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Glenn Kasten99e53b82012-01-19 08:59:58 -08001/*
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9** http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080018//#define LOG_NDEBUG 0
19#define LOG_TAG "AudioTrack"
20
Mark Salyzyn34fb2962014-06-18 16:30:56 -070021#include <inttypes.h>
Glenn Kastenc56f3422014-03-21 17:53:17 -070022#include <math.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080023#include <sys/resource.h>
Andy Hung959b5b82021-09-24 10:46:20 -070024#include <thread>
Mark Salyzyn34fb2962014-06-18 16:30:56 -070025
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -080026#include <android/media/IAudioPolicyService.h>
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -070027#include <android-base/macros.h>
Andy Hung3acde2c2021-11-11 09:18:08 -080028#include <android-base/stringprintf.h>
Andy Hung2b01f002017-07-05 12:01:36 -070029#include <audio_utils/clock.h>
Glenn Kasten9f80dd22012-12-18 15:57:32 -080030#include <audio_utils/primitives.h>
31#include <binder/IPCThreadState.h>
32#include <media/AudioTrack.h>
33#include <utils/Log.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080034#include <private/media/AudioTrackShared.h>
Suren Baghdasaryan7435e7d2018-12-19 17:09:28 -080035#include <processgroup/sched_policy.h>
Glenn Kasten1ab85ec2013-05-31 09:18:43 -070036#include <media/IAudioFlinger.h>
Dean Wheatleya70eef72018-01-04 14:23:50 +110037#include <media/AudioParameter.h>
Andy Hungcd044842014-08-07 11:04:34 -070038#include <media/AudioResamplerPublic.h>
Michael Chana94fbb22018-04-24 14:31:19 +100039#include <media/AudioSystem.h>
Ray Essickf27e9872019-12-07 06:28:46 -080040#include <media/MediaMetricsItem.h>
Ray Essicked304702017-12-12 14:00:57 -080041#include <media/TypeConverter.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080042
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +010043#define WAIT_PERIOD_MS 10
44#define WAIT_STREAM_END_TIMEOUT_SEC 120
Andy Hung53c3b5f2014-12-15 16:42:05 -080045static const int kMaxLoopCountNotifications = 32;
Glenn Kasten511754b2012-01-11 09:52:19 -080046
Kuowei Lid4adbdb2020-08-13 14:44:25 +080047using ::android::aidl_utils::statusTFromBinderStatus;
Andy Hung3acde2c2021-11-11 09:18:08 -080048using ::android::base::StringPrintf;
Kuowei Lid4adbdb2020-08-13 14:44:25 +080049
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080050namespace android {
Chia-chi Yeh33005a92010-06-16 06:33:13 +080051// ---------------------------------------------------------------------------
52
Ivan Lozano8cf3a072017-08-09 09:01:33 -070053using media::VolumeShaper;
Svet Ganov3e5f14f2021-05-13 22:51:08 +000054using android::content::AttributionSourceState;
Ivan Lozano8cf3a072017-08-09 09:01:33 -070055
Andy Hunga7f03352015-05-31 21:54:49 -070056// TODO: Move to a separate .h
57
Andy Hung4ede21d2014-12-12 15:37:34 -080058template <typename T>
Andy Hunga7f03352015-05-31 21:54:49 -070059static inline const T &min(const T &x, const T &y) {
Andy Hung4ede21d2014-12-12 15:37:34 -080060 return x < y ? x : y;
61}
62
Andy Hunga7f03352015-05-31 21:54:49 -070063template <typename T>
64static inline const T &max(const T &x, const T &y) {
65 return x > y ? x : y;
66}
67
68static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed)
69{
70 return ((double)frames * 1000000000) / ((double)sampleRate * speed);
71}
72
Andy Hung7f1bc8a2014-09-12 14:43:11 -070073static int64_t convertTimespecToUs(const struct timespec &tv)
74{
Chih-Hung Hsieh50fa06c2018-12-11 13:51:36 -080075 return tv.tv_sec * 1000000LL + tv.tv_nsec / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -070076}
77
Andy Hungffa36952017-08-17 10:41:51 -070078// TODO move to audio_utils.
79static inline struct timespec convertNsToTimespec(int64_t ns) {
80 struct timespec tv;
81 tv.tv_sec = static_cast<time_t>(ns / NANOS_PER_SECOND);
Andy Hung06a730b2020-04-09 13:28:31 -070082 tv.tv_nsec = static_cast<int64_t>(ns % NANOS_PER_SECOND);
Andy Hungffa36952017-08-17 10:41:51 -070083 return tv;
84}
85
Andy Hung7f1bc8a2014-09-12 14:43:11 -070086// current monotonic time in microseconds.
87static int64_t getNowUs()
88{
89 struct timespec tv;
90 (void) clock_gettime(CLOCK_MONOTONIC, &tv);
91 return convertTimespecToUs(tv);
92}
93
Andy Hung26145642015-04-15 21:56:53 -070094// FIXME: we don't use the pitch setting in the time stretcher (not working);
95// instead we emulate it using our sample rate converter.
96static const bool kFixPitch = true; // enable pitch fix
97static inline uint32_t adjustSampleRate(uint32_t sampleRate, float pitch)
98{
99 return kFixPitch ? (sampleRate * pitch + 0.5) : sampleRate;
100}
101
102static inline float adjustSpeed(float speed, float pitch)
103{
Ricardo Garcia6c7f0622015-04-30 18:39:16 -0700104 return kFixPitch ? speed / max(pitch, AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) : speed;
Andy Hung26145642015-04-15 21:56:53 -0700105}
106
107static inline float adjustPitch(float pitch)
108{
109 return kFixPitch ? AUDIO_TIMESTRETCH_PITCH_NORMAL : pitch;
110}
111
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800112// static
113status_t AudioTrack::getMinFrameCount(
Glenn Kastene33054e2012-11-14 12:54:39 -0800114 size_t* frameCount,
Glenn Kastenfff6d712012-01-12 16:38:12 -0800115 audio_stream_type_t streamType,
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800116 uint32_t sampleRate)
117{
Glenn Kastend65d73c2012-06-22 17:21:07 -0700118 if (frameCount == NULL) {
119 return BAD_VALUE;
120 }
Glenn Kasten04cd0182012-06-25 11:49:27 -0700121
Andy Hung0e48d252015-01-26 11:43:15 -0800122 // FIXME handle in server, like createTrack_l(), possible missing info:
Glenn Kastene0fa4672012-04-24 14:35:14 -0700123 // audio_io_handle_t output
124 // audio_format_t format
125 // audio_channel_mask_t channelMask
Andy Hung0e48d252015-01-26 11:43:15 -0800126 // audio_output_flags_t flags (FAST)
Glenn Kasten3b16c762012-11-14 08:44:39 -0800127 uint32_t afSampleRate;
Glenn Kasten66a04672014-01-08 08:53:44 -0800128 status_t status;
129 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
130 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700131 ALOGE("%s(): Unable to query output sample rate for stream type %d; status %d",
132 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800133 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800134 }
Glenn Kastene33054e2012-11-14 12:54:39 -0800135 size_t afFrameCount;
Glenn Kasten66a04672014-01-08 08:53:44 -0800136 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
137 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700138 ALOGE("%s(): Unable to query output frame count for stream type %d; status %d",
139 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800140 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800141 }
142 uint32_t afLatency;
Glenn Kasten66a04672014-01-08 08:53:44 -0800143 status = AudioSystem::getOutputLatency(&afLatency, streamType);
144 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700145 ALOGE("%s(): Unable to query output latency for stream type %d; status %d",
146 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800147 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800148 }
149
Andy Hung8edb8dc2015-03-26 19:13:55 -0700150 // When called from createTrack, speed is 1.0f (normal speed).
151 // This is rechecked again on setting playback rate (TODO: on setting sample rate, too).
Eric Laurent21da6472017-11-09 16:29:26 -0800152 *frameCount = AudioSystem::calculateMinFrameCount(afLatency, afFrameCount, afSampleRate,
153 sampleRate, 1.0f /*, 0 notificationsPerBufferReq*/);
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800154
Andy Hung0e48d252015-01-26 11:43:15 -0800155 // The formula above should always produce a non-zero value under normal circumstances:
156 // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX.
157 // Return error in the unlikely event that it does not, as that's part of the API contract.
Glenn Kasten66a04672014-01-08 08:53:44 -0800158 if (*frameCount == 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700159 ALOGE("%s(): failed for streamType %d, sampleRate %u",
160 __func__, streamType, sampleRate);
Glenn Kasten66a04672014-01-08 08:53:44 -0800161 return BAD_VALUE;
162 }
Andy Hungfb8ede22018-09-12 19:03:24 -0700163 ALOGV("%s(): getMinFrameCount=%zu: afFrameCount=%zu, afSampleRate=%u, afLatency=%u",
164 __func__, *frameCount, afFrameCount, afSampleRate, afLatency);
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800165 return NO_ERROR;
166}
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800167
Michael Chana94fbb22018-04-24 14:31:19 +1000168// static
169bool AudioTrack::isDirectOutputSupported(const audio_config_base_t& config,
170 const audio_attributes_t& attributes) {
171 ALOGV("%s()", __FUNCTION__);
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800172 const sp<media::IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
Michael Chana94fbb22018-04-24 14:31:19 +1000173 if (aps == 0) return false;
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800174
175 auto result = [&]() -> ConversionResult<bool> {
Mikhail Naganovdbf03642021-08-25 18:15:32 -0700176 media::audio::common::AudioConfigBase configAidl = VALUE_OR_RETURN(
Mikhail Naganov9dec7012021-07-21 10:30:57 -0700177 legacy2aidl_audio_config_base_t_AudioConfigBase(config, false /*isInput*/));
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800178 media::AudioAttributesInternal attributesAidl = VALUE_OR_RETURN(
179 legacy2aidl_audio_attributes_t_AudioAttributesInternal(attributes));
180 bool retAidl;
181 RETURN_IF_ERROR(aidl_utils::statusTFromBinderStatus(
182 aps->isDirectOutputSupported(configAidl, attributesAidl, &retAidl)));
183 return retAidl;
184 }();
185 return result.value_or(false);
Michael Chana94fbb22018-04-24 14:31:19 +1000186}
187
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800188// ---------------------------------------------------------------------------
189
Ray Essicked304702017-12-12 14:00:57 -0800190void AudioTrack::MediaMetrics::gather(const AudioTrack *track)
191{
Ray Essick88394302018-01-24 14:52:05 -0800192 // only if we're in a good state...
193 // XXX: shall we gather alternative info if failing?
194 const status_t lstatus = track->initCheck();
195 if (lstatus != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700196 ALOGD("%s(): no metrics gathered, track status=%d", __func__, (int) lstatus);
Ray Essick88394302018-01-24 14:52:05 -0800197 return;
198 }
199
Andy Hungd0979812019-02-21 15:51:44 -0800200#define MM_PREFIX "android.media.audiotrack." // avoid cut-n-paste errors.
Ray Essicked304702017-12-12 14:00:57 -0800201
Andy Hungd0979812019-02-21 15:51:44 -0800202 // Java API 28 entries, do not change.
Ray Essickf27e9872019-12-07 06:28:46 -0800203 mMetricsItem->setCString(MM_PREFIX "streamtype", toString(track->streamType()).c_str());
204 mMetricsItem->setCString(MM_PREFIX "type",
Andy Hungd0979812019-02-21 15:51:44 -0800205 toString(track->mAttributes.content_type).c_str());
Ray Essickf27e9872019-12-07 06:28:46 -0800206 mMetricsItem->setCString(MM_PREFIX "usage", toString(track->mAttributes.usage).c_str());
Ray Essicked304702017-12-12 14:00:57 -0800207
Andy Hungd0979812019-02-21 15:51:44 -0800208 // Non-API entries, these can change due to a Java string mistake.
Ray Essickf27e9872019-12-07 06:28:46 -0800209 mMetricsItem->setInt32(MM_PREFIX "sampleRate", (int32_t)track->mSampleRate);
210 mMetricsItem->setInt64(MM_PREFIX "channelMask", (int64_t)track->mChannelMask);
Andy Hungd0979812019-02-21 15:51:44 -0800211 // Non-API entries, these can change.
Ray Essickf27e9872019-12-07 06:28:46 -0800212 mMetricsItem->setInt32(MM_PREFIX "portId", (int32_t)track->mPortId);
213 mMetricsItem->setCString(MM_PREFIX "encoding", toString(track->mFormat).c_str());
214 mMetricsItem->setInt32(MM_PREFIX "frameCount", (int32_t)track->mFrameCount);
215 mMetricsItem->setCString(MM_PREFIX "attributes", toString(track->mAttributes).c_str());
Andy Hung6f451f02021-02-24 21:53:29 -0800216 mMetricsItem->setCString(MM_PREFIX "logSessionId", track->mLogSessionId.c_str());
Ray Essicked304702017-12-12 14:00:57 -0800217}
218
Ray Essick88394302018-01-24 14:52:05 -0800219// hand the user a snapshot of the metrics.
Ray Essickf27e9872019-12-07 06:28:46 -0800220status_t AudioTrack::getMetrics(mediametrics::Item * &item)
Ray Essick88394302018-01-24 14:52:05 -0800221{
222 mMediaMetrics.gather(this);
Ray Essickf27e9872019-12-07 06:28:46 -0800223 mediametrics::Item *tmp = mMediaMetrics.dup();
Ray Essick88394302018-01-24 14:52:05 -0800224 if (tmp == nullptr) {
225 return BAD_VALUE;
226 }
227 item = tmp;
228 return NO_ERROR;
229}
Ray Essicked304702017-12-12 14:00:57 -0800230
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000231AudioTrack::AudioTrack() : AudioTrack(AttributionSourceState())
Colin Crossb8a9dbb2020-08-27 04:12:26 +0000232{
233}
234
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000235AudioTrack::AudioTrack(const AttributionSourceState& attributionSource)
Glenn Kasten87913512011-06-22 16:15:25 -0700236 : mStatus(NO_INIT),
Haynes Mathew George9b3359f2016-03-23 19:09:10 -0700237 mState(STATE_STOPPED),
John Grossman4ff14ba2012-02-08 16:37:41 -0800238 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800239 mPreviousSchedulingGroup(SP_DEFAULT),
Paul McLeanaa981192015-03-21 09:55:15 -0700240 mPausedPosition(0),
Eric Laurent20b9ef02016-12-05 11:03:16 -0800241 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
jiabinf6eb4c32020-02-25 14:06:25 -0800242 mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000243 mClientAttributionSource(attributionSource),
jiabinf6eb4c32020-02-25 14:06:25 -0800244 mAudioTrackCallback(new AudioTrackCallback())
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800245{
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700246 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
247 mAttributes.usage = AUDIO_USAGE_UNKNOWN;
Mikhail Naganov55773032020-10-01 15:08:13 -0700248 mAttributes.flags = AUDIO_FLAG_NONE;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700249 strcpy(mAttributes.tags, "");
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800250}
251
252AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800253 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800254 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800255 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700256 audio_channel_mask_t channelMask,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800257 size_t frameCount,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700258 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400259 const wp<IAudioTrackCallback> & callback,
Atneyaf86d2692021-10-14 14:02:36 -0400260 int32_t notificationFrames,
261 audio_session_t sessionId,
262 transfer_type transferType,
263 const audio_offload_info_t *offloadInfo,
264 const AttributionSourceState& attributionSource,
265 const audio_attributes_t* pAttributes,
266 bool doNotReconnect,
267 float maxRequiredSpeed,
268 audio_port_handle_t selectedDeviceId)
269 : mStatus(NO_INIT),
270 mState(STATE_STOPPED),
271 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
272 mPreviousSchedulingGroup(SP_DEFAULT),
273 mPausedPosition(0),
274 mAudioTrackCallback(new AudioTrackCallback())
275{
276 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Daniel Chapinf13b98a2021-10-25 21:58:31 +0000277
Atneyaf86d2692021-10-14 14:02:36 -0400278 (void)set(streamType, sampleRate, format, channelMask,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400279 frameCount, flags, callback, notificationFrames,
280 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
281 attributionSource, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
282}
283
284namespace {
285 class LegacyCallbackWrapper : public AudioTrack::IAudioTrackCallback {
286 const AudioTrack::legacy_callback_t mCallback;
287 void * const mData;
288 public:
289 LegacyCallbackWrapper(AudioTrack::legacy_callback_t callback, void* user)
290 : mCallback(callback), mData(user) {}
291 size_t onMoreData(const AudioTrack::Buffer & buffer) override {
292 AudioTrack::Buffer copy = buffer;
293 mCallback(AudioTrack::EVENT_MORE_DATA, mData, static_cast<void*>(&copy));
294 return copy.size;
295 }
296 void onUnderrun() override {
297 mCallback(AudioTrack::EVENT_UNDERRUN, mData, nullptr);
298 }
299 void onLoopEnd(int32_t loopsRemaining) override {
300 mCallback(AudioTrack::EVENT_LOOP_END, mData, &loopsRemaining);
301 }
302 void onMarker(uint32_t markerPosition) override {
303 mCallback(AudioTrack::EVENT_MARKER, mData, &markerPosition);
304 }
305 void onNewPos(uint32_t newPos) override {
306 mCallback(AudioTrack::EVENT_NEW_POS, mData, &newPos);
307 }
308 void onBufferEnd() override {
309 mCallback(AudioTrack::EVENT_BUFFER_END, mData, nullptr);
310 }
311 void onNewIAudioTrack() override {
312 mCallback(AudioTrack::EVENT_NEW_IAUDIOTRACK, mData, nullptr);
313 }
314 void onStreamEnd() override {
315 mCallback(AudioTrack::EVENT_STREAM_END, mData, nullptr);
316 }
317 size_t onCanWriteMoreData(const AudioTrack::Buffer & buffer) override {
318 AudioTrack::Buffer copy = buffer;
319 mCallback(AudioTrack::EVENT_CAN_WRITE_MORE_DATA, mData, static_cast<void*>(&copy));
320 return copy.size;
321 }
322 };
323}
324
325AudioTrack::AudioTrack(
326 audio_stream_type_t streamType,
327 uint32_t sampleRate,
328 audio_format_t format,
329 audio_channel_mask_t channelMask,
330 size_t frameCount,
331 audio_output_flags_t flags,
332 legacy_callback_t callback,
333 void* user,
334 int32_t notificationFrames,
335 audio_session_t sessionId,
336 transfer_type transferType,
337 const audio_offload_info_t *offloadInfo,
338 const AttributionSourceState& attributionSource,
339 const audio_attributes_t* pAttributes,
340 bool doNotReconnect,
341 float maxRequiredSpeed,
342 audio_port_handle_t selectedDeviceId)
343 : mStatus(NO_INIT),
344 mState(STATE_STOPPED),
345 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
346 mPreviousSchedulingGroup(SP_DEFAULT),
347 mPausedPosition(0),
348 mAudioTrackCallback(new AudioTrackCallback())
349{
350 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
351 if (callback != nullptr) {
352 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
353 } else if (user) {
354 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
355 }
356 (void)set(streamType, sampleRate, format, channelMask,
357 frameCount, flags, mLegacyCallbackWrapper, notificationFrames,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000358 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
359 attributionSource, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800360}
361
Andreas Huberc8139852012-01-18 10:51:55 -0800362AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800363 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800364 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800365 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700366 audio_channel_mask_t channelMask,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800367 const sp<IMemory>& sharedBuffer,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700368 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400369 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700370 int32_t notificationFrames,
Glenn Kastend848eb42016-03-08 13:42:11 -0800371 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000372 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800373 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000374 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700375 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700376 bool doNotReconnect,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700377 float maxRequiredSpeed)
Glenn Kasten87913512011-06-22 16:15:25 -0700378 : mStatus(NO_INIT),
Haynes Mathew George9b3359f2016-03-23 19:09:10 -0700379 mState(STATE_STOPPED),
John Grossman4ff14ba2012-02-08 16:37:41 -0800380 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800381 mPreviousSchedulingGroup(SP_DEFAULT),
Paul McLeanaa981192015-03-21 09:55:15 -0700382 mPausedPosition(0),
jiabinf6eb4c32020-02-25 14:06:25 -0800383 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
384 mAudioTrackCallback(new AudioTrackCallback())
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800385{
François Gaffie393f0e02019-04-10 09:09:08 +0200386 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Ippei Murata5fa32ed2018-10-01 17:37:50 +0900387
Eric Laurentf32d7812017-11-30 14:44:07 -0800388 (void)set(streamType, sampleRate, format, channelMask,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400389 0 /*frameCount*/, flags, callback, notificationFrames,
Marco Nelissend457c972014-02-11 08:47:07 -0800390 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000391 attributionSource, pAttributes, doNotReconnect, maxRequiredSpeed);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800392}
393
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400394AudioTrack::AudioTrack(
395 audio_stream_type_t streamType,
396 uint32_t sampleRate,
397 audio_format_t format,
398 audio_channel_mask_t channelMask,
399 const sp<IMemory>& sharedBuffer,
400 audio_output_flags_t flags,
401 legacy_callback_t callback,
402 void* user,
403 int32_t notificationFrames,
404 audio_session_t sessionId,
405 transfer_type transferType,
406 const audio_offload_info_t *offloadInfo,
407 const AttributionSourceState& attributionSource,
408 const audio_attributes_t* pAttributes,
409 bool doNotReconnect,
410 float maxRequiredSpeed)
411 : mStatus(NO_INIT),
412 mState(STATE_STOPPED),
413 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
414 mPreviousSchedulingGroup(SP_DEFAULT),
415 mPausedPosition(0),
416 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
417 mAudioTrackCallback(new AudioTrackCallback())
418{
419 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
420 if (callback) {
421 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
422 } else if (user) {
423 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
424 }
425
426 (void)set(streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags,
427 mLegacyCallbackWrapper, notificationFrames, sharedBuffer,
428 false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, attributionSource,
429 pAttributes, doNotReconnect, maxRequiredSpeed);
430}
431
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800432AudioTrack::~AudioTrack()
433{
Ray Essicked304702017-12-12 14:00:57 -0800434 // pull together the numbers, before we clean up our structures
435 mMediaMetrics.gather(this);
436
Andy Hungb68f5eb2019-12-03 16:49:17 -0800437 mediametrics::LogItem(mMetricsId)
438 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DTOR)
Phil Burkd3813f32020-04-23 16:26:15 -0700439 .set(AMEDIAMETRICS_PROP_CALLERNAME,
440 mCallerName.empty()
Andy Hunga6b27032020-04-27 10:34:24 -0700441 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
Phil Burkd3813f32020-04-23 16:26:15 -0700442 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800443 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
444 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)mStatus)
445 .record();
446
Phil Burk7a9577c2021-03-12 20:12:11 +0000447 stopAndJoinCallbacks(); // checks mStatus
448
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800449 if (mStatus == NO_ERROR) {
Marco Nelissenf8880202014-11-14 07:58:25 -0800450 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten53cec222013-08-29 09:01:02 -0700451 mAudioTrack.clear();
Eric Laurent3bcffa12014-06-12 18:38:45 -0700452 mCblkMemory.clear();
453 mSharedBuffer.clear();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800454 IPCThreadState::self()->flushCommands();
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000455 pid_t clientPid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
Andy Hungfb8ede22018-09-12 19:03:24 -0700456 ALOGV("%s(%d), releasing session id %d from %d on behalf of %d",
Eric Laurent973db022018-11-20 14:54:31 -0800457 __func__, mPortId,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700458 mSessionId, IPCThreadState::self()->getCallingPid(), clientPid);
459 AudioSystem::releaseAudioSessionId(mSessionId, clientPid);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800460 }
461}
462
Phil Burk7a9577c2021-03-12 20:12:11 +0000463void AudioTrack::stopAndJoinCallbacks() {
464 // Prevent nullptr crash if it did not open properly.
