified = loopCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001568 mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
Andy Hung3c09c782014-12-29 18:39:32 -08001569
1570 // Waking the AudioTrackThread is not needed as this cannot be called when active.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001571}
1572
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001573status_t AudioTrack::setMarkerPosition(uint32_t marker)
1574{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001575 // The only purpose of setting marker position is to get a callback
Atneya Nair6a8238eb2021-10-26 19:26:41 -04001576 if (!mCallback.promote() || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001577 return INVALID_OPERATION;
1578 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001579
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001580 AutoMutex lock(mLock);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001581 mMarkerPosition = marker;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001582 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001583
Andy Hung3c09c782014-12-29 18:39:32 -08001584 sp<AudioTrackThread> t = mAudioTrackThread;
1585 if (t != 0) {
1586 t->wake();
1587 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001588 return NO_ERROR;
1589}
1590
Glenn Kastena5224f32012-01-04 12:41:44 -08001591status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001592{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001593 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001594 return INVALID_OPERATION;
1595 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001596 if (marker == NULL) {
1597 return BAD_VALUE;
1598 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001599
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001600 AutoMutex lock(mLock);
Andy Hung90e8a972015-11-09 16:42:40 -08001601 mMarkerPosition.getValue(marker);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001602
1603 return NO_ERROR;
1604}
1605
1606status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
1607{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001608 // The only purpose of setting position update period is to get a callback
Atneya Nair6a8238eb2021-10-26 19:26:41 -04001609 if (!mCallback.promote() || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001610 return INVALID_OPERATION;
1611 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001612
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001613 AutoMutex lock(mLock);
Glenn Kasten200092b2014-08-15 15:13:30 -07001614 mNewPosition = updateAndGetPosition_l() + updatePeriod;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001615 mUpdatePeriod = updatePeriod;
Glenn Kasten2b2165c2014-01-13 08:53:36 -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::getPositionUpdatePeriod(uint32_t *updatePeriod) 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 (updatePeriod == 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);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001634 *updatePeriod = mUpdatePeriod;
1635
1636 return NO_ERROR;
1637}
1638
1639status_t AudioTrack::setPosition(uint32_t position)
1640{
Glenn Kastend79072e2016-01-06 08:41:20 -08001641 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001642 return INVALID_OPERATION;
1643 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001644 if (position > mFrameCount) {
1645 return BAD_VALUE;
1646 }
John Grossman4ff14ba2012-02-08 16:37:41 -08001647
Eric Laurent1703cdf2011-03-07 14:52:59 -08001648 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001649 // Currently we require that the player is inactive before setting parameters such as position
1650 // or loop points. Otherwise, there could be a race condition: the application could read the
1651 // current position, compute a new position or loop parameters, and then set that position or
1652 // loop parameters but it would do the "wrong" thing since the position has continued to advance
1653 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app
1654 // to specify how it wants to handle such scenarios.
1655 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001656 return INVALID_OPERATION;
1657 }
Andy Hung9b461582014-12-01 17:56:29 -08001658 // After setting the position, use full update period before notification.
Glenn Kasten200092b2014-08-15 15:13:30 -07001659 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001660 mStaticProxy->setBufferPosition(position);
Andy Hung3c09c782014-12-29 18:39:32 -08001661
1662 // Waking the AudioTrackThread is not needed as this cannot be called when active.
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001663 return NO_ERROR;
1664}
1665
Glenn Kasten200092b2014-08-15 15:13:30 -07001666status_t AudioTrack::getPosition(uint32_t *position)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001667{
Glenn Kastend65d73c2012-06-22 17:21:07 -07001668 if (position == NULL) {
1669 return BAD_VALUE;
1670 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001671
Eric Laurent1703cdf2011-03-07 14:52:59 -08001672 AutoMutex lock(mLock);
Andy Hung7a490e72016-03-23 15:58:10 -07001673 // FIXME: offloaded and direct tracks call into the HAL for render positions
1674 // for compressed/synced data; however, we use proxy position for pure linear pcm data
1675 // as we do not know the capability of the HAL for pcm position support and standby.
1676 // There may be some latency differences between the HAL position and the proxy position.
1677 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001678 uint32_t dspFrames = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001679
Eric Laurentab5cdba2014-06-09 17:22:27 -07001680 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001681 ALOGV("%s(%d): called in paused state, return cached position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001682 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001683 *position = mPausedPosition;
1684 return NO_ERROR;
1685 }
1686
Glenn Kasten142f5192014-03-25 17:44:59 -07001687 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung1f1db832015-06-08 13:26:10 -07001688 uint32_t halFrames; // actually unused
1689 (void) AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames);
1690 // FIXME: on getRenderPosition() error, we return OK with frame position 0.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001691 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001692 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED)
1693 // due to hardware latency. We leave this behavior for now.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001694 *position = dspFrames;
1695 } else {
Eric Laurent275e8e92014-11-30 15:14:47 -08001696 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung1f1db832015-06-08 13:26:10 -07001697 (void) restoreTrack_l("getPosition");
1698 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l()
1699 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position.
Eric Laurent275e8e92014-11-30 15:14:47 -08001700 }
1701
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001702 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
Glenn Kasten200092b2014-08-15 15:13:30 -07001703 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ?
Andy Hung90e8a972015-11-09 16:42:40 -08001704 0 : updateAndGetPosition_l().value();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001705 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001706 return NO_ERROR;
1707}
1708
Kévin PETIT377b2ec2014-02-03 12:35:36 +00001709status_t AudioTrack::getBufferPosition(uint32_t *position)
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001710{
Glenn Kastend79072e2016-01-06 08:41:20 -08001711 if (mSharedBuffer == 0) {
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001712 return INVALID_OPERATION;
1713 }
1714 if (position == NULL) {
1715 return BAD_VALUE;
1716 }
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001717
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001718 AutoMutex lock(mLock);
1719 *position = mStaticProxy->getBufferPosition();
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001720 return NO_ERROR;
1721}
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001722
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001723status_t AudioTrack::reload()
1724{
Glenn Kastend79072e2016-01-06 08:41:20 -08001725 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001726 return INVALID_OPERATION;
1727 }
1728
Eric Laurent1703cdf2011-03-07 14:52:59 -08001729 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001730 // See setPosition() regarding setting parameters such as loop points or position while active
1731 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001732 return INVALID_OPERATION;
1733 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001734 mNewPosition = mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001735 (void) updateAndGetPosition_l();
1736 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07001737 mPreviousTimestampValid = false;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001738#if 0
Andy Hung9b461582014-12-01 17:56:29 -08001739 // The documentation is not clear on the behavior of reload() and the restoration
Andy Hung53c3b5f2014-12-15 16:42:05 -08001740 // of loop count. Historically we have not restored loop count, start, end,
1741 // but it makes sense if one desires to repeat playing a particular sound.
1742 if (mLoopCount != 0) {
1743 mLoopCountNotified = mLoopCount;
1744 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
1745 }
1746#endif
Andy Hung9b461582014-12-01 17:56:29 -08001747 mStaticProxy->setBufferPosition(0);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001748 return NO_ERROR;
1749}
1750
Glenn Kasten38e905b2014-01-13 10:21:48 -08001751audio_io_handle_t AudioTrack::getOutput() const
Eric Laurentc2f1f072009-07-17 12:17:14 -07001752{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001753 AutoMutex lock(mLock);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001754 return mOutput;
Eric Laurent1703cdf2011-03-07 14:52:59 -08001755}
1756
Paul McLeanaa981192015-03-21 09:55:15 -07001757status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) {
1758 AutoMutex lock(mLock);
Eric Laurent2f2c1982021-06-02 14:03:11 +02001759 ALOGV("%s(%d): deviceId=%d mSelectedDeviceId=%d",
1760 __func__, mPortId, deviceId, mSelectedDeviceId);
Paul McLeanaa981192015-03-21 09:55:15 -07001761 if (mSelectedDeviceId != deviceId) {
1762 mSelectedDeviceId = deviceId;
Eric Laurentfb00fc72017-05-25 18:17:12 -07001763 if (mStatus == NO_ERROR) {
1764 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
jiabin156c6872017-10-06 09:47:15 -07001765 mProxy->interrupt();
Eric Laurentfb00fc72017-05-25 18:17:12 -07001766 }
Paul McLeanaa981192015-03-21 09:55:15 -07001767 }
Eric Laurent493404d2015-04-21 15:07:36 -07001768 return NO_ERROR;
Paul McLeanaa981192015-03-21 09:55:15 -07001769}
1770
1771audio_port_handle_t AudioTrack::getOutputDevice() {
1772 AutoMutex lock(mLock);
1773 return mSelectedDeviceId;
1774}
1775
Eric Laurentad2e7b92017-09-14 20:06:42 -07001776// must be called with mLock held
1777void AudioTrack::updateRoutedDeviceId_l()
1778{
1779 // if the track is inactive, do not update actual device as the output stream maybe routed
1780 // to a device not relevant to this client because of other active use cases.
1781 if (mState != STATE_ACTIVE) {
1782 return;
1783 }
1784 if (mOutput != AUDIO_IO_HANDLE_NONE) {
1785 audio_port_handle_t deviceId = AudioSystem::getDeviceIdForIo(mOutput);
1786 if (deviceId != AUDIO_PORT_HANDLE_NONE) {
1787 mRoutedDeviceId = deviceId;
1788 }
1789 }
1790}
1791
Eric Laurent296fb132015-05-01 11:38:42 -07001792audio_port_handle_t AudioTrack::getRoutedDeviceId() {
1793 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001794 updateRoutedDeviceId_l();
1795 return mRoutedDeviceId;
Eric Laurent296fb132015-05-01 11:38:42 -07001796}
1797
Eric Laurentbe916aa2010-06-01 23:49:17 -07001798status_t AudioTrack::attachAuxEffect(int effectId)
1799{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001800 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001801 status_t status;
1802 mAudioTrack->attachAuxEffect(effectId, &status);
Eric Laurent2beeb502010-07-16 07:43:46 -07001803 if (status == NO_ERROR) {
1804 mAuxEffectId = effectId;
1805 }
1806 return status;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001807}
1808
Eric Laurente83b55d2014-11-14 10:06:21 -08001809audio_stream_type_t AudioTrack::streamType() const
1810{
Eric Laurente83b55d2014-11-14 10:06:21 -08001811 return mStreamType;
1812}
1813
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001814uint32_t AudioTrack::latency()
1815{
1816 AutoMutex lock(mLock);
1817 updateLatency_l();
1818 return mLatency;
1819}
1820
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001821// -------------------------------------------------------------------------
1822
Eric Laurent1703cdf2011-03-07 14:52:59 -08001823// must be called with mLock held
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001824void AudioTrack::updateLatency_l()
1825{
1826 status_t status = AudioSystem::getLatency(mOutput, &mAfLatency);
1827 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08001828 ALOGW("%s(%d): getLatency(%d) failed status %d", __func__, mPortId, mOutput, status);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001829 } else {
1830 // FIXME don't believe this lie
Andy Hung13969262017-09-11 17:24:21 -07001831 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001832 }
1833}
1834
Phil Burkadbb75a2017-06-16 12:19:42 -07001835// TODO Move this macro to a common header file for enum to string conversion in audio framework.
1836#define MEDIA_CASE_ENUM(name) case name: return #name
1837const char * AudioTrack::convertTransferToText(transfer_type transferType) {
1838 switch (transferType) {
1839 MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
1840 MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
1841 MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
1842 MEDIA_CASE_ENUM(TRANSFER_SYNC);
1843 MEDIA_CASE_ENUM(TRANSFER_SHARED);
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001844 MEDIA_CASE_ENUM(TRANSFER_SYNC_NOTIF_CALLBACK);
Phil Burkadbb75a2017-06-16 12:19:42 -07001845 default:
1846 return "UNRECOGNIZED";
1847 }
1848}
1849
Glenn Kasten200092b2014-08-15 15:13:30 -07001850status_t AudioTrack::createTrack_l()
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001851{
Eric Laurentf32d7812017-11-30 14:44:07 -08001852 status_t status;
1853 bool callbackAdded = false;
Andy Hung3acde2c2021-11-11 09:18:08 -08001854 std::string errorMessage;
Eric Laurentf32d7812017-11-30 14:44:07 -08001855
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001856 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
1857 if (audioFlinger == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001858 errorMessage = StringPrintf("%s(%d): Could not get audioflinger",
Eric Laurent973db022018-11-20 14:54:31 -08001859 __func__, mPortId);
Andy Hung3acde2c2021-11-11 09:18:08 -08001860 status = DEAD_OBJECT;
Eric Laurentf32d7812017-11-30 14:44:07 -08001861 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001862 }
1863
Eric Laurent21da6472017-11-09 16:29:26 -08001864 {
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08001865 // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
1866 // After fast request is denied, we will request again if IAudioTrack is re-created.
Glenn Kastend79072e2016-01-06 08:41:20 -08001867 // Client can only express a preference for FAST. Server will perform additional tests.
Phil Burk33ff89b2015-11-30 11:16:01 -08001868 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Phil Burkadbb75a2017-06-16 12:19:42 -07001869 // either of these use cases:
1870 // use case 1: shared buffer
1871 bool sharedBuffer = mSharedBuffer != 0;
1872 bool transferAllowed =
Glenn Kastenc6ba8232014-02-27 13:34:29 -08001873 // use case 2: callback transfer mode
Glenn Kasten1dfe2f92015-03-09 12:03:14 -07001874 (mTransfer == TRANSFER_CALLBACK) ||
1875 // use case 3: obtain/release mode
Phil Burk33ff89b2015-11-30 11:16:01 -08001876 (mTransfer == TRANSFER_OBTAIN) ||
1877 // use case 4: synchronous write
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001878 ((mTransfer == TRANSFER_SYNC || mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK)
1879 && mThreadCanCallJava);
Phil Burkadbb75a2017-06-16 12:19:42 -07001880
Eric Laurent21da6472017-11-09 16:29:26 -08001881 bool fastAllowed = sharedBuffer || transferAllowed;
1882 if (!fastAllowed) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001883 ALOGW("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by client,"
1884 " not shared buffer and transfer = %s",
Eric Laurent973db022018-11-20 14:54:31 -08001885 __func__, mPortId,
Phil Burkadbb75a2017-06-16 12:19:42 -07001886 convertTransferToText(mTransfer));
Phil Burk33ff89b2015-11-30 11:16:01 -08001887 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
1888 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001889 }
1890
Eric Laurent21da6472017-11-09 16:29:26 -08001891 IAudioFlinger::CreateTrackInput input;
Andy Hunga2159aa2021-07-20 13:01:52 -07001892 if (mOriginalStreamType != AUDIO_STREAM_DEFAULT) {
1893 // Legacy: This is based on original parameters even if the track is recreated.
