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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 Hung2b01f002017-07-05 12:01:36 -070028#include <audio_utils/clock.h>
Glenn Kasten9f80dd22012-12-18 15:57:32 -080029#include <audio_utils/primitives.h>
30#include <binder/IPCThreadState.h>
31#include <media/AudioTrack.h>
32#include <utils/Log.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080033#include <private/media/AudioTrackShared.h>
Suren Baghdasaryan7435e7d2018-12-19 17:09:28 -080034#include <processgroup/sched_policy.h>
Glenn Kasten1ab85ec2013-05-31 09:18:43 -070035#include <media/IAudioFlinger.h>
Dean Wheatleya70eef72018-01-04 14:23:50 +110036#include <media/AudioParameter.h>
Andy Hungcd044842014-08-07 11:04:34 -070037#include <media/AudioResamplerPublic.h>
Michael Chana94fbb22018-04-24 14:31:19 +100038#include <media/AudioSystem.h>
Ray Essickf27e9872019-12-07 06:28:46 -080039#include <media/MediaMetricsItem.h>
Ray Essicked304702017-12-12 14:00:57 -080040#include <media/TypeConverter.h>
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080041
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +010042#define WAIT_PERIOD_MS 10
43#define WAIT_STREAM_END_TIMEOUT_SEC 120
Andy Hung53c3b5f2014-12-15 16:42:05 -080044static const int kMaxLoopCountNotifications = 32;
Glenn Kasten511754b2012-01-11 09:52:19 -080045
Kuowei Lid4adbdb2020-08-13 14:44:25 +080046using ::android::aidl_utils::statusTFromBinderStatus;
47
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -080048namespace android {
Chia-chi Yeh33005a92010-06-16 06:33:13 +080049// ---------------------------------------------------------------------------
50
Ivan Lozano8cf3a072017-08-09 09:01:33 -070051using media::VolumeShaper;
Svet Ganov3e5f14f2021-05-13 22:51:08 +000052using android::content::AttributionSourceState;
Ivan Lozano8cf3a072017-08-09 09:01:33 -070053
Andy Hunga7f03352015-05-31 21:54:49 -070054// TODO: Move to a separate .h
55
Andy Hung4ede21d2014-12-12 15:37:34 -080056template <typename T>
Andy Hunga7f03352015-05-31 21:54:49 -070057static inline const T &min(const T &x, const T &y) {
Andy Hung4ede21d2014-12-12 15:37:34 -080058 return x < y ? x : y;
59}
60
Andy Hunga7f03352015-05-31 21:54:49 -070061template <typename T>
62static inline const T &max(const T &x, const T &y) {
63 return x > y ? x : y;
64}
65
66static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed)
67{
68 return ((double)frames * 1000000000) / ((double)sampleRate * speed);
69}
70
Andy Hung7f1bc8a2014-09-12 14:43:11 -070071static int64_t convertTimespecToUs(const struct timespec &tv)
72{
Chih-Hung Hsieh50fa06c2018-12-11 13:51:36 -080073 return tv.tv_sec * 1000000LL + tv.tv_nsec / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -070074}
75
Andy Hungffa36952017-08-17 10:41:51 -070076// TODO move to audio_utils.
77static inline struct timespec convertNsToTimespec(int64_t ns) {
78 struct timespec tv;
79 tv.tv_sec = static_cast<time_t>(ns / NANOS_PER_SECOND);
Andy Hung06a730b2020-04-09 13:28:31 -070080 tv.tv_nsec = static_cast<int64_t>(ns % NANOS_PER_SECOND);
Andy Hungffa36952017-08-17 10:41:51 -070081 return tv;
82}
83
Andy Hung7f1bc8a2014-09-12 14:43:11 -070084// current monotonic time in microseconds.
85static int64_t getNowUs()
86{
87 struct timespec tv;
88 (void) clock_gettime(CLOCK_MONOTONIC, &tv);
89 return convertTimespecToUs(tv);
90}
91
Andy Hung26145642015-04-15 21:56:53 -070092// FIXME: we don't use the pitch setting in the time stretcher (not working);
93// instead we emulate it using our sample rate converter.
94static const bool kFixPitch = true; // enable pitch fix
95static inline uint32_t adjustSampleRate(uint32_t sampleRate, float pitch)
96{
97 return kFixPitch ? (sampleRate * pitch + 0.5) : sampleRate;
98}
99
100static inline float adjustSpeed(float speed, float pitch)
101{
Ricardo Garcia6c7f0622015-04-30 18:39:16 -0700102 return kFixPitch ? speed / max(pitch, AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) : speed;
Andy Hung26145642015-04-15 21:56:53 -0700103}
104
105static inline float adjustPitch(float pitch)
106{
107 return kFixPitch ? AUDIO_TIMESTRETCH_PITCH_NORMAL : pitch;
108}
109
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800110// static
111status_t AudioTrack::getMinFrameCount(
Glenn Kastene33054e2012-11-14 12:54:39 -0800112 size_t* frameCount,
Glenn Kastenfff6d712012-01-12 16:38:12 -0800113 audio_stream_type_t streamType,
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800114 uint32_t sampleRate)
115{
Glenn Kastend65d73c2012-06-22 17:21:07 -0700116 if (frameCount == NULL) {
117 return BAD_VALUE;
118 }
Glenn Kasten04cd0182012-06-25 11:49:27 -0700119
Andy Hung0e48d252015-01-26 11:43:15 -0800120 // FIXME handle in server, like createTrack_l(), possible missing info:
Glenn Kastene0fa4672012-04-24 14:35:14 -0700121 // audio_io_handle_t output
122 // audio_format_t format
123 // audio_channel_mask_t channelMask
Andy Hung0e48d252015-01-26 11:43:15 -0800124 // audio_output_flags_t flags (FAST)
Glenn Kasten3b16c762012-11-14 08:44:39 -0800125 uint32_t afSampleRate;
Glenn Kasten66a04672014-01-08 08:53:44 -0800126 status_t status;
127 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType);
128 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700129 ALOGE("%s(): Unable to query output sample rate for stream type %d; status %d",
130 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800131 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800132 }
Glenn Kastene33054e2012-11-14 12:54:39 -0800133 size_t afFrameCount;
Glenn Kasten66a04672014-01-08 08:53:44 -0800134 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType);
135 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700136 ALOGE("%s(): Unable to query output frame count for stream type %d; status %d",
137 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800138 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800139 }
140 uint32_t afLatency;
Glenn Kasten66a04672014-01-08 08:53:44 -0800141 status = AudioSystem::getOutputLatency(&afLatency, streamType);
142 if (status != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700143 ALOGE("%s(): Unable to query output latency for stream type %d; status %d",
144 __func__, streamType, status);
Glenn Kasten66a04672014-01-08 08:53:44 -0800145 return status;
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800146 }
147
Andy Hung8edb8dc2015-03-26 19:13:55 -0700148 // When called from createTrack, speed is 1.0f (normal speed).
149 // This is rechecked again on setting playback rate (TODO: on setting sample rate, too).
Eric Laurent21da6472017-11-09 16:29:26 -0800150 *frameCount = AudioSystem::calculateMinFrameCount(afLatency, afFrameCount, afSampleRate,
151 sampleRate, 1.0f /*, 0 notificationsPerBufferReq*/);
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800152
Andy Hung0e48d252015-01-26 11:43:15 -0800153 // The formula above should always produce a non-zero value under normal circumstances:
154 // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX.
155 // 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 -0800156 if (*frameCount == 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700157 ALOGE("%s(): failed for streamType %d, sampleRate %u",
158 __func__, streamType, sampleRate);
Glenn Kasten66a04672014-01-08 08:53:44 -0800159 return BAD_VALUE;
160 }
Andy Hungfb8ede22018-09-12 19:03:24 -0700161 ALOGV("%s(): getMinFrameCount=%zu: afFrameCount=%zu, afSampleRate=%u, afLatency=%u",
162 __func__, *frameCount, afFrameCount, afSampleRate, afLatency);
Chia-chi Yeh33005a92010-06-16 06:33:13 +0800163 return NO_ERROR;
164}
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800165
Michael Chana94fbb22018-04-24 14:31:19 +1000166// static
167bool AudioTrack::isDirectOutputSupported(const audio_config_base_t& config,
168 const audio_attributes_t& attributes) {
169 ALOGV("%s()", __FUNCTION__);
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800170 const sp<media::IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
Michael Chana94fbb22018-04-24 14:31:19 +1000171 if (aps == 0) return false;
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800172
173 auto result = [&]() -> ConversionResult<bool> {
Mikhail Naganovdbf03642021-08-25 18:15:32 -0700174 media::audio::common::AudioConfigBase configAidl = VALUE_OR_RETURN(
Mikhail Naganov9dec7012021-07-21 10:30:57 -0700175 legacy2aidl_audio_config_base_t_AudioConfigBase(config, false /*isInput*/));
Ytai Ben-Tsvi0a4904a2021-01-06 12:57:05 -0800176 media::AudioAttributesInternal attributesAidl = VALUE_OR_RETURN(
177 legacy2aidl_audio_attributes_t_AudioAttributesInternal(attributes));
178 bool retAidl;
179 RETURN_IF_ERROR(aidl_utils::statusTFromBinderStatus(
180 aps->isDirectOutputSupported(configAidl, attributesAidl, &retAidl)));
181 return retAidl;
182 }();
183 return result.value_or(false);
Michael Chana94fbb22018-04-24 14:31:19 +1000184}
185
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800186// ---------------------------------------------------------------------------
187
Ray Essicked304702017-12-12 14:00:57 -0800188void AudioTrack::MediaMetrics::gather(const AudioTrack *track)
189{
Ray Essick88394302018-01-24 14:52:05 -0800190 // only if we're in a good state...
191 // XXX: shall we gather alternative info if failing?
192 const status_t lstatus = track->initCheck();
193 if (lstatus != NO_ERROR) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700194 ALOGD("%s(): no metrics gathered, track status=%d", __func__, (int) lstatus);
Ray Essick88394302018-01-24 14:52:05 -0800195 return;
196 }
197
Andy Hungd0979812019-02-21 15:51:44 -0800198#define MM_PREFIX "android.media.audiotrack." // avoid cut-n-paste errors.
Ray Essicked304702017-12-12 14:00:57 -0800199
Andy Hungd0979812019-02-21 15:51:44 -0800200 // Java API 28 entries, do not change.
Ray Essickf27e9872019-12-07 06:28:46 -0800201 mMetricsItem->setCString(MM_PREFIX "streamtype", toString(track->streamType()).c_str());
202 mMetricsItem->setCString(MM_PREFIX "type",
Andy Hungd0979812019-02-21 15:51:44 -0800203 toString(track->mAttributes.content_type).c_str());
Ray Essickf27e9872019-12-07 06:28:46 -0800204 mMetricsItem->setCString(MM_PREFIX "usage", toString(track->mAttributes.usage).c_str());
Ray Essicked304702017-12-12 14:00:57 -0800205
Andy Hungd0979812019-02-21 15:51:44 -0800206 // Non-API entries, these can change due to a Java string mistake.
Ray Essickf27e9872019-12-07 06:28:46 -0800207 mMetricsItem->setInt32(MM_PREFIX "sampleRate", (int32_t)track->mSampleRate);
208 mMetricsItem->setInt64(MM_PREFIX "channelMask", (int64_t)track->mChannelMask);
Andy Hungd0979812019-02-21 15:51:44 -0800209 // Non-API entries, these can change.
Ray Essickf27e9872019-12-07 06:28:46 -0800210 mMetricsItem->setInt32(MM_PREFIX "portId", (int32_t)track->mPortId);
211 mMetricsItem->setCString(MM_PREFIX "encoding", toString(track->mFormat).c_str());
212 mMetricsItem->setInt32(MM_PREFIX "frameCount", (int32_t)track->mFrameCount);
213 mMetricsItem->setCString(MM_PREFIX "attributes", toString(track->mAttributes).c_str());
Andy Hung6f451f02021-02-24 21:53:29 -0800214 mMetricsItem->setCString(MM_PREFIX "logSessionId", track->mLogSessionId.c_str());
Ray Essicked304702017-12-12 14:00:57 -0800215}
216
Ray Essick88394302018-01-24 14:52:05 -0800217// hand the user a snapshot of the metrics.
Ray Essickf27e9872019-12-07 06:28:46 -0800218status_t AudioTrack::getMetrics(mediametrics::Item * &item)
Ray Essick88394302018-01-24 14:52:05 -0800219{
220 mMediaMetrics.gather(this);
Ray Essickf27e9872019-12-07 06:28:46 -0800221 mediametrics::Item *tmp = mMediaMetrics.dup();
Ray Essick88394302018-01-24 14:52:05 -0800222 if (tmp == nullptr) {
223 return BAD_VALUE;
224 }
225 item = tmp;
226 return NO_ERROR;
227}
Ray Essicked304702017-12-12 14:00:57 -0800228
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000229AudioTrack::AudioTrack() : AudioTrack(AttributionSourceState())
Colin Crossb8a9dbb2020-08-27 04:12:26 +0000230{
231}
232
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000233AudioTrack::AudioTrack(const AttributionSourceState& attributionSource)
Glenn Kasten87913512011-06-22 16:15:25 -0700234 : mStatus(NO_INIT),
Haynes Mathew George9b3359f2016-03-23 19:09:10 -0700235 mState(STATE_STOPPED),
John Grossman4ff14ba2012-02-08 16:37:41 -0800236 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800237 mPreviousSchedulingGroup(SP_DEFAULT),
Paul McLeanaa981192015-03-21 09:55:15 -0700238 mPausedPosition(0),
Eric Laurent20b9ef02016-12-05 11:03:16 -0800239 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
jiabinf6eb4c32020-02-25 14:06:25 -0800240 mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000241 mClientAttributionSource(attributionSource),
jiabinf6eb4c32020-02-25 14:06:25 -0800242 mAudioTrackCallback(new AudioTrackCallback())
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800243{
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700244 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
245 mAttributes.usage = AUDIO_USAGE_UNKNOWN;
Mikhail Naganov55773032020-10-01 15:08:13 -0700246 mAttributes.flags = AUDIO_FLAG_NONE;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700247 strcpy(mAttributes.tags, "");
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800248}
249
250AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800251 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800252 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800253 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700254 audio_channel_mask_t channelMask,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800255 size_t frameCount,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700256 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400257 const wp<IAudioTrackCallback> & callback,
Atneyaf86d2692021-10-14 14:02:36 -0400258 int32_t notificationFrames,
259 audio_session_t sessionId,
260 transfer_type transferType,
261 const audio_offload_info_t *offloadInfo,
262 const AttributionSourceState& attributionSource,
263 const audio_attributes_t* pAttributes,
264 bool doNotReconnect,
265 float maxRequiredSpeed,
266 audio_port_handle_t selectedDeviceId)
267 : mStatus(NO_INIT),
268 mState(STATE_STOPPED),
269 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
270 mPreviousSchedulingGroup(SP_DEFAULT),
271 mPausedPosition(0),
272 mAudioTrackCallback(new AudioTrackCallback())
273{
274 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Daniel Chapinf13b98a2021-10-25 21:58:31 +0000275
Atneyaf86d2692021-10-14 14:02:36 -0400276 (void)set(streamType, sampleRate, format, channelMask,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400277 frameCount, flags, callback, notificationFrames,
278 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
279 attributionSource, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
280}
281
282namespace {
283 class LegacyCallbackWrapper : public AudioTrack::IAudioTrackCallback {
284 const AudioTrack::legacy_callback_t mCallback;
285 void * const mData;
286 public:
287 LegacyCallbackWrapper(AudioTrack::legacy_callback_t callback, void* user)
288 : mCallback(callback), mData(user) {}
289 size_t onMoreData(const AudioTrack::Buffer & buffer) override {
290 AudioTrack::Buffer copy = buffer;
291 mCallback(AudioTrack::EVENT_MORE_DATA, mData, static_cast<void*>(&copy));
292 return copy.size;
293 }
294 void onUnderrun() override {
295 mCallback(AudioTrack::EVENT_UNDERRUN, mData, nullptr);
296 }
297 void onLoopEnd(int32_t loopsRemaining) override {
298 mCallback(AudioTrack::EVENT_LOOP_END, mData, &loopsRemaining);
299 }
300 void onMarker(uint32_t markerPosition) override {
301 mCallback(AudioTrack::EVENT_MARKER, mData, &markerPosition);
302 }
303 void onNewPos(uint32_t newPos) override {
304 mCallback(AudioTrack::EVENT_NEW_POS, mData, &newPos);
305 }
306 void onBufferEnd() override {
307 mCallback(AudioTrack::EVENT_BUFFER_END, mData, nullptr);
308 }
309 void onNewIAudioTrack() override {
310 mCallback(AudioTrack::EVENT_NEW_IAUDIOTRACK, mData, nullptr);
311 }
312 void onStreamEnd() override {
313 mCallback(AudioTrack::EVENT_STREAM_END, mData, nullptr);
314 }
315 size_t onCanWriteMoreData(const AudioTrack::Buffer & buffer) override {
316 AudioTrack::Buffer copy = buffer;
317 mCallback(AudioTrack::EVENT_CAN_WRITE_MORE_DATA, mData, static_cast<void*>(&copy));
318 return copy.size;
319 }
320 };
321}
322
323AudioTrack::AudioTrack(
324 audio_stream_type_t streamType,
325 uint32_t sampleRate,
326 audio_format_t format,
327 audio_channel_mask_t channelMask,
328 size_t frameCount,
329 audio_output_flags_t flags,
330 legacy_callback_t callback,
331 void* user,
332 int32_t notificationFrames,
333 audio_session_t sessionId,
334 transfer_type transferType,
335 const audio_offload_info_t *offloadInfo,
336 const AttributionSourceState& attributionSource,
337 const audio_attributes_t* pAttributes,
338 bool doNotReconnect,
339 float maxRequiredSpeed,
340 audio_port_handle_t selectedDeviceId)
341 : mStatus(NO_INIT),
342 mState(STATE_STOPPED),
343 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
344 mPreviousSchedulingGroup(SP_DEFAULT),
345 mPausedPosition(0),
346 mAudioTrackCallback(new AudioTrackCallback())
347{
348 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
349 if (callback != nullptr) {
350 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
351 } else if (user) {
352 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
353 }
354 (void)set(streamType, sampleRate, format, channelMask,
355 frameCount, flags, mLegacyCallbackWrapper, notificationFrames,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000356 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
357 attributionSource, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800358}
359
Andreas Huberc8139852012-01-18 10:51:55 -0800360AudioTrack::AudioTrack(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800361 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800362 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800363 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700364 audio_channel_mask_t channelMask,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800365 const sp<IMemory>& sharedBuffer,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700366 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400367 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700368 int32_t notificationFrames,
Glenn Kastend848eb42016-03-08 13:42:11 -0800369 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000370 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800371 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000372 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700373 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700374 bool doNotReconnect,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700375 float maxRequiredSpeed)
Glenn Kasten87913512011-06-22 16:15:25 -0700376 : mStatus(NO_INIT),
Haynes Mathew George9b3359f2016-03-23 19:09:10 -0700377 mState(STATE_STOPPED),
John Grossman4ff14ba2012-02-08 16:37:41 -0800378 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
Haynes Mathew George7064fd22014-01-08 13:59:53 -0800379 mPreviousSchedulingGroup(SP_DEFAULT),
Paul McLeanaa981192015-03-21 09:55:15 -0700380 mPausedPosition(0),
jiabinf6eb4c32020-02-25 14:06:25 -0800381 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
382 mAudioTrackCallback(new AudioTrackCallback())
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800383{
François Gaffie393f0e02019-04-10 09:09:08 +0200384 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
Ippei Murata5fa32ed2018-10-01 17:37:50 +0900385
Eric Laurentf32d7812017-11-30 14:44:07 -0800386 (void)set(streamType, sampleRate, format, channelMask,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400387 0 /*frameCount*/, flags, callback, notificationFrames,
Marco Nelissend457c972014-02-11 08:47:07 -0800388 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000389 attributionSource, pAttributes, doNotReconnect, maxRequiredSpeed);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800390}
391
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400392AudioTrack::AudioTrack(
393 audio_stream_type_t streamType,
394 uint32_t sampleRate,
395 audio_format_t format,
396 audio_channel_mask_t channelMask,
397 const sp<IMemory>& sharedBuffer,
398 audio_output_flags_t flags,
399 legacy_callback_t callback,
400 void* user,
401 int32_t notificationFrames,
402 audio_session_t sessionId,
403 transfer_type transferType,
404 const audio_offload_info_t *offloadInfo,
405 const AttributionSourceState& attributionSource,
406 const audio_attributes_t* pAttributes,
407 bool doNotReconnect,
408 float maxRequiredSpeed)
409 : mStatus(NO_INIT),
410 mState(STATE_STOPPED),
411 mPreviousPriority(ANDROID_PRIORITY_NORMAL),
412 mPreviousSchedulingGroup(SP_DEFAULT),
413 mPausedPosition(0),
414 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
415 mAudioTrackCallback(new AudioTrackCallback())
416{
417 mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
418 if (callback) {
419 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
420 } else if (user) {
421 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
422 }
423
424 (void)set(streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags,
425 mLegacyCallbackWrapper, notificationFrames, sharedBuffer,
426 false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, attributionSource,
427 pAttributes, doNotReconnect, maxRequiredSpeed);
428}
429
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800430AudioTrack::~AudioTrack()
431{
Ray Essicked304702017-12-12 14:00:57 -0800432 // pull together the numbers, before we clean up our structures
433 mMediaMetrics.gather(this);
434
Andy Hungb68f5eb2019-12-03 16:49:17 -0800435 mediametrics::LogItem(mMetricsId)
436 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DTOR)
Phil Burkd3813f32020-04-23 16:26:15 -0700437 .set(AMEDIAMETRICS_PROP_CALLERNAME,
438 mCallerName.empty()
Andy Hunga6b27032020-04-27 10:34:24 -0700439 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
Phil Burkd3813f32020-04-23 16:26:15 -0700440 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800441 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
442 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)mStatus)
443 .record();
444
Phil Burk7a9577c2021-03-12 20:12:11 +0000445 stopAndJoinCallbacks(); // checks mStatus
446
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800447 if (mStatus == NO_ERROR) {
Marco Nelissenf8880202014-11-14 07:58:25 -0800448 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten53cec222013-08-29 09:01:02 -0700449 mAudioTrack.clear();
Eric Laurent3bcffa12014-06-12 18:38:45 -0700450 mCblkMemory.clear();
451 mSharedBuffer.clear();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800452 IPCThreadState::self()->flushCommands();
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000453 pid_t clientPid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
Andy Hungfb8ede22018-09-12 19:03:24 -0700454 ALOGV("%s(%d), releasing session id %d from %d on behalf of %d",
Eric Laurent973db022018-11-20 14:54:31 -0800455 __func__, mPortId,
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700456 mSessionId, IPCThreadState::self()->getCallingPid(), clientPid);
457 AudioSystem::releaseAudioSessionId(mSessionId, clientPid);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800458 }
459}
460
Phil Burk7a9577c2021-03-12 20:12:11 +0000461void AudioTrack::stopAndJoinCallbacks() {
462 // Prevent nullptr crash if it did not open properly.
463 if (mStatus != NO_ERROR) return;
464
465 // Make sure that callback function exits in the case where
466 // it is looping on buffer full condition in obtainBuffer().
467 // Otherwise the callback thread will never exit.
468 stop();
469 if (mAudioTrackThread != 0) { // not thread safe
470 mProxy->interrupt();
471 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
472 mAudioTrackThread->requestExitAndWait();
473 mAudioTrackThread.clear();
474 }
475 // No lock here: worst case we remove a NULL callback which will be a nop
476 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) {
477 // This may not stop all of these device callbacks!
478 // TODO: Add some sort of protection.