465 if (mStatus != NO_ERROR) return;
466
467 // Make sure that callback function exits in the case where
468 // it is looping on buffer full condition in obtainBuffer().
469 // Otherwise the callback thread will never exit.
470 stop();
471 if (mAudioTrackThread != 0) { // not thread safe
472 mProxy->interrupt();
473 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
474 mAudioTrackThread->requestExitAndWait();
475 mAudioTrackThread.clear();
476 }
477 // No lock here: worst case we remove a NULL callback which will be a nop
478 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) {
479 // This may not stop all of these device callbacks!
480 // TODO: Add some sort of protection.
481 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
482 mDeviceCallback.clear();
483 }
484}
485
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800486status_t AudioTrack::set(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800487 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800488 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800489 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700490 audio_channel_mask_t channelMask,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800491 size_t frameCount,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700492 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400493 legacy_callback_t callback,
494 void * user,
495 int32_t notificationFrames,
496 const sp<IMemory>& sharedBuffer,
497 bool threadCanCallJava,
498 audio_session_t sessionId,
499 transfer_type transferType,
500 const audio_offload_info_t *offloadInfo,
501 const AttributionSourceState& attributionSource,
502 const audio_attributes_t* pAttributes,
503 bool doNotReconnect,
504 float maxRequiredSpeed,
505 audio_port_handle_t selectedDeviceId)
506{
507 if (callback) {
508 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
509 } else if (user) {
510 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
511 }
512 return set(streamType, sampleRate,format, channelMask, frameCount, flags,
513 mLegacyCallbackWrapper, notificationFrames, sharedBuffer, threadCanCallJava,
514 sessionId, transferType, offloadInfo, attributionSource, pAttributes,
515 doNotReconnect, maxRequiredSpeed, selectedDeviceId);
516}
517status_t AudioTrack::set(
518 audio_stream_type_t streamType,
519 uint32_t sampleRate,
520 audio_format_t format,
521 audio_channel_mask_t channelMask,
522 size_t frameCount,
523 audio_output_flags_t flags,
524 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700525 int32_t notificationFrames,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800526 const sp<IMemory>& sharedBuffer,
Eric Laurentbe916aa2010-06-01 23:49:17 -0700527 bool threadCanCallJava,
Glenn Kastend848eb42016-03-08 13:42:11 -0800528 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000529 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800530 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000531 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700532 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700533 bool doNotReconnect,
jiabin156c6872017-10-06 09:47:15 -0700534 float maxRequiredSpeed,
535 audio_port_handle_t selectedDeviceId)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800536{
Eric Laurentf32d7812017-11-30 14:44:07 -0800537 status_t status;
538 uint32_t channelCount;
539 pid_t callingPid;
540 pid_t myPid;
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000541 uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
542 pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400543 sp<IAudioTrackCallback> _callback = callback.promote();
Andy Hung3acde2c2021-11-11 09:18:08 -0800544 std::string errorMessage;
Eric Laurent973db022018-11-20 14:54:31 -0800545 // Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
Andy Hungfb8ede22018-09-12 19:03:24 -0700546 ALOGV("%s(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
Glenn Kastenea38ee72016-04-18 11:08:01 -0700547 "flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
Andy Hungfb8ede22018-09-12 19:03:24 -0700548 __func__,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800549 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000550 sessionId, transferType, attributionSource.uid, attributionSource.pid);
Glenn Kasten86f04662014-02-24 15:13:05 -0800551
Phil Burk33ff89b2015-11-30 11:16:01 -0800552 mThreadCanCallJava = threadCanCallJava;
jiabin156c6872017-10-06 09:47:15 -0700553 mSelectedDeviceId = selectedDeviceId;
Eric Laurent21da6472017-11-09 16:29:26 -0800554 mSessionId = sessionId;
Phil Burk33ff89b2015-11-30 11:16:01 -0800555
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800556 switch (transferType) {
557 case TRANSFER_DEFAULT:
558 if (sharedBuffer != 0) {
559 transferType = TRANSFER_SHARED;
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400560 } else if (_callback == nullptr|| threadCanCallJava) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800561 transferType = TRANSFER_SYNC;
562 } else {
563 transferType = TRANSFER_CALLBACK;
564 }
565 break;
566 case TRANSFER_CALLBACK:
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700567 case TRANSFER_SYNC_NOTIF_CALLBACK:
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400568 if (_callback == nullptr || sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800569 errorMessage = StringPrintf(
570 "%s: Transfer type %s but callback == nullptr || sharedBuffer != 0",
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700571 convertTransferToText(transferType), __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800572 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800573 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800574 }
575 break;
576 case TRANSFER_OBTAIN:
577 case TRANSFER_SYNC:
578 if (sharedBuffer != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800579 errorMessage = StringPrintf(
580 "%s: Transfer type TRANSFER_OBTAIN but sharedBuffer != 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800581 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800582 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800583 }
584 break;
585 case TRANSFER_SHARED:
586 if (sharedBuffer == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800587 errorMessage = StringPrintf(
588 "%s: Transfer type TRANSFER_SHARED but sharedBuffer == 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800589 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800590 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800591 }
592 break;
593 default:
Andy Hung3acde2c2021-11-11 09:18:08 -0800594 errorMessage = StringPrintf("%s: Invalid transfer type %d", __func__, transferType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800595 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800596 goto error;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800597 }
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800598 mSharedBuffer = sharedBuffer;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800599 mTransfer = transferType;
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700600 mDoNotReconnect = doNotReconnect;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800601
Andy Hungfb8ede22018-09-12 19:03:24 -0700602 ALOGV_IF(sharedBuffer != 0, "%s(): sharedBuffer: %p, size: %zu",
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -0700603 __func__, sharedBuffer->unsecurePointer(), sharedBuffer->size());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800604
Andy Hungfb8ede22018-09-12 19:03:24 -0700605 ALOGV("%s(): streamType %d frameCount %zu flags %04x",
606 __func__, streamType, frameCount, flags);
Eric Laurent1a9ed112012-03-20 18:36:01 -0700607
Glenn Kasten53cec222013-08-29 09:01:02 -0700608 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Eric Laurent1dd70b92009-04-21 07:56:33 -0700609 if (mAudioTrack != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800610 errorMessage = StringPrintf("%s: Track already in use", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800611 status = INVALID_OPERATION;
Andy Hung3acde2c2021-11-11 09:18:08 -0800612 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800613 }
614
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800615 // handle default values first.
Eric Laurente83b55d2014-11-14 10:06:21 -0800616 if (streamType == AUDIO_STREAM_DEFAULT) {
Dima Zavinfce7a472011-04-19 22:30:36 -0700617 streamType = AUDIO_STREAM_MUSIC;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800618 }
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700619 if (pAttributes == NULL) {
Eric Laurent223fd5c2014-11-11 13:43:36 -0800620 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800621 errorMessage = StringPrintf("%s: Invalid stream type %d", __func__, streamType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800622 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800623 goto error;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700624 }
Andy Hunga2159aa2021-07-20 13:01:52 -0700625 mOriginalStreamType = streamType;
Eric Laurente83b55d2014-11-14 10:06:21 -0800626
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700627 } else {
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700628 // stream type shouldn't be looked at, this track has audio attributes
629 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
Andy Hungfb8ede22018-09-12 19:03:24 -0700630 ALOGV("%s(): Building AudioTrack with attributes:"
631 " usage=%d content=%d flags=0x%x tags=[%s]",
632 __func__,
633 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
Andy Hunga2159aa2021-07-20 13:01:52 -0700634 mOriginalStreamType = AUDIO_STREAM_DEFAULT;
François Gaffie58d4be52018-11-06 15:30:12 +0100635 audio_flags_to_audio_output_flags(mAttributes.flags, &flags);
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800636 }
Glenn Kastenea7939a2012-03-14 12:56:26 -0700637
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800638 // these below should probably come from the audioFlinger too...
Glenn Kastene1c39622012-01-04 09:36:37 -0800639 if (format == AUDIO_FORMAT_DEFAULT) {
Dima Zavinfce7a472011-04-19 22:30:36 -0700640 format = AUDIO_FORMAT_PCM_16_BIT;
Phil Burkfdb3c072016-02-09 10:47:02 -0800641 } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
Mikhail Naganov55773032020-10-01 15:08:13 -0700642 flags = static_cast<audio_output_flags_t>(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800643 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800644
645 // validate parameters
Dima Zavinfce7a472011-04-19 22:30:36 -0700646 if (!audio_is_valid_format(format)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800647 errorMessage = StringPrintf("%s: Invalid format %#x", __func__, format);
Eric Laurentf32d7812017-11-30 14:44:07 -0800648 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800649 goto error;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800650 }
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800651 mFormat = format;
Eric Laurentc2f1f072009-07-17 12:17:14 -0700652
Glenn Kasten8ba90322013-10-30 11:29:27 -0700653 if (!audio_is_output_channel(channelMask)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800654 errorMessage = StringPrintf("%s: Invalid channel mask %#x", __func__, channelMask);
Eric Laurentf32d7812017-11-30 14:44:07 -0800655 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800656 goto error;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700657 }
Glenn Kastene3247bf2014-02-24 15:19:07 -0800658 mChannelMask = channelMask;
Eric Laurentf32d7812017-11-30 14:44:07 -0800659 channelCount = audio_channel_count_from_out_mask(channelMask);
Glenn Kastene3247bf2014-02-24 15:19:07 -0800660 mChannelCount = channelCount;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700661
Eric Laurentc2f1f072009-07-17 12:17:14 -0700662 // force direct flag if format is not linear PCM
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100663 // or offload was requested
664 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
665 || !audio_is_linear_pcm(format)) {
666 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
Andy Hungfb8ede22018-09-12 19:03:24 -0700667 ? "%s(): Offload request, forcing to Direct Output"
668 : "%s(): Not linear PCM, forcing to Direct Output",
669 __func__);
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700670 flags = (audio_output_flags_t)
Glenn Kasten3acbd052012-02-28 10:39:56 -0800671 // FIXME why can't we allow direct AND fast?
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700672 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
Eric Laurentc2f1f072009-07-17 12:17:14 -0700673 }
674
Eric Laurentd1f69b02014-12-15 14:33:13 -0800675 // force direct flag if HW A/V sync requested
676 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
677 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
678 }
679
Glenn Kastenb7730382014-04-30 15:50:31 -0700680 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
Phil Burkfdb3c072016-02-09 10:47:02 -0800681 if (audio_has_proportional_frames(format)) {
Glenn Kastenb7730382014-04-30 15:50:31 -0700682 mFrameSize = channelCount * audio_bytes_per_sample(format);
683 } else {
684 mFrameSize = sizeof(uint8_t);
685 }
Glenn Kastene3aa6592012-12-04 12:22:46 -0800686 } else {
Phil Burkfdb3c072016-02-09 10:47:02 -0800687 ALOG_ASSERT(audio_has_proportional_frames(format));
Glenn Kastenb7730382014-04-30 15:50:31 -0700688 mFrameSize = channelCount * audio_bytes_per_sample(format);
Glenn Kastenb7730382014-04-30 15:50:31 -0700689 // createTrack will return an error if PCM format is not supported by server,
690 // so no need to check for specific PCM formats here
Glenn Kastene3aa6592012-12-04 12:22:46 -0800691 }
692
Eric Laurent0d6db582014-11-12 18:39:44 -0800693 // sampling rate must be specified for direct outputs
694 if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800695 errorMessage = StringPrintf(
696 "%s: sample rate must be specified for direct outputs", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800697 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800698 goto error;
Eric Laurent0d6db582014-11-12 18:39:44 -0800699 }
700 mSampleRate = sampleRate;
Lajos Molnar3a474aa2015-04-24 17:10:07 -0700701 mOriginalSampleRate = sampleRate;
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -0700702 mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
Andy Hungff874dc2016-04-11 16:49:09 -0700703 // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
704 mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
Eric Laurent0d6db582014-11-12 18:39:44 -0800705
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800706 // Make copy of input parameter offloadInfo so that in the future:
707 // (a) createTrack_l doesn't need it as an input parameter
708 // (b) we can support re-creation of offloaded tracks
709 if (offloadInfo != NULL) {
710 mOffloadInfoCopy = *offloadInfo;
711 mOffloadInfo = &mOffloadInfoCopy;
712 } else {
713 mOffloadInfo = NULL;
Eric Laurent20b9ef02016-12-05 11:03:16 -0800714 memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
Ytai Ben-Tsviffa2fd92020-10-20 09:13:53 -0700715 mOffloadInfoCopy = AUDIO_INFO_INITIALIZER;
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800716 }
717
Glenn Kasten66e46352014-01-16 17:44:23 -0800718 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
719 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
Glenn Kasten05632a52012-01-03 14:22:33 -0800720 mSendLevel = 0.0f;
Glenn Kasten396fabd2014-01-08 08:54:23 -0800721 // mFrameCount is initialized in createTrack_l
Glenn Kastenb6037442012-11-14 13:42:25 -0800722 mReqFrameCount = frameCount;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700723 if (notificationFrames >= 0) {
724 mNotificationFramesReq = notificationFrames;
725 mNotificationsPerBufferReq = 0;
726 } else {
727 if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
Andy Hung3acde2c2021-11-11 09:18:08 -0800728 errorMessage = StringPrintf(
729 "%s: notificationFrames=%d not permitted for non-fast track",
Andy Hungfb8ede22018-09-12 19:03:24 -0700730 __func__, notificationFrames);
Eric Laurentf32d7812017-11-30 14:44:07 -0800731 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800732 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700733 }
734 if (frameCount > 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700735 ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
736 __func__, notificationFrames, frameCount);
Eric Laurentf32d7812017-11-30 14:44:07 -0800737 status = BAD_VALUE;
Andy Hung3acde2c2021-11-11 09:18:08 -0800738 goto error;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700739 }
740 mNotificationFramesReq = 0;
741 const uint32_t minNotificationsPerBuffer = 1;
742 const uint32_t maxNotificationsPerBuffer = 8;
743 mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
744 max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
745 ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
Andy Hungfb8ede22018-09-12 19:03:24 -0700746 "%s(): notificationFrames=%d clamped to the range -%u to -%u",
747 __func__,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700748 notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
749 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800750 mNotificationFramesAct = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700751 // TODO b/182392553: refactor or remove
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000752 mClientAttributionSource = AttributionSourceState(attributionSource);
Eric Laurentf32d7812017-11-30 14:44:07 -0800753 callingPid = IPCThreadState::self()->getCallingPid();
754 myPid = getpid();
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700755 if (uid == -1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000756 mClientAttributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700757 IPCThreadState::self()->getCallingUid()));
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800758 }
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700759 if (pid == (pid_t)-1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000760 mClientAttributionSource.pid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(callingPid));
Marco Nelissend457c972014-02-11 08:47:07 -0800761 }
Eric Laurent2beeb502010-07-16 07:43:46 -0700762 mAuxEffectId = 0;
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -0800763 mOrigFlags = mFlags = flags;
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400764 mCallback = callback;
Eric Laurentbe916aa2010-06-01 23:49:17 -0700765
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400766 if (_callback != nullptr) {
767 mAudioTrackThread = sp<AudioTrackThread>::make(*this);
Glenn Kastena997e7a2012-08-07 09:44:19 -0700768 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
Glenn Kastenbfd31842015-03-20 09:01:44 -0700769 // thread begins in paused state, and will not reference us until start()
Glenn Kastena997e7a2012-08-07 09:44:19 -0700770 }
771
Eric Laurent34f1d8e2009-11-04 08:27:26 -0800772 // create the IAudioTrack
Francois Gaffie24a9fb02019-01-18 17:51:34 +0100773 {
774 AutoMutex lock(mLock);
775 status = createTrack_l();
776 }
Glenn Kastena997e7a2012-08-07 09:44:19 -0700777 if (status != NO_ERROR) {
778 if (mAudioTrackThread != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100779 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
780 mAudioTrackThread->requestExitAndWait();
Glenn Kastena997e7a2012-08-07 09:44:19 -0700781 mAudioTrackThread.clear();
782 }
Andy Hung3acde2c2021-11-11 09:18:08 -0800783 // We do not goto error to prevent double-logging errors.
Eric Laurentf32d7812017-11-30 14:44:07 -0800784 goto exit;
Glenn Kasten5d464eb2012-06-22 17:19:53 -0700785 }
786
Andy Hung4ede21d2014-12-12 15:37:34 -0800787 mLoopCount = 0;
788 mLoopStart = 0;
789 mLoopEnd = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -0800790 mLoopCountNotified = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800791 mMarkerPosition = 0;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -0700792 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800793 mNewPosition = 0;
794 mUpdatePeriod = 0;
Glenn Kasten200092b2014-08-15 15:13:30 -0700795 mPosition = 0;
796 mReleased = 0;
Andy Hungffa36952017-08-17 10:41:51 -0700797 mStartNs = 0;
798 mStartFromZeroUs = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700799 AudioSystem::acquireAudioSessionId(mSessionId, pid, uid);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800800 mSequence = 1;
801 mObservedSequence = mSequence;
802 mInUnderrun = false;
Phil Burk1b420972015-04-22 10:52:21 -0700803 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700804 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700805 mTimestampRetrogradePositionReported = false;
806 mTimestampRetrogradeTimeReported = false;
807 mTimestampStallReported = false;
808 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700809 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Andy Hung65ffdfc2016-10-10 15:52:11 -0700810 mStartTs.mPosition = 0;
Phil Burk2812d9e2016-01-04 10:34:30 -0800811 mUnderrunCountOffset = 0;
Andy Hungea2b9c02016-02-12 17:06:53 -0800812 mFramesWritten = 0;
813 mFramesWrittenServerOffset = 0;
Andy Hungf20a4e92016-08-15 19:10:34 -0700814 mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
Ivan Lozano8cf3a072017-08-09 09:01:33 -0700815 mVolumeHandler = new media::VolumeHandler();
Eric Laurentf32d7812017-11-30 14:44:07 -0800816
Andy Hung3acde2c2021-11-11 09:18:08 -0800817error:
818 if (status != NO_ERROR) {
819 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
820 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
821 }
822 // fall through
Eric Laurentf32d7812017-11-30 14:44:07 -0800823exit:
824 mStatus = status;
825 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800826}
827
Mikhail Naganov55773032020-10-01 15:08:13 -0700828
829status_t AudioTrack::set(
830 audio_stream_type_t streamType,
831 uint32_t sampleRate,
832 audio_format_t format,
833 uint32_t channelMask,
834 size_t frameCount,
835 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400836 legacy_callback_t callback,
Mikhail Naganov55773032020-10-01 15:08:13 -0700837 void* user,
838 int32_t notificationFrames,
839 const sp<IMemory>& sharedBuffer,
840 bool threadCanCallJava,
841 audio_session_t sessionId,
842 transfer_type transferType,
843 const audio_offload_info_t *offloadInfo,
844 uid_t uid,
845 pid_t pid,
846 const audio_attributes_t* pAttributes,
847 bool doNotReconnect,
848 float maxRequiredSpeed,
849 audio_port_handle_t selectedDeviceId)
850{
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000851 AttributionSourceState attributionSource;
852 attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(uid));
853 attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(pid));
854 attributionSource.token = sp<BBinder>::make();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400855 if (callback) {
856 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
857 } else if (user) {
858 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
859 }
860 return set(streamType, sampleRate, format, static_cast<audio_channel_mask_t>(channelMask),
861 frameCount, flags, mLegacyCallbackWrapper, notificationFrames, sharedBuffer,
862 threadCanCallJava, sessionId, transferType, offloadInfo, attributionSource,
863 pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
Mikhail Naganov55773032020-10-01 15:08:13 -0700864}
865
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800866// -------------------------------------------------------------------------
867
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100868status_t AudioTrack::start()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800869{
Eric Laurentf5aafb22010-11-18 08:40:16 -0800870 AutoMutex lock(mLock);
Andy Hungb68f5eb2019-12-03 16:49:17 -0800871
Andy Hung10fb4be2020-05-27 22:22:22 -0700872 if (mState == STATE_ACTIVE) {
873 return INVALID_OPERATION;
874 }
875
876 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
877
878 // Defer logging here due to OpenSL ES repeated start calls.
879 // TODO(b/154868033) after fix, restore this logging back to the beginning of start().
880 const int64_t beginNs = systemTime();
Andy Hungb68f5eb2019-12-03 16:49:17 -0800881 status_t status = NO_ERROR; // logged: make sure to set this before returning.
Andy Hung06a730b2020-04-09 13:28:31 -0700882 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800883 mediametrics::LogItem(mMetricsId)
Andy Hunga6b27032020-04-27 10:34:24 -0700884 .set(AMEDIAMETRICS_PROP_CALLERNAME,
885 mCallerName.empty()
886 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
887 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800888 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_START)
Andy Hungea840382020-05-05 21:50:17 -0700889 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800890 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
891 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
892 .record(); });
893
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800894
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800895 mInUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800896
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800897 State previousState = mState;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100898 if (previousState == STATE_PAUSED_STOPPING) {
899 mState = STATE_STOPPING;
900 } else {
901 mState = STATE_ACTIVE;
902 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700903 (void) updateAndGetPosition_l();
Andy Hung65ffdfc2016-10-10 15:52:11 -0700904
905 // save start timestamp
906 if (isOffloadedOrDirect_l()) {
907 if (getTimestamp_l(mStartTs) != OK) {
908 mStartTs.mPosition = 0;
909 }
910 } else {
911 if (getTimestamp_l(&mStartEts) != OK) {
912 mStartEts.clear();
913 }
914 }
Andy Hungffa36952017-08-17 10:41:51 -0700915 mStartNs = systemTime(); // save this for timestamp adjustment after starting.
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800916 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
917 // reset current position as seen by client to 0
Glenn Kasten200092b2014-08-15 15:13:30 -0700918 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -0700919 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700920 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700921 mTimestampRetrogradePositionReported = false;
922 mTimestampRetrogradeTimeReported = false;
923 mTimestampStallReported = false;
924 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700925 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Phil Burk1b420972015-04-22 10:52:21 -0700926
Andy Hung65ffdfc2016-10-10 15:52:11 -0700927 if (!isOffloadedOrDirect_l()
928 && mStartEts.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) {
Andy Hunge1e98462016-04-12 10:18:51 -0700929 // Server side has consumed something, but is it finished consuming?
930 // It is possible since flush and stop are asynchronous that the server
931 // is still active at this point.
Andy Hungfb8ede22018-09-12 19:03:24 -0700932 ALOGV("%s(%d): server read:%lld cumulative flushed:%lld client written:%lld",
Eric Laurent973db022018-11-20 14:54:31 -0800933 __func__, mPortId,
Andy Hunge1e98462016-04-12 10:18:51 -0700934 (long long)(mFramesWrittenServerOffset
Andy Hung65ffdfc2016-10-10 15:52:11 -0700935 + mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]),
936 (long long)mStartEts.mFlushed,
Andy Hunge1e98462016-04-12 10:18:51 -0700937 (long long)mFramesWritten);
Andy Hungc4e60eb2017-09-06 11:14:57 -0700938 // mStartEts is already adjusted by mFramesWrittenServerOffset, so we delta adjust.