1894 input.attr = AudioSystem::streamTypeToAttributes(mOriginalStreamType);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001895 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08001896 input.attr = mAttributes;
Eric Laurentd1b449a2010-05-14 03:26:45 -07001897 }
Eric Laurent21da6472017-11-09 16:29:26 -08001898 input.config = AUDIO_CONFIG_INITIALIZER;
1899 input.config.sample_rate = mSampleRate;
1900 input.config.channel_mask = mChannelMask;
1901 input.config.format = mFormat;
1902 input.config.offload_info = mOffloadInfoCopy;
Svet Ganov3e5f14f2021-05-13 22:51:08 +00001903 input.clientInfo.attributionSource = mClientAttributionSource;
Eric Laurent21da6472017-11-09 16:29:26 -08001904 input.clientInfo.clientTid = -1;
Glenn Kasten363fb752014-01-15 12:27:31 -08001905 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Glenn Kasten47d55172017-05-23 11:19:30 -07001906 // It is currently meaningless to request SCHED_FIFO for a Java thread. Even if the
1907 // application-level code follows all non-blocking design rules, the language runtime
1908 // doesn't also follow those rules, so the thread will not benefit overall.
Phil Burk33ff89b2015-11-30 11:16:01 -08001909 if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
Eric Laurent21da6472017-11-09 16:29:26 -08001910 input.clientInfo.clientTid = mAudioTrackThread->getTid();
Glenn Kasten3acbd052012-02-28 10:39:56 -08001911 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001912 }
Eric Laurent21da6472017-11-09 16:29:26 -08001913 input.sharedBuffer = mSharedBuffer;
1914 input.notificationsPerBuffer = mNotificationsPerBufferReq;
1915 input.speed = 1.0;
1916 if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
1917 (mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
1918 input.speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
1919 max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
1920 }
1921 input.flags = mFlags;
1922 input.frameCount = mReqFrameCount;
1923 input.notificationFrameCount = mNotificationFramesReq;
1924 input.selectedDeviceId = mSelectedDeviceId;
1925 input.sessionId = mSessionId;
jiabinf6eb4c32020-02-25 14:06:25 -08001926 input.audioTrackCallback = mAudioTrackCallback;
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001927
Ytai Ben-Tsvi4dfeb622020-11-02 12:47:30 -08001928 media::CreateTrackResponse response;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001929 status = audioFlinger->createTrack(VALUE_OR_FATAL(input.toAidl()), response);
Ytai Ben-Tsvi357e26a2021-01-05 13:21:19 -08001930
1931 IAudioFlinger::CreateTrackOutput output{};
1932 if (status == NO_ERROR) {
1933 output = VALUE_OR_FATAL(IAudioFlinger::CreateTrackOutput::fromAidl(response));
1934 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001935
Eric Laurent21da6472017-11-09 16:29:26 -08001936 if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001937 errorMessage = StringPrintf(
1938 "%s(%d): AudioFlinger could not create track, status: %d output %d",
Eric Laurent973db022018-11-20 14:54:31 -08001939 __func__, mPortId, status, output.outputId);
Eric Laurentf32d7812017-11-30 14:44:07 -08001940 if (status == NO_ERROR) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001941 status = INVALID_OPERATION; // device not ready
Eric Laurentf32d7812017-11-30 14:44:07 -08001942 }
1943 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001944 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001945 ALOG_ASSERT(output.audioTrack != 0);
Glenn Kastenc08d20b2014-02-24 15:21:10 -08001946
Eric Laurent21da6472017-11-09 16:29:26 -08001947 mFrameCount = output.frameCount;
1948 mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
1949 mRoutedDeviceId = output.selectedDeviceId;
1950 mSessionId = output.sessionId;
Andy Hunga2159aa2021-07-20 13:01:52 -07001951 mStreamType = output.streamType;
Eric Laurent21da6472017-11-09 16:29:26 -08001952
1953 mSampleRate = output.sampleRate;
1954 if (mOriginalSampleRate == 0) {
1955 mOriginalSampleRate = mSampleRate;
1956 }
1957
1958 mAfFrameCount = output.afFrameCount;
1959 mAfSampleRate = output.afSampleRate;
1960 mAfLatency = output.afLatencyMs;
1961
1962 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
1963
Glenn Kasten38e905b2014-01-13 10:21:48 -08001964 // AudioFlinger now owns the reference to the I/O handle,
1965 // so we are no longer responsible for releasing it.
1966
Glenn Kasten7fd04222016-02-02 12:38:16 -08001967 // FIXME compare to AudioRecord
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001968 std::optional<media::SharedFileRegion> sfr;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001969 output.audioTrack->getCblk(&sfr);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001970 sp<IMemory> iMem = VALUE_OR_FATAL(aidl2legacy_NullableSharedFileRegion_IMemory(sfr));
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001971 if (iMem == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001972 errorMessage = StringPrintf("%s(%d): Could not get control block", __func__, mPortId);
1973 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001974 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001975 }
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001976 // TODO: Using unsecurePointer() has some associated security pitfalls
1977 // (see declaration for details).
1978 // Either document why it is safe in this case or address the
1979 // issue (e.g. by copying).
1980 void *iMemPointer = iMem->unsecurePointer();
Glenn Kasten0cde0762014-01-16 15:06:36 -08001981 if (iMemPointer == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001982 errorMessage = StringPrintf(
1983 "%s(%d): Could not get control block pointer", __func__, mPortId);
1984 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001985 goto exit;
Glenn Kasten0cde0762014-01-16 15:06:36 -08001986 }
Glenn Kasten53cec222013-08-29 09:01:02 -07001987 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001988 if (mAudioTrack != 0) {
Marco Nelissenf8880202014-11-14 07:58:25 -08001989 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001990 mDeathNotifier.clear();
1991 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001992 mAudioTrack = output.audioTrack;
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001993 mCblkMemory = iMem;
Eric Laurent3bcffa12014-06-12 18:38:45 -07001994 IPCThreadState::self()->flushCommands();
1995
Glenn Kasten0cde0762014-01-16 15:06:36 -08001996 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001997 mCblk = cblk;
Glenn Kasten5f631512014-02-24 15:16:07 -08001998
Glenn Kastena07f17c2013-04-23 12:39:37 -07001999 mAwaitBoost = false;
Glenn Kasten363fb752014-01-15 12:27:31 -08002000 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Eric Laurent21da6472017-11-09 16:29:26 -08002001 if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002002 ALOGI("%s(%d): AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08002003 __func__, mPortId, mReqFrameCount, mFrameCount);
Phil Burk33ff89b2015-11-30 11:16:01 -08002004 if (!mThreadCanCallJava) {
2005 mAwaitBoost = true;
2006 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08002007 } else {
Robert Wuc121cd12021-08-13 17:51:40 +00002008 ALOGV("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08002009 __func__, mPortId, mReqFrameCount, mFrameCount);
Glenn Kastene0fa4672012-04-24 14:35:14 -07002010 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08002011 }
Eric Laurent21da6472017-11-09 16:29:26 -08002012 mFlags = output.flags;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002013
Eric Laurentad2e7b92017-09-14 20:06:42 -07002014 //mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
Eric Laurent09f1ed22019-04-24 17:45:17 -07002015 if (mDeviceCallback != 0) {
Eric Laurentad2e7b92017-09-14 20:06:42 -07002016 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07002017 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07002018 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07002019 AudioSystem::addAudioDeviceCallback(this, output.outputId, output.portId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07002020 callbackAdded = true;
2021 }
2022
Eric Laurent09f1ed22019-04-24 17:45:17 -07002023 mPortId = output.portId;
Glenn Kasten38e905b2014-01-13 10:21:48 -08002024 // We retain a copy of the I/O handle, but don't own the reference
Eric Laurent21da6472017-11-09 16:29:26 -08002025 mOutput = output.outputId;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002026 mRefreshRemaining = true;
2027
2028 // Starting address of buffers in shared memory. If there is a shared buffer, buffers
2029 // is the value of pointer() for the shared buffer, otherwise buffers points
2030 // immediately after the control block. This address is for the mapping within client
2031 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
2032 void* buffers;
Glenn Kasten363fb752014-01-15 12:27:31 -08002033 if (mSharedBuffer == 0) {
Glenn Kasten138d6f92015-03-20 10:54:51 -07002034 buffers = cblk + 1;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002035 } else {
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07002036 // TODO: Using unsecurePointer() has some associated security pitfalls
2037 // (see declaration for details).
2038 // Either document why it is safe in this case or address the
2039 // issue (e.g. by copying).
2040 buffers = mSharedBuffer->unsecurePointer();
Glenn Kasten138d6f92015-03-20 10:54:51 -07002041 if (buffers == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08002042 errorMessage = StringPrintf(
2043 "%s(%d): Could not get buffer pointer", __func__, mPortId);
2044 ALOGE("%s", errorMessage.c_str());
2045 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08002046 goto exit;
Glenn Kasten138d6f92015-03-20 10:54:51 -07002047 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002048 }
2049
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002050 mAudioTrack->attachAuxEffect(mAuxEffectId, &status);
Glenn Kasten5f631512014-02-24 15:16:07 -08002051
Glenn Kasten093000f2012-05-03 09:35:36 -07002052 // If IAudioTrack is re-created, don't let the requested frameCount
2053 // decrease. This can confuse clients that cache frameCount().
Eric Laurent21da6472017-11-09 16:29:26 -08002054 if (mFrameCount > mReqFrameCount) {
2055 mReqFrameCount = mFrameCount;
Glenn Kasten093000f2012-05-03 09:35:36 -07002056 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08002057
Andy Hungd7bd69e2015-07-24 07:52:41 -07002058 // reset server position to 0 as we have new cblk.
2059 mServer = 0;
2060
Glenn Kastene3aa6592012-12-04 12:22:46 -08002061 // update proxy
Glenn Kasten363fb752014-01-15 12:27:31 -08002062 if (mSharedBuffer == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002063 mStaticProxy.clear();
Eric Laurent21da6472017-11-09 16:29:26 -08002064 mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002065 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08002066 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002067 mProxy = mStaticProxy;
2068 }
seunghak.hane6a9d6582014-11-22 15:22:35 +09002069
2070 mProxy->setVolumeLR(gain_minifloat_pack(
2071 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
2072 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
2073
Glenn Kastene3aa6592012-12-04 12:22:46 -08002074 mProxy->setSendLevel(mSendLevel);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002075 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
2076 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
2077 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07002078 mProxy->setSampleRate(effectiveSampleRate);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002079
2080 AudioPlaybackRate playbackRateTemp = mPlaybackRate;
2081 playbackRateTemp.mSpeed = effectiveSpeed;
2082 playbackRateTemp.mPitch = effectivePitch;
2083 mProxy->setPlaybackRate(playbackRateTemp);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002084 mProxy->setMinimum(mNotificationFramesAct);
2085
Kuowei Lid4adbdb2020-08-13 14:44:25 +08002086 if (mDualMonoMode != AUDIO_DUAL_MONO_MODE_OFF) {
2087 setDualMonoMode_l(mDualMonoMode);
2088 }
2089 if (mAudioDescriptionMixLeveldB != -std::numeric_limits<float>::infinity()) {
2090 setAudioDescriptionMixLevel_l(mAudioDescriptionMixLeveldB);
2091 }
2092
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002093 mDeathNotifier = new DeathNotifier(this);
Marco Nelissenf8880202014-11-14 07:58:25 -08002094 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
Glenn Kastene3aa6592012-12-04 12:22:46 -08002095
Andy Hungb68f5eb2019-12-03 16:49:17 -08002096 // This is the first log sent from the AudioTrack client.
2097 // The creation of the audio track by AudioFlinger (in the code above)
2098 // is the first log of the AudioTrack and must be present before
2099 // any AudioTrack client logs will be accepted.
Andy Hungea840382020-05-05 21:50:17 -07002100
Andy Hungb68f5eb2019-12-03 16:49:17 -08002101 mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(mPortId);
2102 mediametrics::LogItem(mMetricsId)
2103 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE)
2104 // the following are immutable
Andy Hunga629bd12020-06-05 16:03:53 -07002105 .set(AMEDIAMETRICS_PROP_FLAGS, toString(mFlags).c_str())
2106 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -08002107 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
Andy Hung3a5c2f32021-02-17 15:06:42 -08002108 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, mLogSessionId)
Andy Hung839a3062021-02-17 11:15:16 -08002109 .set(AMEDIAMETRICS_PROP_PLAYERIID, mPlayerIId)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002110 .set(AMEDIAMETRICS_PROP_TRACKID, mPortId) // dup from key
Andy Hungb68f5eb2019-12-03 16:49:17 -08002111 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2112 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2113 .set(AMEDIAMETRICS_PROP_THREADID, (int32_t)output.outputId)
2114 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2115 .set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)mRoutedDeviceId)
2116 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2117 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
2118 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
2119 // the following are NOT immutable
2120 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)mVolume[AUDIO_INTERLEAVE_LEFT])
2121 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)mVolume[AUDIO_INTERLEAVE_RIGHT])
2122 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
2123 .set(AMEDIAMETRICS_PROP_AUXEFFECTID, (int32_t)mAuxEffectId)
2124 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2125 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2126 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2127 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2128 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveSampleRate)
2129 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2130 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)effectiveSpeed)
2131 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2132 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)effectivePitch)
2133 .record();
2134
2135 // mSendLevel
2136 // mReqFrameCount?