479 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
480 mDeviceCallback.clear();
481 }
482}
483
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800484status_t AudioTrack::set(
Glenn Kastenfff6d712012-01-12 16:38:12 -0800485 audio_stream_type_t streamType,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800486 uint32_t sampleRate,
Glenn Kastene1c39622012-01-04 09:36:37 -0800487 audio_format_t format,
Glenn Kasten28b76b32012-07-03 17:24:41 -0700488 audio_channel_mask_t channelMask,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800489 size_t frameCount,
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700490 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400491 legacy_callback_t callback,
492 void * user,
493 int32_t notificationFrames,
494 const sp<IMemory>& sharedBuffer,
495 bool threadCanCallJava,
496 audio_session_t sessionId,
497 transfer_type transferType,
498 const audio_offload_info_t *offloadInfo,
499 const AttributionSourceState& attributionSource,
500 const audio_attributes_t* pAttributes,
501 bool doNotReconnect,
502 float maxRequiredSpeed,
503 audio_port_handle_t selectedDeviceId)
504{
505 if (callback) {
506 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
507 } else if (user) {
508 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
509 }
510 return set(streamType, sampleRate,format, channelMask, frameCount, flags,
511 mLegacyCallbackWrapper, notificationFrames, sharedBuffer, threadCanCallJava,
512 sessionId, transferType, offloadInfo, attributionSource, pAttributes,
513 doNotReconnect, maxRequiredSpeed, selectedDeviceId);
514}
515status_t AudioTrack::set(
516 audio_stream_type_t streamType,
517 uint32_t sampleRate,
518 audio_format_t format,
519 audio_channel_mask_t channelMask,
520 size_t frameCount,
521 audio_output_flags_t flags,
522 const wp<IAudioTrackCallback>& callback,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700523 int32_t notificationFrames,
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800524 const sp<IMemory>& sharedBuffer,
Eric Laurentbe916aa2010-06-01 23:49:17 -0700525 bool threadCanCallJava,
Glenn Kastend848eb42016-03-08 13:42:11 -0800526 audio_session_t sessionId,
Richard Fitzgeraldad3af332013-03-25 16:54:37 +0000527 transfer_type transferType,
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800528 const audio_offload_info_t *offloadInfo,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000529 const AttributionSourceState& attributionSource,
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700530 const audio_attributes_t* pAttributes,
Andy Hungff874dc2016-04-11 16:49:09 -0700531 bool doNotReconnect,
jiabin156c6872017-10-06 09:47:15 -0700532 float maxRequiredSpeed,
533 audio_port_handle_t selectedDeviceId)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800534{
Eric Laurentf32d7812017-11-30 14:44:07 -0800535 status_t status;
536 uint32_t channelCount;
537 pid_t callingPid;
538 pid_t myPid;
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000539 uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
540 pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400541 sp<IAudioTrackCallback> _callback = callback.promote();
Eric Laurent973db022018-11-20 14:54:31 -0800542 // Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
Andy Hungfb8ede22018-09-12 19:03:24 -0700543 ALOGV("%s(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
Glenn Kastenea38ee72016-04-18 11:08:01 -0700544 "flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
Andy Hungfb8ede22018-09-12 19:03:24 -0700545 __func__,
Glenn Kastenbce50bf2014-02-27 15:29:51 -0800546 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000547 sessionId, transferType, attributionSource.uid, attributionSource.pid);
Glenn Kasten86f04662014-02-24 15:13:05 -0800548
Phil Burk33ff89b2015-11-30 11:16:01 -0800549 mThreadCanCallJava = threadCanCallJava;
jiabin156c6872017-10-06 09:47:15 -0700550 mSelectedDeviceId = selectedDeviceId;
Eric Laurent21da6472017-11-09 16:29:26 -0800551 mSessionId = sessionId;
Phil Burk33ff89b2015-11-30 11:16:01 -0800552
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800553 switch (transferType) {
554 case TRANSFER_DEFAULT:
555 if (sharedBuffer != 0) {
556 transferType = TRANSFER_SHARED;
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400557 } else if (_callback == nullptr|| threadCanCallJava) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800558 transferType = TRANSFER_SYNC;
559 } else {
560 transferType = TRANSFER_CALLBACK;
561 }
562 break;
563 case TRANSFER_CALLBACK:
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700564 case TRANSFER_SYNC_NOTIF_CALLBACK:
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400565 if (_callback == nullptr || sharedBuffer != 0) {
566 ALOGE("%s(): Transfer type %s but callback == nullptr || sharedBuffer != 0",
Jean-Michel Trivif4158d82018-09-25 12:43:58 -0700567 convertTransferToText(transferType), __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800568 status = BAD_VALUE;
569 goto exit;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800570 }
571 break;
572 case TRANSFER_OBTAIN:
573 case TRANSFER_SYNC:
574 if (sharedBuffer != 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700575 ALOGE("%s(): Transfer type TRANSFER_OBTAIN but sharedBuffer != 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800576 status = BAD_VALUE;
577 goto exit;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800578 }
579 break;
580 case TRANSFER_SHARED:
581 if (sharedBuffer == 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700582 ALOGE("%s(): Transfer type TRANSFER_SHARED but sharedBuffer == 0", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800583 status = BAD_VALUE;
584 goto exit;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800585 }
586 break;
587 default:
Andy Hungfb8ede22018-09-12 19:03:24 -0700588 ALOGE("%s(): Invalid transfer type %d",
589 __func__, transferType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800590 status = BAD_VALUE;
591 goto exit;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800592 }
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800593 mSharedBuffer = sharedBuffer;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800594 mTransfer = transferType;
Ronghua Wufaeb0f22015-05-21 12:20:21 -0700595 mDoNotReconnect = doNotReconnect;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800596
Andy Hungfb8ede22018-09-12 19:03:24 -0700597 ALOGV_IF(sharedBuffer != 0, "%s(): sharedBuffer: %p, size: %zu",
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -0700598 __func__, sharedBuffer->unsecurePointer(), sharedBuffer->size());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800599
Andy Hungfb8ede22018-09-12 19:03:24 -0700600 ALOGV("%s(): streamType %d frameCount %zu flags %04x",
601 __func__, streamType, frameCount, flags);
Eric Laurent1a9ed112012-03-20 18:36:01 -0700602
Glenn Kasten53cec222013-08-29 09:01:02 -0700603 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Eric Laurent1dd70b92009-04-21 07:56:33 -0700604 if (mAudioTrack != 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700605 ALOGE("%s(): Track already in use", __func__);
Eric Laurentf32d7812017-11-30 14:44:07 -0800606 status = INVALID_OPERATION;
607 goto exit;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800608 }
609
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800610 // handle default values first.
Eric Laurente83b55d2014-11-14 10:06:21 -0800611 if (streamType == AUDIO_STREAM_DEFAULT) {
Dima Zavinfce7a472011-04-19 22:30:36 -0700612 streamType = AUDIO_STREAM_MUSIC;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800613 }
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700614 if (pAttributes == NULL) {
Eric Laurent223fd5c2014-11-11 13:43:36 -0800615 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700616 ALOGE("%s(): Invalid stream type %d", __func__, streamType);
Eric Laurentf32d7812017-11-30 14:44:07 -0800617 status = BAD_VALUE;
618 goto exit;
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700619 }
Andy Hunga2159aa2021-07-20 13:01:52 -0700620 mOriginalStreamType = streamType;
Eric Laurente83b55d2014-11-14 10:06:21 -0800621
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700622 } else {
Jean-Michel Trivifaabb512014-06-11 16:55:06 -0700623 // stream type shouldn't be looked at, this track has audio attributes
624 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
Andy Hungfb8ede22018-09-12 19:03:24 -0700625 ALOGV("%s(): Building AudioTrack with attributes:"
626 " usage=%d content=%d flags=0x%x tags=[%s]",
627 __func__,
628 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
Andy Hunga2159aa2021-07-20 13:01:52 -0700629 mOriginalStreamType = AUDIO_STREAM_DEFAULT;
François Gaffie58d4be52018-11-06 15:30:12 +0100630 audio_flags_to_audio_output_flags(mAttributes.flags, &flags);
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800631 }
Glenn Kastenea7939a2012-03-14 12:56:26 -0700632
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800633 // these below should probably come from the audioFlinger too...
Glenn Kastene1c39622012-01-04 09:36:37 -0800634 if (format == AUDIO_FORMAT_DEFAULT) {
Dima Zavinfce7a472011-04-19 22:30:36 -0700635 format = AUDIO_FORMAT_PCM_16_BIT;
Phil Burkfdb3c072016-02-09 10:47:02 -0800636 } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
Mikhail Naganov55773032020-10-01 15:08:13 -0700637 flags = static_cast<audio_output_flags_t>(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800638 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800639
640 // validate parameters
Dima Zavinfce7a472011-04-19 22:30:36 -0700641 if (!audio_is_valid_format(format)) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700642 ALOGE("%s(): Invalid format %#x", __func__, format);
Eric Laurentf32d7812017-11-30 14:44:07 -0800643 status = BAD_VALUE;
644 goto exit;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800645 }
Glenn Kastendd5f4c82014-01-13 10:26:32 -0800646 mFormat = format;
Eric Laurentc2f1f072009-07-17 12:17:14 -0700647
Glenn Kasten8ba90322013-10-30 11:29:27 -0700648 if (!audio_is_output_channel(channelMask)) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700649 ALOGE("%s(): Invalid channel mask %#x", __func__, channelMask);
Eric Laurentf32d7812017-11-30 14:44:07 -0800650 status = BAD_VALUE;
651 goto exit;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700652 }
Glenn Kastene3247bf2014-02-24 15:19:07 -0800653 mChannelMask = channelMask;
Eric Laurentf32d7812017-11-30 14:44:07 -0800654 channelCount = audio_channel_count_from_out_mask(channelMask);
Glenn Kastene3247bf2014-02-24 15:19:07 -0800655 mChannelCount = channelCount;
Glenn Kasten8ba90322013-10-30 11:29:27 -0700656
Eric Laurentc2f1f072009-07-17 12:17:14 -0700657 // force direct flag if format is not linear PCM
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100658 // or offload was requested
659 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
660 || !audio_is_linear_pcm(format)) {
661 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
Andy Hungfb8ede22018-09-12 19:03:24 -0700662 ? "%s(): Offload request, forcing to Direct Output"
663 : "%s(): Not linear PCM, forcing to Direct Output",
664 __func__);
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700665 flags = (audio_output_flags_t)
Glenn Kasten3acbd052012-02-28 10:39:56 -0800666 // FIXME why can't we allow direct AND fast?
Eric Laurent0ca3cf92012-04-18 09:24:29 -0700667 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
Eric Laurentc2f1f072009-07-17 12:17:14 -0700668 }
669
Eric Laurentd1f69b02014-12-15 14:33:13 -0800670 // force direct flag if HW A/V sync requested
671 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
672 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
673 }
674
Glenn Kastenb7730382014-04-30 15:50:31 -0700675 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
Phil Burkfdb3c072016-02-09 10:47:02 -0800676 if (audio_has_proportional_frames(format)) {
Glenn Kastenb7730382014-04-30 15:50:31 -0700677 mFrameSize = channelCount * audio_bytes_per_sample(format);
678 } else {
679 mFrameSize = sizeof(uint8_t);
680 }
Glenn Kastene3aa6592012-12-04 12:22:46 -0800681 } else {
Phil Burkfdb3c072016-02-09 10:47:02 -0800682 ALOG_ASSERT(audio_has_proportional_frames(format));
Glenn Kastenb7730382014-04-30 15:50:31 -0700683 mFrameSize = channelCount * audio_bytes_per_sample(format);
Glenn Kastenb7730382014-04-30 15:50:31 -0700684 // createTrack will return an error if PCM format is not supported by server,
685 // so no need to check for specific PCM formats here
Glenn Kastene3aa6592012-12-04 12:22:46 -0800686 }
687
Eric Laurent0d6db582014-11-12 18:39:44 -0800688 // sampling rate must be specified for direct outputs
689 if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
Eric Laurentf32d7812017-11-30 14:44:07 -0800690 status = BAD_VALUE;
691 goto exit;
Eric Laurent0d6db582014-11-12 18:39:44 -0800692 }
693 mSampleRate = sampleRate;
Lajos Molnar3a474aa2015-04-24 17:10:07 -0700694 mOriginalSampleRate = sampleRate;
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -0700695 mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
Andy Hungff874dc2016-04-11 16:49:09 -0700696 // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
697 mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
Eric Laurent0d6db582014-11-12 18:39:44 -0800698
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800699 // Make copy of input parameter offloadInfo so that in the future:
700 // (a) createTrack_l doesn't need it as an input parameter
701 // (b) we can support re-creation of offloaded tracks
702 if (offloadInfo != NULL) {
703 mOffloadInfoCopy = *offloadInfo;
704 mOffloadInfo = &mOffloadInfoCopy;
705 } else {
706 mOffloadInfo = NULL;
Eric Laurent20b9ef02016-12-05 11:03:16 -0800707 memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
Ytai Ben-Tsviffa2fd92020-10-20 09:13:53 -0700708 mOffloadInfoCopy = AUDIO_INFO_INITIALIZER;
Glenn Kastenb5ccb2d2014-01-13 14:42:43 -0800709 }
710
Glenn Kasten66e46352014-01-16 17:44:23 -0800711 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
712 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
Glenn Kasten05632a52012-01-03 14:22:33 -0800713 mSendLevel = 0.0f;
Glenn Kasten396fabd2014-01-08 08:54:23 -0800714 // mFrameCount is initialized in createTrack_l
Glenn Kastenb6037442012-11-14 13:42:25 -0800715 mReqFrameCount = frameCount;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700716 if (notificationFrames >= 0) {
717 mNotificationFramesReq = notificationFrames;
718 mNotificationsPerBufferReq = 0;
719 } else {
720 if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700721 ALOGE("%s(): notificationFrames=%d not permitted for non-fast track",
722 __func__, notificationFrames);
Eric Laurentf32d7812017-11-30 14:44:07 -0800723 status = BAD_VALUE;
724 goto exit;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700725 }
726 if (frameCount > 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -0700727 ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
728 __func__, notificationFrames, frameCount);
Eric Laurentf32d7812017-11-30 14:44:07 -0800729 status = BAD_VALUE;
730 goto exit;
Glenn Kastenea38ee72016-04-18 11:08:01 -0700731 }
732 mNotificationFramesReq = 0;
733 const uint32_t minNotificationsPerBuffer = 1;
734 const uint32_t maxNotificationsPerBuffer = 8;
735 mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
736 max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
737 ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
Andy Hungfb8ede22018-09-12 19:03:24 -0700738 "%s(): notificationFrames=%d clamped to the range -%u to -%u",
739 __func__,
Glenn Kastenea38ee72016-04-18 11:08:01 -0700740 notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
741 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800742 mNotificationFramesAct = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700743 // TODO b/182392553: refactor or remove
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000744 mClientAttributionSource = AttributionSourceState(attributionSource);
Eric Laurentf32d7812017-11-30 14:44:07 -0800745 callingPid = IPCThreadState::self()->getCallingPid();
746 myPid = getpid();
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700747 if (uid == -1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000748 mClientAttributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700749 IPCThreadState::self()->getCallingUid()));
Marco Nelissen462fd2f2013-01-14 14:12:05 -0800750 }
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700751 if (pid == (pid_t)-1 || (callingPid != myPid)) {
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000752 mClientAttributionSource.pid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(callingPid));
Marco Nelissend457c972014-02-11 08:47:07 -0800753 }
Eric Laurent2beeb502010-07-16 07:43:46 -0700754 mAuxEffectId = 0;
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -0800755 mOrigFlags = mFlags = flags;
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400756 mCallback = callback;
Eric Laurentbe916aa2010-06-01 23:49:17 -0700757
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400758 if (_callback != nullptr) {
759 mAudioTrackThread = sp<AudioTrackThread>::make(*this);
Glenn Kastena997e7a2012-08-07 09:44:19 -0700760 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
Glenn Kastenbfd31842015-03-20 09:01:44 -0700761 // thread begins in paused state, and will not reference us until start()
Glenn Kastena997e7a2012-08-07 09:44:19 -0700762 }
763
Eric Laurent34f1d8e2009-11-04 08:27:26 -0800764 // create the IAudioTrack
Francois Gaffie24a9fb02019-01-18 17:51:34 +0100765 {
766 AutoMutex lock(mLock);
767 status = createTrack_l();
768 }
Glenn Kastena997e7a2012-08-07 09:44:19 -0700769 if (status != NO_ERROR) {
770 if (mAudioTrackThread != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100771 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h
772 mAudioTrackThread->requestExitAndWait();
Glenn Kastena997e7a2012-08-07 09:44:19 -0700773 mAudioTrackThread.clear();
774 }
Eric Laurentf32d7812017-11-30 14:44:07 -0800775 goto exit;
Glenn Kasten5d464eb2012-06-22 17:19:53 -0700776 }
777
Andy Hung4ede21d2014-12-12 15:37:34 -0800778 mLoopCount = 0;
779 mLoopStart = 0;
780 mLoopEnd = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -0800781 mLoopCountNotified = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800782 mMarkerPosition = 0;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -0700783 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800784 mNewPosition = 0;
785 mUpdatePeriod = 0;
Glenn Kasten200092b2014-08-15 15:13:30 -0700786 mPosition = 0;
787 mReleased = 0;
Andy Hungffa36952017-08-17 10:41:51 -0700788 mStartNs = 0;
789 mStartFromZeroUs = 0;
Philip P. Moltmannbda45752020-07-17 16:41:18 -0700790 AudioSystem::acquireAudioSessionId(mSessionId, pid, uid);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800791 mSequence = 1;
792 mObservedSequence = mSequence;
793 mInUnderrun = false;
Phil Burk1b420972015-04-22 10:52:21 -0700794 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700795 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700796 mTimestampRetrogradePositionReported = false;
797 mTimestampRetrogradeTimeReported = false;
798 mTimestampStallReported = false;
799 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700800 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Andy Hung65ffdfc2016-10-10 15:52:11 -0700801 mStartTs.mPosition = 0;
Phil Burk2812d9e2016-01-04 10:34:30 -0800802 mUnderrunCountOffset = 0;
Andy Hungea2b9c02016-02-12 17:06:53 -0800803 mFramesWritten = 0;
804 mFramesWrittenServerOffset = 0;
Andy Hungf20a4e92016-08-15 19:10:34 -0700805 mFramesWrittenAtRestore = -1; // -1 is a unique initializer.
Ivan Lozano8cf3a072017-08-09 09:01:33 -0700806 mVolumeHandler = new media::VolumeHandler();
Eric Laurentf32d7812017-11-30 14:44:07 -0800807
808exit:
809 mStatus = status;
810 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800811}
812
Mikhail Naganov55773032020-10-01 15:08:13 -0700813
814status_t AudioTrack::set(
815 audio_stream_type_t streamType,
816 uint32_t sampleRate,
817 audio_format_t format,
818 uint32_t channelMask,
819 size_t frameCount,
820 audio_output_flags_t flags,
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400821 legacy_callback_t callback,
Mikhail Naganov55773032020-10-01 15:08:13 -0700822 void* user,
823 int32_t notificationFrames,
824 const sp<IMemory>& sharedBuffer,
825 bool threadCanCallJava,
826 audio_session_t sessionId,
827 transfer_type transferType,
828 const audio_offload_info_t *offloadInfo,
829 uid_t uid,
830 pid_t pid,
831 const audio_attributes_t* pAttributes,
832 bool doNotReconnect,
833 float maxRequiredSpeed,
834 audio_port_handle_t selectedDeviceId)
835{
Svet Ganov3e5f14f2021-05-13 22:51:08 +0000836 AttributionSourceState attributionSource;
837 attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(uid));
838 attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(pid));
839 attributionSource.token = sp<BBinder>::make();
Atneya Nair6a8238eb2021-10-26 19:26:41 -0400840 if (callback) {
841 mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
842 } else if (user) {
843 LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
844 }
845 return set(streamType, sampleRate, format, static_cast<audio_channel_mask_t>(channelMask),
846 frameCount, flags, mLegacyCallbackWrapper, notificationFrames, sharedBuffer,
847 threadCanCallJava, sessionId, transferType, offloadInfo, attributionSource,
848 pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
Mikhail Naganov55773032020-10-01 15:08:13 -0700849}
850
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800851// -------------------------------------------------------------------------
852
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100853status_t AudioTrack::start()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800854{
Eric Laurentf5aafb22010-11-18 08:40:16 -0800855 AutoMutex lock(mLock);
Andy Hungb68f5eb2019-12-03 16:49:17 -0800856
Andy Hung10fb4be2020-05-27 22:22:22 -0700857 if (mState == STATE_ACTIVE) {
858 return INVALID_OPERATION;
859 }
860
861 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
862
863 // Defer logging here due to OpenSL ES repeated start calls.
864 // TODO(b/154868033) after fix, restore this logging back to the beginning of start().
865 const int64_t beginNs = systemTime();
Andy Hungb68f5eb2019-12-03 16:49:17 -0800866 status_t status = NO_ERROR; // logged: make sure to set this before returning.
Andy Hung06a730b2020-04-09 13:28:31 -0700867 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800868 mediametrics::LogItem(mMetricsId)
Andy Hunga6b27032020-04-27 10:34:24 -0700869 .set(AMEDIAMETRICS_PROP_CALLERNAME,
870 mCallerName.empty()
871 ? AMEDIAMETRICS_PROP_CALLERNAME_VALUE_UNKNOWN
872 : mCallerName.c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -0800873 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_START)
Andy Hungea840382020-05-05 21:50:17 -0700874 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -0800875 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
876 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)status)
877 .record(); });
878
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800879
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800880 mInUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800881
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800882 State previousState = mState;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100883 if (previousState == STATE_PAUSED_STOPPING) {
884 mState = STATE_STOPPING;
885 } else {
886 mState = STATE_ACTIVE;
887 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700888 (void) updateAndGetPosition_l();
Andy Hung65ffdfc2016-10-10 15:52:11 -0700889
890 // save start timestamp
891 if (isOffloadedOrDirect_l()) {
892 if (getTimestamp_l(mStartTs) != OK) {
893 mStartTs.mPosition = 0;
894 }
895 } else {
896 if (getTimestamp_l(&mStartEts) != OK) {
897 mStartEts.clear();
898 }
899 }
Andy Hungffa36952017-08-17 10:41:51 -0700900 mStartNs = systemTime(); // save this for timestamp adjustment after starting.
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800901 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) {
902 // reset current position as seen by client to 0
Glenn Kasten200092b2014-08-15 15:13:30 -0700903 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -0700904 mPreviousTimestampValid = false;
Andy Hungc8e09c62015-06-03 23:43:36 -0700905 mTimestampStartupGlitchReported = false;
Andy Hungcf3b7152019-04-19 18:29:21 -0700906 mTimestampRetrogradePositionReported = false;
907 mTimestampRetrogradeTimeReported = false;
908 mTimestampStallReported = false;
909 mTimestampStaleTimeReported = false;
Andy Hungb01faa32016-04-27 12:51:32 -0700910 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID;
Phil Burk1b420972015-04-22 10:52:21 -0700911
Andy Hung65ffdfc2016-10-10 15:52:11 -0700912 if (!isOffloadedOrDirect_l()
913 && mStartEts.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) {
Andy Hunge1e98462016-04-12 10:18:51 -0700914 // Server side has consumed something, but is it finished consuming?
915 // It is possible since flush and stop are asynchronous that the server
916 // is still active at this point.
Andy Hungfb8ede22018-09-12 19:03:24 -0700917 ALOGV("%s(%d): server read:%lld cumulative flushed:%lld client written:%lld",
Eric Laurent973db022018-11-20 14:54:31 -0800918 __func__, mPortId,
Andy Hunge1e98462016-04-12 10:18:51 -0700919 (long long)(mFramesWrittenServerOffset
Andy Hung65ffdfc2016-10-10 15:52:11 -0700920 + mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]),
921 (long long)mStartEts.mFlushed,
Andy Hunge1e98462016-04-12 10:18:51 -0700922 (long long)mFramesWritten);
Andy Hungc4e60eb2017-09-06 11:14:57 -0700923 // mStartEts is already adjusted by mFramesWrittenServerOffset, so we delta adjust.
924 mFramesWrittenServerOffset -= mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER];
Andy Hung61be8412015-10-06 10:51:09 -0700925 }
Andy Hunge1e98462016-04-12 10:18:51 -0700926 mFramesWritten = 0;
927 mProxy->clearTimestamp(); // need new server push for valid timestamp
928 mMarkerReached = false;
Andy Hung61be8412015-10-06 10:51:09 -0700929
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700930 // For offloaded tracks, we don't know if the hardware counters are really zero here,
931 // since the flush is asynchronous and stop may not fully drain.
932 // We save the time when the track is started to later verify whether
933 // the counters are realistic (i.e. start from zero after this time).
Andy Hungffa36952017-08-17 10:41:51 -0700934 mStartFromZeroUs = mStartNs / 1000;
Andy Hung7f1bc8a2014-09-12 14:43:11 -0700935
Eric Laurentec9a0322013-08-28 10:23:01 -0700936 // force refresh of remaining frames by processAudioBuffer() as last
937 // write before stop could be partial.
938 mRefreshRemaining = true;
Tomoharu Kasahara6e8bf982017-06-09 16:17:33 +0900939
940 // for static track, clear the old flags when starting from stopped state
941 if (mSharedBuffer != 0) {
942 android_atomic_and(
943 ~(CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
944 &mCblk->mFlags);
945 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800946 }
Glenn Kasten200092b2014-08-15 15:13:30 -0700947 mNewPosition = mPosition + mUpdatePeriod;
Andy Hung4be3b832016-10-13 17:51:43 -0700948 int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800949
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800950 if (!(flags & CBLK_INVALID)) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -0800951 mAudioTrack->start(&status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800952 if (status == DEAD_OBJECT) {
953 flags |= CBLK_INVALID;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800954 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800955 }
956 if (flags & CBLK_INVALID) {
957 status = restoreTrack_l("start");
958 }
959
Andy Hung79629f02016-03-24 13:57:40 -0700960 // resume or pause the callback thread as needed.