939 mFramesWrittenServerOffset -= mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER];
Andy Hung61be8412015-10-06 10:51:09 -0700940 }
Andy Hunge1e98462016-04-12 10:18:51 -0700941 mFramesWritten = 0;
942 mProxy->clearTimestamp(); // need new server push for valid timestamp
943 mMarkerReached = false;
Andy Hung61be8412015-10-06 10:51:09 -0700944
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700945 // For offloaded tracks, we don't know if the hardware counters are really zero here,
946 // since the flush is asynchronous and stop may not fully drain.
947 // We save the time when the track is started to later verify whether
948 // the counters are realistic (i.e. start from zero after this time).
Andy Hungffa36952017-08-17 10:41:51 -0700949 mStartFromZeroUs = mStartNs / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700950
Eric Laurentec9a0322013-08-28 10:23:01 -0700951 // force refresh of remaining frames by processAudioBuffer() as last
952 // write before stop could be partial.
953 mRefreshRemaining = true;
Tomoharu Kasahara6e8bf982017-06-09 16:17:33 +0900954
955 // for static track, clear the old flags when starting from stopped state
956 if (mSharedBuffer != 0) {
957 android_atomic_and(
958 ~(CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
959 &mCblk->mFlags);
960 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800961 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700962 mNewPosition = mPosition + mUpdatePeriod;
Andy Hung4be3b832016-10-13 17:51:43 -0700963 int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800964
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800965 if (!(flags & CBLK_INVALID)) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -0800966 mAudioTrack->start(&status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800967 if (status == DEAD_OBJECT) {
968 flags |= CBLK_INVALID;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800969 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800970 }
971 if (flags & CBLK_INVALID) {
972 status = restoreTrack_l("start");
973 }
974
Andy Hung79629f02016-03-24 13:57:40 -0700975 // resume or pause the callback thread as needed.
976 sp<AudioTrackThread> t = mAudioTrackThread;
977 if (status == NO_ERROR) {
978 if (t != 0) {
979 if (previousState == STATE_STOPPING) {
980 mProxy->interrupt();
981 } else {
982 t->resume();
983 }
984 } else {
985 mPreviousPriority = getpriority(PRIO_PROCESS, 0);
986 get_sched_policy(0, &mPreviousSchedulingGroup);
987 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
988 }
Andy Hung39399b62017-04-21 15:07:45 -0700989
990 // Start our local VolumeHandler for restoration purposes.
991 mVolumeHandler->setStarted();
Andy Hung79629f02016-03-24 13:57:40 -0700992 } else {
Eric Laurent973db022018-11-20 14:54:31 -0800993 ALOGE("%s(%d): status %d", __func__, mPortId, status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800994 mState = previousState;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800995 if (t != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100996 if (previousState != STATE_STOPPING) {
997 t->pause();
998 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800999 } else {
Glenn Kasten87913512011-06-22 16:15:25 -07001000 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
Glenn Kastena6364332012-04-19 09:35:04 -07001001 set_sched_policy(0, mPreviousSchedulingGroup);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001002 }
1003 }
1004
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001005 return status;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001006}
1007
1008void AudioTrack::stop()
1009{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001010 const int64_t beginNs = systemTime();
1011
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001012 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -07001013 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001014 mediametrics::LogItem(mMetricsId)
1015 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_STOP)
Andy Hungea840382020-05-05 21:50:17 -07001016 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001017 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Phil Burk64e16a72020-06-01 13:25:51 -07001018 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1019 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getUnderrunCount_l())
Phil Burka9876702020-04-20 18:16:15 -07001020 .record();
Phil Burka9876702020-04-20 18:16:15 -07001021 });
Andy Hungb68f5eb2019-12-03 16:49:17 -08001022
Eric Laurent973db022018-11-20 14:54:31 -08001023 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001024
Glenn Kasten397edb32013-08-30 15:10:13 -07001025 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001026 return;
1027 }
1028
Glenn Kasten23a75452014-01-13 10:37:17 -08001029 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001030 mState = STATE_STOPPING;
1031 } else {
1032 mState = STATE_STOPPED;
Andy Hung2148bf02016-11-28 19:01:02 -08001033 ALOGD_IF(mSharedBuffer == nullptr,
Eric Laurent973db022018-11-20 14:54:31 -08001034 "%s(%d): called with %u frames delivered", __func__, mPortId, mReleased.value());
Andy Hungc2813e52014-10-16 17:54:34 -07001035 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001036 }
1037
Andy Hung1d3556d2018-03-29 16:30:14 -07001038 mProxy->stop(); // notify server not to read beyond current client position until start().
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001039 mProxy->interrupt();
1040 mAudioTrack->stop();
Andy Hung61be8412015-10-06 10:51:09 -07001041
1042 // Note: legacy handling - stop does not clear playback marker
1043 // and periodic update counter, but flush does for streaming tracks.
Andy Hung9b461582014-12-01 17:56:29 -08001044
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001045 if (mSharedBuffer != 0) {
Andy Hung9b461582014-12-01 17:56:29 -08001046 // clear buffer position and loop count.
Andy Hung9b461582014-12-01 17:56:29 -08001047 mStaticProxy->setBufferPositionAndLoop(0 /* position */,
1048 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001049 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001050
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001051 sp<AudioTrackThread> t = mAudioTrackThread;
1052 if (t != 0) {
Glenn Kasten23a75452014-01-13 10:37:17 -08001053 if (!isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001054 t->pause();
Jean-Michel Trivia60e9a22018-11-15 16:13:36 -08001055 } else if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
Jean-Michel Trivi0248a2a2018-11-21 10:05:57 -08001056 // causes wake up of the playback thread, that will callback the client for
1057 // EVENT_STREAM_END in processAudioBuffer()
1058 t->wake();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001059 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001060 } else {
1061 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
1062 set_sched_policy(0, mPreviousSchedulingGroup);
1063 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001064}
1065
1066bool AudioTrack::stopped() const
1067{
Glenn Kasten9a2aaf92012-01-03 09:42:47 -08001068 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001069 return mState != STATE_ACTIVE;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001070}
1071
1072void AudioTrack::flush()
1073{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001074 const int64_t beginNs = systemTime();
Andy Hungfb8ede22018-09-12 19:03:24 -07001075 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -07001076 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001077 mediametrics::LogItem(mMetricsId)
1078 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_FLUSH)
Andy Hungea840382020-05-05 21:50:17 -07001079 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001080 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
1081 .record(); });
1082
Eric Laurent973db022018-11-20 14:54:31 -08001083 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001084
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001085 if (mSharedBuffer != 0) {
1086 return;
Glenn Kasten4bae3642012-11-30 13:41:12 -08001087 }
Andy Hung4c5ed302018-05-09 11:16:21 -07001088 if (mState == STATE_ACTIVE) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001089 return;
1090 }
1091 flush_l();
Eric Laurent1703cdf2011-03-07 14:52:59 -08001092}
1093
Eric Laurent1703cdf2011-03-07 14:52:59 -08001094void AudioTrack::flush_l()
1095{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001096 ALOG_ASSERT(mState != STATE_ACTIVE);
Eric Laurentc2f1f072009-07-17 12:17:14 -07001097
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001098 // clear playback marker and periodic update counter
1099 mMarkerPosition = 0;
1100 mMarkerReached = false;
1101 mUpdatePeriod = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001102 mRefreshRemaining = true;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001103
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001104 mState = STATE_FLUSHED;
Andy Hungc2813e52014-10-16 17:54:34 -07001105 mReleased = 0;
Glenn Kasten23a75452014-01-13 10:37:17 -08001106 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001107 mProxy->interrupt();
1108 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001109 mProxy->flush();
Glenn Kasten4bae3642012-11-30 13:41:12 -08001110 mAudioTrack->flush();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001111}
1112
Andy Hung959b5b82021-09-24 10:46:20 -07001113bool AudioTrack::pauseAndWait(const std::chrono::milliseconds& timeout)
1114{
1115 using namespace std::chrono_literals;
1116
1117 pause();
1118
1119 AutoMutex lock(mLock);
1120 // offload and direct tracks do not wait because pause volume ramp is handled by hardware.
1121 if (isOffloadedOrDirect_l()) return true;
1122
1123 // Wait for the track state to be anything besides pausing.
1124 // This ensures that the volume has ramped down.
1125 constexpr auto SLEEP_INTERVAL_MS = 10ms;
1126 auto begin = std::chrono::steady_clock::now();
1127 while (true) {
1128 // wait for state to change
1129 const int state = mProxy->getState();
1130
1131 mLock.unlock(); // only local variables accessed until lock.
1132 auto elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(
1133 std::chrono::steady_clock::now() - begin);
1134 if (state != CBLK_STATE_PAUSING) {
1135 ALOGV("%s: success state:%d after %lld ms", __func__, state, elapsed.count());
1136 return true;
1137 }
1138 std::chrono::milliseconds remaining = timeout - elapsed;
1139 if (remaining.count() <= 0) {
1140 ALOGW("%s: timeout expired state:%d still pausing:%d after %lld ms",
1141 __func__, state, CBLK_STATE_PAUSING, elapsed.count());
1142 return false;
1143 }
1144 // It is conceivable that the track is restored while sleeping;
1145 // as this logic is advisory, we allow that.
1146 std::this_thread::sleep_for(std::min(remaining, SLEEP_INTERVAL_MS));
1147 mLock.lock();
1148 }
1149}
1150
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001151void AudioTrack::pause()
1152{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001153 const int64_t beginNs = systemTime();
Eric Laurentf5aafb22010-11-18 08:40:16 -08001154 AutoMutex lock(mLock);
Andy Hunga6b27032020-04-27 10:34:24 -07001155 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001156 mediametrics::LogItem(mMetricsId)
1157 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_PAUSE)
Andy Hungea840382020-05-05 21:50:17 -07001158 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001159 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
1160 .record(); });
1161
Eric Laurent973db022018-11-20 14:54:31 -08001162 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001163
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001164 if (mState == STATE_ACTIVE) {
1165 mState = STATE_PAUSED;
1166 } else if (mState == STATE_STOPPING) {
1167 mState = STATE_PAUSED_STOPPING;
1168 } else {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001169 return;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001170 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001171 mProxy->interrupt();
1172 mAudioTrack->pause();
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001173
Marco Nelissen3a90f282014-03-10 11:21:43 -07001174 if (isOffloaded_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001175 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001176 // An offload output can be re-used between two audio tracks having
1177 // the same configuration. A timestamp query for a paused track
1178 // while the other is running would return an incorrect time.
1179 // To fix this, cache the playback position on a pause() and return
1180 // this time when requested until the track is resumed.
1181
1182 // OffloadThread sends HAL pause in its threadLoop. Time saved
1183 // here can be slightly off.
1184
1185 // TODO: check return code for getRenderPosition.
1186
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001187 uint32_t halFrames;
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001188 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition);
Andy Hungfb8ede22018-09-12 19:03:24 -07001189 ALOGV("%s(%d): for offload, cache current position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001190 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001191 }
1192 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001193}
1194
Eric Laurentbe916aa2010-06-01 23:49:17 -07001195status_t AudioTrack::setVolume(float left, float right)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001196{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001197 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1198 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY ||
1199 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001200 return BAD_VALUE;
1201 }
1202
Andy Hungb68f5eb2019-12-03 16:49:17 -08001203 mediametrics::LogItem(mMetricsId)
1204 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOLUME)
1205 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)left)
1206 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)right)
1207 .record();
1208
Eric Laurent1703cdf2011-03-07 14:52:59 -08001209 AutoMutex lock(mLock);
Glenn Kasten66e46352014-01-16 17:44:23 -08001210 mVolume[AUDIO_INTERLEAVE_LEFT] = left;
1211 mVolume[AUDIO_INTERLEAVE_RIGHT] = right;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001212
Glenn Kastenc56f3422014-03-21 17:53:17 -07001213 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right)));
Eric Laurentbe916aa2010-06-01 23:49:17 -07001214
Glenn Kasten23a75452014-01-13 10:37:17 -08001215 if (isOffloaded_l()) {
Eric Laurent59fe0102013-09-27 18:48:26 -07001216 mAudioTrack->signal();
1217 }
Eric Laurentbe916aa2010-06-01 23:49:17 -07001218 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001219}
1220
Glenn Kastenb1c09932012-02-27 16:21:04 -08001221status_t AudioTrack::setVolume(float volume)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001222{
Glenn Kastenb1c09932012-02-27 16:21:04 -08001223 return setVolume(volume, volume);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001224}
1225
Eric Laurent2beeb502010-07-16 07:43:46 -07001226status_t AudioTrack::setAuxEffectSendLevel(float level)
Eric Laurentbe916aa2010-06-01 23:49:17 -07001227{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001228 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1229 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001230 return BAD_VALUE;
1231 }
1232
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001233 AutoMutex lock(mLock);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001234 mSendLevel = level;
Glenn Kastene3aa6592012-12-04 12:22:46 -08001235 mProxy->setSendLevel(level);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001236
1237 return NO_ERROR;
1238}
1239
Glenn Kastena5224f32012-01-04 12:41:44 -08001240void AudioTrack::getAuxEffectSendLevel(float* level) const
Eric Laurentbe916aa2010-06-01 23:49:17 -07001241{
1242 if (level != NULL) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001243 *level = mSendLevel;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001244 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001245}
1246
Glenn Kasten3b16c762012-11-14 08:44:39 -08001247status_t AudioTrack::setSampleRate(uint32_t rate)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001248{
Andy Hung5cbb5782015-03-27 18:39:59 -07001249 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08001250 ALOGV("%s(%d): prior state:%s rate:%u", __func__, mPortId, stateToString(mState), rate);
Andy Hungfb8ede22018-09-12 19:03:24 -07001251
Andy Hung5cbb5782015-03-27 18:39:59 -07001252 if (rate == mSampleRate) {
1253 return NO_ERROR;
1254 }
jiabinf4de6112018-12-19 12:40:08 -08001255 if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)
1256 || (mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL)) {
John Grossman4ff14ba2012-02-08 16:37:41 -08001257 return INVALID_OPERATION;
1258 }
Eric Laurent0d6db582014-11-12 18:39:44 -08001259 if (mOutput == AUDIO_IO_HANDLE_NONE) {
1260 return NO_INIT;
1261 }
Andy Hung5cbb5782015-03-27 18:39:59 -07001262 // NOTE: it is theoretically possible, but highly unlikely, that a device change
1263 // could mean a previously allowed sampling rate is no longer allowed.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001264 uint32_t afSamplingRate;
Eric Laurent0d6db582014-11-12 18:39:44 -08001265 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
Eric Laurent57326622009-07-07 07:10:45 -07001266 return NO_INIT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001267 }
Andy Hung26145642015-04-15 21:56:53 -07001268 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001269 const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001270 if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001271 return BAD_VALUE;
1272 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001273 // TODO: Should we also check if the buffer size is compatible?
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001274
Glenn Kastene3aa6592012-12-04 12:22:46 -08001275 mSampleRate = rate;
Andy Hung26145642015-04-15 21:56:53 -07001276 mProxy->setSampleRate(effectiveSampleRate);
Glenn Kastene3aa6592012-12-04 12:22:46 -08001277
Eric Laurent57326622009-07-07 07:10:45 -07001278 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001279}
1280
Glenn Kastena5224f32012-01-04 12:41:44 -08001281uint32_t AudioTrack::getSampleRate() const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001282{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001283 AutoMutex lock(mLock);
Eric Laurent6f59db12013-07-26 17:16:50 -07001284
1285 // sample rate can be updated during playback by the offloaded decoder so we need to
1286 // query the HAL and update if needed.
1287// FIXME use Proxy return channel to update the rate from server and avoid polling here
Eric Laurentab5cdba2014-06-09 17:22:27 -07001288 if (isOffloadedOrDirect_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001289 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent6f59db12013-07-26 17:16:50 -07001290 uint32_t sampleRate = 0;
Jean-Michel Trivib7f24b12014-06-11 10:05:30 -07001291 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
Eric Laurent6f59db12013-07-26 17:16:50 -07001292 if (status == NO_ERROR) {
1293 mSampleRate = sampleRate;
1294 }
1295 }
1296 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001297 return mSampleRate;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001298}
1299
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001300uint32_t AudioTrack::getOriginalSampleRate() const
1301{
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001302 return mOriginalSampleRate;
1303}
1304
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001305status_t AudioTrack::setDualMonoMode(audio_dual_mono_mode_t mode)
1306{
1307 AutoMutex lock(mLock);
1308 return setDualMonoMode_l(mode);
1309}
1310
1311status_t AudioTrack::setDualMonoMode_l(audio_dual_mono_mode_t mode)
1312{
1313 const status_t status = statusTFromBinderStatus(
1314 mAudioTrack->setDualMonoMode(VALUE_OR_RETURN_STATUS(
1315 legacy2aidl_audio_dual_mono_mode_t_AudioDualMonoMode(mode))));
1316 if (status == NO_ERROR) mDualMonoMode = mode;
1317 return status;
1318}
1319
1320status_t AudioTrack::getDualMonoMode(audio_dual_mono_mode_t* mode) const
1321{
1322 AutoMutex lock(mLock);
1323 media::AudioDualMonoMode mediaMode;
1324 const status_t status = statusTFromBinderStatus(mAudioTrack->getDualMonoMode(&mediaMode));
1325 if (status == NO_ERROR) {
1326 *mode = VALUE_OR_RETURN_STATUS(
1327 aidl2legacy_AudioDualMonoMode_audio_dual_mono_mode_t(mediaMode));
1328 }
1329 return status;
1330}
1331
1332status_t AudioTrack::setAudioDescriptionMixLevel(float leveldB)
1333{
1334 AutoMutex lock(mLock);
1335 return setAudioDescriptionMixLevel_l(leveldB);
1336}
1337
1338status_t AudioTrack::setAudioDescriptionMixLevel_l(float leveldB)
1339{
1340 const status_t status = statusTFromBinderStatus(
1341 mAudioTrack->setAudioDescriptionMixLevel(leveldB));
1342 if (status == NO_ERROR) mAudioDescriptionMixLeveldB = leveldB;
1343 return status;
1344}
1345
1346status_t AudioTrack::getAudioDescriptionMixLevel(float* leveldB) const
1347{
1348 AutoMutex lock(mLock);
1349 return statusTFromBinderStatus(mAudioTrack->getAudioDescriptionMixLevel(leveldB));
1350}
1351
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001352status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
Andy Hung8edb8dc2015-03-26 19:13:55 -07001353{
Andy Hung8edb8dc2015-03-26 19:13:55 -07001354 AutoMutex lock(mLock);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001355 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
Andy Hung8edb8dc2015-03-26 19:13:55 -07001356 return NO_ERROR;
1357 }
Glenn Kastend79072e2016-01-06 08:41:20 -08001358 if (isOffloadedOrDirect_l()) {
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001359 const status_t status = statusTFromBinderStatus(mAudioTrack->setPlaybackRateParameters(
1360 VALUE_OR_RETURN_STATUS(
1361 legacy2aidl_audio_playback_rate_t_AudioPlaybackRate(playbackRate))));
1362 if (status == NO_ERROR) {
1363 mPlaybackRate = playbackRate;
1364 }
1365 return status;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001366 }
1367 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
1368 return INVALID_OPERATION;
1369 }
Andy Hungff874dc2016-04-11 16:49:09 -07001370
Andy Hungfb8ede22018-09-12 19:03:24 -07001371 ALOGV("%s(%d): mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001372 __func__, mPortId, mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001373 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001374 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
1375 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
1376 const float effectivePitch = adjustPitch(playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001377 AudioPlaybackRate playbackRateTemp = playbackRate;
1378 playbackRateTemp.mSpeed = effectiveSpeed;
1379 playbackRateTemp.mPitch = effectivePitch;
1380
Andy Hungfb8ede22018-09-12 19:03:24 -07001381 ALOGV("%s(%d) (effective) mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001382 __func__, mPortId, effectiveRate, effectiveSpeed, effectivePitch);
Andy Hungff874dc2016-04-11 16:49:09 -07001383
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001384 if (!isAudioPlaybackRateValid(playbackRateTemp)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001385 ALOGW("%s(%d) (%f, %f) failed (effective rate out of bounds)",
Eric Laurent973db022018-11-20 14:54:31 -08001386 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001387 return BAD_VALUE;
1388 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001389 // Check if the buffer size is compatible.
Andy Hung26145642015-04-15 21:56:53 -07001390 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001391 ALOGW("%s(%d) (%f, %f) failed (buffer size)",
Eric Laurent973db022018-11-20 14:54:31 -08001392 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung8edb8dc2015-03-26 19:13:55 -07001393 return BAD_VALUE;
1394 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001395
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001396 // Check resampler ratios are within bounds
Glenn Kastend3bb6452016-12-05 18:14:37 -08001397 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
1398 (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001399 ALOGW("%s(%d) (%f, %f) failed. Resample rate exceeds max accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001400 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001401 return BAD_VALUE;
1402 }
1403
Dan Austine34eae22015-10-27 16:14:52 -07001404 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001405 ALOGW("%s(%d) (%f, %f) failed. Resample rate below min accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001406 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001407 return BAD_VALUE;
1408 }
1409 mPlaybackRate = playbackRate;
1410 //set effective rates
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001411 mProxy->setPlaybackRate(playbackRateTemp);
Andy Hung26145642015-04-15 21:56:53 -07001412 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
Andy Hungb68f5eb2019-12-03 16:49:17 -08001413
1414 mediametrics::LogItem(mMetricsId)
1415 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYBACKPARAM)
1416 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
1417 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
1418 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
1419 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1420 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveRate)
1421 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1422 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)playbackRateTemp.mSpeed)
1423 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1424 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)playbackRateTemp.mPitch)
1425 .record();
1426
Andy Hung8edb8dc2015-03-26 19:13:55 -07001427 return NO_ERROR;
1428}
1429
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001430const AudioPlaybackRate& AudioTrack::getPlaybackRate()
Andy Hung8edb8dc2015-03-26 19:13:55 -07001431{
1432 AutoMutex lock(mLock);
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001433 if (isOffloadedOrDirect_l()) {
1434 media::AudioPlaybackRate playbackRateTemp;
1435 const status_t status = statusTFromBinderStatus(
1436 mAudioTrack->getPlaybackRateParameters(&playbackRateTemp));
1437 if (status == NO_ERROR) { // update local version if changed.