2137 // mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq
2138 // mLatency, mAfLatency, mAfFrameCount, mAfSampleRate
2139
Glenn Kasten38e905b2014-01-13 10:21:48 -08002140 }
2141
Eric Laurentf32d7812017-11-30 14:44:07 -08002142exit:
Andy Hung3acde2c2021-11-11 09:18:08 -08002143 if (status != NO_ERROR) {
2144 if (callbackAdded) {
2145 // note: mOutput is always valid is callbackAdded is true
2146 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
2147 }
2148 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
2149 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
Eric Laurentad2e7b92017-09-14 20:06:42 -07002150 }
Eric Laurentf32d7812017-11-30 14:44:07 -08002151 mStatus = status;
Eric Laurent21da6472017-11-09 16:29:26 -08002152
2153 // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
Glenn Kasten38e905b2014-01-13 10:21:48 -08002154 return status;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002155}
2156
Andy Hung3acde2c2021-11-11 09:18:08 -08002157void AudioTrack::reportError(status_t status, const char *event, const char *message) const
2158{
2159 if (status == NO_ERROR) return;
2160 // We report error on the native side because some callers do not come
2161 // from Java.
Andy Hungde602302021-12-07 21:35:49 -08002162 // Ensure these variables are initialized in set().
2163 mediametrics::LogItem(AMEDIAMETRICS_KEY_AUDIO_TRACK_ERROR)
Andy Hung3acde2c2021-11-11 09:18:08 -08002164 .set(AMEDIAMETRICS_PROP_EVENT, event)
2165 .set(AMEDIAMETRICS_PROP_ERROR, mediametrics::statusToErrorString(status))
2166 .set(AMEDIAMETRICS_PROP_ERRORMESSAGE, message)
2167 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
2168 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
2169 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2170 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2171 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2172 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2173 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
Andy Hung3acde2c2021-11-11 09:18:08 -08002174 // the following are NOT immutable
Andy Hungde602302021-12-07 21:35:49 -08002175 // frame count is initially the requested frame count, but may be adjusted
2176 // by AudioFlinger after creation.
2177 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
Andy Hung3acde2c2021-11-11 09:18:08 -08002178 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2179 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2180 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2181 .record();
2182}
2183
Glenn Kastenb46f3942015-03-09 12:00:30 -07002184status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002185{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002186 if (audioBuffer == NULL) {
Glenn Kasten551b5352015-03-20 11:30:28 -07002187 if (nonContig != NULL) {
2188 *nonContig = 0;
2189 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002190 return BAD_VALUE;
Eric Laurent9b7d9502011-03-21 11:49:00 -07002191 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002192 if (mTransfer != TRANSFER_OBTAIN) {
2193 audioBuffer->frameCount = 0;
2194 audioBuffer->size = 0;
2195 audioBuffer->raw = NULL;
Glenn Kasten551b5352015-03-20 11:30:28 -07002196 if (nonContig != NULL) {
2197 *nonContig = 0;
2198 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002199 return INVALID_OPERATION;
2200 }
Eric Laurent9b7d9502011-03-21 11:49:00 -07002201
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002202 const struct timespec *requested;
Eric Laurentdf576992014-01-27 18:13:39 -08002203 struct timespec timeout;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002204 if (waitCount == -1) {
2205 requested = &ClientProxy::kForever;
2206 } else if (waitCount == 0) {
2207 requested = &ClientProxy::kNonBlocking;
2208 } else if (waitCount > 0) {
Chih-Hung Hsiehbca74292018-08-10 16:06:07 -07002209 time_t ms = WAIT_PERIOD_MS * (time_t) waitCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002210 timeout.tv_sec = ms / 1000;
Andy Hung06a730b2020-04-09 13:28:31 -07002211 timeout.tv_nsec = (ms % 1000) * 1000000;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002212 requested = &timeout;
2213 } else {
Eric Laurent973db022018-11-20 14:54:31 -08002214 ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002215 requested = NULL;
2216 }
Glenn Kastenb46f3942015-03-09 12:00:30 -07002217 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002218}
Eric Laurent1703cdf2011-03-07 14:52:59 -08002219
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002220status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
2221 struct timespec *elapsed, size_t *nonContig)
2222{
2223 // previous and new IAudioTrack sequence numbers are used to detect track re-creation
2224 uint32_t oldSequence = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002225
2226 Proxy::Buffer buffer;
2227 status_t status = NO_ERROR;
2228
2229 static const int32_t kMaxTries = 5;
2230 int32_t tryCounter = kMaxTries;
2231
2232 do {
2233 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
2234 // keep them from going away if another thread re-creates the track during obtainBuffer()
2235 sp<AudioTrackClientProxy> proxy;
2236 sp<IMemory> iMem;
2237
2238 { // start of lock scope
2239 AutoMutex lock(mLock);
2240
Glenn Kasten305996c2020-01-27 08:03:37 -08002241 uint32_t newSequence = mSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002242 // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
2243 if (status == DEAD_OBJECT) {
2244 // re-create track, unless someone else has already done so
2245 if (newSequence == oldSequence) {
2246 status = restoreTrack_l("obtainBuffer");
2247 if (status != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002248 buffer.mFrameCount = 0;
2249 buffer.mRaw = NULL;
2250 buffer.mNonContig = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002251 break;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002252 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002253 }
2254 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002255 oldSequence = newSequence;
2256
Eric Laurent4d231dc2016-03-11 18:38:23 -08002257 if (status == NOT_ENOUGH_DATA) {
2258 restartIfDisabled();
2259 }
2260
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002261 // Keep the extra references
2262 proxy = mProxy;
2263 iMem = mCblkMemory;
2264
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002265 if (mState == STATE_STOPPING) {
2266 status = -EINTR;
2267 buffer.mFrameCount = 0;
2268 buffer.mRaw = NULL;
2269 buffer.mNonContig = 0;
2270 break;
2271 }
2272
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002273 // Non-blocking if track is stopped or paused
2274 if (mState != STATE_ACTIVE) {
2275 requested = &ClientProxy::kNonBlocking;
2276 }
2277
2278 } // end of lock scope
2279
2280 buffer.mFrameCount = audioBuffer->frameCount;
2281 // FIXME starts the requested timeout and elapsed over from scratch
2282 status = proxy->obtainBuffer(&buffer, requested, elapsed);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002283 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002284
2285 audioBuffer->frameCount = buffer.mFrameCount;
Andy Hungabdb9902015-01-12 15:08:22 -08002286 audioBuffer->size = buffer.mFrameCount * mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002287 audioBuffer->raw = buffer.mRaw;
Glenn Kasten305996c2020-01-27 08:03:37 -08002288 audioBuffer->sequence = oldSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002289 if (nonContig != NULL) {
2290 *nonContig = buffer.mNonContig;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002291 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002292 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002293}
2294
Glenn Kasten54a8a452015-03-09 12:03:00 -07002295void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002296{
Glenn Kasten3f02be22015-03-09 11:59:04 -07002297 // FIXME add error checking on mode, by adding an internal version
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002298 if (mTransfer == TRANSFER_SHARED) {
2299 return;
2300 }
2301
Andy Hungabdb9902015-01-12 15:08:22 -08002302 size_t stepCount = audioBuffer->size / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002303 if (stepCount == 0) {
2304 return;
2305 }
2306
2307 Proxy::Buffer buffer;
2308 buffer.mFrameCount = stepCount;
2309 buffer.mRaw = audioBuffer->raw;
Glenn Kastene3aa6592012-12-04 12:22:46 -08002310
Eric Laurent1703cdf2011-03-07 14:52:59 -08002311 AutoMutex lock(mLock);
Glenn Kasten305996c2020-01-27 08:03:37 -08002312 if (audioBuffer->sequence != mSequence) {
2313 // This Buffer came from a different IAudioTrack instance, so ignore the releaseBuffer
2314 ALOGD("%s is no-op due to IAudioTrack sequence mismatch %u != %u",
2315 __func__, audioBuffer->sequence, mSequence);
2316 return;
2317 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002318 mReleased += stepCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002319 mInUnderrun = false;
2320 mProxy->releaseBuffer(&buffer);
2321
2322 // restart track if it was disabled by audioflinger due to previous underrun
Eric Laurent4d231dc2016-03-11 18:38:23 -08002323 restartIfDisabled();
2324}
2325
2326void AudioTrack::restartIfDisabled()
2327{
2328 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
2329 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002330 ALOGW("%s(%d): releaseBuffer() track %p disabled due to previous underrun, restarting",
Eric Laurent973db022018-11-20 14:54:31 -08002331 __func__, mPortId, this);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002332 // FIXME ignoring status
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002333 status_t status;
2334 mAudioTrack->start(&status);
Eric Laurentdf839842012-05-31 14:27:14 -07002335 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002336}
2337
2338// -------------------------------------------------------------------------
2339
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002340ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002341{
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002342 if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07002343 return INVALID_OPERATION;
2344 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002345
Eric Laurentab5cdba2014-06-09 17:22:27 -07002346 if (isDirect()) {
2347 AutoMutex lock(mLock);
2348 int32_t flags = android_atomic_and(
2349 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
2350 &mCblk->mFlags);
2351 if (flags & CBLK_INVALID) {
2352 return DEAD_OBJECT;
2353 }
2354 }
2355
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002356 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
Jiabin Huang447cea72020-07-28 22:35:18 +00002357 // Validation: user is most-likely passing an error code, and it would
Glenn Kasten99e53b82012-01-19 08:59:58 -08002358 // make the return value ambiguous (actualSize vs error).
Andy Hungfb8ede22018-09-12 19:03:24 -07002359 ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
Eric Laurent973db022018-11-20 14:54:31 -08002360 __func__, mPortId, buffer, userSize, userSize);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002361 return BAD_VALUE;
2362 }
2363
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002364 size_t written = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002365 Buffer audioBuffer;
2366
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002367 while (userSize >= mFrameSize) {
2368 audioBuffer.frameCount = userSize / mFrameSize;
Eric Laurentc2f1f072009-07-17 12:17:14 -07002369
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002370 status_t err = obtainBuffer(&audioBuffer,
2371 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002372 if (err < 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002373 if (written > 0) {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002374 break;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002375 }
Glenn Kasten0a2f1512016-07-22 08:06:37 -07002376 if (err == TIMED_OUT || err == -EINTR) {
2377 err = WOULD_BLOCK;
2378 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002379 return ssize_t(err);
2380 }
2381
Glenn Kastenae4b8792015-03-20 09:04:21 -07002382 size_t toWrite = audioBuffer.size;
Andy Hungabdb9902015-01-12 15:08:22 -08002383 memcpy(audioBuffer.i8, buffer, toWrite);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002384 buffer = ((const char *) buffer) + toWrite;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002385 userSize -= toWrite;
2386 written += toWrite;
2387
2388 releaseBuffer(&audioBuffer);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002389 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002390
Andy Hungea2b9c02016-02-12 17:06:53 -08002391 if (written > 0) {
2392 mFramesWritten += written / mFrameSize;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002393
2394 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2395 const sp<AudioTrackThread> t = mAudioTrackThread;
2396 if (t != 0) {
2397 // causes wake up of the playback thread, that will callback the client for
2398 // more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
2399 t->wake();
2400 }
2401 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002402 }
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002403
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002404 return written;
2405}
2406
2407// -------------------------------------------------------------------------
2408
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08002409nsecs_t AudioTrack::processAudioBuffer()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002410{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002411 // Currently the AudioTrack thread is not created if there are no callbacks.
2412 // Would it ever make sense to run the thread, even without callbacks?
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002413 // If so, then replace this by checks at each use for mCallback != NULL.
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002414 LOG_ALWAYS_FATAL_IF(mCblk == NULL);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002415 mLock.lock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002416 sp<IAudioTrackCallback> callback = mCallback.promote();
2417 if (!callback) {
2418 mCallback = nullptr;
2419 return NS_NEVER;
2420 }
Glenn Kastena07f17c2013-04-23 12:39:37 -07002421 if (mAwaitBoost) {
2422 mAwaitBoost = false;
2423 mLock.unlock();
2424 static const int32_t kMaxTries = 5;
2425 int32_t tryCounter = kMaxTries;
2426 uint32_t pollUs = 10000;
2427 do {
Glenn Kasten8255ba72016-08-23 13:54:23 -07002428 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002429 if (policy == SCHED_FIFO || policy == SCHED_RR) {
2430 break;
2431 }
2432 usleep(pollUs);
2433 pollUs <<= 1;
2434 } while (tryCounter-- > 0);
2435 if (tryCounter < 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002436 ALOGE("%s(%d): did not receive expected priority boost on time",
Eric Laurent973db022018-11-20 14:54:31 -08002437 __func__, mPortId);
Glenn Kastena07f17c2013-04-23 12:39:37 -07002438 }
Glenn Kastenb0dfd462013-07-10 16:52:47 -07002439 // Run again immediately
2440 return 0;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002441 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002442
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002443 // Can only reference mCblk while locked
2444 int32_t flags = android_atomic_and(
Glenn Kasten96f60d82013-07-12 10:21:18 -07002445 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);
Glenn Kastena47f3162012-11-07 10:13:08 -08002446
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002447 // Check for track invalidation
2448 if (flags & CBLK_INVALID) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002449 // for offloaded tracks restoreTrack_l() will just update the sequence and clear
2450 // AudioSystem cache. We should not exit here but after calling the callback so
2451 // that the upper layers can recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002452 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
Lajos Molnarf1063e22015-04-17 15:19:42 -07002453 status_t status __unused = restoreTrack_l("processAudioBuffer");
2454 // FIXME unused status
Andy Hung53c3b5f2014-12-15 16:42:05 -08002455 // after restoration, continue below to make sure that the loop and buffer events
2456 // are notified because they have been cleared from mCblk->mFlags above.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002457 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002458 }
2459
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002460 bool waitStreamEnd = mState == STATE_STOPPING;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002461 bool active = mState == STATE_ACTIVE;
2462
2463 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
2464 bool newUnderrun = false;
2465 if (flags & CBLK_UNDERRUN) {
2466#if 0
2467 // Currently in shared buffer mode, when the server reaches the end of buffer,
2468 // the track stays active in continuous underrun state. It's up to the application
2469 // to pause or stop the track, or set the position to a new offset within buffer.