961 sp<AudioTrackThread> t = mAudioTrackThread;
962 if (status == NO_ERROR) {
963 if (t != 0) {
964 if (previousState == STATE_STOPPING) {
965 mProxy->interrupt();
966 } else {
967 t->resume();
968 }
969 } else {
970 mPreviousPriority = getpriority(PRIO_PROCESS, 0);
971 get_sched_policy(0, &mPreviousSchedulingGroup);
972 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
973 }
Andy Hung39399b62017-04-21 15:07:45 -0700974
975 // Start our local VolumeHandler for restoration purposes.
976 mVolumeHandler->setStarted();
Andy Hung79629f02016-03-24 13:57:40 -0700977 } else {
Eric Laurent973db022018-11-20 14:54:31 -0800978 ALOGE("%s(%d): status %d", __func__, mPortId, status);
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800979 mState = previousState;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800980 if (t != 0) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100981 if (previousState != STATE_STOPPING) {
982 t->pause();
983 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800984 } else {
Glenn Kasten87913512011-06-22 16:15:25 -0700985 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
Glenn Kastena6364332012-04-19 09:35:04 -0700986 set_sched_policy(0, mPreviousSchedulingGroup);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -0800987 }
988 }
989
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +0100990 return status;
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800991}
992
993void AudioTrack::stop()
994{
Andy Hungb68f5eb2019-12-03 16:49:17 -0800995 const int64_t beginNs = systemTime();
996
Glenn Kasten9f80dd22012-12-18 15:57:32 -0800997 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -0700998 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -0800999 mediametrics::LogItem(mMetricsId)
1000 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_STOP)
Andy Hungea840382020-05-05 21:50:17 -07001001 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001002 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
Phil Burk64e16a72020-06-01 13:25:51 -07001003 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1004 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getUnderrunCount_l())
Phil Burka9876702020-04-20 18:16:15 -07001005 .record();
Phil Burka9876702020-04-20 18:16:15 -07001006 });
Andy Hungb68f5eb2019-12-03 16:49:17 -08001007
Eric Laurent973db022018-11-20 14:54:31 -08001008 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001009
Glenn Kasten397edb32013-08-30 15:10:13 -07001010 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001011 return;
1012 }
1013
Glenn Kasten23a75452014-01-13 10:37:17 -08001014 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001015 mState = STATE_STOPPING;
1016 } else {
1017 mState = STATE_STOPPED;
Andy Hung2148bf02016-11-28 19:01:02 -08001018 ALOGD_IF(mSharedBuffer == nullptr,
Eric Laurent973db022018-11-20 14:54:31 -08001019 "%s(%d): called with %u frames delivered", __func__, mPortId, mReleased.value());
Andy Hungc2813e52014-10-16 17:54:34 -07001020 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001021 }
1022
Andy Hung1d3556d2018-03-29 16:30:14 -07001023 mProxy->stop(); // notify server not to read beyond current client position until start().
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001024 mProxy->interrupt();
1025 mAudioTrack->stop();
Andy Hung61be8412015-10-06 10:51:09 -07001026
1027 // Note: legacy handling - stop does not clear playback marker
1028 // and periodic update counter, but flush does for streaming tracks.
Andy Hung9b461582014-12-01 17:56:29 -08001029
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001030 if (mSharedBuffer != 0) {
Andy Hung9b461582014-12-01 17:56:29 -08001031 // clear buffer position and loop count.
Andy Hung9b461582014-12-01 17:56:29 -08001032 mStaticProxy->setBufferPositionAndLoop(0 /* position */,
1033 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001034 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001035
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001036 sp<AudioTrackThread> t = mAudioTrackThread;
1037 if (t != 0) {
Glenn Kasten23a75452014-01-13 10:37:17 -08001038 if (!isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001039 t->pause();
Jean-Michel Trivia60e9a22018-11-15 16:13:36 -08001040 } else if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
Jean-Michel Trivi0248a2a2018-11-21 10:05:57 -08001041 // causes wake up of the playback thread, that will callback the client for
1042 // EVENT_STREAM_END in processAudioBuffer()
1043 t->wake();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001044 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001045 } else {
1046 setpriority(PRIO_PROCESS, 0, mPreviousPriority);
1047 set_sched_policy(0, mPreviousSchedulingGroup);
1048 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001049}
1050
1051bool AudioTrack::stopped() const
1052{
Glenn Kasten9a2aaf92012-01-03 09:42:47 -08001053 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001054 return mState != STATE_ACTIVE;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001055}
1056
1057void AudioTrack::flush()
1058{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001059 const int64_t beginNs = systemTime();
Andy Hungfb8ede22018-09-12 19:03:24 -07001060 AutoMutex lock(mLock);
Andy Hung06a730b2020-04-09 13:28:31 -07001061 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001062 mediametrics::LogItem(mMetricsId)
1063 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_FLUSH)
Andy Hungea840382020-05-05 21:50:17 -07001064 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001065 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
1066 .record(); });
1067
Eric Laurent973db022018-11-20 14:54:31 -08001068 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001069
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001070 if (mSharedBuffer != 0) {
1071 return;
Glenn Kasten4bae3642012-11-30 13:41:12 -08001072 }
Andy Hung4c5ed302018-05-09 11:16:21 -07001073 if (mState == STATE_ACTIVE) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001074 return;
1075 }
1076 flush_l();
Eric Laurent1703cdf2011-03-07 14:52:59 -08001077}
1078
Eric Laurent1703cdf2011-03-07 14:52:59 -08001079void AudioTrack::flush_l()
1080{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001081 ALOG_ASSERT(mState != STATE_ACTIVE);
Eric Laurentc2f1f072009-07-17 12:17:14 -07001082
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001083 // clear playback marker and periodic update counter
1084 mMarkerPosition = 0;
1085 mMarkerReached = false;
1086 mUpdatePeriod = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001087 mRefreshRemaining = true;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001088
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001089 mState = STATE_FLUSHED;
Andy Hungc2813e52014-10-16 17:54:34 -07001090 mReleased = 0;
Glenn Kasten23a75452014-01-13 10:37:17 -08001091 if (isOffloaded_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001092 mProxy->interrupt();
1093 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001094 mProxy->flush();
Glenn Kasten4bae3642012-11-30 13:41:12 -08001095 mAudioTrack->flush();
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001096}
1097
Andy Hung959b5b82021-09-24 10:46:20 -07001098bool AudioTrack::pauseAndWait(const std::chrono::milliseconds& timeout)
1099{
1100 using namespace std::chrono_literals;
1101
1102 pause();
1103
1104 AutoMutex lock(mLock);
1105 // offload and direct tracks do not wait because pause volume ramp is handled by hardware.
1106 if (isOffloadedOrDirect_l()) return true;
1107
1108 // Wait for the track state to be anything besides pausing.
1109 // This ensures that the volume has ramped down.
1110 constexpr auto SLEEP_INTERVAL_MS = 10ms;
1111 auto begin = std::chrono::steady_clock::now();
1112 while (true) {
1113 // wait for state to change
1114 const int state = mProxy->getState();
1115
1116 mLock.unlock(); // only local variables accessed until lock.
1117 auto elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(
1118 std::chrono::steady_clock::now() - begin);
1119 if (state != CBLK_STATE_PAUSING) {
1120 ALOGV("%s: success state:%d after %lld ms", __func__, state, elapsed.count());
1121 return true;
1122 }
1123 std::chrono::milliseconds remaining = timeout - elapsed;
1124 if (remaining.count() <= 0) {
1125 ALOGW("%s: timeout expired state:%d still pausing:%d after %lld ms",
1126 __func__, state, CBLK_STATE_PAUSING, elapsed.count());
1127 return false;
1128 }
1129 // It is conceivable that the track is restored while sleeping;
1130 // as this logic is advisory, we allow that.
1131 std::this_thread::sleep_for(std::min(remaining, SLEEP_INTERVAL_MS));
1132 mLock.lock();
1133 }
1134}
1135
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001136void AudioTrack::pause()
1137{
Andy Hungb68f5eb2019-12-03 16:49:17 -08001138 const int64_t beginNs = systemTime();
Eric Laurentf5aafb22010-11-18 08:40:16 -08001139 AutoMutex lock(mLock);
Andy Hunga6b27032020-04-27 10:34:24 -07001140 mediametrics::Defer defer([&]() {
Andy Hungb68f5eb2019-12-03 16:49:17 -08001141 mediametrics::LogItem(mMetricsId)
1142 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_PAUSE)
Andy Hungea840382020-05-05 21:50:17 -07001143 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08001144 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
1145 .record(); });
1146
Eric Laurent973db022018-11-20 14:54:31 -08001147 ALOGV("%s(%d): prior state:%s", __func__, mPortId, stateToString(mState));
Andy Hungfb8ede22018-09-12 19:03:24 -07001148
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001149 if (mState == STATE_ACTIVE) {
1150 mState = STATE_PAUSED;
1151 } else if (mState == STATE_STOPPING) {
1152 mState = STATE_PAUSED_STOPPING;
1153 } else {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001154 return;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001155 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001156 mProxy->interrupt();
1157 mAudioTrack->pause();
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001158
Marco Nelissen3a90f282014-03-10 11:21:43 -07001159 if (isOffloaded_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001160 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001161 // An offload output can be re-used between two audio tracks having
1162 // the same configuration. A timestamp query for a paused track
1163 // while the other is running would return an incorrect time.
1164 // To fix this, cache the playback position on a pause() and return
1165 // this time when requested until the track is resumed.
1166
1167 // OffloadThread sends HAL pause in its threadLoop. Time saved
1168 // here can be slightly off.
1169
1170 // TODO: check return code for getRenderPosition.
1171
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001172 uint32_t halFrames;
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001173 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition);
Andy Hungfb8ede22018-09-12 19:03:24 -07001174 ALOGV("%s(%d): for offload, cache current position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001175 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001176 }
1177 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001178}
1179
Eric Laurentbe916aa2010-06-01 23:49:17 -07001180status_t AudioTrack::setVolume(float left, float right)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001181{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001182 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1183 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY ||
1184 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001185 return BAD_VALUE;
1186 }
1187
Andy Hungb68f5eb2019-12-03 16:49:17 -08001188 mediametrics::LogItem(mMetricsId)
1189 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETVOLUME)
1190 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)left)
1191 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)right)
1192 .record();
1193
Eric Laurent1703cdf2011-03-07 14:52:59 -08001194 AutoMutex lock(mLock);
Glenn Kasten66e46352014-01-16 17:44:23 -08001195 mVolume[AUDIO_INTERLEAVE_LEFT] = left;
1196 mVolume[AUDIO_INTERLEAVE_RIGHT] = right;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001197
Glenn Kastenc56f3422014-03-21 17:53:17 -07001198 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right)));
Eric Laurentbe916aa2010-06-01 23:49:17 -07001199
Glenn Kasten23a75452014-01-13 10:37:17 -08001200 if (isOffloaded_l()) {
Eric Laurent59fe0102013-09-27 18:48:26 -07001201 mAudioTrack->signal();
1202 }
Eric Laurentbe916aa2010-06-01 23:49:17 -07001203 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001204}
1205
Glenn Kastenb1c09932012-02-27 16:21:04 -08001206status_t AudioTrack::setVolume(float volume)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001207{
Glenn Kastenb1c09932012-02-27 16:21:04 -08001208 return setVolume(volume, volume);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001209}
1210
Eric Laurent2beeb502010-07-16 07:43:46 -07001211status_t AudioTrack::setAuxEffectSendLevel(float level)
Eric Laurentbe916aa2010-06-01 23:49:17 -07001212{
Glenn Kastenc56f3422014-03-21 17:53:17 -07001213 // This duplicates a test by AudioTrack JNI, but that is not the only caller
1214 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) {
Eric Laurentbe916aa2010-06-01 23:49:17 -07001215 return BAD_VALUE;
1216 }
1217
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001218 AutoMutex lock(mLock);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001219 mSendLevel = level;
Glenn Kastene3aa6592012-12-04 12:22:46 -08001220 mProxy->setSendLevel(level);
Eric Laurentbe916aa2010-06-01 23:49:17 -07001221
1222 return NO_ERROR;
1223}
1224
Glenn Kastena5224f32012-01-04 12:41:44 -08001225void AudioTrack::getAuxEffectSendLevel(float* level) const
Eric Laurentbe916aa2010-06-01 23:49:17 -07001226{
1227 if (level != NULL) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001228 *level = mSendLevel;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001229 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001230}
1231
Glenn Kasten3b16c762012-11-14 08:44:39 -08001232status_t AudioTrack::setSampleRate(uint32_t rate)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001233{
Andy Hung5cbb5782015-03-27 18:39:59 -07001234 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08001235 ALOGV("%s(%d): prior state:%s rate:%u", __func__, mPortId, stateToString(mState), rate);
Andy Hungfb8ede22018-09-12 19:03:24 -07001236
Andy Hung5cbb5782015-03-27 18:39:59 -07001237 if (rate == mSampleRate) {
1238 return NO_ERROR;
1239 }
jiabinf4de6112018-12-19 12:40:08 -08001240 if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)
1241 || (mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL)) {
John Grossman4ff14ba2012-02-08 16:37:41 -08001242 return INVALID_OPERATION;
1243 }
Eric Laurent0d6db582014-11-12 18:39:44 -08001244 if (mOutput == AUDIO_IO_HANDLE_NONE) {
1245 return NO_INIT;
1246 }
Andy Hung5cbb5782015-03-27 18:39:59 -07001247 // NOTE: it is theoretically possible, but highly unlikely, that a device change
1248 // could mean a previously allowed sampling rate is no longer allowed.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001249 uint32_t afSamplingRate;
Eric Laurent0d6db582014-11-12 18:39:44 -08001250 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
Eric Laurent57326622009-07-07 07:10:45 -07001251 return NO_INIT;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001252 }
Andy Hung26145642015-04-15 21:56:53 -07001253 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001254 const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001255 if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001256 return BAD_VALUE;
1257 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001258 // TODO: Should we also check if the buffer size is compatible?
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001259
Glenn Kastene3aa6592012-12-04 12:22:46 -08001260 mSampleRate = rate;
Andy Hung26145642015-04-15 21:56:53 -07001261 mProxy->setSampleRate(effectiveSampleRate);
Glenn Kastene3aa6592012-12-04 12:22:46 -08001262
Eric Laurent57326622009-07-07 07:10:45 -07001263 return NO_ERROR;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001264}
1265
Glenn Kastena5224f32012-01-04 12:41:44 -08001266uint32_t AudioTrack::getSampleRate() const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001267{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001268 AutoMutex lock(mLock);
Eric Laurent6f59db12013-07-26 17:16:50 -07001269
1270 // sample rate can be updated during playback by the offloaded decoder so we need to
1271 // query the HAL and update if needed.
1272// FIXME use Proxy return channel to update the rate from server and avoid polling here
Eric Laurentab5cdba2014-06-09 17:22:27 -07001273 if (isOffloadedOrDirect_l()) {
Glenn Kasten142f5192014-03-25 17:44:59 -07001274 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent6f59db12013-07-26 17:16:50 -07001275 uint32_t sampleRate = 0;
Jean-Michel Trivib7f24b12014-06-11 10:05:30 -07001276 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
Eric Laurent6f59db12013-07-26 17:16:50 -07001277 if (status == NO_ERROR) {
1278 mSampleRate = sampleRate;
1279 }
1280 }
1281 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08001282 return mSampleRate;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001283}
1284
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001285uint32_t AudioTrack::getOriginalSampleRate() const
1286{
Lajos Molnar3a474aa2015-04-24 17:10:07 -07001287 return mOriginalSampleRate;
1288}
1289
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001290status_t AudioTrack::setDualMonoMode(audio_dual_mono_mode_t mode)
1291{
1292 AutoMutex lock(mLock);
1293 return setDualMonoMode_l(mode);
1294}
1295
1296status_t AudioTrack::setDualMonoMode_l(audio_dual_mono_mode_t mode)
1297{
1298 const status_t status = statusTFromBinderStatus(
1299 mAudioTrack->setDualMonoMode(VALUE_OR_RETURN_STATUS(
1300 legacy2aidl_audio_dual_mono_mode_t_AudioDualMonoMode(mode))));
1301 if (status == NO_ERROR) mDualMonoMode = mode;
1302 return status;
1303}
1304
1305status_t AudioTrack::getDualMonoMode(audio_dual_mono_mode_t* mode) const
1306{
1307 AutoMutex lock(mLock);
1308 media::AudioDualMonoMode mediaMode;
1309 const status_t status = statusTFromBinderStatus(mAudioTrack->getDualMonoMode(&mediaMode));
1310 if (status == NO_ERROR) {
1311 *mode = VALUE_OR_RETURN_STATUS(
1312 aidl2legacy_AudioDualMonoMode_audio_dual_mono_mode_t(mediaMode));
1313 }
1314 return status;
1315}
1316
1317status_t AudioTrack::setAudioDescriptionMixLevel(float leveldB)
1318{
1319 AutoMutex lock(mLock);
1320 return setAudioDescriptionMixLevel_l(leveldB);
1321}
1322
1323status_t AudioTrack::setAudioDescriptionMixLevel_l(float leveldB)
1324{
1325 const status_t status = statusTFromBinderStatus(
1326 mAudioTrack->setAudioDescriptionMixLevel(leveldB));
1327 if (status == NO_ERROR) mAudioDescriptionMixLeveldB = leveldB;
1328 return status;
1329}
1330
1331status_t AudioTrack::getAudioDescriptionMixLevel(float* leveldB) const
1332{
1333 AutoMutex lock(mLock);
1334 return statusTFromBinderStatus(mAudioTrack->getAudioDescriptionMixLevel(leveldB));
1335}
1336
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001337status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
Andy Hung8edb8dc2015-03-26 19:13:55 -07001338{
Andy Hung8edb8dc2015-03-26 19:13:55 -07001339 AutoMutex lock(mLock);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001340 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) {
Andy Hung8edb8dc2015-03-26 19:13:55 -07001341 return NO_ERROR;
1342 }
Glenn Kastend79072e2016-01-06 08:41:20 -08001343 if (isOffloadedOrDirect_l()) {
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001344 const status_t status = statusTFromBinderStatus(mAudioTrack->setPlaybackRateParameters(
1345 VALUE_OR_RETURN_STATUS(
1346 legacy2aidl_audio_playback_rate_t_AudioPlaybackRate(playbackRate))));
1347 if (status == NO_ERROR) {
1348 mPlaybackRate = playbackRate;
1349 }
1350 return status;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001351 }
1352 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
1353 return INVALID_OPERATION;
1354 }
Andy Hungff874dc2016-04-11 16:49:09 -07001355
Andy Hungfb8ede22018-09-12 19:03:24 -07001356 ALOGV("%s(%d): mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001357 __func__, mPortId, mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001358 // pitch is emulated by adjusting speed and sampleRate
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001359 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
1360 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
1361 const float effectivePitch = adjustPitch(playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001362 AudioPlaybackRate playbackRateTemp = playbackRate;
1363 playbackRateTemp.mSpeed = effectiveSpeed;
1364 playbackRateTemp.mPitch = effectivePitch;
1365
Andy Hungfb8ede22018-09-12 19:03:24 -07001366 ALOGV("%s(%d) (effective) mSampleRate:%u mSpeed:%f mPitch:%f",
Eric Laurent973db022018-11-20 14:54:31 -08001367 __func__, mPortId, effectiveRate, effectiveSpeed, effectivePitch);
Andy Hungff874dc2016-04-11 16:49:09 -07001368
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001369 if (!isAudioPlaybackRateValid(playbackRateTemp)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001370 ALOGW("%s(%d) (%f, %f) failed (effective rate out of bounds)",
Eric Laurent973db022018-11-20 14:54:31 -08001371 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07001372 return BAD_VALUE;
1373 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001374 // Check if the buffer size is compatible.
Andy Hung26145642015-04-15 21:56:53 -07001375 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001376 ALOGW("%s(%d) (%f, %f) failed (buffer size)",
Eric Laurent973db022018-11-20 14:54:31 -08001377 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Andy Hung8edb8dc2015-03-26 19:13:55 -07001378 return BAD_VALUE;
1379 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001380
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001381 // Check resampler ratios are within bounds
Glenn Kastend3bb6452016-12-05 18:14:37 -08001382 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
1383 (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001384 ALOGW("%s(%d) (%f, %f) failed. Resample rate exceeds max accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001385 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001386 return BAD_VALUE;
1387 }
1388
Dan Austine34eae22015-10-27 16:14:52 -07001389 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001390 ALOGW("%s(%d) (%f, %f) failed. Resample rate below min accepted value",
Eric Laurent973db022018-11-20 14:54:31 -08001391 __func__, mPortId, playbackRate.mSpeed, playbackRate.mPitch);
Ricardo Garcia6c7f0622015-04-30 18:39:16 -07001392 return BAD_VALUE;
1393 }
1394 mPlaybackRate = playbackRate;
1395 //set effective rates
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001396 mProxy->setPlaybackRate(playbackRateTemp);
Andy Hung26145642015-04-15 21:56:53 -07001397 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
Andy Hungb68f5eb2019-12-03 16:49:17 -08001398
1399 mediametrics::LogItem(mMetricsId)
1400 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYBACKPARAM)
1401 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
1402 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
1403 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
1404 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1405 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveRate)
1406 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1407 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)playbackRateTemp.mSpeed)
1408 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
1409 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)playbackRateTemp.mPitch)
1410 .record();
1411
Andy Hung8edb8dc2015-03-26 19:13:55 -07001412 return NO_ERROR;
1413}
1414
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001415const AudioPlaybackRate& AudioTrack::getPlaybackRate()
Andy Hung8edb8dc2015-03-26 19:13:55 -07001416{
1417 AutoMutex lock(mLock);
Kuowei Lid4adbdb2020-08-13 14:44:25 +08001418 if (isOffloadedOrDirect_l()) {
1419 media::AudioPlaybackRate playbackRateTemp;
1420 const status_t status = statusTFromBinderStatus(
1421 mAudioTrack->getPlaybackRateParameters(&playbackRateTemp));
1422 if (status == NO_ERROR) { // update local version if changed.
1423 mPlaybackRate =
1424 aidl2legacy_AudioPlaybackRate_audio_playback_rate_t(playbackRateTemp).value();
1425 }
1426 }
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07001427 return mPlaybackRate;
Andy Hung8edb8dc2015-03-26 19:13:55 -07001428}
1429
Phil Burkc0adecb2016-01-08 12:44:11 -08001430ssize_t AudioTrack::getBufferSizeInFrames()
1431{
1432 AutoMutex lock(mLock);
1433 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1434 return NO_INIT;
1435 }
Phil Burka9876702020-04-20 18:16:15 -07001436
Phil Burke8972b02016-03-04 11:29:57 -08001437 return (ssize_t) mProxy->getBufferSizeInFrames();
Phil Burkc0adecb2016-01-08 12:44:11 -08001438}
1439
Andy Hungf2c87b32016-04-07 19:49:29 -07001440status_t AudioTrack::getBufferDurationInUs(int64_t *duration)
1441{
1442 if (duration == nullptr) {
1443 return BAD_VALUE;
1444 }
1445 AutoMutex lock(mLock);
1446 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1447 return NO_INIT;
1448 }
1449 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames();
1450 if (bufferSizeInFrames < 0) {
1451 return (status_t)bufferSizeInFrames;
1452 }
1453 *duration = (int64_t)((double)bufferSizeInFrames * 1000000
1454 / ((double)mSampleRate * mPlaybackRate.mSpeed));
1455 return NO_ERROR;
1456}
1457
Phil Burkc0adecb2016-01-08 12:44:11 -08001458ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames)
1459{
1460 AutoMutex lock(mLock);
1461 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1462 return NO_INIT;
1463 }
Phil Burka9876702020-04-20 18:16:15 -07001464
1465 ssize_t originalBufferSize = mProxy->getBufferSizeInFrames();
1466 ssize_t finalBufferSize = mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
1467 if (originalBufferSize != finalBufferSize) {
Phil Burk64e16a72020-06-01 13:25:51 -07001468 android::mediametrics::LogItem(mMetricsId)
1469 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
1470 .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, (int32_t)mProxy->getBufferSizeInFrames())
1471 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t)getUnderrunCount_l())
1472 .record();
Phil Burka9876702020-04-20 18:16:15 -07001473 }
1474 return finalBufferSize;
Phil Burkc0adecb2016-01-08 12:44:11 -08001475}
1476
Andy Hung3c7f47a2021-03-16 17:30:09 -07001477ssize_t AudioTrack::getStartThresholdInFrames() const
1478{
1479 AutoMutex lock(mLock);
1480 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1481 return NO_INIT;
1482 }
1483 return (ssize_t) mProxy->getStartThresholdInFrames();
1484}
1485
1486ssize_t AudioTrack::setStartThresholdInFrames(size_t startThresholdInFrames)
1487{
1488 if (startThresholdInFrames > INT32_MAX || startThresholdInFrames == 0) {
1489 // contractually we could simply return the current threshold in frames
1490 // to indicate the request was ignored, but we return an error here.