1438 mPlaybackRate =
1439 aidl2legacy_AudioPlaybackRate_audio_playback_rate_t(playbackRateTemp).value();
1440 }
1441 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001442 return mPlaybackRate;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001443}
1444
Phil Burkc0adecb2016-01-08 12:44:11 -08001445ssize_t AudioTrack::getBufferSizeInFrames()
1446{
1447 AutoMutex lock(mLock);
1448 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1449 return NO_INIT;
1450 }
Phil Burka9876702020-04-20 18:16:15 -07001451
Phil Burke8972b02016-03-04 11:29:57 -08001452 return (ssize_t) mProxy->getBufferSizeInFrames();
Phil Burkc0adecb2016-01-08 12:44:11 -08001453}
1454
Andy Hungf2c87b32016-04-07 19:49:29 -07001455status_t AudioTrack::getBufferDurationInUs(int64_t *duration)
1456{
1457 if (duration == nullptr) {
1458 return BAD_VALUE;
1459 }
1460 AutoMutex lock(mLock);
1461 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1462 return NO_INIT;
1463 }
1464 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames();
1465 if (bufferSizeInFrames < 0) {
1466 return (status_t)bufferSizeInFrames;
1467 }
1468 *duration = (int64_t)((double)bufferSizeInFrames * 1000000
1469 / ((double)mSampleRate * mPlaybackRate.mSpeed));
1470 return NO_ERROR;
1471}
1472
Phil Burkc0adecb2016-01-08 12:44:11 -08001473ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames)
1474{
1475 AutoMutex lock(mLock);
1476 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1477 return NO_INIT;
1478 }
Phil Burka9876702020-04-20 18:16:15 -07001479
1480 ssize_t originalBufferSize = mProxy->getBufferSizeInFrames();
1481 ssize_t finalBufferSize = mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
1482 if (originalBufferSize != finalBufferSize) {
Phil Burk64e16a72020-06-01 13:25:51 -07001483 android::mediametrics::LogItem(mMetricsId)
1484 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
1485 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1486 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t)getUnderrunCount_l())
1487 .record();
Phil Burka9876702020-04-20 18:16:15 -07001488 }
1489 return finalBufferSize;
Phil Burkc0adecb2016-01-08 12:44:11 -08001490}
1491
Andy Hung3c7f47a2021-03-16 17:30:09 -07001492ssize_t AudioTrack::getStartThresholdInFrames() const
1493{
1494 AutoMutex lock(mLock);
1495 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1496 return NO_INIT;
1497 }
1498 return (ssize_t) mProxy->getStartThresholdInFrames();
1499}
1500
1501ssize_t AudioTrack::setStartThresholdInFrames(size_t startThresholdInFrames)
1502{
1503 if (startThresholdInFrames > INT32_MAX || startThresholdInFrames == 0) {
1504 // contractually we could simply return the current threshold in frames
1505 // to indicate the request was ignored, but we return an error here.
1506 return BAD_VALUE;
1507 }
1508 AutoMutex lock(mLock);
1509 // We do not permit calling setStartThresholdInFrames() between the AudioTrack
1510 // default ctor AudioTrack() and set(...) but rather fail such an attempt.
1511 // (To do so would require a cached mOrigStartThresholdInFrames and we may
1512 // not have proper validation for the actual set value).
1513 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1514 return NO_INIT;
1515 }
1516 const uint32_t original = mProxy->getStartThresholdInFrames();
1517 const uint32_t final = mProxy->setStartThresholdInFrames(startThresholdInFrames);
1518 if (original != final) {
1519 android::mediametrics::LogItem(mMetricsId)
1520 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD)
1521 .set(AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES, (int32_t)final)
1522 .record();
1523 if (original > final) {
1524 // restart track if it was disabled by audioflinger due to previous underrun
1525 // and we reduced the number of frames for the threshold.
1526 restartIfDisabled();
1527 }
1528 }
1529 return final;
1530}
1531
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001532status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1533{
Glenn Kastend79072e2016-01-06 08:41:20 -08001534 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001535 return INVALID_OPERATION;
1536 }
1537
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001538 if (loopCount == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001539 ;
1540 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount &&
1541 loopEnd - loopStart >= MIN_LOOP) {
1542 ;
1543 } else {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001544 return BAD_VALUE;
1545 }
1546
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001547 AutoMutex lock(mLock);
1548 // See setPosition() regarding setting parameters such as loop points or position while active
1549 if (mState == STATE_ACTIVE) {
1550 return INVALID_OPERATION;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001551 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001552 setLoop_l(loopStart, loopEnd, loopCount);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001553 return NO_ERROR;
1554}
1555
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001556void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1557{
Andy Hung4ede21d2014-12-12 15:37:34 -08001558 // We do not update the periodic notification point.
1559 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
1560 mLoopCount = loopCount;
1561 mLoopEnd = loopEnd;
1562 mLoopStart = loopStart;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001563 mLoopCountNotified = loopCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001564 mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
Andy Hung3c09c782014-12-29 18:39:32 -08001565
1566 // Waking the AudioTrackThread is not needed as this cannot be called when active.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001567}
1568
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001569status_t AudioTrack::setMarkerPosition(uint32_t marker)
1570{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001571 // The only purpose of setting marker position is to get a callback
Atneya Nair6a8238eb2021-10-26 19:26:41 -04001572 if (!mCallback.promote() || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001573 return INVALID_OPERATION;
1574 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001575
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001576 AutoMutex lock(mLock);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001577 mMarkerPosition = marker;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001578 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001579
Andy Hung3c09c782014-12-29 18:39:32 -08001580 sp<AudioTrackThread> t = mAudioTrackThread;
1581 if (t != 0) {
1582 t->wake();
1583 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001584 return NO_ERROR;
1585}
1586
Glenn Kastena5224f32012-01-04 12:41:44 -08001587status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001588{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001589 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001590 return INVALID_OPERATION;
1591 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001592 if (marker == NULL) {
1593 return BAD_VALUE;
1594 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001595
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001596 AutoMutex lock(mLock);
Andy Hung90e8a972015-11-09 16:42:40 -08001597 mMarkerPosition.getValue(marker);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001598
1599 return NO_ERROR;
1600}
1601
1602status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
1603{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001604 // The only purpose of setting position update period is to get a callback
Atneya Nair6a8238eb2021-10-26 19:26:41 -04001605 if (!mCallback.promote() || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001606 return INVALID_OPERATION;
1607 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001608
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001609 AutoMutex lock(mLock);
Glenn Kasten200092b2014-08-15 15:13:30 -07001610 mNewPosition = updateAndGetPosition_l() + updatePeriod;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001611 mUpdatePeriod = updatePeriod;
Glenn Kasten2b2165c2014-01-13 08:53:36 -08001612
Andy Hung3c09c782014-12-29 18:39:32 -08001613 sp<AudioTrackThread> t = mAudioTrackThread;
1614 if (t != 0) {
1615 t->wake();
1616 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001617 return NO_ERROR;
1618}
1619
Glenn Kastena5224f32012-01-04 12:41:44 -08001620status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001621{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001622 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001623 return INVALID_OPERATION;
1624 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001625 if (updatePeriod == NULL) {
1626 return BAD_VALUE;
1627 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001628
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001629 AutoMutex lock(mLock);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001630 *updatePeriod = mUpdatePeriod;
1631
1632 return NO_ERROR;
1633}
1634
1635status_t AudioTrack::setPosition(uint32_t position)
1636{
Glenn Kastend79072e2016-01-06 08:41:20 -08001637 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001638 return INVALID_OPERATION;
1639 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001640 if (position > mFrameCount) {
1641 return BAD_VALUE;
1642 }
John Grossman4ff14ba2012-02-08 16:37:41 -08001643
Eric Laurent1703cdf2011-03-07 14:52:59 -08001644 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001645 // Currently we require that the player is inactive before setting parameters such as position
1646 // or loop points. Otherwise, there could be a race condition: the application could read the
1647 // current position, compute a new position or loop parameters, and then set that position or
1648 // loop parameters but it would do the "wrong" thing since the position has continued to advance
1649 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app
1650 // to specify how it wants to handle such scenarios.
1651 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001652 return INVALID_OPERATION;
1653 }
Andy Hung9b461582014-12-01 17:56:29 -08001654 // After setting the position, use full update period before notification.
Glenn Kasten200092b2014-08-15 15:13:30 -07001655 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001656 mStaticProxy->setBufferPosition(position);
Andy Hung3c09c782014-12-29 18:39:32 -08001657
1658 // Waking the AudioTrackThread is not needed as this cannot be called when active.
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001659 return NO_ERROR;
1660}
1661
Glenn Kasten200092b2014-08-15 15:13:30 -07001662status_t AudioTrack::getPosition(uint32_t *position)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001663{
Glenn Kastend65d73c2012-06-22 17:21:07 -07001664 if (position == NULL) {
1665 return BAD_VALUE;
1666 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001667
Eric Laurent1703cdf2011-03-07 14:52:59 -08001668 AutoMutex lock(mLock);
Andy Hung7a490e72016-03-23 15:58:10 -07001669 // FIXME: offloaded and direct tracks call into the HAL for render positions
1670 // for compressed/synced data; however, we use proxy position for pure linear pcm data
1671 // as we do not know the capability of the HAL for pcm position support and standby.
1672 // There may be some latency differences between the HAL position and the proxy position.
1673 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001674 uint32_t dspFrames = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001675
Eric Laurentab5cdba2014-06-09 17:22:27 -07001676 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001677 ALOGV("%s(%d): called in paused state, return cached position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001678 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001679 *position = mPausedPosition;
1680 return NO_ERROR;
1681 }
1682
Glenn Kasten142f5192014-03-25 17:44:59 -07001683 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung1f1db832015-06-08 13:26:10 -07001684 uint32_t halFrames; // actually unused
1685 (void) AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames);
1686 // FIXME: on getRenderPosition() error, we return OK with frame position 0.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001687 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001688 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED)
1689 // due to hardware latency. We leave this behavior for now.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001690 *position = dspFrames;
1691 } else {
Eric Laurent275e8e92014-11-30 15:14:47 -08001692 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung1f1db832015-06-08 13:26:10 -07001693 (void) restoreTrack_l("getPosition");
1694 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l()
1695 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position.
Eric Laurent275e8e92014-11-30 15:14:47 -08001696 }
1697
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001698 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
Glenn Kasten200092b2014-08-15 15:13:30 -07001699 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ?
Andy Hung90e8a972015-11-09 16:42:40 -08001700 0 : updateAndGetPosition_l().value();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001701 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001702 return NO_ERROR;
1703}
1704
Kévin PETIT377b2ec2014-02-03 12:35:36 +00001705status_t AudioTrack::getBufferPosition(uint32_t *position)
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001706{
Glenn Kastend79072e2016-01-06 08:41:20 -08001707 if (mSharedBuffer == 0) {
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001708 return INVALID_OPERATION;
1709 }
1710 if (position == NULL) {
1711 return BAD_VALUE;
1712 }
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001713
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001714 AutoMutex lock(mLock);
1715 *position = mStaticProxy->getBufferPosition();
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001716 return NO_ERROR;
1717}
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001718
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001719status_t AudioTrack::reload()
1720{
Glenn Kastend79072e2016-01-06 08:41:20 -08001721 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001722 return INVALID_OPERATION;
1723 }
1724
Eric Laurent1703cdf2011-03-07 14:52:59 -08001725 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001726 // See setPosition() regarding setting parameters such as loop points or position while active
1727 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001728 return INVALID_OPERATION;
1729 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001730 mNewPosition = mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001731 (void) updateAndGetPosition_l();
1732 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07001733 mPreviousTimestampValid = false;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001734#if 0
Andy Hung9b461582014-12-01 17:56:29 -08001735 // The documentation is not clear on the behavior of reload() and the restoration
Andy Hung53c3b5f2014-12-15 16:42:05 -08001736 // of loop count. Historically we have not restored loop count, start, end,
1737 // but it makes sense if one desires to repeat playing a particular sound.
1738 if (mLoopCount != 0) {
1739 mLoopCountNotified = mLoopCount;
1740 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
1741 }
1742#endif
Andy Hung9b461582014-12-01 17:56:29 -08001743 mStaticProxy->setBufferPosition(0);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001744 return NO_ERROR;
1745}
1746
Glenn Kasten38e905b2014-01-13 10:21:48 -08001747audio_io_handle_t AudioTrack::getOutput() const
Eric Laurentc2f1f072009-07-17 12:17:14 -07001748{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001749 AutoMutex lock(mLock);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001750 return mOutput;
Eric Laurent1703cdf2011-03-07 14:52:59 -08001751}
1752
Paul McLeanaa981192015-03-21 09:55:15 -07001753status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) {
1754 AutoMutex lock(mLock);
Eric Laurent2f2c1982021-06-02 14:03:11 +02001755 ALOGV("%s(%d): deviceId=%d mSelectedDeviceId=%d",
1756 __func__, mPortId, deviceId, mSelectedDeviceId);
Paul McLeanaa981192015-03-21 09:55:15 -07001757 if (mSelectedDeviceId != deviceId) {
1758 mSelectedDeviceId = deviceId;
Eric Laurentfb00fc72017-05-25 18:17:12 -07001759 if (mStatus == NO_ERROR) {
1760 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
jiabin156c6872017-10-06 09:47:15 -07001761 mProxy->interrupt();
Eric Laurentfb00fc72017-05-25 18:17:12 -07001762 }
Paul McLeanaa981192015-03-21 09:55:15 -07001763 }
Eric Laurent493404d2015-04-21 15:07:36 -07001764 return NO_ERROR;
Paul McLeanaa981192015-03-21 09:55:15 -07001765}
1766
1767audio_port_handle_t AudioTrack::getOutputDevice() {
1768 AutoMutex lock(mLock);
1769 return mSelectedDeviceId;
1770}
1771
Eric Laurentad2e7b92017-09-14 20:06:42 -07001772// must be called with mLock held
1773void AudioTrack::updateRoutedDeviceId_l()
1774{
1775 // if the track is inactive, do not update actual device as the output stream maybe routed
1776 // to a device not relevant to this client because of other active use cases.
1777 if (mState != STATE_ACTIVE) {
1778 return;
1779 }
1780 if (mOutput != AUDIO_IO_HANDLE_NONE) {
1781 audio_port_handle_t deviceId = AudioSystem::getDeviceIdForIo(mOutput);
1782 if (deviceId != AUDIO_PORT_HANDLE_NONE) {
1783 mRoutedDeviceId = deviceId;
1784 }
1785 }
1786}
1787
Eric Laurent296fb132015-05-01 11:38:42 -07001788audio_port_handle_t AudioTrack::getRoutedDeviceId() {
1789 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001790 updateRoutedDeviceId_l();
1791 return mRoutedDeviceId;
Eric Laurent296fb132015-05-01 11:38:42 -07001792}
1793
Eric Laurentbe916aa2010-06-01 23:49:17 -07001794status_t AudioTrack::attachAuxEffect(int effectId)
1795{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001796 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001797 status_t status;
1798 mAudioTrack->attachAuxEffect(effectId, &status);
Eric Laurent2beeb502010-07-16 07:43:46 -07001799 if (status == NO_ERROR) {
1800 mAuxEffectId = effectId;
1801 }
1802 return status;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001803}
1804
Eric Laurente83b55d2014-11-14 10:06:21 -08001805audio_stream_type_t AudioTrack::streamType() const
1806{
Eric Laurente83b55d2014-11-14 10:06:21 -08001807 return mStreamType;
1808}
1809
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001810uint32_t AudioTrack::latency()
1811{
1812 AutoMutex lock(mLock);
1813 updateLatency_l();
1814 return mLatency;
1815}
1816
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001817// -------------------------------------------------------------------------
1818
Eric Laurent1703cdf2011-03-07 14:52:59 -08001819// must be called with mLock held
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001820void AudioTrack::updateLatency_l()
1821{
1822 status_t status = AudioSystem::getLatency(mOutput, &mAfLatency);
1823 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08001824 ALOGW("%s(%d): getLatency(%d) failed status %d", __func__, mPortId, mOutput, status);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001825 } else {
1826 // FIXME don't believe this lie
Andy Hung13969262017-09-11 17:24:21 -07001827 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001828 }
1829}
1830
Phil Burkadbb75a2017-06-16 12:19:42 -07001831// TODO Move this macro to a common header file for enum to string conversion in audio framework.
1832#define MEDIA_CASE_ENUM(name) case name: return #name
1833const char * AudioTrack::convertTransferToText(transfer_type transferType) {
1834 switch (transferType) {
1835 MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
1836 MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
1837 MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
1838 MEDIA_CASE_ENUM(TRANSFER_SYNC);
1839 MEDIA_CASE_ENUM(TRANSFER_SHARED);
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001840 MEDIA_CASE_ENUM(TRANSFER_SYNC_NOTIF_CALLBACK);
Phil Burkadbb75a2017-06-16 12:19:42 -07001841 default:
1842 return "UNRECOGNIZED";
1843 }
1844}
1845
Glenn Kasten200092b2014-08-15 15:13:30 -07001846status_t AudioTrack::createTrack_l()
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001847{
Eric Laurentf32d7812017-11-30 14:44:07 -08001848 status_t status;
1849 bool callbackAdded = false;
Andy Hung3acde2c2021-11-11 09:18:08 -08001850 std::string errorMessage;
Eric Laurentf32d7812017-11-30 14:44:07 -08001851
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001852 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
1853 if (audioFlinger == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001854 errorMessage = StringPrintf("%s(%d): Could not get audioflinger",
Eric Laurent973db022018-11-20 14:54:31 -08001855 __func__, mPortId);
Andy Hung3acde2c2021-11-11 09:18:08 -08001856 status = DEAD_OBJECT;
Eric Laurentf32d7812017-11-30 14:44:07 -08001857 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001858 }
1859
Eric Laurent21da6472017-11-09 16:29:26 -08001860 {
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08001861 // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
1862 // After fast request is denied, we will request again if IAudioTrack is re-created.
Glenn Kastend79072e2016-01-06 08:41:20 -08001863 // Client can only express a preference for FAST. Server will perform additional tests.
Phil Burk33ff89b2015-11-30 11:16:01 -08001864 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Phil Burkadbb75a2017-06-16 12:19:42 -07001865 // either of these use cases:
1866 // use case 1: shared buffer
1867 bool sharedBuffer = mSharedBuffer != 0;
1868 bool transferAllowed =
Glenn Kastenc6ba8232014-02-27 13:34:29 -08001869 // use case 2: callback transfer mode
Glenn Kasten1dfe2f92015-03-09 12:03:14 -07001870 (mTransfer == TRANSFER_CALLBACK) ||
1871 // use case 3: obtain/release mode
Phil Burk33ff89b2015-11-30 11:16:01 -08001872 (mTransfer == TRANSFER_OBTAIN) ||
1873 // use case 4: synchronous write
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001874 ((mTransfer == TRANSFER_SYNC || mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK)
1875 && mThreadCanCallJava);
Phil Burkadbb75a2017-06-16 12:19:42 -07001876
Eric Laurent21da6472017-11-09 16:29:26 -08001877 bool fastAllowed = sharedBuffer || transferAllowed;
1878 if (!fastAllowed) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001879 ALOGW("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by client,"
1880 " not shared buffer and transfer = %s",
Eric Laurent973db022018-11-20 14:54:31 -08001881 __func__, mPortId,
Phil Burkadbb75a2017-06-16 12:19:42 -07001882 convertTransferToText(mTransfer));
Phil Burk33ff89b2015-11-30 11:16:01 -08001883 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
1884 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001885 }
1886
Eric Laurent21da6472017-11-09 16:29:26 -08001887 IAudioFlinger::CreateTrackInput input;
Andy Hunga2159aa2021-07-20 13:01:52 -07001888 if (mOriginalStreamType != AUDIO_STREAM_DEFAULT) {
1889 // Legacy: This is based on original parameters even if the track is recreated.
1890 input.attr = AudioSystem::streamTypeToAttributes(mOriginalStreamType);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001891 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08001892 input.attr = mAttributes;
Eric Laurentd1b449a2010-05-14 03:26:45 -07001893 }
Eric Laurent21da6472017-11-09 16:29:26 -08001894 input.config = AUDIO_CONFIG_INITIALIZER;
1895 input.config.sample_rate = mSampleRate;
1896 input.config.channel_mask = mChannelMask;
1897 input.config.format = mFormat;
1898 input.config.offload_info = mOffloadInfoCopy;
Svet Ganov3e5f14f2021-05-13 22:51:08 +00001899 input.clientInfo.attributionSource = mClientAttributionSource;
Eric Laurent21da6472017-11-09 16:29:26 -08001900 input.clientInfo.clientTid = -1;
Glenn Kasten363fb752014-01-15 12:27:31 -08001901 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Glenn Kasten47d55172017-05-23 11:19:30 -07001902 // It is currently meaningless to request SCHED_FIFO for a Java thread. Even if the
1903 // application-level code follows all non-blocking design rules, the language runtime
1904 // doesn't also follow those rules, so the thread will not benefit overall.
Phil Burk33ff89b2015-11-30 11:16:01 -08001905 if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
Eric Laurent21da6472017-11-09 16:29:26 -08001906 input.clientInfo.clientTid = mAudioTrackThread->getTid();
Glenn Kasten3acbd052012-02-28 10:39:56 -08001907 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001908 }
Eric Laurent21da6472017-11-09 16:29:26 -08001909 input.sharedBuffer = mSharedBuffer;
1910 input.notificationsPerBuffer = mNotificationsPerBufferReq;
1911 input.speed = 1.0;
1912 if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
1913 (mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
1914 input.speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
1915 max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
1916 }
1917 input.flags = mFlags;
1918 input.frameCount = mReqFrameCount;
1919 input.notificationFrameCount = mNotificationFramesReq;
1920 input.selectedDeviceId = mSelectedDeviceId;
1921 input.sessionId = mSessionId;
jiabinf6eb4c32020-02-25 14:06:25 -08001922 input.audioTrackCallback = mAudioTrackCallback;
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001923
Ytai Ben-Tsvi4dfeb622020-11-02 12:47:30 -08001924 media::CreateTrackResponse response;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001925 status = audioFlinger->createTrack(VALUE_OR_FATAL(input.toAidl()), response);
Ytai Ben-Tsvi357e26a2021-01-05 13:21:19 -08001926
1927 IAudioFlinger::CreateTrackOutput output{};
1928 if (status == NO_ERROR) {
1929 output = VALUE_OR_FATAL(IAudioFlinger::CreateTrackOutput::fromAidl(response));
1930 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001931
Eric Laurent21da6472017-11-09 16:29:26 -08001932 if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001933 errorMessage = StringPrintf(
1934 "%s(%d): AudioFlinger could not create track, status: %d output %d",
Eric Laurent973db022018-11-20 14:54:31 -08001935 __func__, mPortId, status, output.outputId);
Eric Laurentf32d7812017-11-30 14:44:07 -08001936 if (status == NO_ERROR) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001937 status = INVALID_OPERATION; // device not ready
Eric Laurentf32d7812017-11-30 14:44:07 -08001938 }
1939 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001940 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001941 ALOG_ASSERT(output.audioTrack != 0);
Glenn Kastenc08d20b2014-02-24 15:21:10 -08001942
Eric Laurent21da6472017-11-09 16:29:26 -08001943 mFrameCount = output.frameCount;
1944 mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
1945 mRoutedDeviceId = output.selectedDeviceId;
1946 mSessionId = output.sessionId;
Andy Hunga2159aa2021-07-20 13:01:52 -07001947 mStreamType = output.streamType;
Eric Laurent21da6472017-11-09 16:29:26 -08001948
1949 mSampleRate = output.sampleRate;
1950 if (mOriginalSampleRate == 0) {
1951 mOriginalSampleRate = mSampleRate;
1952 }
1953
1954 mAfFrameCount = output.afFrameCount;
1955 mAfSampleRate = output.afSampleRate;
1956 mAfLatency = output.afLatencyMs;
1957
1958 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
1959
Glenn Kasten38e905b2014-01-13 10:21:48 -08001960 // AudioFlinger now owns the reference to the I/O handle,
1961 // so we are no longer responsible for releasing it.