2470 // This was some experimental code to auto-pause on underrun. Keeping it here
2471 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
2472 if (mTransfer == TRANSFER_SHARED) {
2473 mState = STATE_PAUSED;
2474 active = false;
2475 }
2476#endif
2477 if (!mInUnderrun) {
2478 mInUnderrun = true;
2479 newUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002480 }
2481 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002482
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002483 // Get current position of server
Andy Hung90e8a972015-11-09 16:42:40 -08002484 Modulo<uint32_t> position(updateAndGetPosition_l());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002485
2486 // Manage marker callback
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002487 bool markerReached = false;
Andy Hung90e8a972015-11-09 16:42:40 -08002488 Modulo<uint32_t> markerPosition(mMarkerPosition);
2489 // uses 32 bit wraparound for comparison with position.
2490 if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002491 mMarkerReached = markerReached = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002492 }
2493
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002494 // Determine number of new position callback(s) that will be needed, while locked
2495 size_t newPosCount = 0;
Andy Hung90e8a972015-11-09 16:42:40 -08002496 Modulo<uint32_t> newPosition(mNewPosition);
2497 uint32_t updatePeriod = mUpdatePeriod;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002498 // FIXME fails for wraparound, need 64 bits
2499 if (updatePeriod > 0 && position >= newPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002500 newPosCount = ((position - newPosition).value() / updatePeriod) + 1;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002501 mNewPosition += updatePeriod * newPosCount;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002502 }
2503
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002504 // Cache other fields that will be needed soon
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002505 uint32_t sampleRate = mSampleRate;
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002506 float speed = mPlaybackRate.mSpeed;
Andy Hunga7f03352015-05-31 21:54:49 -07002507 const uint32_t notificationFrames = mNotificationFramesAct;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002508 if (mRefreshRemaining) {
2509 mRefreshRemaining = false;
2510 mRemainingFrames = notificationFrames;
2511 mRetryOnPartialBuffer = false;
2512 }
2513 size_t misalignment = mProxy->getMisalignment();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002514 uint32_t sequence = mSequence;
Glenn Kasten96f04882013-09-20 09:28:56 -07002515 sp<AudioTrackClientProxy> proxy = mProxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002516
Andy Hung53c3b5f2014-12-15 16:42:05 -08002517 // Determine the number of new loop callback(s) that will be needed, while locked.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002518 uint32_t loopCountNotifications = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -08002519 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
2520
2521 if (mLoopCount > 0) {
2522 int loopCount;
2523 size_t bufferPosition;
2524 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
2525 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
2526 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
2527 mLoopCountNotified = loopCount; // discard any excess notifications
2528 } else if (mLoopCount < 0) {
2529 // FIXME: We're not accurate with notification count and position with infinite looping
2530 // since loopCount from server side will always return -1 (we could decrement it).
2531 size_t bufferPosition = mStaticProxy->getBufferPosition();
2532 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
2533 loopPeriod = mLoopEnd - bufferPosition;
2534 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
2535 size_t bufferPosition = mStaticProxy->getBufferPosition();
2536 loopPeriod = mFrameCount - bufferPosition;
2537 }
2538
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002539 // These fields don't need to be cached, because they are assigned only by set():
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002540 // mTransfer, mCallback, mUserData, mFormat, mFrameSize, mFlags
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002541 // mFlags is also assigned by createTrack_l(), but not the bit we care about.
2542
2543 mLock.unlock();
2544
Andy Hunga7f03352015-05-31 21:54:49 -07002545 // get anchor time to account for callbacks.
2546 const nsecs_t timeBeforeCallbacks = systemTime();
2547
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002548 if (waitStreamEnd) {
Andy Hunga7f03352015-05-31 21:54:49 -07002549 // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread
2550 // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
2551 // (and make sure we don't callback for more data while we're stopping).
2552 // This helps with position, marker notifications, and track invalidation.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002553 struct timespec timeout;
2554 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
2555 timeout.tv_nsec = 0;
2556
Glenn Kasten96f04882013-09-20 09:28:56 -07002557 status_t status = proxy->waitStreamEndDone(&timeout);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002558 switch (status) {
2559 case NO_ERROR:
2560 case DEAD_OBJECT:
2561 case TIMED_OUT:
Andy Hung39609a02015-09-03 16:38:38 -07002562 if (status != DEAD_OBJECT) {
2563 // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop();
2564 // instead, the application should handle the EVENT_NEW_IAUDIOTRACK.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002565 callback->onStreamEnd();
Andy Hung39609a02015-09-03 16:38:38 -07002566 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002567 {
2568 AutoMutex lock(mLock);
2569 // The previously assigned value of waitStreamEnd is no longer valid,
2570 // since the mutex has been unlocked and either the callback handler
2571 // or another thread could have re-started the AudioTrack during that time.
2572 waitStreamEnd = mState == STATE_STOPPING;
2573 if (waitStreamEnd) {
2574 mState = STATE_STOPPED;
Andy Hungc2813e52014-10-16 17:54:34 -07002575 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002576 }
2577 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002578 if (waitStreamEnd && status != DEAD_OBJECT) {
2579 return NS_INACTIVE;
2580 }
2581 break;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002582 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002583 return 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002584 }
2585
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002586 // perform callbacks while unlocked
2587 if (newUnderrun) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002588 callback->onUnderrun();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002589 }
Andy Hung53c3b5f2014-12-15 16:42:05 -08002590 while (loopCountNotifications > 0) {
Andy Hung53c3b5f2014-12-15 16:42:05 -08002591 --loopCountNotifications;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002592 callback->onLoopEnd(mLoopCount > 0 ? loopCountNotifications + mLoopCountNotified : -1);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002593 }
2594 if (flags & CBLK_BUFFER_END) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002595 callback->onBufferEnd();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002596 }
2597 if (markerReached) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002598 callback->onMarker(markerPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002599 }
2600 while (newPosCount > 0) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002601 callback->onNewPos(newPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002602 newPosition += updatePeriod;
2603 newPosCount--;
2604 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002605
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002606 if (mObservedSequence != sequence) {
2607 mObservedSequence = sequence;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002608 callback->onNewIAudioTrack();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002609 // for offloaded tracks, just wait for the upper layers to recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002610 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002611 return NS_INACTIVE;
2612 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002613 }
2614
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002615 // if inactive, then don't run me again until re-started
2616 if (!active) {
2617 return NS_INACTIVE;
Eric Laurent2267ba12011-09-07 11:13:23 -07002618 }
2619
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002620 // Compute the estimated time until the next timed event (position, markers, loops)
2621 // FIXME only for non-compressed audio
2622 uint32_t minFrames = ~0;
2623 if (!markerReached && position < markerPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002624 minFrames = (markerPosition - position).value();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002625 }
2626 if (loopPeriod > 0 && loopPeriod < minFrames) {
Andy Hung2d85f092015-01-07 12:45:13 -08002627 // loopPeriod is already adjusted for actual position.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002628 minFrames = loopPeriod;
2629 }
Andy Hung2d85f092015-01-07 12:45:13 -08002630 if (updatePeriod > 0) {
Andy Hung90e8a972015-11-09 16:42:40 -08002631 minFrames = min(minFrames, (newPosition - position).value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002632 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002633
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002634 // If > 0, poll periodically to recover from a stuck server. A good value is 2.
2635 static const uint32_t kPoll = 0;
2636 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
2637 minFrames = kPoll * notificationFrames;
2638 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002639
Andy Hunga7f03352015-05-31 21:54:49 -07002640 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
2641 static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL;
2642 const nsecs_t timeAfterCallbacks = systemTime();
2643
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002644 // Convert frame units to time units
2645 nsecs_t ns = NS_WHENEVER;
2646 if (minFrames != (uint32_t) ~0) {
jiabinc7bb8322017-09-06 18:20:11 -07002647 // AudioFlinger consumption of client data may be irregular when coming out of device
2648 // standby since the kernel buffers require filling. This is throttled to no more than 2x
2649 // the expected rate in the MixerThread. Hence, we reduce the estimated time to wait by one
2650 // half (but no more than half a second) to improve callback accuracy during these temporary
2651 // data surges.
2652 const nsecs_t estimatedNs = framesToNanoseconds(minFrames, sampleRate, speed);
2653 constexpr nsecs_t maxThrottleCompensationNs = 500000000LL;
2654 ns = estimatedNs - min(estimatedNs / 2, maxThrottleCompensationNs) + kWaitPeriodNs;
Andy Hunga7f03352015-05-31 21:54:49 -07002655 ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time
2656 // TODO: Should we warn if the callback time is too long?
2657 if (ns < 0) ns = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002658 }
2659
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002660 // If not supplying data by EVENT_MORE_DATA or EVENT_CAN_WRITE_MORE_DATA, then we're done
2661 if (mTransfer != TRANSFER_CALLBACK && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002662 return ns;
2663 }
2664
Andy Hunga7f03352015-05-31 21:54:49 -07002665 // EVENT_MORE_DATA callback handling.
2666 // Timing for linear pcm audio data formats can be derived directly from the
2667 // buffer fill level.
2668 // Timing for compressed data is not directly available from the buffer fill level,
2669 // rather indirectly from waiting for blocking mode callbacks or waiting for obtain()
2670 // to return a certain fill level.
2671
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002672 struct timespec timeout;
2673 const struct timespec *requested = &ClientProxy::kForever;
2674 if (ns != NS_WHENEVER) {
2675 timeout.tv_sec = ns / 1000000000LL;
2676 timeout.tv_nsec = ns % 1000000000LL;
Andy Hungfb8ede22018-09-12 19:03:24 -07002677 ALOGV("%s(%d): timeout %ld.%03d",
Eric Laurent973db022018-11-20 14:54:31 -08002678 __func__, mPortId, timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002679 requested = &timeout;
2680 }
2681
Andy Hungea2b9c02016-02-12 17:06:53 -08002682 size_t writtenFrames = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002683 while (mRemainingFrames > 0) {
2684
2685 Buffer audioBuffer;
2686 audioBuffer.frameCount = mRemainingFrames;
2687 size_t nonContig;
2688 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
2689 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
Andy Hungfb8ede22018-09-12 19:03:24 -07002690 "%s(%d): obtainBuffer() err=%d frameCount=%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002691 __func__, mPortId, err, audioBuffer.frameCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002692 requested = &ClientProxy::kNonBlocking;
2693 size_t avail = audioBuffer.frameCount + nonContig;
Andy Hungfb8ede22018-09-12 19:03:24 -07002694 ALOGV("%s(%d): obtainBuffer(%u) returned %zu = %zu + %zu err %d",
Eric Laurent973db022018-11-20 14:54:31 -08002695 __func__, mPortId, mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002696 if (err != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002697 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
2698 (isOffloaded() && (err == DEAD_OBJECT))) {
Glenn Kasten606fbc12015-10-22 15:28:15 -07002699 // FIXME bug 25195759
2700 return 1000000;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002701 }
Andy Hungfb8ede22018-09-12 19:03:24 -07002702 ALOGE("%s(%d): Error %d obtaining an audio buffer, giving up.",
Eric Laurent973db022018-11-20 14:54:31 -08002703 __func__, mPortId, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002704 return NS_NEVER;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002705 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002706
Phil Burkfdb3c072016-02-09 10:47:02 -08002707 if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002708 mRetryOnPartialBuffer = false;
2709 if (avail < mRemainingFrames) {
Andy Hunga7f03352015-05-31 21:54:49 -07002710 if (ns > 0) { // account for obtain time
2711 const nsecs_t timeNow = systemTime();
2712 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2713 }
Andy Hungeb6e6502019-04-29 13:47:40 -07002714
2715 // delayNs is first computed by the additional frames required in the buffer.
2716 nsecs_t delayNs = framesToNanoseconds(
2717 mRemainingFrames - avail, sampleRate, speed);
2718
2719 // afNs is the AudioFlinger mixer period in ns.
2720 const nsecs_t afNs = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2721
2722 // If the AudioTrack is double buffered based on the AudioFlinger mixer period,
2723 // we may have a race if we wait based on the number of frames desired.
2724 // This is a possible issue with resampling and AAudio.
2725 //
2726 // The granularity of audioflinger processing is one mixer period; if
2727 // our wait time is less than one mixer period, wait at most half the period.
2728 if (delayNs < afNs) {
2729 delayNs = std::min(delayNs, afNs / 2);
2730 }
2731
2732 // adjust our ns wait by delayNs.
2733 if (ns < 0 /* NS_WHENEVER */ || delayNs < ns) {
2734 ns = delayNs;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002735 }
2736 return ns;
2737 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07002738 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002739
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002740 size_t reqSize = audioBuffer.size;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002741 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2742 // when notifying client it can write more data, pass the total size that can be
2743 // written in the next write() call, since it's not passed through the callback
2744 audioBuffer.size += nonContig;
2745 }
Atneya Nairc2dd1272021-10-26 19:39:51 -04002746 const size_t writtenSize = (mTransfer == TRANSFER_CALLBACK)
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002747 ? callback->onMoreData(audioBuffer)
2748 : callback->onCanWriteMoreData(audioBuffer);
Jiabin Huang447cea72020-07-28 22:35:18 +00002749 // Validate on returned size
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002750 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002751 ALOGE("%s(%d): EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
Eric Laurent973db022018-11-20 14:54:31 -08002752 __func__, mPortId, reqSize, ssize_t(writtenSize));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002753 return NS_NEVER;
2754 }
2755
2756 if (writtenSize == 0) {
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002757 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2758 // The callback EVENT_CAN_WRITE_MORE_DATA was processed in the JNI of
2759 // android.media.AudioTrack. The JNI is not using the callback to provide data,
2760 // it only signals to the Java client that it can provide more data, which
2761 // this track is read to accept now.
2762 // The playback thread will be awaken at the next ::write()
2763 return NS_WHENEVER;
2764 }
The Android Open Source Project8555d082009-03-05 14:34:35 -08002765 // The callback is done filling buffers
2766 // Keep this thread going to handle timed events and
2767 // still try to get more data in intervals of WAIT_PERIOD_MS
2768 // but don't just loop and block the CPU, so wait
Andy Hunga7f03352015-05-31 21:54:49 -07002769
2770 // mCbf(EVENT_MORE_DATA, ...) might either
2771 // (1) Block until it can fill the buffer, returning 0 size on EOS.