1491 return BAD_VALUE;
1492 }
1493 AutoMutex lock(mLock);
1494 // We do not permit calling setStartThresholdInFrames() between the AudioTrack
1495 // default ctor AudioTrack() and set(...) but rather fail such an attempt.
1496 // (To do so would require a cached mOrigStartThresholdInFrames and we may
1497 // not have proper validation for the actual set value).
1498 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
1499 return NO_INIT;
1500 }
1501 const uint32_t original = mProxy->getStartThresholdInFrames();
1502 const uint32_t final = mProxy->setStartThresholdInFrames(startThresholdInFrames);
1503 if (original != final) {
1504 android::mediametrics::LogItem(mMetricsId)
1505 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETSTARTTHRESHOLD)
1506 .set(AMEDIAMETRICS_PROP_STARTTHRESHOLDFRAMES, (int32_t)final)
1507 .record();
1508 if (original > final) {
1509 // restart track if it was disabled by audioflinger due to previous underrun
1510 // and we reduced the number of frames for the threshold.
1511 restartIfDisabled();
1512 }
1513 }
1514 return final;
1515}
1516
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001517status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1518{
Glenn Kastend79072e2016-01-06 08:41:20 -08001519 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001520 return INVALID_OPERATION;
1521 }
1522
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001523 if (loopCount == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001524 ;
1525 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount &&
1526 loopEnd - loopStart >= MIN_LOOP) {
1527 ;
1528 } else {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001529 return BAD_VALUE;
1530 }
1531
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001532 AutoMutex lock(mLock);
1533 // See setPosition() regarding setting parameters such as loop points or position while active
1534 if (mState == STATE_ACTIVE) {
1535 return INVALID_OPERATION;
Eric Laurentc2f1f072009-07-17 12:17:14 -07001536 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001537 setLoop_l(loopStart, loopEnd, loopCount);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001538 return NO_ERROR;
1539}
1540
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001541void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
1542{
Andy Hung4ede21d2014-12-12 15:37:34 -08001543 // We do not update the periodic notification point.
1544 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
1545 mLoopCount = loopCount;
1546 mLoopEnd = loopEnd;
1547 mLoopStart = loopStart;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001548 mLoopCountNotified = loopCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001549 mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
Andy Hung3c09c782014-12-29 18:39:32 -08001550
1551 // Waking the AudioTrackThread is not needed as this cannot be called when active.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001552}
1553
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001554status_t AudioTrack::setMarkerPosition(uint32_t marker)
1555{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001556 // The only purpose of setting marker position is to get a callback
Atneya Nair6a8238eb2021-10-26 19:26:41 -04001557 if (!mCallback.promote() || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001558 return INVALID_OPERATION;
1559 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001560
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001561 AutoMutex lock(mLock);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001562 mMarkerPosition = marker;
Jean-Michel Trivi2c22aeb2009-03-24 18:11:07 -07001563 mMarkerReached = false;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001564
Andy Hung3c09c782014-12-29 18:39:32 -08001565 sp<AudioTrackThread> t = mAudioTrackThread;
1566 if (t != 0) {
1567 t->wake();
1568 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001569 return NO_ERROR;
1570}
1571
Glenn Kastena5224f32012-01-04 12:41:44 -08001572status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001573{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001574 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001575 return INVALID_OPERATION;
1576 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001577 if (marker == NULL) {
1578 return BAD_VALUE;
1579 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001580
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001581 AutoMutex lock(mLock);
Andy Hung90e8a972015-11-09 16:42:40 -08001582 mMarkerPosition.getValue(marker);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001583
1584 return NO_ERROR;
1585}
1586
1587status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
1588{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07001589 // The only purpose of setting position update period is to get a callback
Atneya Nair6a8238eb2021-10-26 19:26:41 -04001590 if (!mCallback.promote() || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001591 return INVALID_OPERATION;
1592 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001593
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001594 AutoMutex lock(mLock);
Glenn Kasten200092b2014-08-15 15:13:30 -07001595 mNewPosition = updateAndGetPosition_l() + updatePeriod;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001596 mUpdatePeriod = updatePeriod;
Glenn Kasten2b2165c2014-01-13 08:53:36 -08001597
Andy Hung3c09c782014-12-29 18:39:32 -08001598 sp<AudioTrackThread> t = mAudioTrackThread;
1599 if (t != 0) {
1600 t->wake();
1601 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001602 return NO_ERROR;
1603}
1604
Glenn Kastena5224f32012-01-04 12:41:44 -08001605status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001606{
Eric Laurentab5cdba2014-06-09 17:22:27 -07001607 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001608 return INVALID_OPERATION;
1609 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07001610 if (updatePeriod == NULL) {
1611 return BAD_VALUE;
1612 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001613
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001614 AutoMutex lock(mLock);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001615 *updatePeriod = mUpdatePeriod;
1616
1617 return NO_ERROR;
1618}
1619
1620status_t AudioTrack::setPosition(uint32_t position)
1621{
Glenn Kastend79072e2016-01-06 08:41:20 -08001622 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001623 return INVALID_OPERATION;
1624 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001625 if (position > mFrameCount) {
1626 return BAD_VALUE;
1627 }
John Grossman4ff14ba2012-02-08 16:37:41 -08001628
Eric Laurent1703cdf2011-03-07 14:52:59 -08001629 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001630 // Currently we require that the player is inactive before setting parameters such as position
1631 // or loop points. Otherwise, there could be a race condition: the application could read the
1632 // current position, compute a new position or loop parameters, and then set that position or
1633 // loop parameters but it would do the "wrong" thing since the position has continued to advance
1634 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app
1635 // to specify how it wants to handle such scenarios.
1636 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001637 return INVALID_OPERATION;
1638 }
Andy Hung9b461582014-12-01 17:56:29 -08001639 // After setting the position, use full update period before notification.
Glenn Kasten200092b2014-08-15 15:13:30 -07001640 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001641 mStaticProxy->setBufferPosition(position);
Andy Hung3c09c782014-12-29 18:39:32 -08001642
1643 // Waking the AudioTrackThread is not needed as this cannot be called when active.
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001644 return NO_ERROR;
1645}
1646
Glenn Kasten200092b2014-08-15 15:13:30 -07001647status_t AudioTrack::getPosition(uint32_t *position)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001648{
Glenn Kastend65d73c2012-06-22 17:21:07 -07001649 if (position == NULL) {
1650 return BAD_VALUE;
1651 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001652
Eric Laurent1703cdf2011-03-07 14:52:59 -08001653 AutoMutex lock(mLock);
Andy Hung7a490e72016-03-23 15:58:10 -07001654 // FIXME: offloaded and direct tracks call into the HAL for render positions
1655 // for compressed/synced data; however, we use proxy position for pure linear pcm data
1656 // as we do not know the capability of the HAL for pcm position support and standby.
1657 // There may be some latency differences between the HAL position and the proxy position.
1658 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001659 uint32_t dspFrames = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001660
Eric Laurentab5cdba2014-06-09 17:22:27 -07001661 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001662 ALOGV("%s(%d): called in paused state, return cached position %u",
Eric Laurent973db022018-11-20 14:54:31 -08001663 __func__, mPortId, mPausedPosition);
Haynes Mathew George7064fd22014-01-08 13:59:53 -08001664 *position = mPausedPosition;
1665 return NO_ERROR;
1666 }
1667
Glenn Kasten142f5192014-03-25 17:44:59 -07001668 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Andy Hung1f1db832015-06-08 13:26:10 -07001669 uint32_t halFrames; // actually unused
1670 (void) AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames);
1671 // FIXME: on getRenderPosition() error, we return OK with frame position 0.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001672 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07001673 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED)
1674 // due to hardware latency. We leave this behavior for now.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001675 *position = dspFrames;
1676 } else {
Eric Laurent275e8e92014-11-30 15:14:47 -08001677 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung1f1db832015-06-08 13:26:10 -07001678 (void) restoreTrack_l("getPosition");
1679 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l()
1680 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position.
Eric Laurent275e8e92014-11-30 15:14:47 -08001681 }
1682
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001683 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
Glenn Kasten200092b2014-08-15 15:13:30 -07001684 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ?
Andy Hung90e8a972015-11-09 16:42:40 -08001685 0 : updateAndGetPosition_l().value();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001686 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001687 return NO_ERROR;
1688}
1689
Kévin PETIT377b2ec2014-02-03 12:35:36 +00001690status_t AudioTrack::getBufferPosition(uint32_t *position)
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001691{
Glenn Kastend79072e2016-01-06 08:41:20 -08001692 if (mSharedBuffer == 0) {
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001693 return INVALID_OPERATION;
1694 }
1695 if (position == NULL) {
1696 return BAD_VALUE;
1697 }
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001698
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001699 AutoMutex lock(mLock);
1700 *position = mStaticProxy->getBufferPosition();
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001701 return NO_ERROR;
1702}
Glenn Kasten9c6745f2012-11-30 13:35:29 -08001703
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001704status_t AudioTrack::reload()
1705{
Glenn Kastend79072e2016-01-06 08:41:20 -08001706 if (mSharedBuffer == 0 || isOffloadedOrDirect()) {
Glenn Kasten083d1c12012-11-30 15:00:36 -08001707 return INVALID_OPERATION;
1708 }
1709
Eric Laurent1703cdf2011-03-07 14:52:59 -08001710 AutoMutex lock(mLock);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001711 // See setPosition() regarding setting parameters such as loop points or position while active
1712 if (mState == STATE_ACTIVE) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07001713 return INVALID_OPERATION;
1714 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001715 mNewPosition = mUpdatePeriod;
Andy Hung9b461582014-12-01 17:56:29 -08001716 (void) updateAndGetPosition_l();
1717 mPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07001718 mPreviousTimestampValid = false;
Andy Hung53c3b5f2014-12-15 16:42:05 -08001719#if 0
Andy Hung9b461582014-12-01 17:56:29 -08001720 // The documentation is not clear on the behavior of reload() and the restoration
Andy Hung53c3b5f2014-12-15 16:42:05 -08001721 // of loop count. Historically we have not restored loop count, start, end,
1722 // but it makes sense if one desires to repeat playing a particular sound.
1723 if (mLoopCount != 0) {
1724 mLoopCountNotified = mLoopCount;
1725 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
1726 }
1727#endif
Andy Hung9b461582014-12-01 17:56:29 -08001728 mStaticProxy->setBufferPosition(0);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001729 return NO_ERROR;
1730}
1731
Glenn Kasten38e905b2014-01-13 10:21:48 -08001732audio_io_handle_t AudioTrack::getOutput() const
Eric Laurentc2f1f072009-07-17 12:17:14 -07001733{
Eric Laurent1703cdf2011-03-07 14:52:59 -08001734 AutoMutex lock(mLock);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001735 return mOutput;
Eric Laurent1703cdf2011-03-07 14:52:59 -08001736}
1737
Paul McLeanaa981192015-03-21 09:55:15 -07001738status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) {
1739 AutoMutex lock(mLock);
Eric Laurent2f2c1982021-06-02 14:03:11 +02001740 ALOGV("%s(%d): deviceId=%d mSelectedDeviceId=%d",
1741 __func__, mPortId, deviceId, mSelectedDeviceId);
Paul McLeanaa981192015-03-21 09:55:15 -07001742 if (mSelectedDeviceId != deviceId) {
1743 mSelectedDeviceId = deviceId;
Eric Laurentfb00fc72017-05-25 18:17:12 -07001744 if (mStatus == NO_ERROR) {
1745 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
jiabin156c6872017-10-06 09:47:15 -07001746 mProxy->interrupt();
Eric Laurentfb00fc72017-05-25 18:17:12 -07001747 }
Paul McLeanaa981192015-03-21 09:55:15 -07001748 }
Eric Laurent493404d2015-04-21 15:07:36 -07001749 return NO_ERROR;
Paul McLeanaa981192015-03-21 09:55:15 -07001750}
1751
1752audio_port_handle_t AudioTrack::getOutputDevice() {
1753 AutoMutex lock(mLock);
1754 return mSelectedDeviceId;
1755}
1756
Eric Laurentad2e7b92017-09-14 20:06:42 -07001757// must be called with mLock held
1758void AudioTrack::updateRoutedDeviceId_l()
1759{
1760 // if the track is inactive, do not update actual device as the output stream maybe routed
1761 // to a device not relevant to this client because of other active use cases.
1762 if (mState != STATE_ACTIVE) {
1763 return;
1764 }
1765 if (mOutput != AUDIO_IO_HANDLE_NONE) {
1766 audio_port_handle_t deviceId = AudioSystem::getDeviceIdForIo(mOutput);
1767 if (deviceId != AUDIO_PORT_HANDLE_NONE) {
1768 mRoutedDeviceId = deviceId;
1769 }
1770 }
1771}
1772
Eric Laurent296fb132015-05-01 11:38:42 -07001773audio_port_handle_t AudioTrack::getRoutedDeviceId() {
1774 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001775 updateRoutedDeviceId_l();
1776 return mRoutedDeviceId;
Eric Laurent296fb132015-05-01 11:38:42 -07001777}
1778
Eric Laurentbe916aa2010-06-01 23:49:17 -07001779status_t AudioTrack::attachAuxEffect(int effectId)
1780{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001781 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001782 status_t status;
1783 mAudioTrack->attachAuxEffect(effectId, &status);
Eric Laurent2beeb502010-07-16 07:43:46 -07001784 if (status == NO_ERROR) {
1785 mAuxEffectId = effectId;
1786 }
1787 return status;
Eric Laurentbe916aa2010-06-01 23:49:17 -07001788}
1789
Eric Laurente83b55d2014-11-14 10:06:21 -08001790audio_stream_type_t AudioTrack::streamType() const
1791{
Eric Laurente83b55d2014-11-14 10:06:21 -08001792 return mStreamType;
1793}
1794
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001795uint32_t AudioTrack::latency()
1796{
1797 AutoMutex lock(mLock);
1798 updateLatency_l();
1799 return mLatency;
1800}
1801
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08001802// -------------------------------------------------------------------------
1803
Eric Laurent1703cdf2011-03-07 14:52:59 -08001804// must be called with mLock held
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001805void AudioTrack::updateLatency_l()
1806{
1807 status_t status = AudioSystem::getLatency(mOutput, &mAfLatency);
1808 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08001809 ALOGW("%s(%d): getLatency(%d) failed status %d", __func__, mPortId, mOutput, status);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001810 } else {
1811 // FIXME don't believe this lie
Andy Hung13969262017-09-11 17:24:21 -07001812 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07001813 }
1814}
1815
Phil Burkadbb75a2017-06-16 12:19:42 -07001816// TODO Move this macro to a common header file for enum to string conversion in audio framework.
1817#define MEDIA_CASE_ENUM(name) case name: return #name
1818const char * AudioTrack::convertTransferToText(transfer_type transferType) {
1819 switch (transferType) {
1820 MEDIA_CASE_ENUM(TRANSFER_DEFAULT);
1821 MEDIA_CASE_ENUM(TRANSFER_CALLBACK);
1822 MEDIA_CASE_ENUM(TRANSFER_OBTAIN);
1823 MEDIA_CASE_ENUM(TRANSFER_SYNC);
1824 MEDIA_CASE_ENUM(TRANSFER_SHARED);
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001825 MEDIA_CASE_ENUM(TRANSFER_SYNC_NOTIF_CALLBACK);
Phil Burkadbb75a2017-06-16 12:19:42 -07001826 default:
1827 return "UNRECOGNIZED";
1828 }
1829}
1830
Glenn Kasten200092b2014-08-15 15:13:30 -07001831status_t AudioTrack::createTrack_l()
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001832{
Eric Laurentf32d7812017-11-30 14:44:07 -08001833 status_t status;
1834 bool callbackAdded = false;
1835
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001836 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
1837 if (audioFlinger == 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001838 ALOGE("%s(%d): Could not get audioflinger",
Eric Laurent973db022018-11-20 14:54:31 -08001839 __func__, mPortId);
Eric Laurentf32d7812017-11-30 14:44:07 -08001840 status = NO_INIT;
1841 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001842 }
1843
Eric Laurent21da6472017-11-09 16:29:26 -08001844 {
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08001845 // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
1846 // After fast request is denied, we will request again if IAudioTrack is re-created.
Glenn Kastend79072e2016-01-06 08:41:20 -08001847 // Client can only express a preference for FAST. Server will perform additional tests.
Phil Burk33ff89b2015-11-30 11:16:01 -08001848 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Phil Burkadbb75a2017-06-16 12:19:42 -07001849 // either of these use cases:
1850 // use case 1: shared buffer
1851 bool sharedBuffer = mSharedBuffer != 0;
1852 bool transferAllowed =
Glenn Kastenc6ba8232014-02-27 13:34:29 -08001853 // use case 2: callback transfer mode
Glenn Kasten1dfe2f92015-03-09 12:03:14 -07001854 (mTransfer == TRANSFER_CALLBACK) ||
1855 // use case 3: obtain/release mode
Phil Burk33ff89b2015-11-30 11:16:01 -08001856 (mTransfer == TRANSFER_OBTAIN) ||
1857 // use case 4: synchronous write
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07001858 ((mTransfer == TRANSFER_SYNC || mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK)
1859 && mThreadCanCallJava);
Phil Burkadbb75a2017-06-16 12:19:42 -07001860
Eric Laurent21da6472017-11-09 16:29:26 -08001861 bool fastAllowed = sharedBuffer || transferAllowed;
1862 if (!fastAllowed) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001863 ALOGW("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by client,"
1864 " not shared buffer and transfer = %s",
Eric Laurent973db022018-11-20 14:54:31 -08001865 __func__, mPortId,
Phil Burkadbb75a2017-06-16 12:19:42 -07001866 convertTransferToText(mTransfer));
Phil Burk33ff89b2015-11-30 11:16:01 -08001867 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
1868 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001869 }
1870
Eric Laurent21da6472017-11-09 16:29:26 -08001871 IAudioFlinger::CreateTrackInput input;
Andy Hunga2159aa2021-07-20 13:01:52 -07001872 if (mOriginalStreamType != AUDIO_STREAM_DEFAULT) {
1873 // Legacy: This is based on original parameters even if the track is recreated.
1874 input.attr = AudioSystem::streamTypeToAttributes(mOriginalStreamType);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001875 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08001876 input.attr = mAttributes;
Eric Laurentd1b449a2010-05-14 03:26:45 -07001877 }
Eric Laurent21da6472017-11-09 16:29:26 -08001878 input.config = AUDIO_CONFIG_INITIALIZER;
1879 input.config.sample_rate = mSampleRate;
1880 input.config.channel_mask = mChannelMask;
1881 input.config.format = mFormat;
1882 input.config.offload_info = mOffloadInfoCopy;
Svet Ganov3e5f14f2021-05-13 22:51:08 +00001883 input.clientInfo.attributionSource = mClientAttributionSource;
Eric Laurent21da6472017-11-09 16:29:26 -08001884 input.clientInfo.clientTid = -1;
Glenn Kasten363fb752014-01-15 12:27:31 -08001885 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Glenn Kasten47d55172017-05-23 11:19:30 -07001886 // It is currently meaningless to request SCHED_FIFO for a Java thread. Even if the
1887 // application-level code follows all non-blocking design rules, the language runtime
1888 // doesn't also follow those rules, so the thread will not benefit overall.
Phil Burk33ff89b2015-11-30 11:16:01 -08001889 if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
Eric Laurent21da6472017-11-09 16:29:26 -08001890 input.clientInfo.clientTid = mAudioTrackThread->getTid();
Glenn Kasten3acbd052012-02-28 10:39:56 -08001891 }
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001892 }
Eric Laurent21da6472017-11-09 16:29:26 -08001893 input.sharedBuffer = mSharedBuffer;
1894 input.notificationsPerBuffer = mNotificationsPerBufferReq;
1895 input.speed = 1.0;
1896 if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
1897 (mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
1898 input.speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
1899 max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
1900 }
1901 input.flags = mFlags;
1902 input.frameCount = mReqFrameCount;
1903 input.notificationFrameCount = mNotificationFramesReq;
1904 input.selectedDeviceId = mSelectedDeviceId;
1905 input.sessionId = mSessionId;
jiabinf6eb4c32020-02-25 14:06:25 -08001906 input.audioTrackCallback = mAudioTrackCallback;
Glenn Kasten4a4a0952012-03-19 11:38:14 -07001907
Ytai Ben-Tsvi4dfeb622020-11-02 12:47:30 -08001908 media::CreateTrackResponse response;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001909 status = audioFlinger->createTrack(VALUE_OR_FATAL(input.toAidl()), response);
Ytai Ben-Tsvi357e26a2021-01-05 13:21:19 -08001910
1911 IAudioFlinger::CreateTrackOutput output{};
1912 if (status == NO_ERROR) {
1913 output = VALUE_OR_FATAL(IAudioFlinger::CreateTrackOutput::fromAidl(response));
1914 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001915
Eric Laurent21da6472017-11-09 16:29:26 -08001916 if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001917 ALOGE("%s(%d): AudioFlinger could not create track, status: %d output %d",
Eric Laurent973db022018-11-20 14:54:31 -08001918 __func__, mPortId, status, output.outputId);
Eric Laurentf32d7812017-11-30 14:44:07 -08001919 if (status == NO_ERROR) {
1920 status = NO_INIT;
1921 }
1922 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001923 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001924 ALOG_ASSERT(output.audioTrack != 0);
Glenn Kastenc08d20b2014-02-24 15:21:10 -08001925
Eric Laurent21da6472017-11-09 16:29:26 -08001926 mFrameCount = output.frameCount;
1927 mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
1928 mRoutedDeviceId = output.selectedDeviceId;
1929 mSessionId = output.sessionId;
Andy Hunga2159aa2021-07-20 13:01:52 -07001930 mStreamType = output.streamType;
Eric Laurent21da6472017-11-09 16:29:26 -08001931
1932 mSampleRate = output.sampleRate;
1933 if (mOriginalSampleRate == 0) {
1934 mOriginalSampleRate = mSampleRate;
1935 }
1936
1937 mAfFrameCount = output.afFrameCount;
1938 mAfSampleRate = output.afSampleRate;
1939 mAfLatency = output.afLatencyMs;
1940
1941 mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;
1942
Glenn Kasten38e905b2014-01-13 10:21:48 -08001943 // AudioFlinger now owns the reference to the I/O handle,
1944 // so we are no longer responsible for releasing it.
1945
Glenn Kasten7fd04222016-02-02 12:38:16 -08001946 // FIXME compare to AudioRecord
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001947 std::optional<media::SharedFileRegion> sfr;
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001948 output.audioTrack->getCblk(&sfr);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08001949 sp<IMemory> iMem = VALUE_OR_FATAL(aidl2legacy_NullableSharedFileRegion_IMemory(sfr));
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001950 if (iMem == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08001951 ALOGE("%s(%d): Could not get control block", __func__, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001952 status = NO_INIT;
Eric Laurentf32d7812017-11-30 14:44:07 -08001953 goto exit;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08001954 }
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07001955 // TODO: Using unsecurePointer() has some associated security pitfalls
1956 // (see declaration for details).
1957 // Either document why it is safe in this case or address the
1958 // issue (e.g. by copying).