1962
Glenn Kasten7fd04222016-02-02 12:38:16 -08001963 // FIXME compare to AudioRecord
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001964 std::optional<media::SharedFileRegion> sfr;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001965 output.audioTrack->getCblk(&sfr);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001966 sp<IMemory> iMem = VALUE_OR_FATAL(aidl2legacy_NullableSharedFileRegion_IMemory(sfr));
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001967 if (iMem == 0) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001968 errorMessage = StringPrintf("%s(%d): Could not get control block", __func__, mPortId);
1969 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001970 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001971 }
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001972 // TODO: Using unsecurePointer() has some associated security pitfalls
1973 // (see declaration for details).
1974 // Either document why it is safe in this case or address the
1975 // issue (e.g. by copying).
1976 void *iMemPointer = iMem->unsecurePointer();
Glenn Kasten0cde0762014-01-16 15:06:36 -08001977 if (iMemPointer == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08001978 errorMessage = StringPrintf(
1979 "%s(%d): Could not get control block pointer", __func__, mPortId);
1980 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08001981 goto exit;
Glenn Kasten0cde0762014-01-16 15:06:36 -08001982 }
Glenn Kasten53cec222013-08-29 09:01:02 -07001983 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001984 if (mAudioTrack != 0) {
Marco Nelissenf8880202014-11-14 07:58:25 -08001985 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001986 mDeathNotifier.clear();
1987 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001988 mAudioTrack = output.audioTrack;
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001989 mCblkMemory = iMem;
Eric Laurent3bcffa12014-06-12 18:38:45 -07001990 IPCThreadState::self()->flushCommands();
1991
Glenn Kasten0cde0762014-01-16 15:06:36 -08001992 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001993 mCblk = cblk;
Glenn Kasten5f631512014-02-24 15:16:07 -08001994
Glenn Kastena07f17c2013-04-23 12:39:37 -07001995 mAwaitBoost = false;
Glenn Kasten363fb752014-01-15 12:27:31 -08001996 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Eric Laurent21da6472017-11-09 16:29:26 -08001997 if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001998 ALOGI("%s(%d): AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08001999 __func__, mPortId, mReqFrameCount, mFrameCount);
Phil Burk33ff89b2015-11-30 11:16:01 -08002000 if (!mThreadCanCallJava) {
2001 mAwaitBoost = true;
2002 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08002003 } else {
Robert Wuc121cd12021-08-13 17:51:40 +00002004 ALOGV("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08002005 __func__, mPortId, mReqFrameCount, mFrameCount);
Glenn Kastene0fa4672012-04-24 14:35:14 -07002006 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08002007 }
Eric Laurent21da6472017-11-09 16:29:26 -08002008 mFlags = output.flags;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002009
Eric Laurentad2e7b92017-09-14 20:06:42 -07002010 //mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
Eric Laurent09f1ed22019-04-24 17:45:17 -07002011 if (mDeviceCallback != 0) {
Eric Laurentad2e7b92017-09-14 20:06:42 -07002012 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07002013 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07002014 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07002015 AudioSystem::addAudioDeviceCallback(this, output.outputId, output.portId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07002016 callbackAdded = true;
2017 }
2018
Eric Laurent09f1ed22019-04-24 17:45:17 -07002019 mPortId = output.portId;
Glenn Kasten38e905b2014-01-13 10:21:48 -08002020 // We retain a copy of the I/O handle, but don't own the reference
Eric Laurent21da6472017-11-09 16:29:26 -08002021 mOutput = output.outputId;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002022 mRefreshRemaining = true;
2023
2024 // Starting address of buffers in shared memory. If there is a shared buffer, buffers
2025 // is the value of pointer() for the shared buffer, otherwise buffers points
2026 // immediately after the control block. This address is for the mapping within client
2027 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
2028 void* buffers;
Glenn Kasten363fb752014-01-15 12:27:31 -08002029 if (mSharedBuffer == 0) {
Glenn Kasten138d6f92015-03-20 10:54:51 -07002030 buffers = cblk + 1;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002031 } else {
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07002032 // TODO: Using unsecurePointer() has some associated security pitfalls
2033 // (see declaration for details).
2034 // Either document why it is safe in this case or address the
2035 // issue (e.g. by copying).
2036 buffers = mSharedBuffer->unsecurePointer();
Glenn Kasten138d6f92015-03-20 10:54:51 -07002037 if (buffers == NULL) {
Andy Hung3acde2c2021-11-11 09:18:08 -08002038 errorMessage = StringPrintf(
2039 "%s(%d): Could not get buffer pointer", __func__, mPortId);
2040 ALOGE("%s", errorMessage.c_str());
2041 status = FAILED_TRANSACTION;
Eric Laurentf32d7812017-11-30 14:44:07 -08002042 goto exit;
Glenn Kasten138d6f92015-03-20 10:54:51 -07002043 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002044 }
2045
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002046 mAudioTrack->attachAuxEffect(mAuxEffectId, &status);
Glenn Kasten5f631512014-02-24 15:16:07 -08002047
Glenn Kasten093000f2012-05-03 09:35:36 -07002048 // If IAudioTrack is re-created, don't let the requested frameCount
2049 // decrease. This can confuse clients that cache frameCount().
Eric Laurent21da6472017-11-09 16:29:26 -08002050 if (mFrameCount > mReqFrameCount) {
2051 mReqFrameCount = mFrameCount;
Glenn Kasten093000f2012-05-03 09:35:36 -07002052 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08002053
Andy Hungd7bd69e2015-07-24 07:52:41 -07002054 // reset server position to 0 as we have new cblk.
2055 mServer = 0;
2056
Glenn Kastene3aa6592012-12-04 12:22:46 -08002057 // update proxy
Glenn Kasten363fb752014-01-15 12:27:31 -08002058 if (mSharedBuffer == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002059 mStaticProxy.clear();
Eric Laurent21da6472017-11-09 16:29:26 -08002060 mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002061 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08002062 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002063 mProxy = mStaticProxy;
2064 }
seunghak.hane6a9d6582014-11-22 15:22:35 +09002065
2066 mProxy->setVolumeLR(gain_minifloat_pack(
2067 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
2068 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
2069
Glenn Kastene3aa6592012-12-04 12:22:46 -08002070 mProxy->setSendLevel(mSendLevel);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002071 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
2072 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
2073 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07002074 mProxy->setSampleRate(effectiveSampleRate);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002075
2076 AudioPlaybackRate playbackRateTemp = mPlaybackRate;
2077 playbackRateTemp.mSpeed = effectiveSpeed;
2078 playbackRateTemp.mPitch = effectivePitch;
2079 mProxy->setPlaybackRate(playbackRateTemp);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002080 mProxy->setMinimum(mNotificationFramesAct);
2081
Kuowei Lid4adbdb2020-08-13 14:44:25 +08002082 if (mDualMonoMode != AUDIO_DUAL_MONO_MODE_OFF) {
2083 setDualMonoMode_l(mDualMonoMode);
2084 }
2085 if (mAudioDescriptionMixLeveldB != -std::numeric_limits<float>::infinity()) {
2086 setAudioDescriptionMixLevel_l(mAudioDescriptionMixLeveldB);
2087 }
2088
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002089 mDeathNotifier = new DeathNotifier(this);
Marco Nelissenf8880202014-11-14 07:58:25 -08002090 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
Glenn Kastene3aa6592012-12-04 12:22:46 -08002091
Andy Hungb68f5eb2019-12-03 16:49:17 -08002092 // This is the first log sent from the AudioTrack client.
2093 // The creation of the audio track by AudioFlinger (in the code above)
2094 // is the first log of the AudioTrack and must be present before
2095 // any AudioTrack client logs will be accepted.
Andy Hungea840382020-05-05 21:50:17 -07002096
Andy Hungb68f5eb2019-12-03 16:49:17 -08002097 mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(mPortId);
2098 mediametrics::LogItem(mMetricsId)
2099 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE)
2100 // the following are immutable
Andy Hunga629bd12020-06-05 16:03:53 -07002101 .set(AMEDIAMETRICS_PROP_FLAGS, toString(mFlags).c_str())
2102 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -08002103 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
Andy Hung3a5c2f32021-02-17 15:06:42 -08002104 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, mLogSessionId)
Andy Hung839a3062021-02-17 11:15:16 -08002105 .set(AMEDIAMETRICS_PROP_PLAYERIID, mPlayerIId)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002106 .set(AMEDIAMETRICS_PROP_TRACKID, mPortId) // dup from key
Andy Hungb68f5eb2019-12-03 16:49:17 -08002107 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2108 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2109 .set(AMEDIAMETRICS_PROP_THREADID, (int32_t)output.outputId)
2110 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2111 .set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)mRoutedDeviceId)
2112 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2113 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
2114 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
2115 // the following are NOT immutable
2116 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)mVolume[AUDIO_INTERLEAVE_LEFT])
2117 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)mVolume[AUDIO_INTERLEAVE_RIGHT])
2118 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
2119 .set(AMEDIAMETRICS_PROP_AUXEFFECTID, (int32_t)mAuxEffectId)
2120 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2121 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2122 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2123 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2124 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveSampleRate)
2125 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2126 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)effectiveSpeed)
2127 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2128 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)effectivePitch)
2129 .record();
2130
2131 // mSendLevel
2132 // mReqFrameCount?
2133 // mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq
2134 // mLatency, mAfLatency, mAfFrameCount, mAfSampleRate
2135
Glenn Kasten38e905b2014-01-13 10:21:48 -08002136 }
2137
Eric Laurentf32d7812017-11-30 14:44:07 -08002138exit:
Andy Hung3acde2c2021-11-11 09:18:08 -08002139 if (status != NO_ERROR) {
2140 if (callbackAdded) {
2141 // note: mOutput is always valid is callbackAdded is true
2142 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
2143 }
2144 ALOGE_IF(!errorMessage.empty(), "%s", errorMessage.c_str());
2145 reportError(status, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE, errorMessage.c_str());
Eric Laurentad2e7b92017-09-14 20:06:42 -07002146 }
Eric Laurentf32d7812017-11-30 14:44:07 -08002147 mStatus = status;
Eric Laurent21da6472017-11-09 16:29:26 -08002148
2149 // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
Glenn Kasten38e905b2014-01-13 10:21:48 -08002150 return status;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002151}
2152
Andy Hung3acde2c2021-11-11 09:18:08 -08002153void AudioTrack::reportError(status_t status, const char *event, const char *message) const
2154{
2155 if (status == NO_ERROR) return;
2156 // We report error on the native side because some callers do not come
2157 // from Java.
2158 mediametrics::LogItem(std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + "error")
2159 .set(AMEDIAMETRICS_PROP_EVENT, event)
2160 .set(AMEDIAMETRICS_PROP_ERROR, mediametrics::statusToErrorString(status))
2161 .set(AMEDIAMETRICS_PROP_ERRORMESSAGE, message)
2162 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
2163 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
2164 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2165 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2166 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2167 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2168 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
2169 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mReqFrameCount) // requested frame count
2170 // the following are NOT immutable
2171 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2172 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2173 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2174 .record();
2175}
2176
Glenn Kastenb46f3942015-03-09 12:00:30 -07002177status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002178{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002179 if (audioBuffer == NULL) {
Glenn Kasten551b5352015-03-20 11:30:28 -07002180 if (nonContig != NULL) {
2181 *nonContig = 0;
2182 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002183 return BAD_VALUE;
Eric Laurent9b7d9502011-03-21 11:49:00 -07002184 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002185 if (mTransfer != TRANSFER_OBTAIN) {
2186 audioBuffer->frameCount = 0;
2187 audioBuffer->size = 0;
2188 audioBuffer->raw = NULL;
Glenn Kasten551b5352015-03-20 11:30:28 -07002189 if (nonContig != NULL) {
2190 *nonContig = 0;
2191 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002192 return INVALID_OPERATION;
2193 }
Eric Laurent9b7d9502011-03-21 11:49:00 -07002194
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002195 const struct timespec *requested;
Eric Laurentdf576992014-01-27 18:13:39 -08002196 struct timespec timeout;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002197 if (waitCount == -1) {
2198 requested = &ClientProxy::kForever;
2199 } else if (waitCount == 0) {
2200 requested = &ClientProxy::kNonBlocking;
2201 } else if (waitCount > 0) {
Chih-Hung Hsiehbca74292018-08-10 16:06:07 -07002202 time_t ms = WAIT_PERIOD_MS * (time_t) waitCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002203 timeout.tv_sec = ms / 1000;
Andy Hung06a730b2020-04-09 13:28:31 -07002204 timeout.tv_nsec = (ms % 1000) * 1000000;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002205 requested = &timeout;
2206 } else {
Eric Laurent973db022018-11-20 14:54:31 -08002207 ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002208 requested = NULL;
2209 }
Glenn Kastenb46f3942015-03-09 12:00:30 -07002210 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002211}
Eric Laurent1703cdf2011-03-07 14:52:59 -08002212
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002213status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
2214 struct timespec *elapsed, size_t *nonContig)
2215{
2216 // previous and new IAudioTrack sequence numbers are used to detect track re-creation
2217 uint32_t oldSequence = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002218
2219 Proxy::Buffer buffer;
2220 status_t status = NO_ERROR;
2221
2222 static const int32_t kMaxTries = 5;
2223 int32_t tryCounter = kMaxTries;
2224
2225 do {
2226 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
2227 // keep them from going away if another thread re-creates the track during obtainBuffer()
2228 sp<AudioTrackClientProxy> proxy;
2229 sp<IMemory> iMem;
2230
2231 { // start of lock scope
2232 AutoMutex lock(mLock);
2233
Glenn Kasten305996c2020-01-27 08:03:37 -08002234 uint32_t newSequence = mSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002235 // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
2236 if (status == DEAD_OBJECT) {
2237 // re-create track, unless someone else has already done so
2238 if (newSequence == oldSequence) {
2239 status = restoreTrack_l("obtainBuffer");
2240 if (status != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002241 buffer.mFrameCount = 0;
2242 buffer.mRaw = NULL;
2243 buffer.mNonContig = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002244 break;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002245 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002246 }
2247 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002248 oldSequence = newSequence;
2249
Eric Laurent4d231dc2016-03-11 18:38:23 -08002250 if (status == NOT_ENOUGH_DATA) {
2251 restartIfDisabled();
2252 }
2253
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002254 // Keep the extra references
2255 proxy = mProxy;
2256 iMem = mCblkMemory;
2257
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002258 if (mState == STATE_STOPPING) {
2259 status = -EINTR;
2260 buffer.mFrameCount = 0;
2261 buffer.mRaw = NULL;
2262 buffer.mNonContig = 0;
2263 break;
2264 }
2265
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002266 // Non-blocking if track is stopped or paused
2267 if (mState != STATE_ACTIVE) {
2268 requested = &ClientProxy::kNonBlocking;
2269 }
2270
2271 } // end of lock scope
2272
2273 buffer.mFrameCount = audioBuffer->frameCount;
2274 // FIXME starts the requested timeout and elapsed over from scratch
2275 status = proxy->obtainBuffer(&buffer, requested, elapsed);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002276 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002277
2278 audioBuffer->frameCount = buffer.mFrameCount;
Andy Hungabdb9902015-01-12 15:08:22 -08002279 audioBuffer->size = buffer.mFrameCount * mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002280 audioBuffer->raw = buffer.mRaw;
Glenn Kasten305996c2020-01-27 08:03:37 -08002281 audioBuffer->sequence = oldSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002282 if (nonContig != NULL) {
2283 *nonContig = buffer.mNonContig;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002284 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002285 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002286}
2287
Glenn Kasten54a8a452015-03-09 12:03:00 -07002288void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002289{
Glenn Kasten3f02be22015-03-09 11:59:04 -07002290 // FIXME add error checking on mode, by adding an internal version
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002291 if (mTransfer == TRANSFER_SHARED) {
2292 return;
2293 }
2294
Andy Hungabdb9902015-01-12 15:08:22 -08002295 size_t stepCount = audioBuffer->size / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002296 if (stepCount == 0) {
2297 return;
2298 }
2299
2300 Proxy::Buffer buffer;
2301 buffer.mFrameCount = stepCount;
2302 buffer.mRaw = audioBuffer->raw;
Glenn Kastene3aa6592012-12-04 12:22:46 -08002303
Eric Laurent1703cdf2011-03-07 14:52:59 -08002304 AutoMutex lock(mLock);
Glenn Kasten305996c2020-01-27 08:03:37 -08002305 if (audioBuffer->sequence != mSequence) {
2306 // This Buffer came from a different IAudioTrack instance, so ignore the releaseBuffer
2307 ALOGD("%s is no-op due to IAudioTrack sequence mismatch %u != %u",
2308 __func__, audioBuffer->sequence, mSequence);
2309 return;
2310 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002311 mReleased += stepCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002312 mInUnderrun = false;
2313 mProxy->releaseBuffer(&buffer);
2314
2315 // restart track if it was disabled by audioflinger due to previous underrun
Eric Laurent4d231dc2016-03-11 18:38:23 -08002316 restartIfDisabled();
2317}
2318
2319void AudioTrack::restartIfDisabled()
2320{
2321 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
2322 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002323 ALOGW("%s(%d): releaseBuffer() track %p disabled due to previous underrun, restarting",
Eric Laurent973db022018-11-20 14:54:31 -08002324 __func__, mPortId, this);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002325 // FIXME ignoring status
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002326 status_t status;
2327 mAudioTrack->start(&status);
Eric Laurentdf839842012-05-31 14:27:14 -07002328 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002329}
2330
2331// -------------------------------------------------------------------------
2332
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002333ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002334{
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002335 if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07002336 return INVALID_OPERATION;
2337 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002338
Eric Laurentab5cdba2014-06-09 17:22:27 -07002339 if (isDirect()) {
2340 AutoMutex lock(mLock);
2341 int32_t flags = android_atomic_and(
2342 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
2343 &mCblk->mFlags);
2344 if (flags & CBLK_INVALID) {
2345 return DEAD_OBJECT;
2346 }
2347 }
2348
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002349 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
Jiabin Huang447cea72020-07-28 22:35:18 +00002350 // Validation: user is most-likely passing an error code, and it would
Glenn Kasten99e53b82012-01-19 08:59:58 -08002351 // make the return value ambiguous (actualSize vs error).
Andy Hungfb8ede22018-09-12 19:03:24 -07002352 ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
Eric Laurent973db022018-11-20 14:54:31 -08002353 __func__, mPortId, buffer, userSize, userSize);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002354 return BAD_VALUE;
2355 }
2356
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002357 size_t written = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002358 Buffer audioBuffer;
2359
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002360 while (userSize >= mFrameSize) {
2361 audioBuffer.frameCount = userSize / mFrameSize;
Eric Laurentc2f1f072009-07-17 12:17:14 -07002362
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002363 status_t err = obtainBuffer(&audioBuffer,
2364 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002365 if (err < 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002366 if (written > 0) {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002367 break;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002368 }
Glenn Kasten0a2f1512016-07-22 08:06:37 -07002369 if (err == TIMED_OUT || err == -EINTR) {
2370 err = WOULD_BLOCK;
2371 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002372 return ssize_t(err);
2373 }
2374
Glenn Kastenae4b8792015-03-20 09:04:21 -07002375 size_t toWrite = audioBuffer.size;
Andy Hungabdb9902015-01-12 15:08:22 -08002376 memcpy(audioBuffer.i8, buffer, toWrite);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002377 buffer = ((const char *) buffer) + toWrite;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002378 userSize -= toWrite;
2379 written += toWrite;
2380
2381 releaseBuffer(&audioBuffer);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002382 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002383
Andy Hungea2b9c02016-02-12 17:06:53 -08002384 if (written > 0) {
2385 mFramesWritten += written / mFrameSize;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002386
2387 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2388 const sp<AudioTrackThread> t = mAudioTrackThread;
2389 if (t != 0) {
2390 // causes wake up of the playback thread, that will callback the client for
2391 // more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
2392 t->wake();
2393 }
2394 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002395 }
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002396
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002397 return written;
2398}
2399
2400// -------------------------------------------------------------------------
2401
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08002402nsecs_t AudioTrack::processAudioBuffer()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002403{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002404 // Currently the AudioTrack thread is not created if there are no callbacks.
2405 // Would it ever make sense to run the thread, even without callbacks?
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002406 // If so, then replace this by checks at each use for mCallback != NULL.
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002407 LOG_ALWAYS_FATAL_IF(mCblk == NULL);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002408 mLock.lock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002409 sp<IAudioTrackCallback> callback = mCallback.promote();
2410 if (!callback) {
2411 mCallback = nullptr;
2412 return NS_NEVER;
2413 }
Glenn Kastena07f17c2013-04-23 12:39:37 -07002414 if (mAwaitBoost) {
2415 mAwaitBoost = false;
2416 mLock.unlock();
2417 static const int32_t kMaxTries = 5;
2418 int32_t tryCounter = kMaxTries;
2419 uint32_t pollUs = 10000;
2420 do {
Glenn Kasten8255ba72016-08-23 13:54:23 -07002421 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002422 if (policy == SCHED_FIFO || policy == SCHED_RR) {
2423 break;
2424 }
2425 usleep(pollUs);
2426 pollUs <<= 1;
2427 } while (tryCounter-- > 0);
2428 if (tryCounter < 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002429 ALOGE("%s(%d): did not receive expected priority boost on time",
Eric Laurent973db022018-11-20 14:54:31 -08002430 __func__, mPortId);
Glenn Kastena07f17c2013-04-23 12:39:37 -07002431 }
Glenn Kastenb0dfd462013-07-10 16:52:47 -07002432 // Run again immediately
2433 return 0;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002434 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002435
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002436 // Can only reference mCblk while locked
2437 int32_t flags = android_atomic_and(
Glenn Kasten96f60d82013-07-12 10:21:18 -07002438 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);
Glenn Kastena47f3162012-11-07 10:13:08 -08002439
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002440 // Check for track invalidation
2441 if (flags & CBLK_INVALID) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002442 // for offloaded tracks restoreTrack_l() will just update the sequence and clear
2443 // AudioSystem cache. We should not exit here but after calling the callback so
2444 // that the upper layers can recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002445 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
Lajos Molnarf1063e22015-04-17 15:19:42 -07002446 status_t status __unused = restoreTrack_l("processAudioBuffer");
2447 // FIXME unused status
Andy Hung53c3b5f2014-12-15 16:42:05 -08002448 // after restoration, continue below to make sure that the loop and buffer events
2449 // are notified because they have been cleared from mCblk->mFlags above.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002450 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002451 }
2452
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002453 bool waitStreamEnd = mState == STATE_STOPPING;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002454 bool active = mState == STATE_ACTIVE;
2455
2456 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
2457 bool newUnderrun = false;
2458 if (flags & CBLK_UNDERRUN) {
2459#if 0
2460 // Currently in shared buffer mode, when the server reaches the end of buffer,
2461 // the track stays active in continuous underrun state. It's up to the application
2462 // to pause or stop the track, or set the position to a new offset within buffer.