2772 // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS.
2773 // (3) Return 0 size when no data is available, does not wait for more data.
2774 //
2775 // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer.
2776 // We try to compute the wait time to avoid a tight sleep-wait cycle,
2777 // especially for case (3).
2778 //
2779 // The decision to support (1) and (2) affect the sizing of mRemainingFrames
2780 // and this loop; whereas for case (3) we could simply check once with the full
2781 // buffer size and skip the loop entirely.
2782
2783 nsecs_t myns;
Phil Burkfdb3c072016-02-09 10:47:02 -08002784 if (audio_has_proportional_frames(mFormat)) {
Andy Hunga7f03352015-05-31 21:54:49 -07002785 // time to wait based on buffer occupancy
2786 const nsecs_t datans = mRemainingFrames <= avail ? 0 :
2787 framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
2788 // audio flinger thread buffer size (TODO: adjust for fast tracks)
Glenn Kastenea38ee72016-04-18 11:08:01 -07002789 // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks.
Andy Hunga7f03352015-05-31 21:54:49 -07002790 const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2791 // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0.
2792 myns = datans + (afns / 2);
2793 } else {
2794 // FIXME: This could ping quite a bit if the buffer isn't full.
2795 // Note that when mState is stopping we waitStreamEnd, so it never gets here.
2796 myns = kWaitPeriodNs;
2797 }
2798 if (ns > 0) { // account for obtain and callback time
2799 const nsecs_t timeNow = systemTime();
2800 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2801 }
2802 if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
2803 ns = myns;
2804 }
2805 return ns;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002806 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002807
Atneya Nairc2dd1272021-10-26 19:39:51 -04002808 // releaseBuffer reads from audioBuffer.size
2809 audioBuffer.size = writtenSize;
2810
Glenn Kasten138d6f92015-03-20 10:54:51 -07002811 size_t releasedFrames = writtenSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002812 audioBuffer.frameCount = releasedFrames;
2813 mRemainingFrames -= releasedFrames;
2814 if (misalignment >= releasedFrames) {
2815 misalignment -= releasedFrames;
2816 } else {
2817 misalignment = 0;
2818 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002819
2820 releaseBuffer(&audioBuffer);
Andy Hungea2b9c02016-02-12 17:06:53 -08002821 writtenFrames += releasedFrames;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002822
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002823 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
2824 // if callback doesn't like to accept the full chunk
2825 if (writtenSize < reqSize) {
2826 continue;
2827 }
2828
2829 // There could be enough non-contiguous frames available to satisfy the remaining request
2830 if (mRemainingFrames <= nonContig) {
2831 continue;
2832 }
2833
2834#if 0
2835 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
2836 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
2837 // that total to a sum == notificationFrames.
2838 if (0 < misalignment && misalignment <= mRemainingFrames) {
2839 mRemainingFrames = misalignment;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002840 return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002841 }
2842#endif
2843
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002844 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002845 if (writtenFrames > 0) {
2846 AutoMutex lock(mLock);
2847 mFramesWritten += writtenFrames;
2848 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002849 mRemainingFrames = notificationFrames;
2850 mRetryOnPartialBuffer = true;
2851
2852 // A lot has transpired since ns was calculated, so run again immediately and re-calculate
2853 return 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002854}
2855
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002856status_t AudioTrack::restoreTrack_l(const char *from)
Eric Laurent1703cdf2011-03-07 14:52:59 -08002857{
Andy Hungb68f5eb2019-12-03 16:49:17 -08002858 status_t result = NO_ERROR; // logged: make sure to set this before returning.
2859 const int64_t beginNs = systemTime();
Andy Hunga6b27032020-04-27 10:34:24 -07002860 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -08002861 mediametrics::LogItem(mMetricsId)
2862 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_RESTORE)
Andy Hungea840382020-05-05 21:50:17 -07002863 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08002864 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
2865 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
2866 .set(AMEDIAMETRICS_PROP_WHERE, from)
2867 .record(); });
2868
Andy Hungfb8ede22018-09-12 19:03:24 -07002869 ALOGW("%s(%d): dead IAudioTrack, %s, creating a new one from %s()",
Eric Laurent973db022018-11-20 14:54:31 -08002870 __func__, mPortId, isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002871 ++mSequence;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002872
Glenn Kastena47f3162012-11-07 10:13:08 -08002873 // refresh the audio configuration cache in this process to make sure we get new
Glenn Kastend2d089f2014-11-05 11:48:12 -08002874 // output parameters and new IAudioFlinger in createTrack_l()
Glenn Kastena47f3162012-11-07 10:13:08 -08002875 AudioSystem::clearAudioConfigCache();
Eric Laurent9f6530f2011-08-30 10:18:54 -07002876
Ronghua Wufaeb0f22015-05-21 12:20:21 -07002877 if (isOffloadedOrDirect_l() || mDoNotReconnect) {
Andy Hung1f1db832015-06-08 13:26:10 -07002878 // FIXME re-creation of offloaded and direct tracks is not yet implemented;
2879 // reconsider enabling for linear PCM encodings when position can be preserved.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002880 result = DEAD_OBJECT;
2881 return result;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002882 }
2883
Phil Burk2812d9e2016-01-04 10:34:30 -08002884 // Save so we can return count since creation.
2885 mUnderrunCountOffset = getUnderrunCount_l();
2886
Glenn Kasten200092b2014-08-15 15:13:30 -07002887 // save the old static buffer position
Andy Hungf20a4e92016-08-15 19:10:34 -07002888 uint32_t staticPosition = 0;
Andy Hung4ede21d2014-12-12 15:37:34 -08002889 size_t bufferPosition = 0;
2890 int loopCount = 0;
2891 if (mStaticProxy != 0) {
2892 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
Andy Hungf20a4e92016-08-15 19:10:34 -07002893 staticPosition = mStaticProxy->getPosition().unsignedValue();
Andy Hung4ede21d2014-12-12 15:37:34 -08002894 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002895
Andy Hung3c7f47a2021-03-16 17:30:09 -07002896 // save the old startThreshold and framecount
2897 const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
2898 const uint32_t originalFrameCount = mProxy->frameCount();
2899
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002900 // See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
2901 // causes a lot of churn on the service side, and it can reject starting
2902 // playback of a previously created track. May also apply to other cases.
2903 const int INITIAL_RETRIES = 3;
2904 int retries = INITIAL_RETRIES;
2905retry:
2906 if (retries < INITIAL_RETRIES) {
2907 // See the comment for clearAudioConfigCache at the start of the function.
2908 AudioSystem::clearAudioConfigCache();
2909 }
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08002910 mFlags = mOrigFlags;
2911
Glenn Kasten200092b2014-08-15 15:13:30 -07002912 // If a new IAudioTrack is successfully created, createTrack_l() will modify the
Glenn Kastena47f3162012-11-07 10:13:08 -08002913 // following member variables: mAudioTrack, mCblkMemory and mCblk.
Glenn Kasten200092b2014-08-15 15:13:30 -07002914 // It will also delete the strong references on previous IAudioTrack and IMemory.
2915 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002916 result = createTrack_l();
Eric Laurentcc21e4f2013-10-16 15:12:32 -07002917
Eric Laurent6ec546d2018-10-10 16:52:14 -07002918 if (result == NO_ERROR) {
Andy Hungd7bd69e2015-07-24 07:52:41 -07002919 // take the frames that will be lost by track recreation into account in saved position
2920 // For streaming tracks, this is the amount we obtained from the user/client
2921 // (not the number actually consumed at the server - those are already lost).
2922 if (mStaticProxy == 0) {
2923 mPosition = mReleased;
2924 }
Andy Hung4ede21d2014-12-12 15:37:34 -08002925 // Continue playback from last known position and restore loop.
2926 if (mStaticProxy != 0) {
2927 if (loopCount != 0) {
2928 mStaticProxy->setBufferPositionAndLoop(bufferPosition,
2929 mLoopStart, mLoopEnd, loopCount);
2930 } else {
2931 mStaticProxy->setBufferPosition(bufferPosition);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002932 if (bufferPosition == mFrameCount) {
Eric Laurent973db022018-11-20 14:54:31 -08002933 ALOGD("%s(%d): restoring track at end of static buffer", __func__, mPortId);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002934 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002935 }
2936 }
Andy Hung4ef88d72017-02-21 19:47:53 -08002937 // restore volume handler
Andy Hung39399b62017-04-21 15:07:45 -07002938 mVolumeHandler->forall([this](const VolumeShaper &shaper) -> VolumeShaper::Status {
2939 sp<VolumeShaper::Operation> operationToEnd =
2940 new VolumeShaper::Operation(shaper.mOperation);
Andy Hung4ef88d72017-02-21 19:47:53 -08002941 // TODO: Ideally we would restore to the exact xOffset position
2942 // as returned by getVolumeShaperState(), but we don't have that
2943 // information when restoring at the client unless we periodically poll
2944 // the server or create shared memory state.
2945 //
Andy Hung39399b62017-04-21 15:07:45 -07002946 // For now, we simply advance to the end of the VolumeShaper effect
2947 // if it has been started.
2948 if (shaper.isStarted()) {
Andy Hungf3702642017-05-05 17:33:32 -07002949 operationToEnd->setNormalizedTime(1.f);
Andy Hung39399b62017-04-21 15:07:45 -07002950 }
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002951 media::VolumeShaperConfiguration config;
2952 shaper.mConfiguration->writeToParcelable(&config);
2953 media::VolumeShaperOperation operation;
2954 operationToEnd->writeToParcelable(&operation);
2955 status_t status;
2956 mAudioTrack->applyVolumeShaper(config, operation, &status);
2957 return status;
Andy Hung4ef88d72017-02-21 19:47:53 -08002958 });
2959
Andy Hung3c7f47a2021-03-16 17:30:09 -07002960 // restore the original start threshold if different than frameCount.
2961 if (originalStartThresholdInFrames != originalFrameCount) {
2962 // Note: mProxy->setStartThresholdInFrames() call is in the Proxy
2963 // and does not trigger a restart.
2964 // (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
2965 // Any start would be triggered on the mState == ACTIVE check below.
2966 const uint32_t currentThreshold =
2967 mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
2968 ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
2969 "%s(%d) startThresholdInFrames changing from %u to %u",
2970 __func__, mPortId, originalStartThresholdInFrames, currentThreshold);
2971 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002972 if (mState == STATE_ACTIVE) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002973 mAudioTrack->start(&result);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002974 }
Andy Hungf20a4e92016-08-15 19:10:34 -07002975 // server resets to zero so we offset
2976 mFramesWrittenServerOffset =
2977 mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten;
2978 mFramesWrittenAtRestore = mFramesWrittenServerOffset;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002979 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002980 if (result != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08002981 ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002982 if (--retries > 0) {
Eric Laurent6ec546d2018-10-10 16:52:14 -07002983 // leave time for an eventual race condition to clear before retrying
2984 usleep(500000);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002985 goto retry;
2986 }
Eric Laurent6ec546d2018-10-10 16:52:14 -07002987 // if no retries left, set invalid bit to force restoring at next occasion
2988 // and avoid inconsistent active state on client and server sides
2989 if (mCblk != nullptr) {
2990 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
2991 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002992 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002993 return result;
2994}
2995
Andy Hung90e8a972015-11-09 16:42:40 -08002996Modulo<uint32_t> AudioTrack::updateAndGetPosition_l()
Glenn Kasten200092b2014-08-15 15:13:30 -07002997{
2998 // This is the sole place to read server consumed frames
Andy Hung90e8a972015-11-09 16:42:40 -08002999 Modulo<uint32_t> newServer(mProxy->getPosition());
3000 const int32_t delta = (newServer - mServer).signedValue();
Glenn Kasten200092b2014-08-15 15:13:30 -07003001 // TODO There is controversy about whether there can be "negative jitter" in server position.
3002 // This should be investigated further, and if possible, it should be addressed.
3003 // A more definite failure mode is infrequent polling by client.
3004 // One could call (void)getPosition_l() in releaseBuffer(),
3005 // so mReleased and mPosition are always lock-step as best possible.
3006 // That should ensure delta never goes negative for infrequent polling
3007 // unless the server has more than 2^31 frames in its buffer,
3008 // in which case the use of uint32_t for these counters has bigger issues.
Andy Hung90e8a972015-11-09 16:42:40 -08003009 ALOGE_IF(delta < 0,
Andy Hungfb8ede22018-09-12 19:03:24 -07003010 "%s(%d): detected illegal retrograde motion by the server: mServer advanced by %d",
Eric Laurent973db022018-11-20 14:54:31 -08003011 __func__, mPortId, delta);
Chad Brubaker039c27a2015-09-23 15:17:29 -07003012 mServer = newServer;
Andy Hung90e8a972015-11-09 16:42:40 -08003013 if (delta > 0) { // avoid retrograde
3014 mPosition += delta;
3015 }
3016 return mPosition;
Glenn Kasten200092b2014-08-15 15:13:30 -07003017}
3018
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003019bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed)
Andy Hung8edb8dc2015-03-26 19:13:55 -07003020{
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003021 updateLatency_l();
Andy Hung8edb8dc2015-03-26 19:13:55 -07003022 // applicable for mixing tracks only (not offloaded or direct)
3023 if (mStaticProxy != 0) {
3024 return true; // static tracks do not have issues with buffer sizing.