1959 void *iMemPointer = iMem->unsecurePointer();
Glenn Kasten0cde0762014-01-16 15:06:36 -08001960 if (iMemPointer == NULL) {
Eric Laurent973db022018-11-20 14:54:31 -08001961 ALOGE("%s(%d): Could not get control block pointer", __func__, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001962 status = NO_INIT;
Eric Laurentf32d7812017-11-30 14:44:07 -08001963 goto exit;
Glenn Kasten0cde0762014-01-16 15:06:36 -08001964 }
Glenn Kasten53cec222013-08-29 09:01:02 -07001965 // invariant that mAudioTrack != 0 is true only after set() returns successfully
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001966 if (mAudioTrack != 0) {
Marco Nelissenf8880202014-11-14 07:58:25 -08001967 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08001968 mDeathNotifier.clear();
1969 }
Ytai Ben-Tsvi16d87612020-11-03 16:32:36 -08001970 mAudioTrack = output.audioTrack;
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001971 mCblkMemory = iMem;
Eric Laurent3bcffa12014-06-12 18:38:45 -07001972 IPCThreadState::self()->flushCommands();
1973
Glenn Kasten0cde0762014-01-16 15:06:36 -08001974 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
Glenn Kastend2c38fc2012-11-01 14:58:02 -07001975 mCblk = cblk;
Glenn Kasten5f631512014-02-24 15:16:07 -08001976
Glenn Kastena07f17c2013-04-23 12:39:37 -07001977 mAwaitBoost = false;
Glenn Kasten363fb752014-01-15 12:27:31 -08001978 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
Eric Laurent21da6472017-11-09 16:29:26 -08001979 if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
Andy Hungfb8ede22018-09-12 19:03:24 -07001980 ALOGI("%s(%d): AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08001981 __func__, mPortId, mReqFrameCount, mFrameCount);
Phil Burk33ff89b2015-11-30 11:16:01 -08001982 if (!mThreadCanCallJava) {
1983 mAwaitBoost = true;
1984 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08001985 } else {
Robert Wuc121cd12021-08-13 17:51:40 +00001986 ALOGV("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
Eric Laurent973db022018-11-20 14:54:31 -08001987 __func__, mPortId, mReqFrameCount, mFrameCount);
Glenn Kastene0fa4672012-04-24 14:35:14 -07001988 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08001989 }
Eric Laurent21da6472017-11-09 16:29:26 -08001990 mFlags = output.flags;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01001991
Eric Laurentad2e7b92017-09-14 20:06:42 -07001992 //mOutput != output includes the case where mOutput == AUDIO_IO_HANDLE_NONE for first creation
Eric Laurent09f1ed22019-04-24 17:45:17 -07001993 if (mDeviceCallback != 0) {
Eric Laurentad2e7b92017-09-14 20:06:42 -07001994 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07001995 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001996 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07001997 AudioSystem::addAudioDeviceCallback(this, output.outputId, output.portId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07001998 callbackAdded = true;
1999 }
2000
Eric Laurent09f1ed22019-04-24 17:45:17 -07002001 mPortId = output.portId;
Glenn Kasten38e905b2014-01-13 10:21:48 -08002002 // We retain a copy of the I/O handle, but don't own the reference
Eric Laurent21da6472017-11-09 16:29:26 -08002003 mOutput = output.outputId;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002004 mRefreshRemaining = true;
2005
2006 // Starting address of buffers in shared memory. If there is a shared buffer, buffers
2007 // is the value of pointer() for the shared buffer, otherwise buffers points
2008 // immediately after the control block. This address is for the mapping within client
2009 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
2010 void* buffers;
Glenn Kasten363fb752014-01-15 12:27:31 -08002011 if (mSharedBuffer == 0) {
Glenn Kasten138d6f92015-03-20 10:54:51 -07002012 buffers = cblk + 1;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002013 } else {
Ytai Ben-Tsvi7dd39722019-09-05 15:14:30 -07002014 // TODO: Using unsecurePointer() has some associated security pitfalls
2015 // (see declaration for details).
2016 // Either document why it is safe in this case or address the
2017 // issue (e.g. by copying).
2018 buffers = mSharedBuffer->unsecurePointer();
Glenn Kasten138d6f92015-03-20 10:54:51 -07002019 if (buffers == NULL) {
Eric Laurent973db022018-11-20 14:54:31 -08002020 ALOGE("%s(%d): Could not get buffer pointer", __func__, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07002021 status = NO_INIT;
Eric Laurentf32d7812017-11-30 14:44:07 -08002022 goto exit;
Glenn Kasten138d6f92015-03-20 10:54:51 -07002023 }
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002024 }
2025
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002026 mAudioTrack->attachAuxEffect(mAuxEffectId, &status);
Glenn Kasten5f631512014-02-24 15:16:07 -08002027
Glenn Kasten093000f2012-05-03 09:35:36 -07002028 // If IAudioTrack is re-created, don't let the requested frameCount
2029 // decrease. This can confuse clients that cache frameCount().
Eric Laurent21da6472017-11-09 16:29:26 -08002030 if (mFrameCount > mReqFrameCount) {
2031 mReqFrameCount = mFrameCount;
Glenn Kasten093000f2012-05-03 09:35:36 -07002032 }
Glenn Kastene3aa6592012-12-04 12:22:46 -08002033
Andy Hungd7bd69e2015-07-24 07:52:41 -07002034 // reset server position to 0 as we have new cblk.
2035 mServer = 0;
2036
Glenn Kastene3aa6592012-12-04 12:22:46 -08002037 // update proxy
Glenn Kasten363fb752014-01-15 12:27:31 -08002038 if (mSharedBuffer == 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002039 mStaticProxy.clear();
Eric Laurent21da6472017-11-09 16:29:26 -08002040 mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002041 } else {
Eric Laurent21da6472017-11-09 16:29:26 -08002042 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002043 mProxy = mStaticProxy;
2044 }
seunghak.hane6a9d6582014-11-22 15:22:35 +09002045
2046 mProxy->setVolumeLR(gain_minifloat_pack(
2047 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
2048 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
2049
Glenn Kastene3aa6592012-12-04 12:22:46 -08002050 mProxy->setSendLevel(mSendLevel);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002051 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
2052 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
2053 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
Andy Hung26145642015-04-15 21:56:53 -07002054 mProxy->setSampleRate(effectiveSampleRate);
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002055
2056 AudioPlaybackRate playbackRateTemp = mPlaybackRate;
2057 playbackRateTemp.mSpeed = effectiveSpeed;
2058 playbackRateTemp.mPitch = effectivePitch;
2059 mProxy->setPlaybackRate(playbackRateTemp);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002060 mProxy->setMinimum(mNotificationFramesAct);
2061
Kuowei Lid4adbdb2020-08-13 14:44:25 +08002062 if (mDualMonoMode != AUDIO_DUAL_MONO_MODE_OFF) {
2063 setDualMonoMode_l(mDualMonoMode);
2064 }
2065 if (mAudioDescriptionMixLeveldB != -std::numeric_limits<float>::infinity()) {
2066 setAudioDescriptionMixLevel_l(mAudioDescriptionMixLeveldB);
2067 }
2068
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002069 mDeathNotifier = new DeathNotifier(this);
Marco Nelissenf8880202014-11-14 07:58:25 -08002070 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);
Glenn Kastene3aa6592012-12-04 12:22:46 -08002071
Andy Hungb68f5eb2019-12-03 16:49:17 -08002072 // This is the first log sent from the AudioTrack client.
2073 // The creation of the audio track by AudioFlinger (in the code above)
2074 // is the first log of the AudioTrack and must be present before
2075 // any AudioTrack client logs will be accepted.
Andy Hungea840382020-05-05 21:50:17 -07002076
Andy Hungb68f5eb2019-12-03 16:49:17 -08002077 mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(mPortId);
2078 mediametrics::LogItem(mMetricsId)
2079 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_CREATE)
2080 // the following are immutable
Andy Hunga629bd12020-06-05 16:03:53 -07002081 .set(AMEDIAMETRICS_PROP_FLAGS, toString(mFlags).c_str())
2082 .set(AMEDIAMETRICS_PROP_ORIGINALFLAGS, toString(mOrigFlags).c_str())
Andy Hungb68f5eb2019-12-03 16:49:17 -08002083 .set(AMEDIAMETRICS_PROP_SESSIONID, (int32_t)mSessionId)
Andy Hung3a5c2f32021-02-17 15:06:42 -08002084 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, mLogSessionId)
Andy Hung839a3062021-02-17 11:15:16 -08002085 .set(AMEDIAMETRICS_PROP_PLAYERIID, mPlayerIId)
Andy Hungb68f5eb2019-12-03 16:49:17 -08002086 .set(AMEDIAMETRICS_PROP_TRACKID, mPortId) // dup from key
Andy Hungb68f5eb2019-12-03 16:49:17 -08002087 .set(AMEDIAMETRICS_PROP_CONTENTTYPE, toString(mAttributes.content_type).c_str())
2088 .set(AMEDIAMETRICS_PROP_USAGE, toString(mAttributes.usage).c_str())
2089 .set(AMEDIAMETRICS_PROP_THREADID, (int32_t)output.outputId)
2090 .set(AMEDIAMETRICS_PROP_SELECTEDDEVICEID, (int32_t)mSelectedDeviceId)
2091 .set(AMEDIAMETRICS_PROP_ROUTEDDEVICEID, (int32_t)mRoutedDeviceId)
2092 .set(AMEDIAMETRICS_PROP_ENCODING, toString(mFormat).c_str())
2093 .set(AMEDIAMETRICS_PROP_CHANNELMASK, (int32_t)mChannelMask)
2094 .set(AMEDIAMETRICS_PROP_FRAMECOUNT, (int32_t)mFrameCount)
2095 // the following are NOT immutable
2096 .set(AMEDIAMETRICS_PROP_VOLUME_LEFT, (double)mVolume[AUDIO_INTERLEAVE_LEFT])
2097 .set(AMEDIAMETRICS_PROP_VOLUME_RIGHT, (double)mVolume[AUDIO_INTERLEAVE_RIGHT])
2098 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
2099 .set(AMEDIAMETRICS_PROP_AUXEFFECTID, (int32_t)mAuxEffectId)
2100 .set(AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)mSampleRate)
2101 .set(AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)mPlaybackRate.mSpeed)
2102 .set(AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)mPlaybackRate.mPitch)
2103 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2104 AMEDIAMETRICS_PROP_SAMPLERATE, (int32_t)effectiveSampleRate)
2105 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2106 AMEDIAMETRICS_PROP_PLAYBACK_SPEED, (double)effectiveSpeed)
2107 .set(AMEDIAMETRICS_PROP_PREFIX_EFFECTIVE
2108 AMEDIAMETRICS_PROP_PLAYBACK_PITCH, (double)effectivePitch)
2109 .record();
2110
2111 // mSendLevel
2112 // mReqFrameCount?
2113 // mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq
2114 // mLatency, mAfLatency, mAfFrameCount, mAfSampleRate
2115
Glenn Kasten38e905b2014-01-13 10:21:48 -08002116 }
2117
Eric Laurentf32d7812017-11-30 14:44:07 -08002118exit:
2119 if (status != NO_ERROR && callbackAdded) {
Eric Laurentad2e7b92017-09-14 20:06:42 -07002120 // note: mOutput is always valid is callbackAdded is true
Eric Laurent09f1ed22019-04-24 17:45:17 -07002121 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurentad2e7b92017-09-14 20:06:42 -07002122 }
Eric Laurentf32d7812017-11-30 14:44:07 -08002123
2124 mStatus = status;
Eric Laurent21da6472017-11-09 16:29:26 -08002125
2126 // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
Glenn Kasten38e905b2014-01-13 10:21:48 -08002127 return status;
Eric Laurent34f1d8e2009-11-04 08:27:26 -08002128}
2129
Glenn Kastenb46f3942015-03-09 12:00:30 -07002130status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002131{
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002132 if (audioBuffer == NULL) {
Glenn Kasten551b5352015-03-20 11:30:28 -07002133 if (nonContig != NULL) {
2134 *nonContig = 0;
2135 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002136 return BAD_VALUE;
Eric Laurent9b7d9502011-03-21 11:49:00 -07002137 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002138 if (mTransfer != TRANSFER_OBTAIN) {
2139 audioBuffer->frameCount = 0;
2140 audioBuffer->size = 0;
2141 audioBuffer->raw = NULL;
Glenn Kasten551b5352015-03-20 11:30:28 -07002142 if (nonContig != NULL) {
2143 *nonContig = 0;
2144 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002145 return INVALID_OPERATION;
2146 }
Eric Laurent9b7d9502011-03-21 11:49:00 -07002147
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002148 const struct timespec *requested;
Eric Laurentdf576992014-01-27 18:13:39 -08002149 struct timespec timeout;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002150 if (waitCount == -1) {
2151 requested = &ClientProxy::kForever;
2152 } else if (waitCount == 0) {
2153 requested = &ClientProxy::kNonBlocking;
2154 } else if (waitCount > 0) {
Chih-Hung Hsiehbca74292018-08-10 16:06:07 -07002155 time_t ms = WAIT_PERIOD_MS * (time_t) waitCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002156 timeout.tv_sec = ms / 1000;
Andy Hung06a730b2020-04-09 13:28:31 -07002157 timeout.tv_nsec = (ms % 1000) * 1000000;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002158 requested = &timeout;
2159 } else {
Eric Laurent973db022018-11-20 14:54:31 -08002160 ALOGE("%s(%d): invalid waitCount %d", __func__, mPortId, waitCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002161 requested = NULL;
2162 }
Glenn Kastenb46f3942015-03-09 12:00:30 -07002163 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002164}
Eric Laurent1703cdf2011-03-07 14:52:59 -08002165
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002166status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
2167 struct timespec *elapsed, size_t *nonContig)
2168{
2169 // previous and new IAudioTrack sequence numbers are used to detect track re-creation
2170 uint32_t oldSequence = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002171
2172 Proxy::Buffer buffer;
2173 status_t status = NO_ERROR;
2174
2175 static const int32_t kMaxTries = 5;
2176 int32_t tryCounter = kMaxTries;
2177
2178 do {
2179 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
2180 // keep them from going away if another thread re-creates the track during obtainBuffer()
2181 sp<AudioTrackClientProxy> proxy;
2182 sp<IMemory> iMem;
2183
2184 { // start of lock scope
2185 AutoMutex lock(mLock);
2186
Glenn Kasten305996c2020-01-27 08:03:37 -08002187 uint32_t newSequence = mSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002188 // did previous obtainBuffer() fail due to media server death or voluntary invalidation?
2189 if (status == DEAD_OBJECT) {
2190 // re-create track, unless someone else has already done so
2191 if (newSequence == oldSequence) {
2192 status = restoreTrack_l("obtainBuffer");
2193 if (status != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002194 buffer.mFrameCount = 0;
2195 buffer.mRaw = NULL;
2196 buffer.mNonContig = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002197 break;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002198 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002199 }
2200 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002201 oldSequence = newSequence;
2202
Eric Laurent4d231dc2016-03-11 18:38:23 -08002203 if (status == NOT_ENOUGH_DATA) {
2204 restartIfDisabled();
2205 }
2206
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002207 // Keep the extra references
2208 proxy = mProxy;
2209 iMem = mCblkMemory;
2210
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002211 if (mState == STATE_STOPPING) {
2212 status = -EINTR;
2213 buffer.mFrameCount = 0;
2214 buffer.mRaw = NULL;
2215 buffer.mNonContig = 0;
2216 break;
2217 }
2218
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002219 // Non-blocking if track is stopped or paused
2220 if (mState != STATE_ACTIVE) {
2221 requested = &ClientProxy::kNonBlocking;
2222 }
2223
2224 } // end of lock scope
2225
2226 buffer.mFrameCount = audioBuffer->frameCount;
2227 // FIXME starts the requested timeout and elapsed over from scratch
2228 status = proxy->obtainBuffer(&buffer, requested, elapsed);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002229 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002230
2231 audioBuffer->frameCount = buffer.mFrameCount;
Andy Hungabdb9902015-01-12 15:08:22 -08002232 audioBuffer->size = buffer.mFrameCount * mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002233 audioBuffer->raw = buffer.mRaw;
Glenn Kasten305996c2020-01-27 08:03:37 -08002234 audioBuffer->sequence = oldSequence;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002235 if (nonContig != NULL) {
2236 *nonContig = buffer.mNonContig;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002237 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002238 return status;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002239}
2240
Glenn Kasten54a8a452015-03-09 12:03:00 -07002241void AudioTrack::releaseBuffer(const Buffer* audioBuffer)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002242{
Glenn Kasten3f02be22015-03-09 11:59:04 -07002243 // FIXME add error checking on mode, by adding an internal version
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002244 if (mTransfer == TRANSFER_SHARED) {
2245 return;
2246 }
2247
Andy Hungabdb9902015-01-12 15:08:22 -08002248 size_t stepCount = audioBuffer->size / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002249 if (stepCount == 0) {
2250 return;
2251 }
2252
2253 Proxy::Buffer buffer;
2254 buffer.mFrameCount = stepCount;
2255 buffer.mRaw = audioBuffer->raw;
Glenn Kastene3aa6592012-12-04 12:22:46 -08002256
Eric Laurent1703cdf2011-03-07 14:52:59 -08002257 AutoMutex lock(mLock);
Glenn Kasten305996c2020-01-27 08:03:37 -08002258 if (audioBuffer->sequence != mSequence) {
2259 // This Buffer came from a different IAudioTrack instance, so ignore the releaseBuffer
2260 ALOGD("%s is no-op due to IAudioTrack sequence mismatch %u != %u",
2261 __func__, audioBuffer->sequence, mSequence);
2262 return;
2263 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002264 mReleased += stepCount;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002265 mInUnderrun = false;
2266 mProxy->releaseBuffer(&buffer);
2267
2268 // restart track if it was disabled by audioflinger due to previous underrun
Eric Laurent4d231dc2016-03-11 18:38:23 -08002269 restartIfDisabled();
2270}
2271
2272void AudioTrack::restartIfDisabled()
2273{
2274 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
2275 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002276 ALOGW("%s(%d): releaseBuffer() track %p disabled due to previous underrun, restarting",
Eric Laurent973db022018-11-20 14:54:31 -08002277 __func__, mPortId, this);
Eric Laurent4d231dc2016-03-11 18:38:23 -08002278 // FIXME ignoring status
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002279 status_t status;
2280 mAudioTrack->start(&status);
Eric Laurentdf839842012-05-31 14:27:14 -07002281 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002282}
2283
2284// -------------------------------------------------------------------------
2285
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002286ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002287{
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002288 if (mTransfer != TRANSFER_SYNC && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kastend65d73c2012-06-22 17:21:07 -07002289 return INVALID_OPERATION;
2290 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002291
Eric Laurentab5cdba2014-06-09 17:22:27 -07002292 if (isDirect()) {
2293 AutoMutex lock(mLock);
2294 int32_t flags = android_atomic_and(
2295 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
2296 &mCblk->mFlags);
2297 if (flags & CBLK_INVALID) {
2298 return DEAD_OBJECT;
2299 }
2300 }
2301
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002302 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
Jiabin Huang447cea72020-07-28 22:35:18 +00002303 // Validation: user is most-likely passing an error code, and it would
Glenn Kasten99e53b82012-01-19 08:59:58 -08002304 // make the return value ambiguous (actualSize vs error).
Andy Hungfb8ede22018-09-12 19:03:24 -07002305 ALOGE("%s(%d): AudioTrack::write(buffer=%p, size=%zu (%zd)",
Eric Laurent973db022018-11-20 14:54:31 -08002306 __func__, mPortId, buffer, userSize, userSize);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002307 return BAD_VALUE;
2308 }
2309
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002310 size_t written = 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002311 Buffer audioBuffer;
2312
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002313 while (userSize >= mFrameSize) {
2314 audioBuffer.frameCount = userSize / mFrameSize;
Eric Laurentc2f1f072009-07-17 12:17:14 -07002315
Jean-Michel Trivi720ad9d2014-02-04 11:00:59 -08002316 status_t err = obtainBuffer(&audioBuffer,
2317 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002318 if (err < 0) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002319 if (written > 0) {
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002320 break;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002321 }
Glenn Kasten0a2f1512016-07-22 08:06:37 -07002322 if (err == TIMED_OUT || err == -EINTR) {
2323 err = WOULD_BLOCK;
2324 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002325 return ssize_t(err);
2326 }
2327
Glenn Kastenae4b8792015-03-20 09:04:21 -07002328 size_t toWrite = audioBuffer.size;
Andy Hungabdb9902015-01-12 15:08:22 -08002329 memcpy(audioBuffer.i8, buffer, toWrite);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002330 buffer = ((const char *) buffer) + toWrite;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002331 userSize -= toWrite;
2332 written += toWrite;
2333
2334 releaseBuffer(&audioBuffer);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002335 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002336
Andy Hungea2b9c02016-02-12 17:06:53 -08002337 if (written > 0) {
2338 mFramesWritten += written / mFrameSize;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002339
2340 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2341 const sp<AudioTrackThread> t = mAudioTrackThread;
2342 if (t != 0) {
2343 // causes wake up of the playback thread, that will callback the client for
2344 // more data (with EVENT_CAN_WRITE_MORE_DATA) in processAudioBuffer()
2345 t->wake();
2346 }
2347 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002348 }
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002349
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002350 return written;
2351}
2352
2353// -------------------------------------------------------------------------
2354
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08002355nsecs_t AudioTrack::processAudioBuffer()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002356{
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002357 // Currently the AudioTrack thread is not created if there are no callbacks.
2358 // Would it ever make sense to run the thread, even without callbacks?
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002359 // If so, then replace this by checks at each use for mCallback != NULL.
Glenn Kastenfb1fdc92013-07-10 17:03:19 -07002360 LOG_ALWAYS_FATAL_IF(mCblk == NULL);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002361 mLock.lock();
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002362 sp<IAudioTrackCallback> callback = mCallback.promote();
2363 if (!callback) {
2364 mCallback = nullptr;
2365 return NS_NEVER;
2366 }
Glenn Kastena07f17c2013-04-23 12:39:37 -07002367 if (mAwaitBoost) {
2368 mAwaitBoost = false;
2369 mLock.unlock();
2370 static const int32_t kMaxTries = 5;
2371 int32_t tryCounter = kMaxTries;
2372 uint32_t pollUs = 10000;
2373 do {
Glenn Kasten8255ba72016-08-23 13:54:23 -07002374 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002375 if (policy == SCHED_FIFO || policy == SCHED_RR) {
2376 break;
2377 }
2378 usleep(pollUs);
2379 pollUs <<= 1;
2380 } while (tryCounter-- > 0);
2381 if (tryCounter < 0) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002382 ALOGE("%s(%d): did not receive expected priority boost on time",
Eric Laurent973db022018-11-20 14:54:31 -08002383 __func__, mPortId);
Glenn Kastena07f17c2013-04-23 12:39:37 -07002384 }
Glenn Kastenb0dfd462013-07-10 16:52:47 -07002385 // Run again immediately
2386 return 0;
Glenn Kastena07f17c2013-04-23 12:39:37 -07002387 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002388
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002389 // Can only reference mCblk while locked
2390 int32_t flags = android_atomic_and(
Glenn Kasten96f60d82013-07-12 10:21:18 -07002391 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);
Glenn Kastena47f3162012-11-07 10:13:08 -08002392
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002393 // Check for track invalidation
2394 if (flags & CBLK_INVALID) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002395 // for offloaded tracks restoreTrack_l() will just update the sequence and clear
2396 // AudioSystem cache. We should not exit here but after calling the callback so
2397 // that the upper layers can recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002398 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
Lajos Molnarf1063e22015-04-17 15:19:42 -07002399 status_t status __unused = restoreTrack_l("processAudioBuffer");
2400 // FIXME unused status
Andy Hung53c3b5f2014-12-15 16:42:05 -08002401 // after restoration, continue below to make sure that the loop and buffer events
2402 // are notified because they have been cleared from mCblk->mFlags above.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002403 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002404 }
2405
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002406 bool waitStreamEnd = mState == STATE_STOPPING;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002407 bool active = mState == STATE_ACTIVE;
2408
2409 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
2410 bool newUnderrun = false;
2411 if (flags & CBLK_UNDERRUN) {
2412#if 0
2413 // Currently in shared buffer mode, when the server reaches the end of buffer,
2414 // the track stays active in continuous underrun state. It's up to the application
2415 // to pause or stop the track, or set the position to a new offset within buffer.
2416 // This was some experimental code to auto-pause on underrun. Keeping it here
2417 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
2418 if (mTransfer == TRANSFER_SHARED) {
2419 mState = STATE_PAUSED;
2420 active = false;
2421 }
2422#endif
2423 if (!mInUnderrun) {
2424 mInUnderrun = true;
2425 newUnderrun = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002426 }
2427 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002428
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002429 // Get current position of server
Andy Hung90e8a972015-11-09 16:42:40 -08002430 Modulo<uint32_t> position(updateAndGetPosition_l());
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002431
2432 // Manage marker callback
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002433 bool markerReached = false;
Andy Hung90e8a972015-11-09 16:42:40 -08002434 Modulo<uint32_t> markerPosition(mMarkerPosition);
2435 // uses 32 bit wraparound for comparison with position.