2463 // This was some experimental code to auto-pause on underrun. Keeping it here
2464 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
2465 if (mTransfer == TRANSFER_SHARED) {
2466 mState = STATE_PAUSED;
2467 active = false;
2468 }
2469#endif
2470 if (!mInUnderrun) {
2471 mInUnderrun = true;
2472 newUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002473 }
2474 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002475
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002476 // Get current position of server
Andy Hung90e8a972015-11-09 16:42:40 -08002477 Modulo<uint32_t> position(updateAndGetPosition_l());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002478
2479 // Manage marker callback
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002480 bool markerReached = false;
Andy Hung90e8a972015-11-09 16:42:40 -08002481 Modulo<uint32_t> markerPosition(mMarkerPosition);
2482 // uses 32 bit wraparound for comparison with position.
2483 if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002484 mMarkerReached = markerReached = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002485 }
2486
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002487 // Determine number of new position callback(s) that will be needed, while locked
2488 size_t newPosCount = 0;
Andy Hung90e8a972015-11-09 16:42:40 -08002489 Modulo<uint32_t> newPosition(mNewPosition);
2490 uint32_t updatePeriod = mUpdatePeriod;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002491 // FIXME fails for wraparound, need 64 bits
2492 if (updatePeriod > 0 && position >= newPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002493 newPosCount = ((position - newPosition).value() / updatePeriod) + 1;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002494 mNewPosition += updatePeriod * newPosCount;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002495 }
2496
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002497 // Cache other fields that will be needed soon
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002498 uint32_t sampleRate = mSampleRate;
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002499 float speed = mPlaybackRate.mSpeed;
Andy Hunga7f03352015-05-31 21:54:49 -07002500 const uint32_t notificationFrames = mNotificationFramesAct;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002501 if (mRefreshRemaining) {
2502 mRefreshRemaining = false;
2503 mRemainingFrames = notificationFrames;
2504 mRetryOnPartialBuffer = false;
2505 }
2506 size_t misalignment = mProxy->getMisalignment();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002507 uint32_t sequence = mSequence;
Glenn Kasten96f04882013-09-20 09:28:56 -07002508 sp<AudioTrackClientProxy> proxy = mProxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002509
Andy Hung53c3b5f2014-12-15 16:42:05 -08002510 // Determine the number of new loop callback(s) that will be needed, while locked.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002511 uint32_t loopCountNotifications = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -08002512 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
2513
2514 if (mLoopCount > 0) {
2515 int loopCount;
2516 size_t bufferPosition;
2517 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
2518 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
2519 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
2520 mLoopCountNotified = loopCount; // discard any excess notifications
2521 } else if (mLoopCount < 0) {
2522 // FIXME: We're not accurate with notification count and position with infinite looping
2523 // since loopCount from server side will always return -1 (we could decrement it).
2524 size_t bufferPosition = mStaticProxy->getBufferPosition();
2525 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
2526 loopPeriod = mLoopEnd - bufferPosition;
2527 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
2528 size_t bufferPosition = mStaticProxy->getBufferPosition();
2529 loopPeriod = mFrameCount - bufferPosition;
2530 }
2531
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002532 // These fields don't need to be cached, because they are assigned only by set():
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002533 // mTransfer, mCallback, mUserData, mFormat, mFrameSize, mFlags
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002534 // mFlags is also assigned by createTrack_l(), but not the bit we care about.
2535
2536 mLock.unlock();
2537
Andy Hunga7f03352015-05-31 21:54:49 -07002538 // get anchor time to account for callbacks.
2539 const nsecs_t timeBeforeCallbacks = systemTime();
2540
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002541 if (waitStreamEnd) {
Andy Hunga7f03352015-05-31 21:54:49 -07002542 // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread
2543 // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
2544 // (and make sure we don't callback for more data while we're stopping).
2545 // This helps with position, marker notifications, and track invalidation.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002546 struct timespec timeout;
2547 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
2548 timeout.tv_nsec = 0;
2549
Glenn Kasten96f04882013-09-20 09:28:56 -07002550 status_t status = proxy->waitStreamEndDone(&timeout);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002551 switch (status) {
2552 case NO_ERROR:
2553 case DEAD_OBJECT:
2554 case TIMED_OUT:
Andy Hung39609a02015-09-03 16:38:38 -07002555 if (status != DEAD_OBJECT) {
2556 // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop();
2557 // instead, the application should handle the EVENT_NEW_IAUDIOTRACK.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002558 callback->onStreamEnd();
Andy Hung39609a02015-09-03 16:38:38 -07002559 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002560 {
2561 AutoMutex lock(mLock);
2562 // The previously assigned value of waitStreamEnd is no longer valid,
2563 // since the mutex has been unlocked and either the callback handler
2564 // or another thread could have re-started the AudioTrack during that time.
2565 waitStreamEnd = mState == STATE_STOPPING;
2566 if (waitStreamEnd) {
2567 mState = STATE_STOPPED;
Andy Hungc2813e52014-10-16 17:54:34 -07002568 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002569 }
2570 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002571 if (waitStreamEnd && status != DEAD_OBJECT) {
2572 return NS_INACTIVE;
2573 }
2574 break;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002575 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002576 return 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002577 }
2578
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002579 // perform callbacks while unlocked
2580 if (newUnderrun) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002581 callback->onUnderrun();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002582 }
Andy Hung53c3b5f2014-12-15 16:42:05 -08002583 while (loopCountNotifications > 0) {
Andy Hung53c3b5f2014-12-15 16:42:05 -08002584 --loopCountNotifications;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002585 callback->onLoopEnd(mLoopCount > 0 ? loopCountNotifications + mLoopCountNotified : -1);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002586 }
2587 if (flags & CBLK_BUFFER_END) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002588 callback->onBufferEnd();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002589 }
2590 if (markerReached) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002591 callback->onMarker(markerPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002592 }
2593 while (newPosCount > 0) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002594 callback->onNewPos(newPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002595 newPosition += updatePeriod;
2596 newPosCount--;
2597 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002598
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002599 if (mObservedSequence != sequence) {
2600 mObservedSequence = sequence;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002601 callback->onNewIAudioTrack();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002602 // for offloaded tracks, just wait for the upper layers to recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002603 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002604 return NS_INACTIVE;
2605 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002606 }
2607
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002608 // if inactive, then don't run me again until re-started
2609 if (!active) {
2610 return NS_INACTIVE;
Eric Laurent2267ba12011-09-07 11:13:23 -07002611 }
2612
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002613 // Compute the estimated time until the next timed event (position, markers, loops)
2614 // FIXME only for non-compressed audio
2615 uint32_t minFrames = ~0;
2616 if (!markerReached && position < markerPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002617 minFrames = (markerPosition - position).value();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002618 }
2619 if (loopPeriod > 0 && loopPeriod < minFrames) {
Andy Hung2d85f092015-01-07 12:45:13 -08002620 // loopPeriod is already adjusted for actual position.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002621 minFrames = loopPeriod;
2622 }
Andy Hung2d85f092015-01-07 12:45:13 -08002623 if (updatePeriod > 0) {
Andy Hung90e8a972015-11-09 16:42:40 -08002624 minFrames = min(minFrames, (newPosition - position).value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002625 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002626
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002627 // If > 0, poll periodically to recover from a stuck server. A good value is 2.
2628 static const uint32_t kPoll = 0;
2629 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
2630 minFrames = kPoll * notificationFrames;
2631 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002632
Andy Hunga7f03352015-05-31 21:54:49 -07002633 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
2634 static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL;
2635 const nsecs_t timeAfterCallbacks = systemTime();
2636
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002637 // Convert frame units to time units
2638 nsecs_t ns = NS_WHENEVER;
2639 if (minFrames != (uint32_t) ~0) {
jiabinc7bb8322017-09-06 18:20:11 -07002640 // AudioFlinger consumption of client data may be irregular when coming out of device
2641 // standby since the kernel buffers require filling. This is throttled to no more than 2x
2642 // the expected rate in the MixerThread. Hence, we reduce the estimated time to wait by one
2643 // half (but no more than half a second) to improve callback accuracy during these temporary
2644 // data surges.
2645 const nsecs_t estimatedNs = framesToNanoseconds(minFrames, sampleRate, speed);
2646 constexpr nsecs_t maxThrottleCompensationNs = 500000000LL;
2647 ns = estimatedNs - min(estimatedNs / 2, maxThrottleCompensationNs) + kWaitPeriodNs;
Andy Hunga7f03352015-05-31 21:54:49 -07002648 ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time
2649 // TODO: Should we warn if the callback time is too long?
2650 if (ns < 0) ns = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002651 }
2652
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002653 // If not supplying data by EVENT_MORE_DATA or EVENT_CAN_WRITE_MORE_DATA, then we're done
2654 if (mTransfer != TRANSFER_CALLBACK && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002655 return ns;
2656 }
2657
Andy Hunga7f03352015-05-31 21:54:49 -07002658 // EVENT_MORE_DATA callback handling.
2659 // Timing for linear pcm audio data formats can be derived directly from the
2660 // buffer fill level.
2661 // Timing for compressed data is not directly available from the buffer fill level,
2662 // rather indirectly from waiting for blocking mode callbacks or waiting for obtain()
2663 // to return a certain fill level.
2664
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002665 struct timespec timeout;
2666 const struct timespec *requested = &ClientProxy::kForever;
2667 if (ns != NS_WHENEVER) {
2668 timeout.tv_sec = ns / 1000000000LL;
2669 timeout.tv_nsec = ns % 1000000000LL;
Andy Hungfb8ede22018-09-12 19:03:24 -07002670 ALOGV("%s(%d): timeout %ld.%03d",
Eric Laurent973db022018-11-20 14:54:31 -08002671 __func__, mPortId, timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002672 requested = &timeout;
2673 }
2674
Andy Hungea2b9c02016-02-12 17:06:53 -08002675 size_t writtenFrames = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002676 while (mRemainingFrames > 0) {
2677
2678 Buffer audioBuffer;
2679 audioBuffer.frameCount = mRemainingFrames;
2680 size_t nonContig;
2681 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
2682 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
Andy Hungfb8ede22018-09-12 19:03:24 -07002683 "%s(%d): obtainBuffer() err=%d frameCount=%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002684 __func__, mPortId, err, audioBuffer.frameCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002685 requested = &ClientProxy::kNonBlocking;
2686 size_t avail = audioBuffer.frameCount + nonContig;
Andy Hungfb8ede22018-09-12 19:03:24 -07002687 ALOGV("%s(%d): obtainBuffer(%u) returned %zu = %zu + %zu err %d",
Eric Laurent973db022018-11-20 14:54:31 -08002688 __func__, mPortId, mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002689 if (err != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002690 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
2691 (isOffloaded() && (err == DEAD_OBJECT))) {
Glenn Kasten606fbc12015-10-22 15:28:15 -07002692 // FIXME bug 25195759
2693 return 1000000;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002694 }
Andy Hungfb8ede22018-09-12 19:03:24 -07002695 ALOGE("%s(%d): Error %d obtaining an audio buffer, giving up.",
Eric Laurent973db022018-11-20 14:54:31 -08002696 __func__, mPortId, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002697 return NS_NEVER;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002698 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002699
Phil Burkfdb3c072016-02-09 10:47:02 -08002700 if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002701 mRetryOnPartialBuffer = false;
2702 if (avail < mRemainingFrames) {
Andy Hunga7f03352015-05-31 21:54:49 -07002703 if (ns > 0) { // account for obtain time
2704 const nsecs_t timeNow = systemTime();
2705 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2706 }
Andy Hungeb6e6502019-04-29 13:47:40 -07002707
2708 // delayNs is first computed by the additional frames required in the buffer.
2709 nsecs_t delayNs = framesToNanoseconds(
2710 mRemainingFrames - avail, sampleRate, speed);
2711
2712 // afNs is the AudioFlinger mixer period in ns.
2713 const nsecs_t afNs = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2714
2715 // If the AudioTrack is double buffered based on the AudioFlinger mixer period,
2716 // we may have a race if we wait based on the number of frames desired.
2717 // This is a possible issue with resampling and AAudio.
2718 //
2719 // The granularity of audioflinger processing is one mixer period; if
2720 // our wait time is less than one mixer period, wait at most half the period.
2721 if (delayNs < afNs) {
2722 delayNs = std::min(delayNs, afNs / 2);
2723 }
2724
2725 // adjust our ns wait by delayNs.
2726 if (ns < 0 /* NS_WHENEVER */ || delayNs < ns) {
2727 ns = delayNs;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002728 }
2729 return ns;
2730 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07002731 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002732
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002733 size_t reqSize = audioBuffer.size;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002734 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2735 // when notifying client it can write more data, pass the total size that can be
2736 // written in the next write() call, since it's not passed through the callback
2737 audioBuffer.size += nonContig;
2738 }
Atneya Nairc2dd1272021-10-26 19:39:51 -04002739 const size_t writtenSize = (mTransfer == TRANSFER_CALLBACK)
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002740 ? callback->onMoreData(audioBuffer)
2741 : callback->onCanWriteMoreData(audioBuffer);
Jiabin Huang447cea72020-07-28 22:35:18 +00002742 // Validate on returned size
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002743 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002744 ALOGE("%s(%d): EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
Eric Laurent973db022018-11-20 14:54:31 -08002745 __func__, mPortId, reqSize, ssize_t(writtenSize));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002746 return NS_NEVER;
2747 }
2748
2749 if (writtenSize == 0) {
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002750 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2751 // The callback EVENT_CAN_WRITE_MORE_DATA was processed in the JNI of
2752 // android.media.AudioTrack. The JNI is not using the callback to provide data,
2753 // it only signals to the Java client that it can provide more data, which
2754 // this track is read to accept now.
2755 // The playback thread will be awaken at the next ::write()
2756 return NS_WHENEVER;
2757 }
The Android Open Source Project8555d082009-03-05 14:34:35 -08002758 // The callback is done filling buffers
2759 // Keep this thread going to handle timed events and
2760 // still try to get more data in intervals of WAIT_PERIOD_MS
2761 // but don't just loop and block the CPU, so wait
Andy Hunga7f03352015-05-31 21:54:49 -07002762
2763 // mCbf(EVENT_MORE_DATA, ...) might either
2764 // (1) Block until it can fill the buffer, returning 0 size on EOS.
2765 // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS.
2766 // (3) Return 0 size when no data is available, does not wait for more data.
2767 //
2768 // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer.
2769 // We try to compute the wait time to avoid a tight sleep-wait cycle,
2770 // especially for case (3).
2771 //
2772 // The decision to support (1) and (2) affect the sizing of mRemainingFrames
2773 // and this loop; whereas for case (3) we could simply check once with the full
2774 // buffer size and skip the loop entirely.
2775
2776 nsecs_t myns;
Phil Burkfdb3c072016-02-09 10:47:02 -08002777 if (audio_has_proportional_frames(mFormat)) {
Andy Hunga7f03352015-05-31 21:54:49 -07002778 // time to wait based on buffer occupancy
2779 const nsecs_t datans = mRemainingFrames <= avail ? 0 :
2780 framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
2781 // audio flinger thread buffer size (TODO: adjust for fast tracks)
Glenn Kastenea38ee72016-04-18 11:08:01 -07002782 // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks.
Andy Hunga7f03352015-05-31 21:54:49 -07002783 const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2784 // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0.
2785 myns = datans + (afns / 2);
2786 } else {
2787 // FIXME: This could ping quite a bit if the buffer isn't full.
2788 // Note that when mState is stopping we waitStreamEnd, so it never gets here.
2789 myns = kWaitPeriodNs;
2790 }
2791 if (ns > 0) { // account for obtain and callback time
2792 const nsecs_t timeNow = systemTime();
2793 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2794 }
2795 if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
2796 ns = myns;
2797 }
2798 return ns;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002799 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002800
Atneya Nairc2dd1272021-10-26 19:39:51 -04002801 // releaseBuffer reads from audioBuffer.size
2802 audioBuffer.size = writtenSize;
2803
Glenn Kasten138d6f92015-03-20 10:54:51 -07002804 size_t releasedFrames = writtenSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002805 audioBuffer.frameCount = releasedFrames;
2806 mRemainingFrames -= releasedFrames;
2807 if (misalignment >= releasedFrames) {
2808 misalignment -= releasedFrames;
2809 } else {
2810 misalignment = 0;
2811 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002812
2813 releaseBuffer(&audioBuffer);
Andy Hungea2b9c02016-02-12 17:06:53 -08002814 writtenFrames += releasedFrames;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002815
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002816 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
2817 // if callback doesn't like to accept the full chunk
2818 if (writtenSize < reqSize) {
2819 continue;
2820 }
2821
2822 // There could be enough non-contiguous frames available to satisfy the remaining request
2823 if (mRemainingFrames <= nonContig) {
2824 continue;
2825 }
2826
2827#if 0
2828 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
2829 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
2830 // that total to a sum == notificationFrames.
2831 if (0 < misalignment && misalignment <= mRemainingFrames) {
2832 mRemainingFrames = misalignment;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002833 return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002834 }
2835#endif
2836
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002837 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002838 if (writtenFrames > 0) {
2839 AutoMutex lock(mLock);
2840 mFramesWritten += writtenFrames;
2841 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002842 mRemainingFrames = notificationFrames;
2843 mRetryOnPartialBuffer = true;
2844
2845 // A lot has transpired since ns was calculated, so run again immediately and re-calculate
2846 return 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002847}
2848
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002849status_t AudioTrack::restoreTrack_l(const char *from)
Eric Laurent1703cdf2011-03-07 14:52:59 -08002850{
Andy Hungb68f5eb2019-12-03 16:49:17 -08002851 status_t result = NO_ERROR; // logged: make sure to set this before returning.
2852 const int64_t beginNs = systemTime();
Andy Hunga6b27032020-04-27 10:34:24 -07002853 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -08002854 mediametrics::LogItem(mMetricsId)
2855 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_RESTORE)
Andy Hungea840382020-05-05 21:50:17 -07002856 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08002857 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
2858 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
2859 .set(AMEDIAMETRICS_PROP_WHERE, from)
2860 .record(); });
2861
Andy Hungfb8ede22018-09-12 19:03:24 -07002862 ALOGW("%s(%d): dead IAudioTrack, %s, creating a new one from %s()",
Eric Laurent973db022018-11-20 14:54:31 -08002863 __func__, mPortId, isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002864 ++mSequence;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002865
Glenn Kastena47f3162012-11-07 10:13:08 -08002866 // refresh the audio configuration cache in this process to make sure we get new
Glenn Kastend2d089f2014-11-05 11:48:12 -08002867 // output parameters and new IAudioFlinger in createTrack_l()
Glenn Kastena47f3162012-11-07 10:13:08 -08002868 AudioSystem::clearAudioConfigCache();
Eric Laurent9f6530f2011-08-30 10:18:54 -07002869
Ronghua Wufaeb0f22015-05-21 12:20:21 -07002870 if (isOffloadedOrDirect_l() || mDoNotReconnect) {
Andy Hung1f1db832015-06-08 13:26:10 -07002871 // FIXME re-creation of offloaded and direct tracks is not yet implemented;
2872 // reconsider enabling for linear PCM encodings when position can be preserved.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002873 result = DEAD_OBJECT;
2874 return result;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002875 }
2876
Phil Burk2812d9e2016-01-04 10:34:30 -08002877 // Save so we can return count since creation.
2878 mUnderrunCountOffset = getUnderrunCount_l();
2879
Glenn Kasten200092b2014-08-15 15:13:30 -07002880 // save the old static buffer position
Andy Hungf20a4e92016-08-15 19:10:34 -07002881 uint32_t staticPosition = 0;
Andy Hung4ede21d2014-12-12 15:37:34 -08002882 size_t bufferPosition = 0;
2883 int loopCount = 0;
2884 if (mStaticProxy != 0) {
2885 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
Andy Hungf20a4e92016-08-15 19:10:34 -07002886 staticPosition = mStaticProxy->getPosition().unsignedValue();
Andy Hung4ede21d2014-12-12 15:37:34 -08002887 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002888
Andy Hung3c7f47a2021-03-16 17:30:09 -07002889 // save the old startThreshold and framecount
2890 const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
2891 const uint32_t originalFrameCount = mProxy->frameCount();
2892
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002893 // See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
2894 // causes a lot of churn on the service side, and it can reject starting
2895 // playback of a previously created track. May also apply to other cases.
2896 const int INITIAL_RETRIES = 3;
2897 int retries = INITIAL_RETRIES;
2898retry:
2899 if (retries < INITIAL_RETRIES) {
2900 // See the comment for clearAudioConfigCache at the start of the function.
2901 AudioSystem::clearAudioConfigCache();
2902 }
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08002903 mFlags = mOrigFlags;
2904
Glenn Kasten200092b2014-08-15 15:13:30 -07002905 // If a new IAudioTrack is successfully created, createTrack_l() will modify the
Glenn Kastena47f3162012-11-07 10:13:08 -08002906 // following member variables: mAudioTrack, mCblkMemory and mCblk.
Glenn Kasten200092b2014-08-15 15:13:30 -07002907 // It will also delete the strong references on previous IAudioTrack and IMemory.
2908 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002909 result = createTrack_l();
Eric Laurentcc21e4f2013-10-16 15:12:32 -07002910
Eric Laurent6ec546d2018-10-10 16:52:14 -07002911 if (result == NO_ERROR) {
Andy Hungd7bd69e2015-07-24 07:52:41 -07002912 // take the frames that will be lost by track recreation into account in saved position
2913 // For streaming tracks, this is the amount we obtained from the user/client
2914 // (not the number actually consumed at the server - those are already lost).
2915 if (mStaticProxy == 0) {
2916 mPosition = mReleased;
2917 }
Andy Hung4ede21d2014-12-12 15:37:34 -08002918 // Continue playback from last known position and restore loop.
2919 if (mStaticProxy != 0) {
2920 if (loopCount != 0) {
2921 mStaticProxy->setBufferPositionAndLoop(bufferPosition,
2922 mLoopStart, mLoopEnd, loopCount);
2923 } else {
2924 mStaticProxy->setBufferPosition(bufferPosition);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002925 if (bufferPosition == mFrameCount) {
Eric Laurent973db022018-11-20 14:54:31 -08002926 ALOGD("%s(%d): restoring track at end of static buffer", __func__, mPortId);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002927 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002928 }
2929 }
Andy Hung4ef88d72017-02-21 19:47:53 -08002930 // restore volume handler
Andy Hung39399b62017-04-21 15:07:45 -07002931 mVolumeHandler->forall([this](const VolumeShaper &shaper) -> VolumeShaper::Status {
2932 sp<VolumeShaper::Operation> operationToEnd =
2933 new VolumeShaper::Operation(shaper.mOperation);
Andy Hung4ef88d72017-02-21 19:47:53 -08002934 // TODO: Ideally we would restore to the exact xOffset position
2935 // as returned by getVolumeShaperState(), but we don't have that
2936 // information when restoring at the client unless we periodically poll
2937 // the server or create shared memory state.
2938 //
Andy Hung39399b62017-04-21 15:07:45 -07002939 // For now, we simply advance to the end of the VolumeShaper effect
2940 // if it has been started.
2941 if (shaper.isStarted()) {
Andy Hungf3702642017-05-05 17:33:32 -07002942 operationToEnd->setNormalizedTime(1.f);
Andy Hung39399b62017-04-21 15:07:45 -07002943 }
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002944 media::VolumeShaperConfiguration config;
2945 shaper.mConfiguration->writeToParcelable(&config);
2946 media::VolumeShaperOperation operation;
2947 operationToEnd->writeToParcelable(&operation);
2948 status_t status;
2949 mAudioTrack->applyVolumeShaper(config, operation, &status);
2950 return status;
Andy Hung4ef88d72017-02-21 19:47:53 -08002951 });
2952
Andy Hung3c7f47a2021-03-16 17:30:09 -07002953 // restore the original start threshold if different than frameCount.