3025 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07003026 const size_t minFrameCount =
Eric Laurent21da6472017-11-09 16:29:26 -08003027 AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
3028 sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003029 const bool allowed = mFrameCount >= minFrameCount;
3030 ALOGD_IF(!allowed,
Andy Hungfb8ede22018-09-12 19:03:24 -07003031 "%s(%d): denied "
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003032 "mAfLatency:%u mAfFrameCount:%zu mAfSampleRate:%u sampleRate:%u speed:%f "
3033 "mFrameCount:%zu < minFrameCount:%zu",
Eric Laurent973db022018-11-20 14:54:31 -08003034 __func__, mPortId,
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003035 mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed,
Andy Hung8edb8dc2015-03-26 19:13:55 -07003036 mFrameCount, minFrameCount);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003037 return allowed;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003038}
3039
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003040status_t AudioTrack::setParameters(const String8& keyValuePairs)
3041{
3042 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003043 status_t status;
3044 mAudioTrack->setParameters(keyValuePairs.c_str(), &status);
3045 return status;
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003046}
3047
Dean Wheatleya70eef72018-01-04 14:23:50 +11003048status_t AudioTrack::selectPresentation(int presentationId, int programId)
3049{
3050 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08003051 AudioParameter param = AudioParameter();
3052 param.addInt(String8(AudioParameter::keyPresentationId), presentationId);
3053 param.addInt(String8(AudioParameter::keyProgramId), programId);
3054 ALOGV("%s(%d): PresentationId/ProgramId[%s]",
3055 __func__, mPortId, param.toString().string());
3056
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003057 status_t status;
3058 mAudioTrack->setParameters(param.toString().c_str(), &status);
3059 return status;
Dean Wheatleya70eef72018-01-04 14:23:50 +11003060}
3061
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003062VolumeShaper::Status AudioTrack::applyVolumeShaper(
3063 const sp<VolumeShaper::Configuration>& configuration,
3064 const sp<VolumeShaper::Operation>& operation)
3065{
3066 AutoMutex lock(mLock);
Andy Hung4ef88d72017-02-21 19:47:53 -08003067 mVolumeHandler->setIdIfNecessary(configuration);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003068 media::VolumeShaperConfiguration config;
3069 configuration->writeToParcelable(&config);
3070 media::VolumeShaperOperation op;
3071 operation->writeToParcelable(&op);
3072 VolumeShaper::Status status;
3073 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003074
3075 if (status == DEAD_OBJECT) {
3076 if (restoreTrack_l("applyVolumeShaper") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003077 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003078 }
3079 }
Andy Hung4ef88d72017-02-21 19:47:53 -08003080 if (status >= 0) {
3081 // save VolumeShaper for restore
3082 mVolumeHandler->applyVolumeShaper(configuration, operation);
Andy Hung39399b62017-04-21 15:07:45 -07003083 if (mState == STATE_ACTIVE || mState == STATE_STOPPING) {
3084 mVolumeHandler->setStarted();
3085 }
3086 } else {
3087 // warn only if not an expected restore failure.
3088 ALOGW_IF(!((isOffloadedOrDirect_l() || mDoNotReconnect) && status == DEAD_OBJECT),
Eric Laurent973db022018-11-20 14:54:31 -08003089 "%s(%d): applyVolumeShaper failed: %d", __func__, mPortId, status);
Andy Hung4ef88d72017-02-21 19:47:53 -08003090 }
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003091 return status;
3092}
3093
3094sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id)
3095{
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003096 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003097 std::optional<media::VolumeShaperState> vss;
3098 mAudioTrack->getVolumeShaperState(id, &vss);
3099 sp<VolumeShaper::State> state;
3100 if (vss.has_value()) {
3101 state = new VolumeShaper::State();
3102 state->readFromParcelable(vss.value());
3103 }
Andy Hung39399b62017-04-21 15:07:45 -07003104 if (state.get() == nullptr && (mCblk->mFlags & CBLK_INVALID) != 0) {
3105 if (restoreTrack_l("getVolumeShaperState") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003106 mAudioTrack->getVolumeShaperState(id, &vss);
3107 if (vss.has_value()) {
3108 state = new VolumeShaper::State();
3109 state->readFromParcelable(vss.value());
3110 }
Andy Hung39399b62017-04-21 15:07:45 -07003111 }
3112 }
3113 return state;
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003114}
3115
Andy Hungea2b9c02016-02-12 17:06:53 -08003116status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp)
3117{
3118 if (timestamp == nullptr) {
3119 return BAD_VALUE;
3120 }
3121 AutoMutex lock(mLock);
Andy Hunge13f8a62016-03-30 14:20:42 -07003122 return getTimestamp_l(timestamp);
3123}
3124
3125status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp)
3126{
Andy Hungea2b9c02016-02-12 17:06:53 -08003127 if (mCblk->mFlags & CBLK_INVALID) {
3128 const status_t status = restoreTrack_l("getTimestampExtended");
3129 if (status != OK) {
3130 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3131 // recommending that the track be recreated.
3132 return DEAD_OBJECT;
3133 }
3134 }
3135 // check for offloaded/direct here in case restoring somehow changed those flags.
3136 if (isOffloadedOrDirect_l()) {
3137 return INVALID_OPERATION; // not supported
3138 }
3139 status_t status = mProxy->getTimestamp(timestamp);
Andy Hungfb8ede22018-09-12 19:03:24 -07003140 LOG_ALWAYS_FATAL_IF(status != OK, "%s(%d): status %d not allowed from proxy getTimestamp",
Eric Laurent973db022018-11-20 14:54:31 -08003141 __func__, mPortId, status);
Andy Hungea2b9c02016-02-12 17:06:53 -08003142 bool found = false;
Andy Hunge1e98462016-04-12 10:18:51 -07003143 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten;
3144 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
3145 // server side frame offset in case AudioTrack has been restored.
3146 for (int i = ExtendedTimestamp::LOCATION_SERVER;
3147 i < ExtendedTimestamp::LOCATION_MAX; ++i) {
3148 if (timestamp->mTimeNs[i] >= 0) {
3149 // apply server offset (frames flushed is ignored
3150 // so we don't report the jump when the flush occurs).
3151 timestamp->mPosition[i] += mFramesWrittenServerOffset;
3152 found = true;
Andy Hungea2b9c02016-02-12 17:06:53 -08003153 }
3154 }
3155 return found ? OK : WOULD_BLOCK;
3156}
3157
Glenn Kastence703742013-07-19 16:33:58 -07003158status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
3159{
Glenn Kasten53cec222013-08-29 09:01:02 -07003160 AutoMutex lock(mLock);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003161 return getTimestamp_l(timestamp);
3162}
Phil Burk1b420972015-04-22 10:52:21 -07003163
Andy Hung65ffdfc2016-10-10 15:52:11 -07003164status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp)
3165{
Phil Burk1b420972015-04-22 10:52:21 -07003166 bool previousTimestampValid = mPreviousTimestampValid;
3167 // Set false here to cover all the error return cases.
3168 mPreviousTimestampValid = false;
3169
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003170 switch (mState) {
3171 case STATE_ACTIVE:
3172 case STATE_PAUSED:
3173 break; // handle below
3174 case STATE_FLUSHED:
3175 case STATE_STOPPED:
3176 return WOULD_BLOCK;
3177 case STATE_STOPPING:
3178 case STATE_PAUSED_STOPPING:
3179 if (!isOffloaded_l()) {
3180 return INVALID_OPERATION;
3181 }
3182 break; // offloaded tracks handled below
3183 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003184 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in getTimestamp(): %d",
Eric Laurent973db022018-11-20 14:54:31 -08003185 __func__, mPortId, mState);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003186 break;
Glenn Kastenfe346c72013-08-30 13:28:22 -07003187 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003188
Eric Laurent275e8e92014-11-30 15:14:47 -08003189 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung6653c932015-06-08 13:27:48 -07003190 const status_t status = restoreTrack_l("getTimestamp");
3191 if (status != OK) {
3192 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3193 // recommending that the track be recreated.
3194 return DEAD_OBJECT;
3195 }
Eric Laurent275e8e92014-11-30 15:14:47 -08003196 }
3197
Glenn Kasten200092b2014-08-15 15:13:30 -07003198 // The presented frame count must always lag behind the consumed frame count.
3199 // To avoid a race, read the presented frames first. This ensures that presented <= consumed.
Andy Hung6ae58432016-02-16 18:32:24 -08003200
3201 status_t status;
Andy Hung818e7a32016-02-16 18:08:07 -08003202 if (isOffloadedOrDirect_l()) {
Andy Hung6ae58432016-02-16 18:32:24 -08003203 // use Binder to get timestamp
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003204 media::AudioTimestampInternal ts;
3205 mAudioTrack->getTimestamp(&ts, &status);
3206 if (status == OK) {
Andy Hung973638a2020-12-08 20:47:45 -08003207 timestamp = VALUE_OR_FATAL(aidl2legacy_AudioTimestampInternal_AudioTimestamp(ts));
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003208 }
Andy Hung6ae58432016-02-16 18:32:24 -08003209 } else {
3210 // read timestamp from shared memory
3211 ExtendedTimestamp ets;
3212 status = mProxy->getTimestamp(&ets);
3213 if (status == OK) {
Andy Hungb01faa32016-04-27 12:51:32 -07003214 ExtendedTimestamp::Location location;
3215 status = ets.getBestTimestamp(&timestamp, &location);
3216
3217 if (status == OK) {
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003218 updateLatency_l();
Andy Hungb01faa32016-04-27 12:51:32 -07003219 // It is possible that the best location has moved from the kernel to the server.
3220 // In this case we adjust the position from the previous computed latency.
3221 if (location == ExtendedTimestamp::LOCATION_SERVER) {
3222 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL,
Andy Hungfb8ede22018-09-12 19:03:24 -07003223 "%s(%d): location moved from kernel to server",
Eric Laurent973db022018-11-20 14:54:31 -08003224 __func__, mPortId);
Andy Hung07eee802016-06-21 16:47:16 -07003225 // check that the last kernel OK time info exists and the positions
3226 // are valid (if they predate the current track, the positions may
3227 // be zero or negative).
Andy Hungb01faa32016-04-27 12:51:32 -07003228 const int64_t frames =
Andy Hung6d7b1192016-05-07 22:59:48 -07003229 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
Andy Hung07eee802016-06-21 16:47:16 -07003230 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 ||
3231 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 ||
3232 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0)
Andy Hung6d7b1192016-05-07 22:59:48 -07003233 ?
3234 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed
3235 / 1000)
3236 :
3237 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3238 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]);
Andy Hungfb8ede22018-09-12 19:03:24 -07003239 ALOGV("%s(%d): frame adjustment:%lld timestamp:%s",
Eric Laurent973db022018-11-20 14:54:31 -08003240 __func__, mPortId, (long long)frames, ets.toString().c_str());
Andy Hungb01faa32016-04-27 12:51:32 -07003241 if (frames >= ets.mPosition[location]) {
3242 timestamp.mPosition = 0;
3243 } else {
3244 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames);
3245 }
Andy Hung69488c42016-05-16 18:43:33 -07003246 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) {
3247 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER,
Andy Hungfb8ede22018-09-12 19:03:24 -07003248 "%s(%d): location moved from server to kernel",
Eric Laurent973db022018-11-20 14:54:31 -08003249 __func__, mPortId);
Andy Hung98731a22019-04-08 19:19:07 -07003250
3251 if (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER] ==
3252 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL]) {
3253 // In Q, we don't return errors as an invalid time
3254 // but instead we leave the last kernel good timestamp alone.
3255 //
3256 // If server is identical to kernel, the device data pipeline is idle.
3257 // A better start time is now. The retrograde check ensures
3258 // timestamp monotonicity.
3259 const int64_t nowNs = systemTime();
Andy Hungcf3b7152019-04-19 18:29:21 -07003260 if (!mTimestampStallReported) {
3261 ALOGD("%s(%d): device stall time corrected using current time %lld",
3262 __func__, mPortId, (long long)nowNs);
3263 mTimestampStallReported = true;
3264 }
Andy Hung98731a22019-04-08 19:19:07 -07003265 timestamp.mTime = convertNsToTimespec(nowNs);
Andy Hungcf3b7152019-04-19 18:29:21 -07003266 } else {
3267 mTimestampStallReported = false;
Andy Hung98731a22019-04-08 19:19:07 -07003268 }
Andy Hungb01faa32016-04-27 12:51:32 -07003269 }
Andy Hung5d313802016-10-10 15:09:39 -07003270
3271 // We update the timestamp time even when paused.
3272 if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) {
3273 const int64_t now = systemTime();
Andy Hung2b01f002017-07-05 12:01:36 -07003274 const int64_t at = audio_utils_ns_from_timespec(&timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003275 const int64_t lag =
3276 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
3277 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0)
3278 ? int64_t(mAfLatency * 1000000LL)
3279 : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3280 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK])
3281 * NANOS_PER_SECOND / mSampleRate;
3282 const int64_t limit = now - lag; // no earlier than this limit
3283 if (at < limit) {
3284 ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld",
3285 (long long)lag, (long long)at, (long long)limit);
Andy Hungffa36952017-08-17 10:41:51 -07003286 timestamp.mTime = convertNsToTimespec(limit);
Andy Hung5d313802016-10-10 15:09:39 -07003287 }
3288 }
Andy Hungb01faa32016-04-27 12:51:32 -07003289 mPreviousLocation = location;
3290 } else {
3291 // right after AudioTrack is started, one may not find a timestamp
Eric Laurent973db022018-11-20 14:54:31 -08003292 ALOGV("%s(%d): getBestTimestamp did not find timestamp", __func__, mPortId);
Andy Hungb01faa32016-04-27 12:51:32 -07003293 }
Andy Hung6ae58432016-02-16 18:32:24 -08003294 }
3295 if (status == INVALID_OPERATION) {
Andy Hungf20a4e92016-08-15 19:10:34 -07003296 // INVALID_OPERATION occurs when no timestamp has been issued by the server;
3297 // other failures are signaled by a negative time.
3298 // If we come out of FLUSHED or STOPPED where the position is known
3299 // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of
3300 // "zero" for NuPlayer). We don't convert for track restoration as position
3301 // does not reset.
Andy Hungfb8ede22018-09-12 19:03:24 -07003302 ALOGV("%s(%d): timestamp server offset:%lld restore frames:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003303 __func__, mPortId,
Andy Hungf20a4e92016-08-15 19:10:34 -07003304 (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore);
3305 if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) {
3306 status = WOULD_BLOCK;
3307 }
Andy Hung6ae58432016-02-16 18:32:24 -08003308 }
3309 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003310 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08003311 ALOGV_IF(status != WOULD_BLOCK, "%s(%d): getTimestamp error:%#x", __func__, mPortId, status);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003312 return status;
3313 }
3314 if (isOffloadedOrDirect_l()) {
3315 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) {
3316 // use cached paused position in case another offloaded track is running.