2436 if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002437 mMarkerReached = markerReached = true;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002438 }
2439
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002440 // Determine number of new position callback(s) that will be needed, while locked
2441 size_t newPosCount = 0;
Andy Hung90e8a972015-11-09 16:42:40 -08002442 Modulo<uint32_t> newPosition(mNewPosition);
2443 uint32_t updatePeriod = mUpdatePeriod;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002444 // FIXME fails for wraparound, need 64 bits
2445 if (updatePeriod > 0 && position >= newPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002446 newPosCount = ((position - newPosition).value() / updatePeriod) + 1;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002447 mNewPosition += updatePeriod * newPosCount;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002448 }
2449
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002450 // Cache other fields that will be needed soon
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002451 uint32_t sampleRate = mSampleRate;
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07002452 float speed = mPlaybackRate.mSpeed;
Andy Hunga7f03352015-05-31 21:54:49 -07002453 const uint32_t notificationFrames = mNotificationFramesAct;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002454 if (mRefreshRemaining) {
2455 mRefreshRemaining = false;
2456 mRemainingFrames = notificationFrames;
2457 mRetryOnPartialBuffer = false;
2458 }
2459 size_t misalignment = mProxy->getMisalignment();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002460 uint32_t sequence = mSequence;
Glenn Kasten96f04882013-09-20 09:28:56 -07002461 sp<AudioTrackClientProxy> proxy = mProxy;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002462
Andy Hung53c3b5f2014-12-15 16:42:05 -08002463 // Determine the number of new loop callback(s) that will be needed, while locked.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002464 uint32_t loopCountNotifications = 0;
Andy Hung53c3b5f2014-12-15 16:42:05 -08002465 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
2466
2467 if (mLoopCount > 0) {
2468 int loopCount;
2469 size_t bufferPosition;
2470 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
2471 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
2472 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
2473 mLoopCountNotified = loopCount; // discard any excess notifications
2474 } else if (mLoopCount < 0) {
2475 // FIXME: We're not accurate with notification count and position with infinite looping
2476 // since loopCount from server side will always return -1 (we could decrement it).
2477 size_t bufferPosition = mStaticProxy->getBufferPosition();
2478 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
2479 loopPeriod = mLoopEnd - bufferPosition;
2480 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
2481 size_t bufferPosition = mStaticProxy->getBufferPosition();
2482 loopPeriod = mFrameCount - bufferPosition;
2483 }
2484
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002485 // These fields don't need to be cached, because they are assigned only by set():
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002486 // mTransfer, mCallback, mUserData, mFormat, mFrameSize, mFlags
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002487 // mFlags is also assigned by createTrack_l(), but not the bit we care about.
2488
2489 mLock.unlock();
2490
Andy Hunga7f03352015-05-31 21:54:49 -07002491 // get anchor time to account for callbacks.
2492 const nsecs_t timeBeforeCallbacks = systemTime();
2493
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002494 if (waitStreamEnd) {
Andy Hunga7f03352015-05-31 21:54:49 -07002495 // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread
2496 // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
2497 // (and make sure we don't callback for more data while we're stopping).
2498 // This helps with position, marker notifications, and track invalidation.
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002499 struct timespec timeout;
2500 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
2501 timeout.tv_nsec = 0;
2502
Glenn Kasten96f04882013-09-20 09:28:56 -07002503 status_t status = proxy->waitStreamEndDone(&timeout);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002504 switch (status) {
2505 case NO_ERROR:
2506 case DEAD_OBJECT:
2507 case TIMED_OUT:
Andy Hung39609a02015-09-03 16:38:38 -07002508 if (status != DEAD_OBJECT) {
2509 // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop();
2510 // instead, the application should handle the EVENT_NEW_IAUDIOTRACK.
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002511 callback->onStreamEnd();
Andy Hung39609a02015-09-03 16:38:38 -07002512 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002513 {
2514 AutoMutex lock(mLock);
2515 // The previously assigned value of waitStreamEnd is no longer valid,
2516 // since the mutex has been unlocked and either the callback handler
2517 // or another thread could have re-started the AudioTrack during that time.
2518 waitStreamEnd = mState == STATE_STOPPING;
2519 if (waitStreamEnd) {
2520 mState = STATE_STOPPED;
Andy Hungc2813e52014-10-16 17:54:34 -07002521 mReleased = 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002522 }
2523 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002524 if (waitStreamEnd && status != DEAD_OBJECT) {
2525 return NS_INACTIVE;
2526 }
2527 break;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002528 }
Glenn Kasten96f04882013-09-20 09:28:56 -07002529 return 0;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002530 }
2531
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002532 // perform callbacks while unlocked
2533 if (newUnderrun) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002534 callback->onUnderrun();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002535 }
Andy Hung53c3b5f2014-12-15 16:42:05 -08002536 while (loopCountNotifications > 0) {
Andy Hung53c3b5f2014-12-15 16:42:05 -08002537 --loopCountNotifications;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002538 callback->onLoopEnd(mLoopCount > 0 ? loopCountNotifications + mLoopCountNotified : -1);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002539 }
2540 if (flags & CBLK_BUFFER_END) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002541 callback->onBufferEnd();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002542 }
2543 if (markerReached) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002544 callback->onMarker(markerPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002545 }
2546 while (newPosCount > 0) {
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002547 callback->onNewPos(newPosition.value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002548 newPosition += updatePeriod;
2549 newPosCount--;
2550 }
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002551
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002552 if (mObservedSequence != sequence) {
2553 mObservedSequence = sequence;
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002554 callback->onNewIAudioTrack();
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002555 // for offloaded tracks, just wait for the upper layers to recreate the track
Eric Laurentab5cdba2014-06-09 17:22:27 -07002556 if (isOffloadedOrDirect()) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002557 return NS_INACTIVE;
2558 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002559 }
2560
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002561 // if inactive, then don't run me again until re-started
2562 if (!active) {
2563 return NS_INACTIVE;
Eric Laurent2267ba12011-09-07 11:13:23 -07002564 }
2565
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002566 // Compute the estimated time until the next timed event (position, markers, loops)
2567 // FIXME only for non-compressed audio
2568 uint32_t minFrames = ~0;
2569 if (!markerReached && position < markerPosition) {
Andy Hung90e8a972015-11-09 16:42:40 -08002570 minFrames = (markerPosition - position).value();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002571 }
2572 if (loopPeriod > 0 && loopPeriod < minFrames) {
Andy Hung2d85f092015-01-07 12:45:13 -08002573 // loopPeriod is already adjusted for actual position.
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002574 minFrames = loopPeriod;
2575 }
Andy Hung2d85f092015-01-07 12:45:13 -08002576 if (updatePeriod > 0) {
Andy Hung90e8a972015-11-09 16:42:40 -08002577 minFrames = min(minFrames, (newPosition - position).value());
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002578 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002579
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002580 // If > 0, poll periodically to recover from a stuck server. A good value is 2.
2581 static const uint32_t kPoll = 0;
2582 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) {
2583 minFrames = kPoll * notificationFrames;
2584 }
Eric Laurentc2f1f072009-07-17 12:17:14 -07002585
Andy Hunga7f03352015-05-31 21:54:49 -07002586 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server
2587 static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL;
2588 const nsecs_t timeAfterCallbacks = systemTime();
2589
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002590 // Convert frame units to time units
2591 nsecs_t ns = NS_WHENEVER;
2592 if (minFrames != (uint32_t) ~0) {
jiabinc7bb8322017-09-06 18:20:11 -07002593 // AudioFlinger consumption of client data may be irregular when coming out of device
2594 // standby since the kernel buffers require filling. This is throttled to no more than 2x
2595 // the expected rate in the MixerThread. Hence, we reduce the estimated time to wait by one
2596 // half (but no more than half a second) to improve callback accuracy during these temporary
2597 // data surges.
2598 const nsecs_t estimatedNs = framesToNanoseconds(minFrames, sampleRate, speed);
2599 constexpr nsecs_t maxThrottleCompensationNs = 500000000LL;
2600 ns = estimatedNs - min(estimatedNs / 2, maxThrottleCompensationNs) + kWaitPeriodNs;
Andy Hunga7f03352015-05-31 21:54:49 -07002601 ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time
2602 // TODO: Should we warn if the callback time is too long?
2603 if (ns < 0) ns = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002604 }
2605
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002606 // If not supplying data by EVENT_MORE_DATA or EVENT_CAN_WRITE_MORE_DATA, then we're done
2607 if (mTransfer != TRANSFER_CALLBACK && mTransfer != TRANSFER_SYNC_NOTIF_CALLBACK) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002608 return ns;
2609 }
2610
Andy Hunga7f03352015-05-31 21:54:49 -07002611 // EVENT_MORE_DATA callback handling.
2612 // Timing for linear pcm audio data formats can be derived directly from the
2613 // buffer fill level.
2614 // Timing for compressed data is not directly available from the buffer fill level,
2615 // rather indirectly from waiting for blocking mode callbacks or waiting for obtain()
2616 // to return a certain fill level.
2617
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002618 struct timespec timeout;
2619 const struct timespec *requested = &ClientProxy::kForever;
2620 if (ns != NS_WHENEVER) {
2621 timeout.tv_sec = ns / 1000000000LL;
2622 timeout.tv_nsec = ns % 1000000000LL;
Andy Hungfb8ede22018-09-12 19:03:24 -07002623 ALOGV("%s(%d): timeout %ld.%03d",
Eric Laurent973db022018-11-20 14:54:31 -08002624 __func__, mPortId, timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002625 requested = &timeout;
2626 }
2627
Andy Hungea2b9c02016-02-12 17:06:53 -08002628 size_t writtenFrames = 0;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002629 while (mRemainingFrames > 0) {
2630
2631 Buffer audioBuffer;
2632 audioBuffer.frameCount = mRemainingFrames;
2633 size_t nonContig;
2634 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
2635 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
Andy Hungfb8ede22018-09-12 19:03:24 -07002636 "%s(%d): obtainBuffer() err=%d frameCount=%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002637 __func__, mPortId, err, audioBuffer.frameCount);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002638 requested = &ClientProxy::kNonBlocking;
2639 size_t avail = audioBuffer.frameCount + nonContig;
Andy Hungfb8ede22018-09-12 19:03:24 -07002640 ALOGV("%s(%d): obtainBuffer(%u) returned %zu = %zu + %zu err %d",
Eric Laurent973db022018-11-20 14:54:31 -08002641 __func__, mPortId, mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002642 if (err != NO_ERROR) {
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002643 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
2644 (isOffloaded() && (err == DEAD_OBJECT))) {
Glenn Kasten606fbc12015-10-22 15:28:15 -07002645 // FIXME bug 25195759
2646 return 1000000;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002647 }
Andy Hungfb8ede22018-09-12 19:03:24 -07002648 ALOGE("%s(%d): Error %d obtaining an audio buffer, giving up.",
Eric Laurent973db022018-11-20 14:54:31 -08002649 __func__, mPortId, err);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002650 return NS_NEVER;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002651 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002652
Phil Burkfdb3c072016-02-09 10:47:02 -08002653 if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002654 mRetryOnPartialBuffer = false;
2655 if (avail < mRemainingFrames) {
Andy Hunga7f03352015-05-31 21:54:49 -07002656 if (ns > 0) { // account for obtain time
2657 const nsecs_t timeNow = systemTime();
2658 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2659 }
Andy Hungeb6e6502019-04-29 13:47:40 -07002660
2661 // delayNs is first computed by the additional frames required in the buffer.
2662 nsecs_t delayNs = framesToNanoseconds(
2663 mRemainingFrames - avail, sampleRate, speed);
2664
2665 // afNs is the AudioFlinger mixer period in ns.
2666 const nsecs_t afNs = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2667
2668 // If the AudioTrack is double buffered based on the AudioFlinger mixer period,
2669 // we may have a race if we wait based on the number of frames desired.
2670 // This is a possible issue with resampling and AAudio.
2671 //
2672 // The granularity of audioflinger processing is one mixer period; if
2673 // our wait time is less than one mixer period, wait at most half the period.
2674 if (delayNs < afNs) {
2675 delayNs = std::min(delayNs, afNs / 2);
2676 }
2677
2678 // adjust our ns wait by delayNs.
2679 if (ns < 0 /* NS_WHENEVER */ || delayNs < ns) {
2680 ns = delayNs;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002681 }
2682 return ns;
2683 }
Glenn Kastend65d73c2012-06-22 17:21:07 -07002684 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002685
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002686 size_t reqSize = audioBuffer.size;
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002687 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2688 // when notifying client it can write more data, pass the total size that can be
2689 // written in the next write() call, since it's not passed through the callback
2690 audioBuffer.size += nonContig;
2691 }
Atneya Nair6a8238eb2021-10-26 19:26:41 -04002692 size_t writtenSize = (mTransfer == TRANSFER_CALLBACK)
2693 ? callback->onMoreData(audioBuffer)
2694 : callback->onCanWriteMoreData(audioBuffer);
Jiabin Huang447cea72020-07-28 22:35:18 +00002695 // Validate on returned size
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002696 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
Andy Hungfb8ede22018-09-12 19:03:24 -07002697 ALOGE("%s(%d): EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
Eric Laurent973db022018-11-20 14:54:31 -08002698 __func__, mPortId, reqSize, ssize_t(writtenSize));
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002699 return NS_NEVER;
2700 }
2701
2702 if (writtenSize == 0) {
Jean-Michel Trivif4158d82018-09-25 12:43:58 -07002703 if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
2704 // The callback EVENT_CAN_WRITE_MORE_DATA was processed in the JNI of
2705 // android.media.AudioTrack. The JNI is not using the callback to provide data,
2706 // it only signals to the Java client that it can provide more data, which
2707 // this track is read to accept now.
2708 // The playback thread will be awaken at the next ::write()
2709 return NS_WHENEVER;
2710 }
The Android Open Source Project8555d082009-03-05 14:34:35 -08002711 // The callback is done filling buffers
2712 // Keep this thread going to handle timed events and
2713 // still try to get more data in intervals of WAIT_PERIOD_MS
2714 // but don't just loop and block the CPU, so wait
Andy Hunga7f03352015-05-31 21:54:49 -07002715
2716 // mCbf(EVENT_MORE_DATA, ...) might either
2717 // (1) Block until it can fill the buffer, returning 0 size on EOS.
2718 // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS.
2719 // (3) Return 0 size when no data is available, does not wait for more data.
2720 //
2721 // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer.
2722 // We try to compute the wait time to avoid a tight sleep-wait cycle,
2723 // especially for case (3).
2724 //
2725 // The decision to support (1) and (2) affect the sizing of mRemainingFrames
2726 // and this loop; whereas for case (3) we could simply check once with the full
2727 // buffer size and skip the loop entirely.
2728
2729 nsecs_t myns;
Phil Burkfdb3c072016-02-09 10:47:02 -08002730 if (audio_has_proportional_frames(mFormat)) {
Andy Hunga7f03352015-05-31 21:54:49 -07002731 // time to wait based on buffer occupancy
2732 const nsecs_t datans = mRemainingFrames <= avail ? 0 :
2733 framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);
2734 // audio flinger thread buffer size (TODO: adjust for fast tracks)
Glenn Kastenea38ee72016-04-18 11:08:01 -07002735 // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks.
Andy Hunga7f03352015-05-31 21:54:49 -07002736 const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed);
2737 // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0.
2738 myns = datans + (afns / 2);
2739 } else {
2740 // FIXME: This could ping quite a bit if the buffer isn't full.
2741 // Note that when mState is stopping we waitStreamEnd, so it never gets here.
2742 myns = kWaitPeriodNs;
2743 }
2744 if (ns > 0) { // account for obtain and callback time
2745 const nsecs_t timeNow = systemTime();
2746 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
2747 }
2748 if (ns < 0 /* NS_WHENEVER */ || myns < ns) {
2749 ns = myns;
2750 }
2751 return ns;
Glenn Kastend65d73c2012-06-22 17:21:07 -07002752 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002753
Glenn Kasten138d6f92015-03-20 10:54:51 -07002754 size_t releasedFrames = writtenSize / mFrameSize;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002755 audioBuffer.frameCount = releasedFrames;
2756 mRemainingFrames -= releasedFrames;
2757 if (misalignment >= releasedFrames) {
2758 misalignment -= releasedFrames;
2759 } else {
2760 misalignment = 0;
2761 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002762
2763 releaseBuffer(&audioBuffer);
Andy Hungea2b9c02016-02-12 17:06:53 -08002764 writtenFrames += releasedFrames;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002765
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002766 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
2767 // if callback doesn't like to accept the full chunk
2768 if (writtenSize < reqSize) {
2769 continue;
2770 }
2771
2772 // There could be enough non-contiguous frames available to satisfy the remaining request
2773 if (mRemainingFrames <= nonContig) {
2774 continue;
2775 }
2776
2777#if 0
2778 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a
2779 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA
2780 // that total to a sum == notificationFrames.
2781 if (0 < misalignment && misalignment <= mRemainingFrames) {
2782 mRemainingFrames = misalignment;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002783 return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002784 }
2785#endif
2786
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002787 }
Andy Hungea2b9c02016-02-12 17:06:53 -08002788 if (writtenFrames > 0) {
2789 AutoMutex lock(mLock);
2790 mFramesWritten += writtenFrames;
2791 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002792 mRemainingFrames = notificationFrames;
2793 mRetryOnPartialBuffer = true;
2794
2795 // A lot has transpired since ns was calculated, so run again immediately and re-calculate
2796 return 0;
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08002797}
2798
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002799status_t AudioTrack::restoreTrack_l(const char *from)
Eric Laurent1703cdf2011-03-07 14:52:59 -08002800{
Andy Hungb68f5eb2019-12-03 16:49:17 -08002801 status_t result = NO_ERROR; // logged: make sure to set this before returning.
2802 const int64_t beginNs = systemTime();
Andy Hunga6b27032020-04-27 10:34:24 -07002803 mediametrics::Defer defer([&] {
Andy Hungb68f5eb2019-12-03 16:49:17 -08002804 mediametrics::LogItem(mMetricsId)
2805 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_RESTORE)
Andy Hungea840382020-05-05 21:50:17 -07002806 .set(AMEDIAMETRICS_PROP_EXECUTIONTIMENS, (int64_t)(systemTime() - beginNs))
Andy Hungb68f5eb2019-12-03 16:49:17 -08002807 .set(AMEDIAMETRICS_PROP_STATE, stateToString(mState))
2808 .set(AMEDIAMETRICS_PROP_STATUS, (int32_t)result)
2809 .set(AMEDIAMETRICS_PROP_WHERE, from)
2810 .record(); });
2811
Andy Hungfb8ede22018-09-12 19:03:24 -07002812 ALOGW("%s(%d): dead IAudioTrack, %s, creating a new one from %s()",
Eric Laurent973db022018-11-20 14:54:31 -08002813 __func__, mPortId, isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002814 ++mSequence;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002815
Glenn Kastena47f3162012-11-07 10:13:08 -08002816 // refresh the audio configuration cache in this process to make sure we get new
Glenn Kastend2d089f2014-11-05 11:48:12 -08002817 // output parameters and new IAudioFlinger in createTrack_l()
Glenn Kastena47f3162012-11-07 10:13:08 -08002818 AudioSystem::clearAudioConfigCache();
Eric Laurent9f6530f2011-08-30 10:18:54 -07002819
Ronghua Wufaeb0f22015-05-21 12:20:21 -07002820 if (isOffloadedOrDirect_l() || mDoNotReconnect) {
Andy Hung1f1db832015-06-08 13:26:10 -07002821 // FIXME re-creation of offloaded and direct tracks is not yet implemented;
2822 // reconsider enabling for linear PCM encodings when position can be preserved.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002823 result = DEAD_OBJECT;
2824 return result;
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01002825 }
2826
Phil Burk2812d9e2016-01-04 10:34:30 -08002827 // Save so we can return count since creation.
2828 mUnderrunCountOffset = getUnderrunCount_l();
2829
Glenn Kasten200092b2014-08-15 15:13:30 -07002830 // save the old static buffer position
Andy Hungf20a4e92016-08-15 19:10:34 -07002831 uint32_t staticPosition = 0;
Andy Hung4ede21d2014-12-12 15:37:34 -08002832 size_t bufferPosition = 0;
2833 int loopCount = 0;
2834 if (mStaticProxy != 0) {
2835 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
Andy Hungf20a4e92016-08-15 19:10:34 -07002836 staticPosition = mStaticProxy->getPosition().unsignedValue();
Andy Hung4ede21d2014-12-12 15:37:34 -08002837 }
Glenn Kasten200092b2014-08-15 15:13:30 -07002838
Andy Hung3c7f47a2021-03-16 17:30:09 -07002839 // save the old startThreshold and framecount
2840 const uint32_t originalStartThresholdInFrames = mProxy->getStartThresholdInFrames();
2841 const uint32_t originalFrameCount = mProxy->frameCount();
2842
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002843 // See b/74409267. Connecting to a BT A2DP device supporting multiple codecs
2844 // causes a lot of churn on the service side, and it can reject starting
2845 // playback of a previously created track. May also apply to other cases.
2846 const int INITIAL_RETRIES = 3;
2847 int retries = INITIAL_RETRIES;
2848retry:
2849 if (retries < INITIAL_RETRIES) {
2850 // See the comment for clearAudioConfigCache at the start of the function.
2851 AudioSystem::clearAudioConfigCache();
2852 }
Haynes Mathew Georgeae34ed22016-01-28 11:58:39 -08002853 mFlags = mOrigFlags;
2854
Glenn Kasten200092b2014-08-15 15:13:30 -07002855 // If a new IAudioTrack is successfully created, createTrack_l() will modify the
Glenn Kastena47f3162012-11-07 10:13:08 -08002856 // following member variables: mAudioTrack, mCblkMemory and mCblk.
Glenn Kasten200092b2014-08-15 15:13:30 -07002857 // It will also delete the strong references on previous IAudioTrack and IMemory.
2858 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact.
Andy Hungb68f5eb2019-12-03 16:49:17 -08002859 result = createTrack_l();
Eric Laurentcc21e4f2013-10-16 15:12:32 -07002860
Eric Laurent6ec546d2018-10-10 16:52:14 -07002861 if (result == NO_ERROR) {
Andy Hungd7bd69e2015-07-24 07:52:41 -07002862 // take the frames that will be lost by track recreation into account in saved position
2863 // For streaming tracks, this is the amount we obtained from the user/client
2864 // (not the number actually consumed at the server - those are already lost).
2865 if (mStaticProxy == 0) {
2866 mPosition = mReleased;
2867 }
Andy Hung4ede21d2014-12-12 15:37:34 -08002868 // Continue playback from last known position and restore loop.
2869 if (mStaticProxy != 0) {
2870 if (loopCount != 0) {
2871 mStaticProxy->setBufferPositionAndLoop(bufferPosition,
2872 mLoopStart, mLoopEnd, loopCount);
2873 } else {
2874 mStaticProxy->setBufferPosition(bufferPosition);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002875 if (bufferPosition == mFrameCount) {
Eric Laurent973db022018-11-20 14:54:31 -08002876 ALOGD("%s(%d): restoring track at end of static buffer", __func__, mPortId);
Andy Hung53c3b5f2014-12-15 16:42:05 -08002877 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002878 }
2879 }
Andy Hung4ef88d72017-02-21 19:47:53 -08002880 // restore volume handler
Andy Hung39399b62017-04-21 15:07:45 -07002881 mVolumeHandler->forall([this](const VolumeShaper &shaper) -> VolumeShaper::Status {
2882 sp<VolumeShaper::Operation> operationToEnd =
2883 new VolumeShaper::Operation(shaper.mOperation);
Andy Hung4ef88d72017-02-21 19:47:53 -08002884 // TODO: Ideally we would restore to the exact xOffset position
2885 // as returned by getVolumeShaperState(), but we don't have that
2886 // information when restoring at the client unless we periodically poll
2887 // the server or create shared memory state.
2888 //
Andy Hung39399b62017-04-21 15:07:45 -07002889 // For now, we simply advance to the end of the VolumeShaper effect
2890 // if it has been started.
2891 if (shaper.isStarted()) {
Andy Hungf3702642017-05-05 17:33:32 -07002892 operationToEnd->setNormalizedTime(1.f);
Andy Hung39399b62017-04-21 15:07:45 -07002893 }
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002894 media::VolumeShaperConfiguration config;
2895 shaper.mConfiguration->writeToParcelable(&config);
2896 media::VolumeShaperOperation operation;
2897 operationToEnd->writeToParcelable(&operation);
2898 status_t status;
2899 mAudioTrack->applyVolumeShaper(config, operation, &status);
2900 return status;
Andy Hung4ef88d72017-02-21 19:47:53 -08002901 });
2902
Andy Hung3c7f47a2021-03-16 17:30:09 -07002903 // restore the original start threshold if different than frameCount.
2904 if (originalStartThresholdInFrames != originalFrameCount) {
2905 // Note: mProxy->setStartThresholdInFrames() call is in the Proxy
2906 // and does not trigger a restart.
2907 // (Also CBLK_DISABLED is not set, buffers are empty after track recreation).
2908 // Any start would be triggered on the mState == ACTIVE check below.