2954 if (originalStartThresholdInFrames != originalFrameCount) {
2955 // Note: mProxy->setStartThresholdInFrames() call is in the Proxy
2956 // and does not trigger a restart.
2957 // (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
2958 // Any start would be triggered on the mState == ACTIVE check below.
2959 const uint32_t currentThreshold =
2960 mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
2961 ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
2962 "%s(%d) startThresholdInFrames changing from %u to %u",
2963 __func__, mPortId, originalStartThresholdInFrames, currentThreshold);
2964 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002965 if (mState == STATE_ACTIVE) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002966 mAudioTrack->start(&result);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002967 }
Andy Hungf20a4e92016-08-15 19:10:34 -07002968 // server resets to zero so we offset
2969 mFramesWrittenServerOffset =
2970 mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten;
2971 mFramesWrittenAtRestore = mFramesWrittenServerOffset;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002972 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002973 if (result != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08002974 ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002975 if (--retries > 0) {
Eric Laurent6ec546d2018-10-10 16:52:14 -07002976 // leave time for an eventual race condition to clear before retrying
2977 usleep(500000);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002978 goto retry;
2979 }
Eric Laurent6ec546d2018-10-10 16:52:14 -07002980 // if no retries left, set invalid bit to force restoring at next occasion
2981 // and avoid inconsistent active state on client and server sides
2982 if (mCblk != nullptr) {
2983 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
2984 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002985 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002986 return result;
2987}
2988
Andy Hung90e8a972015-11-09 16:42:40 -08002989Modulo<uint32_t> AudioTrack::updateAndGetPosition_l()
Glenn Kasten200092b2014-08-15 15:13:30 -07002990{
2991 // This is the sole place to read server consumed frames
Andy Hung90e8a972015-11-09 16:42:40 -08002992 Modulo<uint32_t> newServer(mProxy->getPosition());
2993 const int32_t delta = (newServer - mServer).signedValue();
Glenn Kasten200092b2014-08-15 15:13:30 -07002994 // TODO There is controversy about whether there can be "negative jitter" in server position.
2995 // This should be investigated further, and if possible, it should be addressed.
2996 // A more definite failure mode is infrequent polling by client.
2997 // One could call (void)getPosition_l() in releaseBuffer(),
2998 // so mReleased and mPosition are always lock-step as best possible.
2999 // That should ensure delta never goes negative for infrequent polling
3000 // unless the server has more than 2^31 frames in its buffer,
3001 // in which case the use of uint32_t for these counters has bigger issues.
Andy Hung90e8a972015-11-09 16:42:40 -08003002 ALOGE_IF(delta < 0,
Andy Hungfb8ede22018-09-12 19:03:24 -07003003 "%s(%d): detected illegal retrograde motion by the server: mServer advanced by %d",
Eric Laurent973db022018-11-20 14:54:31 -08003004 __func__, mPortId, delta);
Chad Brubaker039c27a2015-09-23 15:17:29 -07003005 mServer = newServer;
Andy Hung90e8a972015-11-09 16:42:40 -08003006 if (delta > 0) { // avoid retrograde
3007 mPosition += delta;
3008 }
3009 return mPosition;
Glenn Kasten200092b2014-08-15 15:13:30 -07003010}
3011
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003012bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed)
Andy Hung8edb8dc2015-03-26 19:13:55 -07003013{
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003014 updateLatency_l();
Andy Hung8edb8dc2015-03-26 19:13:55 -07003015 // applicable for mixing tracks only (not offloaded or direct)
3016 if (mStaticProxy != 0) {
3017 return true; // static tracks do not have issues with buffer sizing.
3018 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07003019 const size_t minFrameCount =
Eric Laurent21da6472017-11-09 16:29:26 -08003020 AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
3021 sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003022 const bool allowed = mFrameCount >= minFrameCount;
3023 ALOGD_IF(!allowed,
Andy Hungfb8ede22018-09-12 19:03:24 -07003024 "%s(%d): denied "
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003025 "mAfLatency:%u mAfFrameCount:%zu mAfSampleRate:%u sampleRate:%u speed:%f "
3026 "mFrameCount:%zu < minFrameCount:%zu",
Eric Laurent973db022018-11-20 14:54:31 -08003027 __func__, mPortId,
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003028 mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed,
Andy Hung8edb8dc2015-03-26 19:13:55 -07003029 mFrameCount, minFrameCount);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003030 return allowed;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003031}
3032
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003033status_t AudioTrack::setParameters(const String8& keyValuePairs)
3034{
3035 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003036 status_t status;
3037 mAudioTrack->setParameters(keyValuePairs.c_str(), &status);
3038 return status;
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003039}
3040
Dean Wheatleya70eef72018-01-04 14:23:50 +11003041status_t AudioTrack::selectPresentation(int presentationId, int programId)
3042{
3043 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08003044 AudioParameter param = AudioParameter();
3045 param.addInt(String8(AudioParameter::keyPresentationId), presentationId);
3046 param.addInt(String8(AudioParameter::keyProgramId), programId);
3047 ALOGV("%s(%d): PresentationId/ProgramId[%s]",
3048 __func__, mPortId, param.toString().string());
3049
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003050 status_t status;
3051 mAudioTrack->setParameters(param.toString().c_str(), &status);
3052 return status;
Dean Wheatleya70eef72018-01-04 14:23:50 +11003053}
3054
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003055VolumeShaper::Status AudioTrack::applyVolumeShaper(
3056 const sp<VolumeShaper::Configuration>& configuration,
3057 const sp<VolumeShaper::Operation>& operation)
3058{
3059 AutoMutex lock(mLock);
Andy Hung4ef88d72017-02-21 19:47:53 -08003060 mVolumeHandler->setIdIfNecessary(configuration);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003061 media::VolumeShaperConfiguration config;
3062 configuration->writeToParcelable(&config);
3063 media::VolumeShaperOperation op;
3064 operation->writeToParcelable(&op);
3065 VolumeShaper::Status status;
3066 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003067
3068 if (status == DEAD_OBJECT) {
3069 if (restoreTrack_l("applyVolumeShaper") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003070 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003071 }
3072 }
Andy Hung4ef88d72017-02-21 19:47:53 -08003073 if (status >= 0) {
3074 // save VolumeShaper for restore
3075 mVolumeHandler->applyVolumeShaper(configuration, operation);
Andy Hung39399b62017-04-21 15:07:45 -07003076 if (mState == STATE_ACTIVE || mState == STATE_STOPPING) {
3077 mVolumeHandler->setStarted();
3078 }
3079 } else {
3080 // warn only if not an expected restore failure.
3081 ALOGW_IF(!((isOffloadedOrDirect_l() || mDoNotReconnect) && status == DEAD_OBJECT),
Eric Laurent973db022018-11-20 14:54:31 -08003082 "%s(%d): applyVolumeShaper failed: %d", __func__, mPortId, status);
Andy Hung4ef88d72017-02-21 19:47:53 -08003083 }
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003084 return status;
3085}
3086
3087sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id)
3088{
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003089 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003090 std::optional<media::VolumeShaperState> vss;
3091 mAudioTrack->getVolumeShaperState(id, &vss);
3092 sp<VolumeShaper::State> state;
3093 if (vss.has_value()) {
3094 state = new VolumeShaper::State();
3095 state->readFromParcelable(vss.value());
3096 }
Andy Hung39399b62017-04-21 15:07:45 -07003097 if (state.get() == nullptr && (mCblk->mFlags & CBLK_INVALID) != 0) {
3098 if (restoreTrack_l("getVolumeShaperState") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003099 mAudioTrack->getVolumeShaperState(id, &vss);
3100 if (vss.has_value()) {
3101 state = new VolumeShaper::State();
3102 state->readFromParcelable(vss.value());
3103 }
Andy Hung39399b62017-04-21 15:07:45 -07003104 }
3105 }
3106 return state;
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003107}
3108
Andy Hungea2b9c02016-02-12 17:06:53 -08003109status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp)
3110{
3111 if (timestamp == nullptr) {
3112 return BAD_VALUE;
3113 }
3114 AutoMutex lock(mLock);
Andy Hunge13f8a62016-03-30 14:20:42 -07003115 return getTimestamp_l(timestamp);
3116}
3117
3118status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp)
3119{
Andy Hungea2b9c02016-02-12 17:06:53 -08003120 if (mCblk->mFlags & CBLK_INVALID) {
3121 const status_t status = restoreTrack_l("getTimestampExtended");
3122 if (status != OK) {
3123 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3124 // recommending that the track be recreated.
3125 return DEAD_OBJECT;
3126 }
3127 }
3128 // check for offloaded/direct here in case restoring somehow changed those flags.
3129 if (isOffloadedOrDirect_l()) {
3130 return INVALID_OPERATION; // not supported
3131 }
3132 status_t status = mProxy->getTimestamp(timestamp);
Andy Hungfb8ede22018-09-12 19:03:24 -07003133 LOG_ALWAYS_FATAL_IF(status != OK, "%s(%d): status %d not allowed from proxy getTimestamp",
Eric Laurent973db022018-11-20 14:54:31 -08003134 __func__, mPortId, status);
Andy Hungea2b9c02016-02-12 17:06:53 -08003135 bool found = false;
Andy Hunge1e98462016-04-12 10:18:51 -07003136 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten;
3137 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
3138 // server side frame offset in case AudioTrack has been restored.
3139 for (int i = ExtendedTimestamp::LOCATION_SERVER;
3140 i < ExtendedTimestamp::LOCATION_MAX; ++i) {
3141 if (timestamp->mTimeNs[i] >= 0) {
3142 // apply server offset (frames flushed is ignored
3143 // so we don't report the jump when the flush occurs).
3144 timestamp->mPosition[i] += mFramesWrittenServerOffset;
3145 found = true;
Andy Hungea2b9c02016-02-12 17:06:53 -08003146 }
3147 }
3148 return found ? OK : WOULD_BLOCK;
3149}
3150
Glenn Kastence703742013-07-19 16:33:58 -07003151status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
3152{
Glenn Kasten53cec222013-08-29 09:01:02 -07003153 AutoMutex lock(mLock);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003154 return getTimestamp_l(timestamp);
3155}
Phil Burk1b420972015-04-22 10:52:21 -07003156
Andy Hung65ffdfc2016-10-10 15:52:11 -07003157status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp)
3158{
Phil Burk1b420972015-04-22 10:52:21 -07003159 bool previousTimestampValid = mPreviousTimestampValid;
3160 // Set false here to cover all the error return cases.
3161 mPreviousTimestampValid = false;
3162
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003163 switch (mState) {
3164 case STATE_ACTIVE:
3165 case STATE_PAUSED:
3166 break; // handle below
3167 case STATE_FLUSHED:
3168 case STATE_STOPPED:
3169 return WOULD_BLOCK;
3170 case STATE_STOPPING:
3171 case STATE_PAUSED_STOPPING:
3172 if (!isOffloaded_l()) {
3173 return INVALID_OPERATION;
3174 }
3175 break; // offloaded tracks handled below
3176 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003177 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in getTimestamp(): %d",
Eric Laurent973db022018-11-20 14:54:31 -08003178 __func__, mPortId, mState);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003179 break;
Glenn Kastenfe346c72013-08-30 13:28:22 -07003180 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003181
Eric Laurent275e8e92014-11-30 15:14:47 -08003182 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung6653c932015-06-08 13:27:48 -07003183 const status_t status = restoreTrack_l("getTimestamp");
3184 if (status != OK) {
3185 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3186 // recommending that the track be recreated.
3187 return DEAD_OBJECT;
3188 }
Eric Laurent275e8e92014-11-30 15:14:47 -08003189 }
3190
Glenn Kasten200092b2014-08-15 15:13:30 -07003191 // The presented frame count must always lag behind the consumed frame count.
3192 // To avoid a race, read the presented frames first. This ensures that presented <= consumed.
Andy Hung6ae58432016-02-16 18:32:24 -08003193
3194 status_t status;
Andy Hung818e7a32016-02-16 18:08:07 -08003195 if (isOffloadedOrDirect_l()) {
Andy Hung6ae58432016-02-16 18:32:24 -08003196 // use Binder to get timestamp
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003197 media::AudioTimestampInternal ts;
3198 mAudioTrack->getTimestamp(&ts, &status);
3199 if (status == OK) {
Andy Hung973638a2020-12-08 20:47:45 -08003200 timestamp = VALUE_OR_FATAL(aidl2legacy_AudioTimestampInternal_AudioTimestamp(ts));
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003201 }
Andy Hung6ae58432016-02-16 18:32:24 -08003202 } else {
3203 // read timestamp from shared memory
3204 ExtendedTimestamp ets;
3205 status = mProxy->getTimestamp(&ets);
3206 if (status == OK) {
Andy Hungb01faa32016-04-27 12:51:32 -07003207 ExtendedTimestamp::Location location;
3208 status = ets.getBestTimestamp(&timestamp, &location);
3209
3210 if (status == OK) {
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003211 updateLatency_l();
Andy Hungb01faa32016-04-27 12:51:32 -07003212 // It is possible that the best location has moved from the kernel to the server.
3213 // In this case we adjust the position from the previous computed latency.
3214 if (location == ExtendedTimestamp::LOCATION_SERVER) {
3215 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL,
Andy Hungfb8ede22018-09-12 19:03:24 -07003216 "%s(%d): location moved from kernel to server",
Eric Laurent973db022018-11-20 14:54:31 -08003217 __func__, mPortId);
Andy Hung07eee802016-06-21 16:47:16 -07003218 // check that the last kernel OK time info exists and the positions
3219 // are valid (if they predate the current track, the positions may
3220 // be zero or negative).
Andy Hungb01faa32016-04-27 12:51:32 -07003221 const int64_t frames =
Andy Hung6d7b1192016-05-07 22:59:48 -07003222 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
Andy Hung07eee802016-06-21 16:47:16 -07003223 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 ||
3224 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 ||
3225 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0)
Andy Hung6d7b1192016-05-07 22:59:48 -07003226 ?
3227 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed
3228 / 1000)
3229 :
3230 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3231 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]);
Andy Hungfb8ede22018-09-12 19:03:24 -07003232 ALOGV("%s(%d): frame adjustment:%lld timestamp:%s",
Eric Laurent973db022018-11-20 14:54:31 -08003233 __func__, mPortId, (long long)frames, ets.toString().c_str());
Andy Hungb01faa32016-04-27 12:51:32 -07003234 if (frames >= ets.mPosition[location]) {
3235 timestamp.mPosition = 0;
3236 } else {
3237 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames);
3238 }
Andy Hung69488c42016-05-16 18:43:33 -07003239 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) {
3240 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER,
Andy Hungfb8ede22018-09-12 19:03:24 -07003241 "%s(%d): location moved from server to kernel",
Eric Laurent973db022018-11-20 14:54:31 -08003242 __func__, mPortId);
Andy Hung98731a22019-04-08 19:19:07 -07003243
3244 if (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER] ==
3245 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL]) {
3246 // In Q, we don't return errors as an invalid time
3247 // but instead we leave the last kernel good timestamp alone.
3248 //
3249 // If server is identical to kernel, the device data pipeline is idle.
3250 // A better start time is now. The retrograde check ensures
3251 // timestamp monotonicity.
3252 const int64_t nowNs = systemTime();
Andy Hungcf3b7152019-04-19 18:29:21 -07003253 if (!mTimestampStallReported) {
3254 ALOGD("%s(%d): device stall time corrected using current time %lld",
3255 __func__, mPortId, (long long)nowNs);
3256 mTimestampStallReported = true;
3257 }
Andy Hung98731a22019-04-08 19:19:07 -07003258 timestamp.mTime = convertNsToTimespec(nowNs);
Andy Hungcf3b7152019-04-19 18:29:21 -07003259 } else {
3260 mTimestampStallReported = false;
Andy Hung98731a22019-04-08 19:19:07 -07003261 }
Andy Hungb01faa32016-04-27 12:51:32 -07003262 }
Andy Hung5d313802016-10-10 15:09:39 -07003263
3264 // We update the timestamp time even when paused.
3265 if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) {
3266 const int64_t now = systemTime();
Andy Hung2b01f002017-07-05 12:01:36 -07003267 const int64_t at = audio_utils_ns_from_timespec(&timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003268 const int64_t lag =
3269 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
3270 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0)
3271 ? int64_t(mAfLatency * 1000000LL)
3272 : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3273 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK])
3274 * NANOS_PER_SECOND / mSampleRate;
3275 const int64_t limit = now - lag; // no earlier than this limit
3276 if (at < limit) {
3277 ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld",
3278 (long long)lag, (long long)at, (long long)limit);
Andy Hungffa36952017-08-17 10:41:51 -07003279 timestamp.mTime = convertNsToTimespec(limit);
Andy Hung5d313802016-10-10 15:09:39 -07003280 }
3281 }
Andy Hungb01faa32016-04-27 12:51:32 -07003282 mPreviousLocation = location;
3283 } else {
3284 // right after AudioTrack is started, one may not find a timestamp
Eric Laurent973db022018-11-20 14:54:31 -08003285 ALOGV("%s(%d): getBestTimestamp did not find timestamp", __func__, mPortId);
Andy Hungb01faa32016-04-27 12:51:32 -07003286 }
Andy Hung6ae58432016-02-16 18:32:24 -08003287 }
3288 if (status == INVALID_OPERATION) {
Andy Hungf20a4e92016-08-15 19:10:34 -07003289 // INVALID_OPERATION occurs when no timestamp has been issued by the server;
3290 // other failures are signaled by a negative time.
3291 // If we come out of FLUSHED or STOPPED where the position is known
3292 // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of
3293 // "zero" for NuPlayer). We don't convert for track restoration as position
3294 // does not reset.
Andy Hungfb8ede22018-09-12 19:03:24 -07003295 ALOGV("%s(%d): timestamp server offset:%lld restore frames:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003296 __func__, mPortId,
Andy Hungf20a4e92016-08-15 19:10:34 -07003297 (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore);
3298 if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) {
3299 status = WOULD_BLOCK;
3300 }
Andy Hung6ae58432016-02-16 18:32:24 -08003301 }
3302 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003303 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08003304 ALOGV_IF(status != WOULD_BLOCK, "%s(%d): getTimestamp error:%#x", __func__, mPortId, status);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003305 return status;
3306 }
3307 if (isOffloadedOrDirect_l()) {
3308 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) {
3309 // use cached paused position in case another offloaded track is running.
3310 timestamp.mPosition = mPausedPosition;
3311 clock_gettime(CLOCK_MONOTONIC, &timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003312 // TODO: adjust for delay
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003313 return NO_ERROR;
3314 }
3315
3316 // Check whether a pending flush or stop has completed, as those commands may
Andy Hungc8e09c62015-06-03 23:43:36 -07003317 // be asynchronous or return near finish or exhibit glitchy behavior.
3318 //
3319 // Originally this showed up as the first timestamp being a continuation of
3320 // the previous song under gapless playback.
3321 // However, we sometimes see zero timestamps, then a glitch of
3322 // the previous song's position, and then correct timestamps afterwards.
Andy Hungffa36952017-08-17 10:41:51 -07003323 if (mStartFromZeroUs != 0 && mSampleRate != 0) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003324 static const int kTimeJitterUs = 100000; // 100 ms
3325 static const int k1SecUs = 1000000;
3326
3327 const int64_t timeNow = getNowUs();
3328
Andy Hungffa36952017-08-17 10:41:51 -07003329 if (timeNow < mStartFromZeroUs + k1SecUs) { // within first second of starting
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003330 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003331 if (timestampTimeUs < mStartFromZeroUs) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003332 return WOULD_BLOCK; // stale timestamp time, occurs before start.
3333 }
Andy Hungffa36952017-08-17 10:41:51 -07003334 const int64_t deltaTimeUs = timestampTimeUs - mStartFromZeroUs;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003335 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07003336 / ((double)mSampleRate * mPlaybackRate.mSpeed);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003337
3338 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
3339 // Verify that the counter can't count faster than the sample rate
Andy Hungc8e09c62015-06-03 23:43:36 -07003340 // since the start time. If greater, then that means we may have failed
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003341 // to completely flush or stop the previous playing track.
Andy Hungc8e09c62015-06-03 23:43:36 -07003342 ALOGW_IF(!mTimestampStartupGlitchReported,
Andy Hungfb8ede22018-09-12 19:03:24 -07003343 "%s(%d): startup glitch detected"
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003344 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)",
Eric Laurent973db022018-11-20 14:54:31 -08003345 __func__, mPortId,
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003346 (long long)deltaTimeUs, (long long)deltaPositionByUs,
3347 timestamp.mPosition);
Andy Hungc8e09c62015-06-03 23:43:36 -07003348 mTimestampStartupGlitchReported = true;
3349 if (previousTimestampValid
3350 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) {
3351 timestamp = mPreviousTimestamp;
3352 mPreviousTimestampValid = true;
3353 return NO_ERROR;
3354 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003355 return WOULD_BLOCK;
3356 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003357 if (deltaPositionByUs != 0) {
Andy Hungffa36952017-08-17 10:41:51 -07003358 mStartFromZeroUs = 0; // don't check again, we got valid nonzero position.
Andy Hungc8e09c62015-06-03 23:43:36 -07003359 }
3360 } else {
Andy Hungffa36952017-08-17 10:41:51 -07003361 mStartFromZeroUs = 0; // don't check again, start time expired.
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003362 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003363 mTimestampStartupGlitchReported = false;
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003364 }
3365 } else {
Glenn Kasten200092b2014-08-15 15:13:30 -07003366 // Update the mapping between local consumed (mPosition) and server consumed (mServer)
3367 (void) updateAndGetPosition_l();
3368 // Server consumed (mServer) and presented both use the same server time base,
3369 // and server consumed is always >= presented.
3370 // The delta between these represents the number of frames in the buffer pipeline.
3371 // If this delta between these is greater than the client position, it means that
3372 // actually presented is still stuck at the starting line (figuratively speaking),
3373 // waiting for the first frame to go by. So we can't report a valid timestamp yet.
Andy Hung90e8a972015-11-09 16:42:40 -08003374 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when
3375 // mPosition exceeds 32 bits.
3376 // TODO Remove when timestamp is updated to contain pipeline status info.
3377 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue();
3378 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */
3379 && (uint32_t)pipelineDepthInFrames > mPosition.value()) {
Glenn Kasten200092b2014-08-15 15:13:30 -07003380 return INVALID_OPERATION;
3381 }
3382 // Convert timestamp position from server time base to client time base.
3383 // TODO The following code should work OK now because timestamp.mPosition is 32-bit.
3384 // But if we change it to 64-bit then this could fail.
Andy Hung90e8a972015-11-09 16:42:40 -08003385 // Use Modulo computation here.
3386 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value();
Glenn Kasten200092b2014-08-15 15:13:30 -07003387 // Immediately after a call to getPosition_l(), mPosition and
3388 // mServer both represent the same frame position. mPosition is
3389 // in client's point of view, and mServer is in server's point of
3390 // view. So the difference between them is the "fudge factor"
3391 // between client and server views due to stop() and/or new
3392 // IAudioTrack. And timestamp.mPosition is initially in server's
3393 // point of view, so we need to apply the same fudge factor to it.