3317 timestamp.mPosition = mPausedPosition;
3318 clock_gettime(CLOCK_MONOTONIC, &timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003319 // TODO: adjust for delay
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003320 return NO_ERROR;
3321 }
3322
3323 // Check whether a pending flush or stop has completed, as those commands may
Andy Hungc8e09c62015-06-03 23:43:36 -07003324 // be asynchronous or return near finish or exhibit glitchy behavior.
3325 //
3326 // Originally this showed up as the first timestamp being a continuation of
3327 // the previous song under gapless playback.
3328 // However, we sometimes see zero timestamps, then a glitch of
3329 // the previous song's position, and then correct timestamps afterwards.
Andy Hungffa36952017-08-17 10:41:51 -07003330 if (mStartFromZeroUs != 0 && mSampleRate != 0) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003331 static const int kTimeJitterUs = 100000; // 100 ms
3332 static const int k1SecUs = 1000000;
3333
3334 const int64_t timeNow = getNowUs();
3335
Andy Hungffa36952017-08-17 10:41:51 -07003336 if (timeNow < mStartFromZeroUs + k1SecUs) { // within first second of starting
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003337 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003338 if (timestampTimeUs < mStartFromZeroUs) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003339 return WOULD_BLOCK; // stale timestamp time, occurs before start.
3340 }
Andy Hungffa36952017-08-17 10:41:51 -07003341 const int64_t deltaTimeUs = timestampTimeUs - mStartFromZeroUs;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003342 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07003343 / ((double)mSampleRate * mPlaybackRate.mSpeed);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003344
3345 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
3346 // Verify that the counter can't count faster than the sample rate
Andy Hungc8e09c62015-06-03 23:43:36 -07003347 // since the start time. If greater, then that means we may have failed
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003348 // to completely flush or stop the previous playing track.
Andy Hungc8e09c62015-06-03 23:43:36 -07003349 ALOGW_IF(!mTimestampStartupGlitchReported,
Andy Hungfb8ede22018-09-12 19:03:24 -07003350 "%s(%d): startup glitch detected"
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003351 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)",
Eric Laurent973db022018-11-20 14:54:31 -08003352 __func__, mPortId,
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003353 (long long)deltaTimeUs, (long long)deltaPositionByUs,
3354 timestamp.mPosition);
Andy Hungc8e09c62015-06-03 23:43:36 -07003355 mTimestampStartupGlitchReported = true;
3356 if (previousTimestampValid
3357 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) {
3358 timestamp = mPreviousTimestamp;
3359 mPreviousTimestampValid = true;
3360 return NO_ERROR;
3361 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003362 return WOULD_BLOCK;
3363 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003364 if (deltaPositionByUs != 0) {
Andy Hungffa36952017-08-17 10:41:51 -07003365 mStartFromZeroUs = 0; // don't check again, we got valid nonzero position.
Andy Hungc8e09c62015-06-03 23:43:36 -07003366 }
3367 } else {
Andy Hungffa36952017-08-17 10:41:51 -07003368 mStartFromZeroUs = 0; // don't check again, start time expired.
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003369 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003370 mTimestampStartupGlitchReported = false;
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003371 }
3372 } else {
Glenn Kasten200092b2014-08-15 15:13:30 -07003373 // Update the mapping between local consumed (mPosition) and server consumed (mServer)
3374 (void) updateAndGetPosition_l();
3375 // Server consumed (mServer) and presented both use the same server time base,
3376 // and server consumed is always >= presented.
3377 // The delta between these represents the number of frames in the buffer pipeline.
3378 // If this delta between these is greater than the client position, it means that
3379 // actually presented is still stuck at the starting line (figuratively speaking),
3380 // waiting for the first frame to go by. So we can't report a valid timestamp yet.
Andy Hung90e8a972015-11-09 16:42:40 -08003381 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when
3382 // mPosition exceeds 32 bits.
3383 // TODO Remove when timestamp is updated to contain pipeline status info.
3384 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue();
3385 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */
3386 && (uint32_t)pipelineDepthInFrames > mPosition.value()) {
Glenn Kasten200092b2014-08-15 15:13:30 -07003387 return INVALID_OPERATION;
3388 }
3389 // Convert timestamp position from server time base to client time base.
3390 // TODO The following code should work OK now because timestamp.mPosition is 32-bit.
3391 // But if we change it to 64-bit then this could fail.
Andy Hung90e8a972015-11-09 16:42:40 -08003392 // Use Modulo computation here.
3393 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value();
Glenn Kasten200092b2014-08-15 15:13:30 -07003394 // Immediately after a call to getPosition_l(), mPosition and
3395 // mServer both represent the same frame position. mPosition is
3396 // in client's point of view, and mServer is in server's point of
3397 // view. So the difference between them is the "fudge factor"
3398 // between client and server views due to stop() and/or new
3399 // IAudioTrack. And timestamp.mPosition is initially in server's
3400 // point of view, so we need to apply the same fudge factor to it.
Glenn Kastenfe346c72013-08-30 13:28:22 -07003401 }
Phil Burk1b420972015-04-22 10:52:21 -07003402
3403 // Prevent retrograde motion in timestamp.
3404 // This is sometimes caused by erratic reports of the available space in the ALSA drivers.
3405 if (status == NO_ERROR) {
Andy Hung3b8c6332019-04-03 19:29:36 -07003406 // Fix stale time when checking timestamp right after start().
3407 // The position is at the last reported location but the time can be stale
3408 // due to pause or standby or cold start latency.
3409 //
3410 // We keep advancing the time (but not the position) to ensure that the
3411 // stale value does not confuse the application.
3412 //
3413 // For offload compatibility, use a default lag value here.
3414 // Any time discrepancy between this update and the pause timestamp is handled
3415 // by the retrograde check afterwards.
3416 int64_t currentTimeNanos = audio_utils_ns_from_timespec(&timestamp.mTime);
3417 const int64_t lagNs = int64_t(mAfLatency * 1000000LL);
3418 const int64_t limitNs = mStartNs - lagNs;
3419 if (currentTimeNanos < limitNs) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003420 if (!mTimestampStaleTimeReported) {
3421 ALOGD("%s(%d): stale timestamp time corrected, "
3422 "currentTimeNanos: %lld < limitNs: %lld < mStartNs: %lld",
3423 __func__, mPortId,
3424 (long long)currentTimeNanos, (long long)limitNs, (long long)mStartNs);
3425 mTimestampStaleTimeReported = true;
3426 }
Andy Hung3b8c6332019-04-03 19:29:36 -07003427 timestamp.mTime = convertNsToTimespec(limitNs);
3428 currentTimeNanos = limitNs;
Andy Hungcf3b7152019-04-19 18:29:21 -07003429 } else {
3430 mTimestampStaleTimeReported = false;
Andy Hung3b8c6332019-04-03 19:29:36 -07003431 }
3432
Andy Hungffa36952017-08-17 10:41:51 -07003433 // previousTimestampValid is set to false when starting after a stop or flush.
Phil Burk1b420972015-04-22 10:52:21 -07003434 if (previousTimestampValid) {
Andy Hung2b01f002017-07-05 12:01:36 -07003435 const int64_t previousTimeNanos =
3436 audio_utils_ns_from_timespec(&mPreviousTimestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003437
3438 // retrograde check
Phil Burk1b420972015-04-22 10:52:21 -07003439 if (currentTimeNanos < previousTimeNanos) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003440 if (!mTimestampRetrogradeTimeReported) {
3441 ALOGW("%s(%d): retrograde timestamp time corrected, %lld < %lld",
3442 __func__, mPortId,
3443 (long long)currentTimeNanos, (long long)previousTimeNanos);
3444 mTimestampRetrogradeTimeReported = true;
3445 }
Andy Hung5d313802016-10-10 15:09:39 -07003446 timestamp.mTime = mPreviousTimestamp.mTime;
Andy Hungcf3b7152019-04-19 18:29:21 -07003447 } else {
3448 mTimestampRetrogradeTimeReported = false;
Phil Burk1b420972015-04-22 10:52:21 -07003449 }
3450
3451 // Looking at signed delta will work even when the timestamps
3452 // are wrapping around.
Andy Hung90e8a972015-11-09 16:42:40 -08003453 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition)
3454 - mPreviousTimestamp.mPosition).signedValue();
Phil Burk4c5a3672015-04-30 16:18:53 -07003455 if (deltaPosition < 0) {
3456 // Only report once per position instead of spamming the log.
Andy Hungcf3b7152019-04-19 18:29:21 -07003457 if (!mTimestampRetrogradePositionReported) {
Andy Hungfb8ede22018-09-12 19:03:24 -07003458 ALOGW("%s(%d): retrograde timestamp position corrected, %d = %u - %u",
Eric Laurent973db022018-11-20 14:54:31 -08003459 __func__, mPortId,
Phil Burk4c5a3672015-04-30 16:18:53 -07003460 deltaPosition,
3461 timestamp.mPosition,
3462 mPreviousTimestamp.mPosition);
Andy Hungcf3b7152019-04-19 18:29:21 -07003463 mTimestampRetrogradePositionReported = true;
Phil Burk4c5a3672015-04-30 16:18:53 -07003464 }
3465 } else {
Andy Hungcf3b7152019-04-19 18:29:21 -07003466 mTimestampRetrogradePositionReported = false;
Phil Burk4c5a3672015-04-30 16:18:53 -07003467 }
Andy Hung5d313802016-10-10 15:09:39 -07003468 if (deltaPosition < 0) {
3469 timestamp.mPosition = mPreviousTimestamp.mPosition;
3470 deltaPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07003471 }
Andy Hung5d313802016-10-10 15:09:39 -07003472#if 0
3473 // Uncomment this to verify audio timestamp rate.
3474 const int64_t deltaTime =
Andy Hung2b01f002017-07-05 12:01:36 -07003475 audio_utils_ns_from_timespec(&timestamp.mTime) - previousTimeNanos;
Andy Hung5d313802016-10-10 15:09:39 -07003476 if (deltaTime != 0) {
3477 const int64_t computedSampleRate =
3478 deltaPosition * (long long)NANOS_PER_SECOND / deltaTime;
Andy Hungfb8ede22018-09-12 19:03:24 -07003479 ALOGD("%s(%d): computedSampleRate:%u sampleRate:%u",
Eric Laurent973db022018-11-20 14:54:31 -08003480 __func__, mPortId,
Andy Hung5d313802016-10-10 15:09:39 -07003481 (unsigned)computedSampleRate, mSampleRate);
3482 }
3483#endif
Phil Burk1b420972015-04-22 10:52:21 -07003484 }
3485 mPreviousTimestamp = timestamp;
3486 mPreviousTimestampValid = true;
3487 }
3488
Glenn Kastenfe346c72013-08-30 13:28:22 -07003489 return status;
Glenn Kastence703742013-07-19 16:33:58 -07003490}
3491
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003492String8 AudioTrack::getParameters(const String8& keys)
3493{
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003494 audio_io_handle_t output = getOutput();
Glenn Kasten142f5192014-03-25 17:44:59 -07003495 if (output != AUDIO_IO_HANDLE_NONE) {
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003496 return AudioSystem::getParameters(output, keys);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01003497 } else {
3498 return String8::empty();
3499 }
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003500}
3501
Glenn Kasten23a75452014-01-13 10:37:17 -08003502bool AudioTrack::isOffloaded() const
3503{
3504 AutoMutex lock(mLock);
3505 return isOffloaded_l();
3506}
3507
Eric Laurentab5cdba2014-06-09 17:22:27 -07003508bool AudioTrack::isDirect() const
3509{
3510 AutoMutex lock(mLock);
3511 return isDirect_l();
3512}
3513
3514bool AudioTrack::isOffloadedOrDirect() const
3515{
3516 AutoMutex lock(mLock);
3517 return isOffloadedOrDirect_l();
3518}
3519
3520
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003521status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003522{
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003523 String8 result;
3524
3525 result.append(" AudioTrack::dump\n");
Andy Hungfb8ede22018-09-12 19:03:24 -07003526 result.appendFormat(" id(%d) status(%d), state(%d), session Id(%d), flags(%#x)\n",
Eric Laurent973db022018-11-20 14:54:31 -08003527 mPortId, mStatus, mState, mSessionId, mFlags);
Eric Laurentd114b622017-11-27 18:37:04 -08003528 result.appendFormat(" stream type(%d), left - right volume(%f, %f)\n",
François Gaffie58d4be52018-11-06 15:30:12 +01003529 mStreamType,
Eric Laurentd114b622017-11-27 18:37:04 -08003530 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
Glenn Kasten49f36ba2017-12-06 13:02:02 -08003531 result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
Eric Laurentd114b622017-11-27 18:37:04 -08003532 mFormat, mChannelMask, mChannelCount);
3533 result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
3534 mSampleRate, mOriginalSampleRate, mPlaybackRate.mSpeed);
3535 result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
3536 mFrameCount, mReqFrameCount);
3537 result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
3538 " req. notif. per buff(%u)\n",
3539 mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
3540 result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
3541 mLatency, mSelectedDeviceId, mRoutedDeviceId);
3542 result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
3543 mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003544 ::write(fd, result.string(), result.size());
3545 return NO_ERROR;
3546}
3547
Phil Burk2812d9e2016-01-04 10:34:30 -08003548uint32_t AudioTrack::getUnderrunCount() const
3549{
3550 AutoMutex lock(mLock);
3551 return getUnderrunCount_l();
3552}
3553
3554uint32_t AudioTrack::getUnderrunCount_l() const
3555{
3556 return mProxy->getUnderrunCount() + mUnderrunCountOffset;
3557}
3558
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003559uint32_t AudioTrack::getUnderrunFrames() const
3560{
3561 AutoMutex lock(mLock);
3562 return mProxy->getUnderrunFrames();
3563}
3564
Andy Hung3a5c2f32021-02-17 15:06:42 -08003565void AudioTrack::setLogSessionId(const char *logSessionId)
3566{
3567 AutoMutex lock(mLock);
Andy Hung1a9c21b2021-02-25 20:43:18 -08003568 if (logSessionId == nullptr) logSessionId = ""; // an empty string is an unset session id.