2909 const uint32_t currentThreshold =
2910 mProxy->setStartThresholdInFrames(originalStartThresholdInFrames);
2911 ALOGD_IF(originalStartThresholdInFrames != currentThreshold,
2912 "%s(%d) startThresholdInFrames changing from %u to %u",
2913 __func__, mPortId, originalStartThresholdInFrames, currentThreshold);
2914 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002915 if (mState == STATE_ACTIVE) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002916 mAudioTrack->start(&result);
Eric Laurent1703cdf2011-03-07 14:52:59 -08002917 }
Andy Hungf20a4e92016-08-15 19:10:34 -07002918 // server resets to zero so we offset
2919 mFramesWrittenServerOffset =
2920 mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten;
2921 mFramesWrittenAtRestore = mFramesWrittenServerOffset;
Eric Laurent1703cdf2011-03-07 14:52:59 -08002922 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08002923 if (result != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08002924 ALOGW("%s(%d): failed status %d, retries %d", __func__, mPortId, result, retries);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002925 if (--retries > 0) {
Eric Laurent6ec546d2018-10-10 16:52:14 -07002926 // leave time for an eventual race condition to clear before retrying
2927 usleep(500000);
Mikhail Naganovb13b35d2018-03-30 17:21:14 -07002928 goto retry;
2929 }
Eric Laurent6ec546d2018-10-10 16:52:14 -07002930 // if no retries left, set invalid bit to force restoring at next occasion
2931 // and avoid inconsistent active state on client and server sides
2932 if (mCblk != nullptr) {
2933 android_atomic_or(CBLK_INVALID, &mCblk->mFlags);
2934 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002935 }
Eric Laurent1703cdf2011-03-07 14:52:59 -08002936 return result;
2937}
2938
Andy Hung90e8a972015-11-09 16:42:40 -08002939Modulo<uint32_t> AudioTrack::updateAndGetPosition_l()
Glenn Kasten200092b2014-08-15 15:13:30 -07002940{
2941 // This is the sole place to read server consumed frames
Andy Hung90e8a972015-11-09 16:42:40 -08002942 Modulo<uint32_t> newServer(mProxy->getPosition());
2943 const int32_t delta = (newServer - mServer).signedValue();
Glenn Kasten200092b2014-08-15 15:13:30 -07002944 // TODO There is controversy about whether there can be "negative jitter" in server position.
2945 // This should be investigated further, and if possible, it should be addressed.
2946 // A more definite failure mode is infrequent polling by client.
2947 // One could call (void)getPosition_l() in releaseBuffer(),
2948 // so mReleased and mPosition are always lock-step as best possible.
2949 // That should ensure delta never goes negative for infrequent polling
2950 // unless the server has more than 2^31 frames in its buffer,
2951 // in which case the use of uint32_t for these counters has bigger issues.
Andy Hung90e8a972015-11-09 16:42:40 -08002952 ALOGE_IF(delta < 0,
Andy Hungfb8ede22018-09-12 19:03:24 -07002953 "%s(%d): detected illegal retrograde motion by the server: mServer advanced by %d",
Eric Laurent973db022018-11-20 14:54:31 -08002954 __func__, mPortId, delta);
Chad Brubaker039c27a2015-09-23 15:17:29 -07002955 mServer = newServer;
Andy Hung90e8a972015-11-09 16:42:40 -08002956 if (delta > 0) { // avoid retrograde
2957 mPosition += delta;
2958 }
2959 return mPosition;
Glenn Kasten200092b2014-08-15 15:13:30 -07002960}
2961
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002962bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed)
Andy Hung8edb8dc2015-03-26 19:13:55 -07002963{
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002964 updateLatency_l();
Andy Hung8edb8dc2015-03-26 19:13:55 -07002965 // applicable for mixing tracks only (not offloaded or direct)
2966 if (mStaticProxy != 0) {
2967 return true; // static tracks do not have issues with buffer sizing.
2968 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07002969 const size_t minFrameCount =
Eric Laurent21da6472017-11-09 16:29:26 -08002970 AudioSystem::calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate,
2971 sampleRate, speed /*, 0 mNotificationsPerBufferReq*/);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002972 const bool allowed = mFrameCount >= minFrameCount;
2973 ALOGD_IF(!allowed,
Andy Hungfb8ede22018-09-12 19:03:24 -07002974 "%s(%d): denied "
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002975 "mAfLatency:%u mAfFrameCount:%zu mAfSampleRate:%u sampleRate:%u speed:%f "
2976 "mFrameCount:%zu < minFrameCount:%zu",
Eric Laurent973db022018-11-20 14:54:31 -08002977 __func__, mPortId,
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002978 mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed,
Andy Hung8edb8dc2015-03-26 19:13:55 -07002979 mFrameCount, minFrameCount);
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07002980 return allowed;
Andy Hung8edb8dc2015-03-26 19:13:55 -07002981}
2982
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00002983status_t AudioTrack::setParameters(const String8& keyValuePairs)
2984{
2985 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08002986 status_t status;
2987 mAudioTrack->setParameters(keyValuePairs.c_str(), &status);
2988 return status;
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00002989}
2990
Dean Wheatleya70eef72018-01-04 14:23:50 +11002991status_t AudioTrack::selectPresentation(int presentationId, int programId)
2992{
2993 AutoMutex lock(mLock);
Eric Laurent973db022018-11-20 14:54:31 -08002994 AudioParameter param = AudioParameter();
2995 param.addInt(String8(AudioParameter::keyPresentationId), presentationId);
2996 param.addInt(String8(AudioParameter::keyProgramId), programId);
2997 ALOGV("%s(%d): PresentationId/ProgramId[%s]",
2998 __func__, mPortId, param.toString().string());
2999
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003000 status_t status;
3001 mAudioTrack->setParameters(param.toString().c_str(), &status);
3002 return status;
Dean Wheatleya70eef72018-01-04 14:23:50 +11003003}
3004
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003005VolumeShaper::Status AudioTrack::applyVolumeShaper(
3006 const sp<VolumeShaper::Configuration>& configuration,
3007 const sp<VolumeShaper::Operation>& operation)
3008{
3009 AutoMutex lock(mLock);
Andy Hung4ef88d72017-02-21 19:47:53 -08003010 mVolumeHandler->setIdIfNecessary(configuration);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003011 media::VolumeShaperConfiguration config;
3012 configuration->writeToParcelable(&config);
3013 media::VolumeShaperOperation op;
3014 operation->writeToParcelable(&op);
3015 VolumeShaper::Status status;
3016 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003017
3018 if (status == DEAD_OBJECT) {
3019 if (restoreTrack_l("applyVolumeShaper") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003020 mAudioTrack->applyVolumeShaper(config, op, &status);
Andy Hung39399b62017-04-21 15:07:45 -07003021 }
3022 }
Andy Hung4ef88d72017-02-21 19:47:53 -08003023 if (status >= 0) {
3024 // save VolumeShaper for restore
3025 mVolumeHandler->applyVolumeShaper(configuration, operation);
Andy Hung39399b62017-04-21 15:07:45 -07003026 if (mState == STATE_ACTIVE || mState == STATE_STOPPING) {
3027 mVolumeHandler->setStarted();
3028 }
3029 } else {
3030 // warn only if not an expected restore failure.
3031 ALOGW_IF(!((isOffloadedOrDirect_l() || mDoNotReconnect) && status == DEAD_OBJECT),
Eric Laurent973db022018-11-20 14:54:31 -08003032 "%s(%d): applyVolumeShaper failed: %d", __func__, mPortId, status);
Andy Hung4ef88d72017-02-21 19:47:53 -08003033 }
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003034 return status;
3035}
3036
3037sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id)
3038{
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003039 AutoMutex lock(mLock);
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003040 std::optional<media::VolumeShaperState> vss;
3041 mAudioTrack->getVolumeShaperState(id, &vss);
3042 sp<VolumeShaper::State> state;
3043 if (vss.has_value()) {
3044 state = new VolumeShaper::State();
3045 state->readFromParcelable(vss.value());
3046 }
Andy Hung39399b62017-04-21 15:07:45 -07003047 if (state.get() == nullptr && (mCblk->mFlags & CBLK_INVALID) != 0) {
3048 if (restoreTrack_l("getVolumeShaperState") == OK) {
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003049 mAudioTrack->getVolumeShaperState(id, &vss);
3050 if (vss.has_value()) {
3051 state = new VolumeShaper::State();
3052 state->readFromParcelable(vss.value());
3053 }
Andy Hung39399b62017-04-21 15:07:45 -07003054 }
3055 }
3056 return state;
Andy Hung9fc8b5c2017-01-24 13:36:48 -08003057}
3058
Andy Hungea2b9c02016-02-12 17:06:53 -08003059status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp)
3060{
3061 if (timestamp == nullptr) {
3062 return BAD_VALUE;
3063 }
3064 AutoMutex lock(mLock);
Andy Hunge13f8a62016-03-30 14:20:42 -07003065 return getTimestamp_l(timestamp);
3066}
3067
3068status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp)
3069{
Andy Hungea2b9c02016-02-12 17:06:53 -08003070 if (mCblk->mFlags & CBLK_INVALID) {
3071 const status_t status = restoreTrack_l("getTimestampExtended");
3072 if (status != OK) {
3073 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3074 // recommending that the track be recreated.
3075 return DEAD_OBJECT;
3076 }
3077 }
3078 // check for offloaded/direct here in case restoring somehow changed those flags.
3079 if (isOffloadedOrDirect_l()) {
3080 return INVALID_OPERATION; // not supported
3081 }
3082 status_t status = mProxy->getTimestamp(timestamp);
Andy Hungfb8ede22018-09-12 19:03:24 -07003083 LOG_ALWAYS_FATAL_IF(status != OK, "%s(%d): status %d not allowed from proxy getTimestamp",
Eric Laurent973db022018-11-20 14:54:31 -08003084 __func__, mPortId, status);
Andy Hungea2b9c02016-02-12 17:06:53 -08003085 bool found = false;
Andy Hunge1e98462016-04-12 10:18:51 -07003086 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten;
3087 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
3088 // server side frame offset in case AudioTrack has been restored.
3089 for (int i = ExtendedTimestamp::LOCATION_SERVER;
3090 i < ExtendedTimestamp::LOCATION_MAX; ++i) {
3091 if (timestamp->mTimeNs[i] >= 0) {
3092 // apply server offset (frames flushed is ignored
3093 // so we don't report the jump when the flush occurs).
3094 timestamp->mPosition[i] += mFramesWrittenServerOffset;
3095 found = true;
Andy Hungea2b9c02016-02-12 17:06:53 -08003096 }
3097 }
3098 return found ? OK : WOULD_BLOCK;
3099}
3100
Glenn Kastence703742013-07-19 16:33:58 -07003101status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
3102{
Glenn Kasten53cec222013-08-29 09:01:02 -07003103 AutoMutex lock(mLock);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003104 return getTimestamp_l(timestamp);
3105}
Phil Burk1b420972015-04-22 10:52:21 -07003106
Andy Hung65ffdfc2016-10-10 15:52:11 -07003107status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp)
3108{
Phil Burk1b420972015-04-22 10:52:21 -07003109 bool previousTimestampValid = mPreviousTimestampValid;
3110 // Set false here to cover all the error return cases.
3111 mPreviousTimestampValid = false;
3112
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003113 switch (mState) {
3114 case STATE_ACTIVE:
3115 case STATE_PAUSED:
3116 break; // handle below
3117 case STATE_FLUSHED:
3118 case STATE_STOPPED:
3119 return WOULD_BLOCK;
3120 case STATE_STOPPING:
3121 case STATE_PAUSED_STOPPING:
3122 if (!isOffloaded_l()) {
3123 return INVALID_OPERATION;
3124 }
3125 break; // offloaded tracks handled below
3126 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003127 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in getTimestamp(): %d",
Eric Laurent973db022018-11-20 14:54:31 -08003128 __func__, mPortId, mState);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003129 break;
Glenn Kastenfe346c72013-08-30 13:28:22 -07003130 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003131
Eric Laurent275e8e92014-11-30 15:14:47 -08003132 if (mCblk->mFlags & CBLK_INVALID) {
Andy Hung6653c932015-06-08 13:27:48 -07003133 const status_t status = restoreTrack_l("getTimestamp");
3134 if (status != OK) {
3135 // per getTimestamp() API doc in header, we return DEAD_OBJECT here,
3136 // recommending that the track be recreated.
3137 return DEAD_OBJECT;
3138 }
Eric Laurent275e8e92014-11-30 15:14:47 -08003139 }
3140
Glenn Kasten200092b2014-08-15 15:13:30 -07003141 // The presented frame count must always lag behind the consumed frame count.
3142 // To avoid a race, read the presented frames first. This ensures that presented <= consumed.
Andy Hung6ae58432016-02-16 18:32:24 -08003143
3144 status_t status;
Andy Hung818e7a32016-02-16 18:08:07 -08003145 if (isOffloadedOrDirect_l()) {
Andy Hung6ae58432016-02-16 18:32:24 -08003146 // use Binder to get timestamp
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003147 media::AudioTimestampInternal ts;
3148 mAudioTrack->getTimestamp(&ts, &status);
3149 if (status == OK) {
Andy Hung973638a2020-12-08 20:47:45 -08003150 timestamp = VALUE_OR_FATAL(aidl2legacy_AudioTimestampInternal_AudioTimestamp(ts));
Ytai Ben-Tsvibdc293a2020-11-02 17:01:38 -08003151 }
Andy Hung6ae58432016-02-16 18:32:24 -08003152 } else {
3153 // read timestamp from shared memory
3154 ExtendedTimestamp ets;
3155 status = mProxy->getTimestamp(&ets);
3156 if (status == OK) {
Andy Hungb01faa32016-04-27 12:51:32 -07003157 ExtendedTimestamp::Location location;
3158 status = ets.getBestTimestamp(&timestamp, &location);
3159
3160 if (status == OK) {
Haynes Mathew Georgefb12e202017-04-06 12:24:42 -07003161 updateLatency_l();
Andy Hungb01faa32016-04-27 12:51:32 -07003162 // It is possible that the best location has moved from the kernel to the server.
3163 // In this case we adjust the position from the previous computed latency.
3164 if (location == ExtendedTimestamp::LOCATION_SERVER) {
3165 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL,
Andy Hungfb8ede22018-09-12 19:03:24 -07003166 "%s(%d): location moved from kernel to server",
Eric Laurent973db022018-11-20 14:54:31 -08003167 __func__, mPortId);
Andy Hung07eee802016-06-21 16:47:16 -07003168 // check that the last kernel OK time info exists and the positions
3169 // are valid (if they predate the current track, the positions may
3170 // be zero or negative).
Andy Hungb01faa32016-04-27 12:51:32 -07003171 const int64_t frames =
Andy Hung6d7b1192016-05-07 22:59:48 -07003172 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
Andy Hung07eee802016-06-21 16:47:16 -07003173 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 ||
3174 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 ||
3175 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0)
Andy Hung6d7b1192016-05-07 22:59:48 -07003176 ?
3177 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed
3178 / 1000)
3179 :
3180 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3181 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]);
Andy Hungfb8ede22018-09-12 19:03:24 -07003182 ALOGV("%s(%d): frame adjustment:%lld timestamp:%s",
Eric Laurent973db022018-11-20 14:54:31 -08003183 __func__, mPortId, (long long)frames, ets.toString().c_str());
Andy Hungb01faa32016-04-27 12:51:32 -07003184 if (frames >= ets.mPosition[location]) {
3185 timestamp.mPosition = 0;
3186 } else {
3187 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames);
3188 }
Andy Hung69488c42016-05-16 18:43:33 -07003189 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) {
3190 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER,
Andy Hungfb8ede22018-09-12 19:03:24 -07003191 "%s(%d): location moved from server to kernel",
Eric Laurent973db022018-11-20 14:54:31 -08003192 __func__, mPortId);
Andy Hung98731a22019-04-08 19:19:07 -07003193
3194 if (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER] ==
3195 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL]) {
3196 // In Q, we don't return errors as an invalid time
3197 // but instead we leave the last kernel good timestamp alone.
3198 //
3199 // If server is identical to kernel, the device data pipeline is idle.
3200 // A better start time is now. The retrograde check ensures
3201 // timestamp monotonicity.
3202 const int64_t nowNs = systemTime();
Andy Hungcf3b7152019-04-19 18:29:21 -07003203 if (!mTimestampStallReported) {
3204 ALOGD("%s(%d): device stall time corrected using current time %lld",
3205 __func__, mPortId, (long long)nowNs);
3206 mTimestampStallReported = true;
3207 }
Andy Hung98731a22019-04-08 19:19:07 -07003208 timestamp.mTime = convertNsToTimespec(nowNs);
Andy Hungcf3b7152019-04-19 18:29:21 -07003209 } else {
3210 mTimestampStallReported = false;
Andy Hung98731a22019-04-08 19:19:07 -07003211 }
Andy Hungb01faa32016-04-27 12:51:32 -07003212 }
Andy Hung5d313802016-10-10 15:09:39 -07003213
3214 // We update the timestamp time even when paused.
3215 if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) {
3216 const int64_t now = systemTime();
Andy Hung2b01f002017-07-05 12:01:36 -07003217 const int64_t at = audio_utils_ns_from_timespec(&timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003218 const int64_t lag =
3219 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 ||
3220 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0)
3221 ? int64_t(mAfLatency * 1000000LL)
3222 : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK]
3223 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK])
3224 * NANOS_PER_SECOND / mSampleRate;
3225 const int64_t limit = now - lag; // no earlier than this limit
3226 if (at < limit) {
3227 ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld",
3228 (long long)lag, (long long)at, (long long)limit);
Andy Hungffa36952017-08-17 10:41:51 -07003229 timestamp.mTime = convertNsToTimespec(limit);
Andy Hung5d313802016-10-10 15:09:39 -07003230 }
3231 }
Andy Hungb01faa32016-04-27 12:51:32 -07003232 mPreviousLocation = location;
3233 } else {
3234 // right after AudioTrack is started, one may not find a timestamp
Eric Laurent973db022018-11-20 14:54:31 -08003235 ALOGV("%s(%d): getBestTimestamp did not find timestamp", __func__, mPortId);
Andy Hungb01faa32016-04-27 12:51:32 -07003236 }
Andy Hung6ae58432016-02-16 18:32:24 -08003237 }
3238 if (status == INVALID_OPERATION) {
Andy Hungf20a4e92016-08-15 19:10:34 -07003239 // INVALID_OPERATION occurs when no timestamp has been issued by the server;
3240 // other failures are signaled by a negative time.
3241 // If we come out of FLUSHED or STOPPED where the position is known
3242 // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of
3243 // "zero" for NuPlayer). We don't convert for track restoration as position
3244 // does not reset.
Andy Hungfb8ede22018-09-12 19:03:24 -07003245 ALOGV("%s(%d): timestamp server offset:%lld restore frames:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003246 __func__, mPortId,
Andy Hungf20a4e92016-08-15 19:10:34 -07003247 (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore);
3248 if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) {
3249 status = WOULD_BLOCK;
3250 }
Andy Hung6ae58432016-02-16 18:32:24 -08003251 }
3252 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003253 if (status != NO_ERROR) {
Eric Laurent973db022018-11-20 14:54:31 -08003254 ALOGV_IF(status != WOULD_BLOCK, "%s(%d): getTimestamp error:%#x", __func__, mPortId, status);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003255 return status;
3256 }
3257 if (isOffloadedOrDirect_l()) {
3258 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) {
3259 // use cached paused position in case another offloaded track is running.
3260 timestamp.mPosition = mPausedPosition;
3261 clock_gettime(CLOCK_MONOTONIC, &timestamp.mTime);
Andy Hung5d313802016-10-10 15:09:39 -07003262 // TODO: adjust for delay
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003263 return NO_ERROR;
3264 }
3265
3266 // Check whether a pending flush or stop has completed, as those commands may
Andy Hungc8e09c62015-06-03 23:43:36 -07003267 // be asynchronous or return near finish or exhibit glitchy behavior.
3268 //
3269 // Originally this showed up as the first timestamp being a continuation of
3270 // the previous song under gapless playback.
3271 // However, we sometimes see zero timestamps, then a glitch of
3272 // the previous song's position, and then correct timestamps afterwards.
Andy Hungffa36952017-08-17 10:41:51 -07003273 if (mStartFromZeroUs != 0 && mSampleRate != 0) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003274 static const int kTimeJitterUs = 100000; // 100 ms
3275 static const int k1SecUs = 1000000;
3276
3277 const int64_t timeNow = getNowUs();
3278
Andy Hungffa36952017-08-17 10:41:51 -07003279 if (timeNow < mStartFromZeroUs + k1SecUs) { // within first second of starting
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003280 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003281 if (timestampTimeUs < mStartFromZeroUs) {
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003282 return WOULD_BLOCK; // stale timestamp time, occurs before start.
3283 }
Andy Hungffa36952017-08-17 10:41:51 -07003284 const int64_t deltaTimeUs = timestampTimeUs - mStartFromZeroUs;
Andy Hung8edb8dc2015-03-26 19:13:55 -07003285 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07003286 / ((double)mSampleRate * mPlaybackRate.mSpeed);
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003287
3288 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) {
3289 // Verify that the counter can't count faster than the sample rate
Andy Hungc8e09c62015-06-03 23:43:36 -07003290 // since the start time. If greater, then that means we may have failed
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003291 // to completely flush or stop the previous playing track.
Andy Hungc8e09c62015-06-03 23:43:36 -07003292 ALOGW_IF(!mTimestampStartupGlitchReported,
Andy Hungfb8ede22018-09-12 19:03:24 -07003293 "%s(%d): startup glitch detected"
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003294 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)",
Eric Laurent973db022018-11-20 14:54:31 -08003295 __func__, mPortId,
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003296 (long long)deltaTimeUs, (long long)deltaPositionByUs,
3297 timestamp.mPosition);
Andy Hungc8e09c62015-06-03 23:43:36 -07003298 mTimestampStartupGlitchReported = true;
3299 if (previousTimestampValid
3300 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) {
3301 timestamp = mPreviousTimestamp;
3302 mPreviousTimestampValid = true;
3303 return NO_ERROR;
3304 }
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003305 return WOULD_BLOCK;
3306 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003307 if (deltaPositionByUs != 0) {
Andy Hungffa36952017-08-17 10:41:51 -07003308 mStartFromZeroUs = 0; // don't check again, we got valid nonzero position.
Andy Hungc8e09c62015-06-03 23:43:36 -07003309 }
3310 } else {
Andy Hungffa36952017-08-17 10:41:51 -07003311 mStartFromZeroUs = 0; // don't check again, start time expired.
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003312 }
Andy Hungc8e09c62015-06-03 23:43:36 -07003313 mTimestampStartupGlitchReported = false;
Andy Hung7f1bc8a2014-09-12 14:43:11 -07003314 }
3315 } else {
Glenn Kasten200092b2014-08-15 15:13:30 -07003316 // Update the mapping between local consumed (mPosition) and server consumed (mServer)
3317 (void) updateAndGetPosition_l();
3318 // Server consumed (mServer) and presented both use the same server time base,
3319 // and server consumed is always >= presented.
3320 // The delta between these represents the number of frames in the buffer pipeline.
3321 // If this delta between these is greater than the client position, it means that
3322 // actually presented is still stuck at the starting line (figuratively speaking),
3323 // waiting for the first frame to go by. So we can't report a valid timestamp yet.
Andy Hung90e8a972015-11-09 16:42:40 -08003324 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when
3325 // mPosition exceeds 32 bits.
3326 // TODO Remove when timestamp is updated to contain pipeline status info.
3327 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue();
3328 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */
3329 && (uint32_t)pipelineDepthInFrames > mPosition.value()) {
Glenn Kasten200092b2014-08-15 15:13:30 -07003330 return INVALID_OPERATION;
3331 }
3332 // Convert timestamp position from server time base to client time base.
3333 // TODO The following code should work OK now because timestamp.mPosition is 32-bit.
3334 // But if we change it to 64-bit then this could fail.
Andy Hung90e8a972015-11-09 16:42:40 -08003335 // Use Modulo computation here.
3336 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value();
Glenn Kasten200092b2014-08-15 15:13:30 -07003337 // Immediately after a call to getPosition_l(), mPosition and
3338 // mServer both represent the same frame position. mPosition is
3339 // in client's point of view, and mServer is in server's point of
3340 // view. So the difference between them is the "fudge factor"
3341 // between client and server views due to stop() and/or new
3342 // IAudioTrack. And timestamp.mPosition is initially in server's
3343 // point of view, so we need to apply the same fudge factor to it.
Glenn Kastenfe346c72013-08-30 13:28:22 -07003344 }
Phil Burk1b420972015-04-22 10:52:21 -07003345
3346 // Prevent retrograde motion in timestamp.
3347 // This is sometimes caused by erratic reports of the available space in the ALSA drivers.
3348 if (status == NO_ERROR) {
Andy Hung3b8c6332019-04-03 19:29:36 -07003349 // Fix stale time when checking timestamp right after start().
3350 // The position is at the last reported location but the time can be stale
3351 // due to pause or standby or cold start latency.