Glenn Kastenfe346c72013-08-30 13:28:22 -07003394 }
Phil Burk1b420972015-04-22 10:52:21 -07003395
3396 // Prevent retrograde motion in timestamp.
3397 // This is sometimes caused by erratic reports of the available space in the ALSA drivers.
3398 if (status == NO_ERROR) {
Andy Hung3b8c6332019-04-03 19:29:36 -07003399 // Fix stale time when checking timestamp right after start().
3400 // The position is at the last reported location but the time can be stale
3401 // due to pause or standby or cold start latency.
3402 //
3403 // We keep advancing the time (but not the position) to ensure that the
3404 // stale value does not confuse the application.
3405 //
3406 // For offload compatibility, use a default lag value here.
3407 // Any time discrepancy between this update and the pause timestamp is handled
3408 // by the retrograde check afterwards.
3409 int64_t currentTimeNanos = audio_utils_ns_from_timespec(&timestamp.mTime);
3410 const int64_t lagNs = int64_t(mAfLatency * 1000000LL);
3411 const int64_t limitNs = mStartNs - lagNs;
3412 if (currentTimeNanos < limitNs) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003413 if (!mTimestampStaleTimeReported) {
3414 ALOGD("%s(%d): stale timestamp time corrected, "
3415 "currentTimeNanos: %lld < limitNs: %lld < mStartNs: %lld",
3416 __func__, mPortId,
3417 (long long)currentTimeNanos, (long long)limitNs, (long long)mStartNs);
3418 mTimestampStaleTimeReported = true;
3419 }
Andy Hung3b8c6332019-04-03 19:29:36 -07003420 timestamp.mTime = convertNsToTimespec(limitNs);
3421 currentTimeNanos = limitNs;
Andy Hungcf3b7152019-04-19 18:29:21 -07003422 } else {
3423 mTimestampStaleTimeReported = false;
Andy Hung3b8c6332019-04-03 19:29:36 -07003424 }
3425
Andy Hungffa36952017-08-17 10:41:51 -07003426 // previousTimestampValid is set to false when starting after a stop or flush.
Phil Burk1b420972015-04-22 10:52:21 -07003427 if (previousTimestampValid) {
Andy Hung2b01f002017-07-05 12:01:36 -07003428 const int64_t previousTimeNanos =
3429 audio_utils_ns_from_timespec(&mPreviousTimestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003430
3431 // retrograde check
Phil Burk1b420972015-04-22 10:52:21 -07003432 if (currentTimeNanos < previousTimeNanos) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003433 if (!mTimestampRetrogradeTimeReported) {
3434 ALOGW("%s(%d): retrograde timestamp time corrected, %lld < %lld",
3435 __func__, mPortId,
3436 (long long)currentTimeNanos, (long long)previousTimeNanos);
3437 mTimestampRetrogradeTimeReported = true;
3438 }
Andy Hung5d313802016-10-10 15:09:39 -07003439 timestamp.mTime = mPreviousTimestamp.mTime;
Andy Hungcf3b7152019-04-19 18:29:21 -07003440 } else {
3441 mTimestampRetrogradeTimeReported = false;
Phil Burk1b420972015-04-22 10:52:21 -07003442 }
3443
3444 // Looking at signed delta will work even when the timestamps
3445 // are wrapping around.
Andy Hung90e8a972015-11-09 16:42:40 -08003446 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition)
3447 - mPreviousTimestamp.mPosition).signedValue();
Phil Burk4c5a3672015-04-30 16:18:53 -07003448 if (deltaPosition < 0) {
3449 // Only report once per position instead of spamming the log.
Andy Hungcf3b7152019-04-19 18:29:21 -07003450 if (!mTimestampRetrogradePositionReported) {
Andy Hungfb8ede22018-09-12 19:03:24 -07003451 ALOGW("%s(%d): retrograde timestamp position corrected, %d = %u - %u",
Eric Laurent973db022018-11-20 14:54:31 -08003452 __func__, mPortId,
Phil Burk4c5a3672015-04-30 16:18:53 -07003453 deltaPosition,
3454 timestamp.mPosition,
3455 mPreviousTimestamp.mPosition);
Andy Hungcf3b7152019-04-19 18:29:21 -07003456 mTimestampRetrogradePositionReported = true;
Phil Burk4c5a3672015-04-30 16:18:53 -07003457 }
3458 } else {
Andy Hungcf3b7152019-04-19 18:29:21 -07003459 mTimestampRetrogradePositionReported = false;
Phil Burk4c5a3672015-04-30 16:18:53 -07003460 }
Andy Hung5d313802016-10-10 15:09:39 -07003461 if (deltaPosition < 0) {
3462 timestamp.mPosition = mPreviousTimestamp.mPosition;
3463 deltaPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07003464 }
Andy Hung5d313802016-10-10 15:09:39 -07003465#if 0
3466 // Uncomment this to verify audio timestamp rate.
3467 const int64_t deltaTime =
Andy Hung2b01f002017-07-05 12:01:36 -07003468 audio_utils_ns_from_timespec(&timestamp.mTime) - previousTimeNanos;
Andy Hung5d313802016-10-10 15:09:39 -07003469 if (deltaTime != 0) {
3470 const int64_t computedSampleRate =
3471 deltaPosition * (long long)NANOS_PER_SECOND / deltaTime;
Andy Hungfb8ede22018-09-12 19:03:24 -07003472 ALOGD("%s(%d): computedSampleRate:%u sampleRate:%u",
Eric Laurent973db022018-11-20 14:54:31 -08003473 __func__, mPortId,
Andy Hung5d313802016-10-10 15:09:39 -07003474 (unsigned)computedSampleRate, mSampleRate);
3475 }
3476#endif
Phil Burk1b420972015-04-22 10:52:21 -07003477 }
3478 mPreviousTimestamp = timestamp;
3479 mPreviousTimestampValid = true;
3480 }
3481
Glenn Kastenfe346c72013-08-30 13:28:22 -07003482 return status;
Glenn Kastence703742013-07-19 16:33:58 -07003483}
3484
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003485String8 AudioTrack::getParameters(const String8& keys)
3486{
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003487 audio_io_handle_t output = getOutput();
Glenn Kasten142f5192014-03-25 17:44:59 -07003488 if (output != AUDIO_IO_HANDLE_NONE) {
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003489 return AudioSystem::getParameters(output, keys);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01003490 } else {
3491 return String8::empty();
3492 }
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003493}
3494
Glenn Kasten23a75452014-01-13 10:37:17 -08003495bool AudioTrack::isOffloaded() const
3496{
3497 AutoMutex lock(mLock);
3498 return isOffloaded_l();
3499}
3500
Eric Laurentab5cdba2014-06-09 17:22:27 -07003501bool AudioTrack::isDirect() const
3502{
3503 AutoMutex lock(mLock);
3504 return isDirect_l();
3505}
3506
3507bool AudioTrack::isOffloadedOrDirect() const
3508{
3509 AutoMutex lock(mLock);
3510 return isOffloadedOrDirect_l();
3511}
3512
3513
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003514status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003515{
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003516 String8 result;
3517
3518 result.append(" AudioTrack::dump\n");
Andy Hungfb8ede22018-09-12 19:03:24 -07003519 result.appendFormat(" id(%d) status(%d), state(%d), session Id(%d), flags(%#x)\n",
Eric Laurent973db022018-11-20 14:54:31 -08003520 mPortId, mStatus, mState, mSessionId, mFlags);
Eric Laurentd114b622017-11-27 18:37:04 -08003521 result.appendFormat(" stream type(%d), left - right volume(%f, %f)\n",
François Gaffie58d4be52018-11-06 15:30:12 +01003522 mStreamType,
Eric Laurentd114b622017-11-27 18:37:04 -08003523 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
Glenn Kasten49f36ba2017-12-06 13:02:02 -08003524 result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
Eric Laurentd114b622017-11-27 18:37:04 -08003525 mFormat, mChannelMask, mChannelCount);
3526 result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
3527 mSampleRate, mOriginalSampleRate, mPlaybackRate.mSpeed);
3528 result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
3529 mFrameCount, mReqFrameCount);
3530 result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
3531 " req. notif. per buff(%u)\n",
3532 mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
3533 result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
3534 mLatency, mSelectedDeviceId, mRoutedDeviceId);
3535 result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
3536 mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003537 ::write(fd, result.string(), result.size());
3538 return NO_ERROR;
3539}
3540
Phil Burk2812d9e2016-01-04 10:34:30 -08003541uint32_t AudioTrack::getUnderrunCount() const
3542{
3543 AutoMutex lock(mLock);
3544 return getUnderrunCount_l();
3545}
3546
3547uint32_t AudioTrack::getUnderrunCount_l() const
3548{
3549 return mProxy->getUnderrunCount() + mUnderrunCountOffset;
3550}
3551
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003552uint32_t AudioTrack::getUnderrunFrames() const
3553{
3554 AutoMutex lock(mLock);
3555 return mProxy->getUnderrunFrames();
3556}
3557
Andy Hung3a5c2f32021-02-17 15:06:42 -08003558void AudioTrack::setLogSessionId(const char *logSessionId)
3559{
3560 AutoMutex lock(mLock);
Andy Hung1a9c21b2021-02-25 20:43:18 -08003561 if (logSessionId == nullptr) logSessionId = ""; // an empty string is an unset session id.
Andy Hung3a5c2f32021-02-17 15:06:42 -08003562 if (mLogSessionId == logSessionId) return;
3563
3564 mLogSessionId = logSessionId;
3565 mediametrics::LogItem(mMetricsId)
3566 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETLOGSESSIONID)
3567 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, logSessionId)
3568 .record();
3569}
3570
Andy Hung839a3062021-02-17 11:15:16 -08003571void AudioTrack::setPlayerIId(int playerIId)
3572{
3573 AutoMutex lock(mLock);
3574 if (mPlayerIId == playerIId) return;
3575
3576 mPlayerIId = playerIId;
3577 mediametrics::LogItem(mMetricsId)
3578 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYERIID)
3579 .set(AMEDIAMETRICS_PROP_PLAYERIID, playerIId)
3580 .record();
3581}
3582
Eric Laurent296fb132015-05-01 11:38:42 -07003583status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback)
3584{
Eric Laurent09f1ed22019-04-24 17:45:17 -07003585
Eric Laurent296fb132015-05-01 11:38:42 -07003586 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003587 ALOGW("%s(%d): adding NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003588 return BAD_VALUE;
3589 }
3590 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07003591 if (mDeviceCallback.unsafe_get() == callback.get()) {
Eric Laurent973db022018-11-20 14:54:31 -08003592 ALOGW("%s(%d): adding same callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003593 return INVALID_OPERATION;
3594 }
3595 status_t status = NO_ERROR;
3596 if (mOutput != AUDIO_IO_HANDLE_NONE) {
3597 if (mDeviceCallback != 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003598 ALOGW("%s(%d): callback already present!", __func__, mPortId);
Eric Laurent09f1ed22019-04-24 17:45:17 -07003599 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003600 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003601 status = AudioSystem::addAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003602 }
3603 mDeviceCallback = callback;
3604 return status;
3605}
3606
3607status_t AudioTrack::removeAudioDeviceCallback(
3608 const sp<AudioSystem::AudioDeviceCallback>& callback)
3609{
3610 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003611 ALOGW("%s(%d): removing NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003612 return BAD_VALUE;
3613 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003614 AutoMutex lock(mLock);
3615 if (mDeviceCallback.unsafe_get() != callback.get()) {
3616 ALOGW("%s removing different callback!", __FUNCTION__);
3617 return INVALID_OPERATION;
Eric Laurent296fb132015-05-01 11:38:42 -07003618 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003619 mDeviceCallback.clear();
Eric Laurent296fb132015-05-01 11:38:42 -07003620 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07003621 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003622 }
Eric Laurent296fb132015-05-01 11:38:42 -07003623 return NO_ERROR;
3624}
3625
Eric Laurentad2e7b92017-09-14 20:06:42 -07003626
3627void AudioTrack::onAudioDeviceUpdate(audio_io_handle_t audioIo,
3628 audio_port_handle_t deviceId)
3629{
3630 sp<AudioSystem::AudioDeviceCallback> callback;
3631 {
3632 AutoMutex lock(mLock);
3633 if (audioIo != mOutput) {
3634 return;
3635 }
3636 callback = mDeviceCallback.promote();
3637 // only update device if the track is active as route changes due to other use cases are
3638 // irrelevant for this client
3639 if (mState == STATE_ACTIVE) {
3640 mRoutedDeviceId = deviceId;
3641 }
3642 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003643
Eric Laurentad2e7b92017-09-14 20:06:42 -07003644 if (callback.get() != nullptr) {
3645 callback->onAudioDeviceUpdate(mOutput, mRoutedDeviceId);
3646 }
3647}
3648
Andy Hunge13f8a62016-03-30 14:20:42 -07003649status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location)
3650{
3651 if (msec == nullptr ||
3652 (location != ExtendedTimestamp::LOCATION_SERVER
3653 && location != ExtendedTimestamp::LOCATION_KERNEL)) {
3654 return BAD_VALUE;
3655 }
3656 AutoMutex lock(mLock);
3657 // inclusive of offloaded and direct tracks.
3658 //
3659 // It is possible, but not enabled, to allow duration computation for non-pcm
3660 // audio_has_proportional_frames() formats because currently they have
3661 // the drain rate equivalent to the pcm sample rate * framesize.
3662 if (!isPurePcmData_l()) {
3663 return INVALID_OPERATION;
3664 }
3665 ExtendedTimestamp ets;
3666 if (getTimestamp_l(&ets) == OK
3667 && ets.mTimeNs[location] > 0) {
3668 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
3669 - ets.mPosition[location];
3670 if (diff < 0) {
3671 *msec = 0;
3672 } else {
3673 // ms is the playback time by frames
3674 int64_t ms = (int64_t)((double)diff * 1000 /
3675 ((double)mSampleRate * mPlaybackRate.mSpeed));
3676 // clockdiff is the timestamp age (negative)
3677 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
3678 ets.mTimeNs[location]
3679 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
3680 - systemTime(SYSTEM_TIME_MONOTONIC);
3681
3682 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
3683 static const int NANOS_PER_MILLIS = 1000000;
3684 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
3685 }
3686 return NO_ERROR;
3687 }
3688 if (location != ExtendedTimestamp::LOCATION_SERVER) {
3689 return INVALID_OPERATION; // LOCATION_KERNEL is not available
3690 }
3691 // use server position directly (offloaded and direct arrive here)
3692 updateAndGetPosition_l();
3693 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue();
3694 *msec = (diff <= 0) ? 0
3695 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed));
3696 return NO_ERROR;
3697}
3698
Andy Hung65ffdfc2016-10-10 15:52:11 -07003699bool AudioTrack::hasStarted()
3700{
3701 AutoMutex lock(mLock);
3702 switch (mState) {
3703 case STATE_STOPPED:
3704 if (isOffloadedOrDirect_l()) {
3705 // check if we have started in the past to return true.
Andy Hungffa36952017-08-17 10:41:51 -07003706 return mStartFromZeroUs > 0;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003707 }
3708 // A normal audio track may still be draining, so
3709 // check if stream has ended. This covers fasttrack position
3710 // instability and start/stop without any data written.
3711 if (mProxy->getStreamEndDone()) {
3712 return true;
3713 }
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003714 FALLTHROUGH_INTENDED;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003715 case STATE_ACTIVE:
3716 case STATE_STOPPING:
3717 break;
3718 case STATE_PAUSED:
3719 case STATE_PAUSED_STOPPING:
3720 case STATE_FLUSHED:
3721 return false; // we're not active
3722 default:
Eric Laurent973db022018-11-20 14:54:31 -08003723 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in hasStarted(): %d", __func__, mPortId, mState);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003724 break;
3725 }
3726
3727 // wait indicates whether we need to wait for a timestamp.
3728 // This is conservatively figured - if we encounter an unexpected error
3729 // then we will not wait.
3730 bool wait = false;
3731 if (isOffloadedOrDirect_l()) {
3732 AudioTimestamp ts;
3733 status_t status = getTimestamp_l(ts);
3734 if (status == WOULD_BLOCK) {
3735 wait = true;
3736 } else if (status == OK) {
3737 wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition);
3738 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003739 ALOGV("%s(%d): hasStarted wait:%d ts:%u start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003740 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003741 (int)wait,
3742 ts.mPosition,
3743 (long long)mStartTs.mPosition);
3744 } else {
3745 int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG
3746 ExtendedTimestamp ets;
3747 status_t status = getTimestamp_l(&ets);
3748 if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets
3749 wait = true;
3750 } else if (status == OK) {
3751 for (location = ExtendedTimestamp::LOCATION_KERNEL;
3752 location >= ExtendedTimestamp::LOCATION_SERVER; --location) {
3753 if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) {
3754 continue;
3755 }
3756 wait = ets.mPosition[location] == 0
3757 || ets.mPosition[location] == mStartEts.mPosition[location];
3758 break;
3759 }
3760 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003761 ALOGV("%s(%d): hasStarted wait:%d ets:%lld start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003762 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003763 (int)wait,
3764 (long long)ets.mPosition[location],
3765 (long long)mStartEts.mPosition[location]);
3766 }
3767 return !wait;
3768}
3769
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003770// =========================================================================
3771
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003772void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003773{
3774 sp<AudioTrack> audioTrack = mAudioTrack.promote();
3775 if (audioTrack != 0) {
3776 AutoMutex lock(audioTrack->mLock);
3777 audioTrack->mProxy->binderDied();
3778 }
3779}
3780
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003781// =========================================================================
3782
Andy Hungca353672019-03-06 11:54:38 -08003783AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver)
Andy Hungeb46cf92019-03-06 10:13:38 -08003784 : Thread(true /* bCanCallJava */) // binder recursion on restoreTrack_l() may call Java.
3785 , mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
Glenn Kasten598de6c2013-10-16 17:02:13 -07003786 mIgnoreNextPausedInt(false)
Glenn Kasten3acbd052012-02-28 10:39:56 -08003787{
3788}
3789
3790AudioTrack::AudioTrackThread::~AudioTrackThread()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003791{
3792}
3793
3794bool AudioTrack::AudioTrackThread::threadLoop()
3795{
Glenn Kasten3acbd052012-02-28 10:39:56 -08003796 {
3797 AutoMutex _l(mMyLock);
3798 if (mPaused) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003799 // TODO check return value and handle or log
Glenn Kasten3acbd052012-02-28 10:39:56 -08003800 mMyCond.wait(mMyLock);
3801 // caller will check for exitPending()
3802 return true;
3803 }
Glenn Kasten598de6c2013-10-16 17:02:13 -07003804 if (mIgnoreNextPausedInt) {
3805 mIgnoreNextPausedInt = false;
3806 mPausedInt = false;
3807 }
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003808 if (mPausedInt) {
Glenn Kasten47d55172017-05-23 11:19:30 -07003809 // TODO use futex instead of condition, for event flag "or"
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003810 if (mPausedNs > 0) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003811 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003812 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
3813 } else {
Glenn Kasten75f79032017-05-23 11:22:44 -07003814 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003815 mMyCond.wait(mMyLock);
3816 }
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003817 mPausedInt = false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003818 return true;
3819 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08003820 }
Eric Laurent7985dcb2014-10-07 15:45:14 -07003821 if (exitPending()) {
3822 return false;
3823 }
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003824 nsecs_t ns = mReceiver.processAudioBuffer();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003825 switch (ns) {
3826 case 0:
3827 return true;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003828 case NS_INACTIVE:
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003829 pauseInternal();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003830 return true;
3831 case NS_NEVER:
3832 return false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003833 case NS_WHENEVER:
Andy Hung3c09c782014-12-29 18:39:32 -08003834 // Event driven: call wake() when callback notifications conditions change.
3835 ns = INT64_MAX;
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003836 FALLTHROUGH_INTENDED;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003837 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003838 LOG_ALWAYS_FATAL_IF(ns < 0, "%s(%d): processAudioBuffer() returned %lld",
Eric Laurent973db022018-11-20 14:54:31 -08003839 __func__, mReceiver.mPortId, (long long)ns);
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003840 pauseInternal(ns);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003841 return true;
Glenn Kastenca8b2802012-04-23 13:58:16 -07003842 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003843}
3844
Glenn Kasten3acbd052012-02-28 10:39:56 -08003845void AudioTrack::AudioTrackThread::requestExit()
3846{
3847 // must be in this order to avoid a race condition
3848 Thread::requestExit();
Glenn Kasten598de6c2013-10-16 17:02:13 -07003849 resume();
Glenn Kasten3acbd052012-02-28 10:39:56 -08003850}
3851
3852void AudioTrack::AudioTrackThread::pause()
3853{
3854 AutoMutex _l(mMyLock);
3855 mPaused = true;
3856}
3857
3858void AudioTrack::AudioTrackThread::resume()
3859{
3860 AutoMutex _l(mMyLock);
Glenn Kasten598de6c2013-10-16 17:02:13 -07003861 mIgnoreNextPausedInt = true;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003862 if (mPaused || mPausedInt) {
Glenn Kasten3acbd052012-02-28 10:39:56 -08003863 mPaused = false;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003864 mPausedInt = false;
Glenn Kasten3acbd052012-02-28 10:39:56 -08003865 mMyCond.signal();
3866 }
3867}
3868
Andy Hung3c09c782014-12-29 18:39:32 -08003869void AudioTrack::AudioTrackThread::wake()
3870{
3871 AutoMutex _l(mMyLock);
Andy Hunga8d08902015-07-22 11:52:13 -07003872 if (!mPaused) {
3873 // wake() might be called while servicing a callback - ignore the next
3874 // pause time and call processAudioBuffer.
Andy Hung3c09c782014-12-29 18:39:32 -08003875 mIgnoreNextPausedInt = true;
Andy Hunga8d08902015-07-22 11:52:13 -07003876 if (mPausedInt && mPausedNs > 0) {
3877 // audio track is active and internally paused with timeout.
3878 mPausedInt = false;
3879 mMyCond.signal();
3880 }
Andy Hung3c09c782014-12-29 18:39:32 -08003881 }
3882}
3883
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003884void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
3885{
3886 AutoMutex _l(mMyLock);
3887 mPausedInt = true;
3888 mPausedNs = ns;
3889}
3890
jiabinf6eb4c32020-02-25 14:06:25 -08003891binder::Status AudioTrack::AudioTrackCallback::onCodecFormatChanged(
3892 const std::vector<uint8_t>& audioMetadata)
3893{
3894 AutoMutex _l(mAudioTrackCbLock);
3895 sp<media::IAudioTrackCallback> callback = mCallback.promote();
3896 if (callback.get() != nullptr) {
3897 callback->onCodecFormatChanged(audioMetadata);
3898 } else {
3899 mCallback.clear();
3900 }
3901 return binder::Status::ok();
3902}
3903
3904void AudioTrack::AudioTrackCallback::setAudioTrackCallback(
3905 const sp<media::IAudioTrackCallback> &callback) {
3906 AutoMutex lock(mAudioTrackCbLock);
3907 mCallback = callback;
3908}
3909
Glenn Kasten40bc9062015-03-20 09:09:33 -07003910} // namespace android