Andy Hung3a5c2f32021-02-17 15:06:42 -08003569 if (mLogSessionId == logSessionId) return;
3570
3571 mLogSessionId = logSessionId;
3572 mediametrics::LogItem(mMetricsId)
3573 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETLOGSESSIONID)
3574 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, logSessionId)
3575 .record();
3576}
3577
Andy Hung839a3062021-02-17 11:15:16 -08003578void AudioTrack::setPlayerIId(int playerIId)
3579{
3580 AutoMutex lock(mLock);
3581 if (mPlayerIId == playerIId) return;
3582
3583 mPlayerIId = playerIId;
3584 mediametrics::LogItem(mMetricsId)
3585 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYERIID)
3586 .set(AMEDIAMETRICS_PROP_PLAYERIID, playerIId)
3587 .record();
3588}
3589
Eric Laurent296fb132015-05-01 11:38:42 -07003590status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback)
3591{
Eric Laurent09f1ed22019-04-24 17:45:17 -07003592
Eric Laurent296fb132015-05-01 11:38:42 -07003593 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003594 ALOGW("%s(%d): adding NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003595 return BAD_VALUE;
3596 }
3597 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07003598 if (mDeviceCallback.unsafe_get() == callback.get()) {
Eric Laurent973db022018-11-20 14:54:31 -08003599 ALOGW("%s(%d): adding same callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003600 return INVALID_OPERATION;
3601 }
3602 status_t status = NO_ERROR;
3603 if (mOutput != AUDIO_IO_HANDLE_NONE) {
3604 if (mDeviceCallback != 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003605 ALOGW("%s(%d): callback already present!", __func__, mPortId);
Eric Laurent09f1ed22019-04-24 17:45:17 -07003606 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003607 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003608 status = AudioSystem::addAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003609 }
3610 mDeviceCallback = callback;
3611 return status;
3612}
3613
3614status_t AudioTrack::removeAudioDeviceCallback(
3615 const sp<AudioSystem::AudioDeviceCallback>& callback)
3616{
3617 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003618 ALOGW("%s(%d): removing NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003619 return BAD_VALUE;
3620 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003621 AutoMutex lock(mLock);
3622 if (mDeviceCallback.unsafe_get() != callback.get()) {
3623 ALOGW("%s removing different callback!", __FUNCTION__);
3624 return INVALID_OPERATION;
Eric Laurent296fb132015-05-01 11:38:42 -07003625 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003626 mDeviceCallback.clear();
Eric Laurent296fb132015-05-01 11:38:42 -07003627 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07003628 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003629 }
Eric Laurent296fb132015-05-01 11:38:42 -07003630 return NO_ERROR;
3631}
3632
Eric Laurentad2e7b92017-09-14 20:06:42 -07003633
3634void AudioTrack::onAudioDeviceUpdate(audio_io_handle_t audioIo,
3635 audio_port_handle_t deviceId)
3636{
3637 sp<AudioSystem::AudioDeviceCallback> callback;
3638 {
3639 AutoMutex lock(mLock);
3640 if (audioIo != mOutput) {
3641 return;
3642 }
3643 callback = mDeviceCallback.promote();
3644 // only update device if the track is active as route changes due to other use cases are
3645 // irrelevant for this client
3646 if (mState == STATE_ACTIVE) {
3647 mRoutedDeviceId = deviceId;
3648 }
3649 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003650
Eric Laurentad2e7b92017-09-14 20:06:42 -07003651 if (callback.get() != nullptr) {
3652 callback->onAudioDeviceUpdate(mOutput, mRoutedDeviceId);
3653 }
3654}
3655
Andy Hunge13f8a62016-03-30 14:20:42 -07003656status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location)
3657{
3658 if (msec == nullptr ||
3659 (location != ExtendedTimestamp::LOCATION_SERVER
3660 && location != ExtendedTimestamp::LOCATION_KERNEL)) {
3661 return BAD_VALUE;
3662 }
3663 AutoMutex lock(mLock);
3664 // inclusive of offloaded and direct tracks.
3665 //
3666 // It is possible, but not enabled, to allow duration computation for non-pcm
3667 // audio_has_proportional_frames() formats because currently they have
3668 // the drain rate equivalent to the pcm sample rate * framesize.
3669 if (!isPurePcmData_l()) {
3670 return INVALID_OPERATION;
3671 }
3672 ExtendedTimestamp ets;
3673 if (getTimestamp_l(&ets) == OK
3674 && ets.mTimeNs[location] > 0) {
3675 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
3676 - ets.mPosition[location];
3677 if (diff < 0) {
3678 *msec = 0;
3679 } else {
3680 // ms is the playback time by frames
3681 int64_t ms = (int64_t)((double)diff * 1000 /
3682 ((double)mSampleRate * mPlaybackRate.mSpeed));
3683 // clockdiff is the timestamp age (negative)
3684 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
3685 ets.mTimeNs[location]
3686 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
3687 - systemTime(SYSTEM_TIME_MONOTONIC);
3688
3689 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
3690 static const int NANOS_PER_MILLIS = 1000000;
3691 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
3692 }
3693 return NO_ERROR;
3694 }
3695 if (location != ExtendedTimestamp::LOCATION_SERVER) {
3696 return INVALID_OPERATION; // LOCATION_KERNEL is not available
3697 }
3698 // use server position directly (offloaded and direct arrive here)
3699 updateAndGetPosition_l();
3700 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue();
3701 *msec = (diff <= 0) ? 0
3702 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed));
3703 return NO_ERROR;
3704}
3705
Andy Hung65ffdfc2016-10-10 15:52:11 -07003706bool AudioTrack::hasStarted()
3707{
3708 AutoMutex lock(mLock);
3709 switch (mState) {
3710 case STATE_STOPPED:
3711 if (isOffloadedOrDirect_l()) {
3712 // check if we have started in the past to return true.
Andy Hungffa36952017-08-17 10:41:51 -07003713 return mStartFromZeroUs > 0;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003714 }
3715 // A normal audio track may still be draining, so
3716 // check if stream has ended. This covers fasttrack position
3717 // instability and start/stop without any data written.
3718 if (mProxy->getStreamEndDone()) {
3719 return true;
3720 }
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003721 FALLTHROUGH_INTENDED;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003722 case STATE_ACTIVE:
3723 case STATE_STOPPING:
3724 break;
3725 case STATE_PAUSED:
3726 case STATE_PAUSED_STOPPING:
3727 case STATE_FLUSHED:
3728 return false; // we're not active
3729 default:
Eric Laurent973db022018-11-20 14:54:31 -08003730 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in hasStarted(): %d", __func__, mPortId, mState);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003731 break;
3732 }
3733
3734 // wait indicates whether we need to wait for a timestamp.
3735 // This is conservatively figured - if we encounter an unexpected error
3736 // then we will not wait.
3737 bool wait = false;
3738 if (isOffloadedOrDirect_l()) {
3739 AudioTimestamp ts;
3740 status_t status = getTimestamp_l(ts);
3741 if (status == WOULD_BLOCK) {
3742 wait = true;
3743 } else if (status == OK) {
3744 wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition);
3745 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003746 ALOGV("%s(%d): hasStarted wait:%d ts:%u start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003747 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003748 (int)wait,
3749 ts.mPosition,
3750 (long long)mStartTs.mPosition);
3751 } else {
3752 int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG
3753 ExtendedTimestamp ets;
3754 status_t status = getTimestamp_l(&ets);
3755 if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets
3756 wait = true;
3757 } else if (status == OK) {
3758 for (location = ExtendedTimestamp::LOCATION_KERNEL;
3759 location >= ExtendedTimestamp::LOCATION_SERVER; --location) {
3760 if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) {
3761 continue;
3762 }
3763 wait = ets.mPosition[location] == 0
3764 || ets.mPosition[location] == mStartEts.mPosition[location];
3765 break;
3766 }
3767 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003768 ALOGV("%s(%d): hasStarted wait:%d ets:%lld start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003769 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003770 (int)wait,
3771 (long long)ets.mPosition[location],
3772 (long long)mStartEts.mPosition[location]);
3773 }
3774 return !wait;
3775}
3776
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003777// =========================================================================
3778
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003779void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003780{
3781 sp<AudioTrack> audioTrack = mAudioTrack.promote();
3782 if (audioTrack != 0) {
3783 AutoMutex lock(audioTrack->mLock);
3784 audioTrack->mProxy->binderDied();
3785 }
3786}
3787
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003788// =========================================================================
3789
Andy Hungca353672019-03-06 11:54:38 -08003790AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver)
Andy Hungeb46cf92019-03-06 10:13:38 -08003791 : Thread(true /* bCanCallJava */) // binder recursion on restoreTrack_l() may call Java.
3792 , mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
Glenn Kasten598de6c2013-10-16 17:02:13 -07003793 mIgnoreNextPausedInt(false)
Glenn Kasten3acbd052012-02-28 10:39:56 -08003794{
3795}
3796
3797AudioTrack::AudioTrackThread::~AudioTrackThread()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003798{
3799}
3800
3801bool AudioTrack::AudioTrackThread::threadLoop()
3802{
Glenn Kasten3acbd052012-02-28 10:39:56 -08003803 {
3804 AutoMutex _l(mMyLock);
3805 if (mPaused) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003806 // TODO check return value and handle or log
Glenn Kasten3acbd052012-02-28 10:39:56 -08003807 mMyCond.wait(mMyLock);
3808 // caller will check for exitPending()
3809 return true;
3810 }
Glenn Kasten598de6c2013-10-16 17:02:13 -07003811 if (mIgnoreNextPausedInt) {
3812 mIgnoreNextPausedInt = false;
3813 mPausedInt = false;
3814 }
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003815 if (mPausedInt) {
Glenn Kasten47d55172017-05-23 11:19:30 -07003816 // TODO use futex instead of condition, for event flag "or"
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003817 if (mPausedNs > 0) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003818 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003819 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
3820 } else {
Glenn Kasten75f79032017-05-23 11:22:44 -07003821 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003822 mMyCond.wait(mMyLock);
3823 }
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003824 mPausedInt = false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003825 return true;
3826 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08003827 }
Eric Laurent7985dcb2014-10-07 15:45:14 -07003828 if (exitPending()) {
3829 return false;
3830 }
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003831 nsecs_t ns = mReceiver.processAudioBuffer();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003832 switch (ns) {
3833 case 0:
3834 return true;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003835 case NS_INACTIVE:
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003836 pauseInternal();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003837 return true;
3838 case NS_NEVER:
3839 return false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003840 case NS_WHENEVER:
Andy Hung3c09c782014-12-29 18:39:32 -08003841 // Event driven: call wake() when callback notifications conditions change.
3842 ns = INT64_MAX;
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003843 FALLTHROUGH_INTENDED;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003844 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003845 LOG_ALWAYS_FATAL_IF(ns < 0, "%s(%d): processAudioBuffer() returned %lld",
Eric Laurent973db022018-11-20 14:54:31 -08003846 __func__, mReceiver.mPortId, (long long)ns);
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003847 pauseInternal(ns);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003848 return true;
Glenn Kastenca8b2802012-04-23 13:58:16 -07003849 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003850}
3851
Glenn Kasten3acbd052012-02-28 10:39:56 -08003852void AudioTrack::AudioTrackThread::requestExit()
3853{
3854 // must be in this order to avoid a race condition
3855 Thread::requestExit();
Glenn Kasten598de6c2013-10-16 17:02:13 -07003856 resume();
Glenn Kasten3acbd052012-02-28 10:39:56 -08003857}
3858
3859void AudioTrack::AudioTrackThread::pause()
3860{
3861 AutoMutex _l(mMyLock);
3862 mPaused = true;
3863}
3864
3865void AudioTrack::AudioTrackThread::resume()
3866{
3867 AutoMutex _l(mMyLock);
Glenn Kasten598de6c2013-10-16 17:02:13 -07003868 mIgnoreNextPausedInt = true;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003869 if (mPaused || mPausedInt) {
Glenn Kasten3acbd052012-02-28 10:39:56 -08003870 mPaused = false;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003871 mPausedInt = false;
Glenn Kasten3acbd052012-02-28 10:39:56 -08003872 mMyCond.signal();
3873 }
3874}
3875
Andy Hung3c09c782014-12-29 18:39:32 -08003876void AudioTrack::AudioTrackThread::wake()
3877{
3878 AutoMutex _l(mMyLock);
Andy Hunga8d08902015-07-22 11:52:13 -07003879 if (!mPaused) {
3880 // wake() might be called while servicing a callback - ignore the next
3881 // pause time and call processAudioBuffer.
Andy Hung3c09c782014-12-29 18:39:32 -08003882 mIgnoreNextPausedInt = true;
Andy Hunga8d08902015-07-22 11:52:13 -07003883 if (mPausedInt && mPausedNs > 0) {
3884 // audio track is active and internally paused with timeout.
3885 mPausedInt = false;
3886 mMyCond.signal();
3887 }
Andy Hung3c09c782014-12-29 18:39:32 -08003888 }
3889}
3890
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003891void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
3892{
3893 AutoMutex _l(mMyLock);
3894 mPausedInt = true;
3895 mPausedNs = ns;
3896}
3897
jiabinf6eb4c32020-02-25 14:06:25 -08003898binder::Status AudioTrack::AudioTrackCallback::onCodecFormatChanged(
3899 const std::vector<uint8_t>& audioMetadata)
3900{
3901 AutoMutex _l(mAudioTrackCbLock);
3902 sp<media::IAudioTrackCallback> callback = mCallback.promote();
3903 if (callback.get() != nullptr) {
3904 callback->onCodecFormatChanged(audioMetadata);
3905 } else {
3906 mCallback.clear();
3907 }
3908 return binder::Status::ok();
3909}
3910
3911void AudioTrack::AudioTrackCallback::setAudioTrackCallback(
3912 const sp<media::IAudioTrackCallback> &callback) {
3913 AutoMutex lock(mAudioTrackCbLock);
3914 mCallback = callback;
3915}
3916
Glenn Kasten40bc9062015-03-20 09:09:33 -07003917} // namespace android