3352 //
3353 // We keep advancing the time (but not the position) to ensure that the
3354 // stale value does not confuse the application.
3355 //
3356 // For offload compatibility, use a default lag value here.
3357 // Any time discrepancy between this update and the pause timestamp is handled
3358 // by the retrograde check afterwards.
3359 int64_t currentTimeNanos = audio_utils_ns_from_timespec(&timestamp.mTime);
3360 const int64_t lagNs = int64_t(mAfLatency * 1000000LL);
3361 const int64_t limitNs = mStartNs - lagNs;
3362 if (currentTimeNanos < limitNs) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003363 if (!mTimestampStaleTimeReported) {
3364 ALOGD("%s(%d): stale timestamp time corrected, "
3365 "currentTimeNanos: %lld < limitNs: %lld < mStartNs: %lld",
3366 __func__, mPortId,
3367 (long long)currentTimeNanos, (long long)limitNs, (long long)mStartNs);
3368 mTimestampStaleTimeReported = true;
3369 }
Andy Hung3b8c6332019-04-03 19:29:36 -07003370 timestamp.mTime = convertNsToTimespec(limitNs);
3371 currentTimeNanos = limitNs;
Andy Hungcf3b7152019-04-19 18:29:21 -07003372 } else {
3373 mTimestampStaleTimeReported = false;
Andy Hung3b8c6332019-04-03 19:29:36 -07003374 }
3375
Andy Hungffa36952017-08-17 10:41:51 -07003376 // previousTimestampValid is set to false when starting after a stop or flush.
Phil Burk1b420972015-04-22 10:52:21 -07003377 if (previousTimestampValid) {
Andy Hung2b01f002017-07-05 12:01:36 -07003378 const int64_t previousTimeNanos =
3379 audio_utils_ns_from_timespec(&mPreviousTimestamp.mTime);
Andy Hungffa36952017-08-17 10:41:51 -07003380
3381 // retrograde check
Phil Burk1b420972015-04-22 10:52:21 -07003382 if (currentTimeNanos < previousTimeNanos) {
Andy Hungcf3b7152019-04-19 18:29:21 -07003383 if (!mTimestampRetrogradeTimeReported) {
3384 ALOGW("%s(%d): retrograde timestamp time corrected, %lld < %lld",
3385 __func__, mPortId,
3386 (long long)currentTimeNanos, (long long)previousTimeNanos);
3387 mTimestampRetrogradeTimeReported = true;
3388 }
Andy Hung5d313802016-10-10 15:09:39 -07003389 timestamp.mTime = mPreviousTimestamp.mTime;
Andy Hungcf3b7152019-04-19 18:29:21 -07003390 } else {
3391 mTimestampRetrogradeTimeReported = false;
Phil Burk1b420972015-04-22 10:52:21 -07003392 }
3393
3394 // Looking at signed delta will work even when the timestamps
3395 // are wrapping around.
Andy Hung90e8a972015-11-09 16:42:40 -08003396 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition)
3397 - mPreviousTimestamp.mPosition).signedValue();
Phil Burk4c5a3672015-04-30 16:18:53 -07003398 if (deltaPosition < 0) {
3399 // Only report once per position instead of spamming the log.
Andy Hungcf3b7152019-04-19 18:29:21 -07003400 if (!mTimestampRetrogradePositionReported) {
Andy Hungfb8ede22018-09-12 19:03:24 -07003401 ALOGW("%s(%d): retrograde timestamp position corrected, %d = %u - %u",
Eric Laurent973db022018-11-20 14:54:31 -08003402 __func__, mPortId,
Phil Burk4c5a3672015-04-30 16:18:53 -07003403 deltaPosition,
3404 timestamp.mPosition,
3405 mPreviousTimestamp.mPosition);
Andy Hungcf3b7152019-04-19 18:29:21 -07003406 mTimestampRetrogradePositionReported = true;
Phil Burk4c5a3672015-04-30 16:18:53 -07003407 }
3408 } else {
Andy Hungcf3b7152019-04-19 18:29:21 -07003409 mTimestampRetrogradePositionReported = false;
Phil Burk4c5a3672015-04-30 16:18:53 -07003410 }
Andy Hung5d313802016-10-10 15:09:39 -07003411 if (deltaPosition < 0) {
3412 timestamp.mPosition = mPreviousTimestamp.mPosition;
3413 deltaPosition = 0;
Phil Burk1b420972015-04-22 10:52:21 -07003414 }
Andy Hung5d313802016-10-10 15:09:39 -07003415#if 0
3416 // Uncomment this to verify audio timestamp rate.
3417 const int64_t deltaTime =
Andy Hung2b01f002017-07-05 12:01:36 -07003418 audio_utils_ns_from_timespec(&timestamp.mTime) - previousTimeNanos;
Andy Hung5d313802016-10-10 15:09:39 -07003419 if (deltaTime != 0) {
3420 const int64_t computedSampleRate =
3421 deltaPosition * (long long)NANOS_PER_SECOND / deltaTime;
Andy Hungfb8ede22018-09-12 19:03:24 -07003422 ALOGD("%s(%d): computedSampleRate:%u sampleRate:%u",
Eric Laurent973db022018-11-20 14:54:31 -08003423 __func__, mPortId,
Andy Hung5d313802016-10-10 15:09:39 -07003424 (unsigned)computedSampleRate, mSampleRate);
3425 }
3426#endif
Phil Burk1b420972015-04-22 10:52:21 -07003427 }
3428 mPreviousTimestamp = timestamp;
3429 mPreviousTimestampValid = true;
3430 }
3431
Glenn Kastenfe346c72013-08-30 13:28:22 -07003432 return status;
Glenn Kastence703742013-07-19 16:33:58 -07003433}
3434
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003435String8 AudioTrack::getParameters(const String8& keys)
3436{
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003437 audio_io_handle_t output = getOutput();
Glenn Kasten142f5192014-03-25 17:44:59 -07003438 if (output != AUDIO_IO_HANDLE_NONE) {
Glenn Kasten2c6c5292014-01-13 10:29:08 -08003439 return AudioSystem::getParameters(output, keys);
Richard Fitzgeraldb1a270d2013-05-14 12:12:21 +01003440 } else {
3441 return String8::empty();
3442 }
Richard Fitzgeraldad3af332013-03-25 16:54:37 +00003443}
3444
Glenn Kasten23a75452014-01-13 10:37:17 -08003445bool AudioTrack::isOffloaded() const
3446{
3447 AutoMutex lock(mLock);
3448 return isOffloaded_l();
3449}
3450
Eric Laurentab5cdba2014-06-09 17:22:27 -07003451bool AudioTrack::isDirect() const
3452{
3453 AutoMutex lock(mLock);
3454 return isDirect_l();
3455}
3456
3457bool AudioTrack::isOffloadedOrDirect() const
3458{
3459 AutoMutex lock(mLock);
3460 return isOffloadedOrDirect_l();
3461}
3462
3463
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003464status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003465{
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003466 String8 result;
3467
3468 result.append(" AudioTrack::dump\n");
Andy Hungfb8ede22018-09-12 19:03:24 -07003469 result.appendFormat(" id(%d) status(%d), state(%d), session Id(%d), flags(%#x)\n",
Eric Laurent973db022018-11-20 14:54:31 -08003470 mPortId, mStatus, mState, mSessionId, mFlags);
Eric Laurentd114b622017-11-27 18:37:04 -08003471 result.appendFormat(" stream type(%d), left - right volume(%f, %f)\n",
François Gaffie58d4be52018-11-06 15:30:12 +01003472 mStreamType,
Eric Laurentd114b622017-11-27 18:37:04 -08003473 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]);
Glenn Kasten49f36ba2017-12-06 13:02:02 -08003474 result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
Eric Laurentd114b622017-11-27 18:37:04 -08003475 mFormat, mChannelMask, mChannelCount);
3476 result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
3477 mSampleRate, mOriginalSampleRate, mPlaybackRate.mSpeed);
3478 result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
3479 mFrameCount, mReqFrameCount);
3480 result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
3481 " req. notif. per buff(%u)\n",
3482 mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
3483 result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
3484 mLatency, mSelectedDeviceId, mRoutedDeviceId);
3485 result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
3486 mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003487 ::write(fd, result.string(), result.size());
3488 return NO_ERROR;
3489}
3490
Phil Burk2812d9e2016-01-04 10:34:30 -08003491uint32_t AudioTrack::getUnderrunCount() const
3492{
3493 AutoMutex lock(mLock);
3494 return getUnderrunCount_l();
3495}
3496
3497uint32_t AudioTrack::getUnderrunCount_l() const
3498{
3499 return mProxy->getUnderrunCount() + mUnderrunCountOffset;
3500}
3501
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003502uint32_t AudioTrack::getUnderrunFrames() const
3503{
3504 AutoMutex lock(mLock);
3505 return mProxy->getUnderrunFrames();
3506}
3507
Andy Hung3a5c2f32021-02-17 15:06:42 -08003508void AudioTrack::setLogSessionId(const char *logSessionId)
3509{
3510 AutoMutex lock(mLock);
Andy Hung1a9c21b2021-02-25 20:43:18 -08003511 if (logSessionId == nullptr) logSessionId = ""; // an empty string is an unset session id.
Andy Hung3a5c2f32021-02-17 15:06:42 -08003512 if (mLogSessionId == logSessionId) return;
3513
3514 mLogSessionId = logSessionId;
3515 mediametrics::LogItem(mMetricsId)
3516 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETLOGSESSIONID)
3517 .set(AMEDIAMETRICS_PROP_LOGSESSIONID, logSessionId)
3518 .record();
3519}
3520
Andy Hung839a3062021-02-17 11:15:16 -08003521void AudioTrack::setPlayerIId(int playerIId)
3522{
3523 AutoMutex lock(mLock);
3524 if (mPlayerIId == playerIId) return;
3525
3526 mPlayerIId = playerIId;
3527 mediametrics::LogItem(mMetricsId)
3528 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETPLAYERIID)
3529 .set(AMEDIAMETRICS_PROP_PLAYERIID, playerIId)
3530 .record();
3531}
3532
Eric Laurent296fb132015-05-01 11:38:42 -07003533status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback)
3534{
Eric Laurent09f1ed22019-04-24 17:45:17 -07003535
Eric Laurent296fb132015-05-01 11:38:42 -07003536 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003537 ALOGW("%s(%d): adding NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003538 return BAD_VALUE;
3539 }
3540 AutoMutex lock(mLock);
Eric Laurentad2e7b92017-09-14 20:06:42 -07003541 if (mDeviceCallback.unsafe_get() == callback.get()) {
Eric Laurent973db022018-11-20 14:54:31 -08003542 ALOGW("%s(%d): adding same callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003543 return INVALID_OPERATION;
3544 }
3545 status_t status = NO_ERROR;
3546 if (mOutput != AUDIO_IO_HANDLE_NONE) {
3547 if (mDeviceCallback != 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003548 ALOGW("%s(%d): callback already present!", __func__, mPortId);
Eric Laurent09f1ed22019-04-24 17:45:17 -07003549 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003550 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003551 status = AudioSystem::addAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003552 }
3553 mDeviceCallback = callback;
3554 return status;
3555}
3556
3557status_t AudioTrack::removeAudioDeviceCallback(
3558 const sp<AudioSystem::AudioDeviceCallback>& callback)
3559{
3560 if (callback == 0) {
Eric Laurent973db022018-11-20 14:54:31 -08003561 ALOGW("%s(%d): removing NULL callback!", __func__, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003562 return BAD_VALUE;
3563 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003564 AutoMutex lock(mLock);
3565 if (mDeviceCallback.unsafe_get() != callback.get()) {
3566 ALOGW("%s removing different callback!", __FUNCTION__);
3567 return INVALID_OPERATION;
Eric Laurent296fb132015-05-01 11:38:42 -07003568 }
Eric Laurent4463ff52019-02-07 13:56:09 -08003569 mDeviceCallback.clear();
Eric Laurent296fb132015-05-01 11:38:42 -07003570 if (mOutput != AUDIO_IO_HANDLE_NONE) {
Eric Laurent09f1ed22019-04-24 17:45:17 -07003571 AudioSystem::removeAudioDeviceCallback(this, mOutput, mPortId);
Eric Laurent296fb132015-05-01 11:38:42 -07003572 }
Eric Laurent296fb132015-05-01 11:38:42 -07003573 return NO_ERROR;
3574}
3575
Eric Laurentad2e7b92017-09-14 20:06:42 -07003576
3577void AudioTrack::onAudioDeviceUpdate(audio_io_handle_t audioIo,
3578 audio_port_handle_t deviceId)
3579{
3580 sp<AudioSystem::AudioDeviceCallback> callback;
3581 {
3582 AutoMutex lock(mLock);
3583 if (audioIo != mOutput) {
3584 return;
3585 }
3586 callback = mDeviceCallback.promote();
3587 // only update device if the track is active as route changes due to other use cases are
3588 // irrelevant for this client
3589 if (mState == STATE_ACTIVE) {
3590 mRoutedDeviceId = deviceId;
3591 }
3592 }
Eric Laurent09f1ed22019-04-24 17:45:17 -07003593
Eric Laurentad2e7b92017-09-14 20:06:42 -07003594 if (callback.get() != nullptr) {
3595 callback->onAudioDeviceUpdate(mOutput, mRoutedDeviceId);
3596 }
3597}
3598
Andy Hunge13f8a62016-03-30 14:20:42 -07003599status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location)
3600{
3601 if (msec == nullptr ||
3602 (location != ExtendedTimestamp::LOCATION_SERVER
3603 && location != ExtendedTimestamp::LOCATION_KERNEL)) {
3604 return BAD_VALUE;
3605 }
3606 AutoMutex lock(mLock);
3607 // inclusive of offloaded and direct tracks.
3608 //
3609 // It is possible, but not enabled, to allow duration computation for non-pcm
3610 // audio_has_proportional_frames() formats because currently they have
3611 // the drain rate equivalent to the pcm sample rate * framesize.
3612 if (!isPurePcmData_l()) {
3613 return INVALID_OPERATION;
3614 }
3615 ExtendedTimestamp ets;
3616 if (getTimestamp_l(&ets) == OK
3617 && ets.mTimeNs[location] > 0) {
3618 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
3619 - ets.mPosition[location];
3620 if (diff < 0) {
3621 *msec = 0;
3622 } else {
3623 // ms is the playback time by frames
3624 int64_t ms = (int64_t)((double)diff * 1000 /
3625 ((double)mSampleRate * mPlaybackRate.mSpeed));
3626 // clockdiff is the timestamp age (negative)
3627 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
3628 ets.mTimeNs[location]
3629 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
3630 - systemTime(SYSTEM_TIME_MONOTONIC);
3631
3632 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
3633 static const int NANOS_PER_MILLIS = 1000000;
3634 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
3635 }
3636 return NO_ERROR;
3637 }
3638 if (location != ExtendedTimestamp::LOCATION_SERVER) {
3639 return INVALID_OPERATION; // LOCATION_KERNEL is not available
3640 }
3641 // use server position directly (offloaded and direct arrive here)
3642 updateAndGetPosition_l();
3643 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue();
3644 *msec = (diff <= 0) ? 0
3645 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed));
3646 return NO_ERROR;
3647}
3648
Andy Hung65ffdfc2016-10-10 15:52:11 -07003649bool AudioTrack::hasStarted()
3650{
3651 AutoMutex lock(mLock);
3652 switch (mState) {
3653 case STATE_STOPPED:
3654 if (isOffloadedOrDirect_l()) {
3655 // check if we have started in the past to return true.
Andy Hungffa36952017-08-17 10:41:51 -07003656 return mStartFromZeroUs > 0;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003657 }
3658 // A normal audio track may still be draining, so
3659 // check if stream has ended. This covers fasttrack position
3660 // instability and start/stop without any data written.
3661 if (mProxy->getStreamEndDone()) {
3662 return true;
3663 }
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003664 FALLTHROUGH_INTENDED;
Andy Hung65ffdfc2016-10-10 15:52:11 -07003665 case STATE_ACTIVE:
3666 case STATE_STOPPING:
3667 break;
3668 case STATE_PAUSED:
3669 case STATE_PAUSED_STOPPING:
3670 case STATE_FLUSHED:
3671 return false; // we're not active
3672 default:
Eric Laurent973db022018-11-20 14:54:31 -08003673 LOG_ALWAYS_FATAL("%s(%d): Invalid mState in hasStarted(): %d", __func__, mPortId, mState);
Andy Hung65ffdfc2016-10-10 15:52:11 -07003674 break;
3675 }
3676
3677 // wait indicates whether we need to wait for a timestamp.
3678 // This is conservatively figured - if we encounter an unexpected error
3679 // then we will not wait.
3680 bool wait = false;
3681 if (isOffloadedOrDirect_l()) {
3682 AudioTimestamp ts;
3683 status_t status = getTimestamp_l(ts);
3684 if (status == WOULD_BLOCK) {
3685 wait = true;
3686 } else if (status == OK) {
3687 wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition);
3688 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003689 ALOGV("%s(%d): hasStarted wait:%d ts:%u start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003690 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003691 (int)wait,
3692 ts.mPosition,
3693 (long long)mStartTs.mPosition);
3694 } else {
3695 int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG
3696 ExtendedTimestamp ets;
3697 status_t status = getTimestamp_l(&ets);
3698 if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets
3699 wait = true;
3700 } else if (status == OK) {
3701 for (location = ExtendedTimestamp::LOCATION_KERNEL;
3702 location >= ExtendedTimestamp::LOCATION_SERVER; --location) {
3703 if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) {
3704 continue;
3705 }
3706 wait = ets.mPosition[location] == 0
3707 || ets.mPosition[location] == mStartEts.mPosition[location];
3708 break;
3709 }
3710 }
Andy Hungfb8ede22018-09-12 19:03:24 -07003711 ALOGV("%s(%d): hasStarted wait:%d ets:%lld start position:%lld",
Eric Laurent973db022018-11-20 14:54:31 -08003712 __func__, mPortId,
Andy Hung65ffdfc2016-10-10 15:52:11 -07003713 (int)wait,
3714 (long long)ets.mPosition[location],
3715 (long long)mStartEts.mPosition[location]);
3716 }
3717 return !wait;
3718}
3719
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003720// =========================================================================
3721
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003722void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003723{
3724 sp<AudioTrack> audioTrack = mAudioTrack.promote();
3725 if (audioTrack != 0) {
3726 AutoMutex lock(audioTrack->mLock);
3727 audioTrack->mProxy->binderDied();
3728 }
3729}
3730
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003731// =========================================================================
3732
Andy Hungca353672019-03-06 11:54:38 -08003733AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver)
Andy Hungeb46cf92019-03-06 10:13:38 -08003734 : Thread(true /* bCanCallJava */) // binder recursion on restoreTrack_l() may call Java.
3735 , mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL),
Glenn Kasten598de6c2013-10-16 17:02:13 -07003736 mIgnoreNextPausedInt(false)
Glenn Kasten3acbd052012-02-28 10:39:56 -08003737{
3738}
3739
3740AudioTrack::AudioTrackThread::~AudioTrackThread()
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003741{
3742}
3743
3744bool AudioTrack::AudioTrackThread::threadLoop()
3745{
Glenn Kasten3acbd052012-02-28 10:39:56 -08003746 {
3747 AutoMutex _l(mMyLock);
3748 if (mPaused) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003749 // TODO check return value and handle or log
Glenn Kasten3acbd052012-02-28 10:39:56 -08003750 mMyCond.wait(mMyLock);
3751 // caller will check for exitPending()
3752 return true;
3753 }
Glenn Kasten598de6c2013-10-16 17:02:13 -07003754 if (mIgnoreNextPausedInt) {
3755 mIgnoreNextPausedInt = false;
3756 mPausedInt = false;
3757 }
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003758 if (mPausedInt) {
Glenn Kasten47d55172017-05-23 11:19:30 -07003759 // TODO use futex instead of condition, for event flag "or"
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003760 if (mPausedNs > 0) {
Glenn Kasten75f79032017-05-23 11:22:44 -07003761 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003762 (void) mMyCond.waitRelative(mMyLock, mPausedNs);
3763 } else {
Glenn Kasten75f79032017-05-23 11:22:44 -07003764 // TODO check return value and handle or log
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003765 mMyCond.wait(mMyLock);
3766 }
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003767 mPausedInt = false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003768 return true;
3769 }
Glenn Kasten3acbd052012-02-28 10:39:56 -08003770 }
Eric Laurent7985dcb2014-10-07 15:45:14 -07003771 if (exitPending()) {
3772 return false;
3773 }
Glenn Kasten7c7be1e2013-12-19 16:34:04 -08003774 nsecs_t ns = mReceiver.processAudioBuffer();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003775 switch (ns) {
3776 case 0:
3777 return true;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003778 case NS_INACTIVE:
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003779 pauseInternal();
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003780 return true;
3781 case NS_NEVER:
3782 return false;
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003783 case NS_WHENEVER:
Andy Hung3c09c782014-12-29 18:39:32 -08003784 // Event driven: call wake() when callback notifications conditions change.
3785 ns = INT64_MAX;
Chih-Hung Hsiehffe35582018-09-13 13:59:28 -07003786 FALLTHROUGH_INTENDED;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003787 default:
Andy Hungfb8ede22018-09-12 19:03:24 -07003788 LOG_ALWAYS_FATAL_IF(ns < 0, "%s(%d): processAudioBuffer() returned %lld",
Eric Laurent973db022018-11-20 14:54:31 -08003789 __func__, mReceiver.mPortId, (long long)ns);
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003790 pauseInternal(ns);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08003791 return true;
Glenn Kastenca8b2802012-04-23 13:58:16 -07003792 }
The Android Open Source Project89fa4ad2009-03-03 19:31:44 -08003793}
3794
Glenn Kasten3acbd052012-02-28 10:39:56 -08003795void AudioTrack::AudioTrackThread::requestExit()
3796{
3797 // must be in this order to avoid a race condition
3798 Thread::requestExit();
Glenn Kasten598de6c2013-10-16 17:02:13 -07003799 resume();
Glenn Kasten3acbd052012-02-28 10:39:56 -08003800}
3801
3802void AudioTrack::AudioTrackThread::pause()
3803{
3804 AutoMutex _l(mMyLock);
3805 mPaused = true;
3806}
3807
3808void AudioTrack::AudioTrackThread::resume()
3809{
3810 AutoMutex _l(mMyLock);
Glenn Kasten598de6c2013-10-16 17:02:13 -07003811 mIgnoreNextPausedInt = true;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003812 if (mPaused || mPausedInt) {
Glenn Kasten3acbd052012-02-28 10:39:56 -08003813 mPaused = false;
Eric Laurent9d2c78c2013-09-23 12:29:42 -07003814 mPausedInt = false;
Glenn Kasten3acbd052012-02-28 10:39:56 -08003815 mMyCond.signal();
3816 }
3817}
3818
Andy Hung3c09c782014-12-29 18:39:32 -08003819void AudioTrack::AudioTrackThread::wake()
3820{
3821 AutoMutex _l(mMyLock);
Andy Hunga8d08902015-07-22 11:52:13 -07003822 if (!mPaused) {
3823 // wake() might be called while servicing a callback - ignore the next
3824 // pause time and call processAudioBuffer.
Andy Hung3c09c782014-12-29 18:39:32 -08003825 mIgnoreNextPausedInt = true;
Andy Hunga8d08902015-07-22 11:52:13 -07003826 if (mPausedInt && mPausedNs > 0) {
3827 // audio track is active and internally paused with timeout.
3828 mPausedInt = false;
3829 mMyCond.signal();
3830 }
Andy Hung3c09c782014-12-29 18:39:32 -08003831 }
3832}
3833
Glenn Kasten5a6cd222013-09-20 09:20:45 -07003834void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
3835{
3836 AutoMutex _l(mMyLock);
3837 mPausedInt = true;
3838 mPausedNs = ns;
3839}
3840
jiabinf6eb4c32020-02-25 14:06:25 -08003841binder::Status AudioTrack::AudioTrackCallback::onCodecFormatChanged(
3842 const std::vector<uint8_t>& audioMetadata)
3843{
3844 AutoMutex _l(mAudioTrackCbLock);
3845 sp<media::IAudioTrackCallback> callback = mCallback.promote();
3846 if (callback.get() != nullptr) {
3847 callback->onCodecFormatChanged(audioMetadata);
3848 } else {
3849 mCallback.clear();
3850 }
3851 return binder::Status::ok();
3852}
3853
3854void AudioTrack::AudioTrackCallback::setAudioTrackCallback(
3855 const sp<media::IAudioTrackCallback> &callback) {
3856 AutoMutex lock(mAudioTrackCbLock);
3857 mCallback = callback;
3858}
3859
Glenn Kasten40bc9062015-03-20 09:09:33 -07003860} // namespace android